Is app_meetme broken?
I seem to get invalid conference number all the time :(
Panny
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OK, maybe I need more coffee. Or less. Either way, I'm stumped.
I have a Meetme conference room configured. Meetme(|M|) to enable
the MOH. When you are the first one to go into the conf, you get the
announcement that you are the only one, and then a *male* voice gives
a little talk about 'Why are
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:.
_
I know I should probably post this to the Trixbox forums, but am hoping
someone might have a quick answer here for me. Client with Trixbox 2.6.2
with a recompiled asterisk 1.4.23 and various patches for RPID. A meetme
conference with several people was in progress when the UPS died and the
ma
Hi All
I would Like to run a macro in a meetme conference when a user presses a
certain digit sequence, but I cannot seem to find how to do this , is it
possible?
if so how?
Thanks for you help
Robb
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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
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Hi people,
I have a user that forgets to hangup his conference calls, so they go on
forever. Is there a way of limiting the duration of a conf call?
Thanks in advance,
Pablo
--
Pablo Endres <[EMAIL PROTECTED]>
ComVoz Communications
USA: +1 954 343-2085 Ext 199
Venezuela: +58 212 7713195
I am trying to get a simple MeetMe running between
a few SIP phones here in our office.
Here is a clip from my extensions.conf
exten => 9000,1,Ringingexten =>
9000,2,Answerexten => 9000,3,Wait(1)exten
=> 9000,4,MeetMe(|Md)
Here is what my console says:
-- Executing
Ringin
Can it be that the MeetMe application is not installed by
default even if there is a meetme.conf ?
pbx.c:1280 pbx_extension_helper: No application 'MeetMe' for
extension (from-sip, 550, 4)
Regards,
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Hi,
Can someone see what's wrong here please ?
I've installed the ztdummy driver to enable meetme, put his in my
extension.conf
exten => 550,1,Answer
exten => 550,2,Wait(1)
exten => 550,4,MeetMe(18|Md)
exten => 550,5,Hangup
this in my meetme.conf
[rooms]
;
; Usage is conf => confno[,pin]
;
con
Hey All,
Just finished installing Asterisk and configured all the
necessary parameters to start.
I can’t seem to find the Meetme application in my
asterisk directory.
I downloaded asterisk from CVS and installed it and all my
Snom phones are working and voicemail too.
I am get
I've got a problem with MeetMe. I dial the extension that dynamically
creates the new conf, but it just hangs up on me after telling me I'm
the only person in the conference. Here's my extensions.conf and what
its doing:
-- Executing Answer("SIP/101-74c0", "") in new stack
-- Executing Wa
I know it’s on the wiki somewhere, but I’ve been
searching for 2 days on the wiki and google. I just can’t seem to find the web
interface for the meetme application. Does anyone have a link I could use. I found
ASGI’s dynamic conferences which is neat. Thanks in advance.
I have just started using * and have been trying to set up MeetMe. So far I
have not been able to start a conference. When I dial the conference
extension (I am using X-Lite softphone), the call hangs up. I am not using
Zaptel cards so I uncommented the ztdummy in the Zaptel Makefile. My
configurat
I'm presently using meetme extensively on my server. I have a rather
strange question. I'm using it with one person in "talk-only" mode and
everybody else in "monitor" mode. I'm running an athlon xp 2800 with
1gb of ram. I can handle 40 users. Does anybody know of any
adjustments that coul
Anyone notice on latest SVN trunk that meetme no longer asks for a pin when
you try to enter a room?
I want to verify it before I post it as a bug.
John Bittner
Simlab.net
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Hi List, I have the following in my extensions.conf. For some reason if the user enters a room that does not exist instead of going to the next pri. it just says room invalid and dumps the call. Can it be a bug ? Exten => _*5XXX,1,MeetMe(${EXTEN:1},D) Exten => _5XXX,1,MeetMe(${EXTEN},cMrpsq)
I'm having issues getting meetme to work:
Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No
application 'MeetMe' for extension (internal, , 2)
== Spawn extension (internal, , 2) exited non-zero on
'SIP/mileslap-569b'
the only thing I could find was this:
http://72.
I'm having issues getting meetme to work:
Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No
application 'MeetMe' for extension (internal, , 2)
== Spawn extension (internal, , 2) exited non-zero on
'SIP/mileslap-569b'
the only thing I could find was this:
http://72.1
Hi,
when I try to use meetme I always hear this error message
"this is not a valid conference number, please try again",
but my configuration seems to be correct... Here it is:
-- extensions.conf --
exten => 6000,1,MeetMe(1234,ciMp) ; entra nella meetme room 1234
-- meetme.conf --
[rooms]
conf =
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 m
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
What is the interest to supply binary of Asterisk, under debian for example,
while to use MeetMe it is necessary to COMPILE Asterisk ??? :-))
Mickael.
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Hi guys,
i'm trying to use the "featuremap" of features.conf inside the app meetme,
but it's no working.
like:
_5XXX => {
Set(DYNAMIC_FEATURES=toca_macaco);
MeetMe(${EXTEN},F); //F forces the meetme to pass DTMF
Hangup();
};
in features.conf:
toca_macaco => 123, peer
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
MSN:flavi
Just curious if anyone has ran into this problem before, and if so if they
were able to work around it. When I dial into a MeetMe conference via an
IAX connection, my call is dropped if I enter an invalid meeting ID. I am
first played the greeting conf-invalid, then the call is dropped. If I
cal
I noticed in the source for app_meetme.c,
that there is an arg to allow someone to be an admin.
What exactly does that mean? What extra privileges do they gain? And
furthermore, how is one made an admin in a room?
Ok.. I got * and SIP working internally now .. still wrestling with
connecting two remote * pbx's together.. I'll save that for another
day though :)
I setup Meetme on this new * PBX, but when I try to dial to join the
conference,
I hear a recording saying the conference is invalid or something to
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Panny
> Malialis
> Sent: Wednesday, October 22, 2003 11:43 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Meetme
>
>
> Is app_meetme broken?
>
> I seem to get in
Panny Malialis wrote:
Is app_meetme broken?
I seem to get invalid conference number all the time :(
You have to have some Zaptel device installed. Like wcfxo or ztdummy
Jeremy McNamara
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http:
On Wed, 22 Oct 2003 17:43:08 +0100, Panny Malialis <[EMAIL PROTECTED]>
wrote:
Is app_meetme broken?
I seem to get invalid conference number all the time :(
(Is there a FAQ for this yet?)
1) Did it work before? If so, what changed?
2) Do you have zaptel hardware installed and working, or the ztd
Jeremy McNamara wrote:
Panny Malialis wrote:
Is app_meetme broken?
I seem to get invalid conference number all the time :(
You have to have some Zaptel device installed. Like wcfxo or ztdummy
See
http://www.voip-info.org/tiki-index.php?page=Asterisk+timer
for more information!
/O
__
03 6:47 PM
Subject: Re: [Asterisk-Users] Meetme
> Jeremy McNamara wrote:
>
> > Panny Malialis wrote:
> >
> >> Is app_meetme broken?
> >>
> >> I seem to get invalid conference number all the time :(
>
> > You have to have some Zaptel dev
Do you have ztdummy or zaptel device in your system?
- Original Message -
From: "Panny Malialis" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, October 22, 2003 1:43 PM
Subject: [Asterisk-Users] Meetme
> Is app_meetme broken?
>
> I seem t
Yes.
Tim Thompson
http://www.amatechtel.com
(806) 722-2227
-Original Message-
From: CW_ASN - Gus [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 22, 2003 1:12 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Meetme
Do you have ztdummy or zaptel device in your system
Hi guys,
Having a bit of a problem trying to get conference bridges working. In my
meetme.conf file I have the following line
[rooms]
conf => 6000
In my extensions.conf file I have:
exten => 1000,1,MeetMe,6000
My problem is that when I dial into extension 1000 it is telling me "this
is not a
Hi,
Can anybody explain me in configuring Asterisk to record a conference?
Regards...
Girish
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___
Hi All,
Earlier today I tried using my new * server to host a simple 3 party
conference in a MeetMe conf room. I was on a snom 200, one other party
called inbound from PSTN via X100p, while the third party was a call
that I placed outbound via VoicePulse Connect. I transfered both
external parties
Hi all,
i have a setup with chatrooms, several MeetMe conferences wich users can
change inbetween. 10 users maximum in each room.
It seems like when i have more than 40-45 users on the system at the same
time asterisk drops abt 20 and continnues buisness as usual.
Is there anyone else who have
On Thu, 2004-01-15 at 19:18, [EMAIL PROTECTED] wrote:
>> I do not have any zaptel hardware on the Asterisk box, I could not have
>> meetme functioning. I did modify the Makefile in zaptel directory on
>> line 168 by including ztdummy as one of the modules to compile in.
try modprobe ztdummy
This
I'm looking into deploying * for an internal conference call server (using
MeetMe) and had a couple of quick questions for those of you who have used
it. I checked the Wiki but there weren't a lot of details for MeetMe.
- Can you limit the size of a conference "room", ie max 8 people, etc.
- Is t
I am trying to setup MeetMe conference.
In my MeetMe.conf file I have
[rooms]
conf => 4001,4001
In my extension.conf file I have the following:
exten =>4001,1,MeetMe(4001|p|4001)
When I try to call the extension 4001 it gives me the following error message. I am
using SIP and I have not created
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
***
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
Hi All,
Is it possible to specify the context of a meetme conference under 1.4.x?
By default all meeting rooms are generated under the context rooms, I would
like to use other contexts depending on what extension number is used to
call the meetme application.
If it is possible can some
Hi
I have reading the voiip side i found some document says
"
The conference bridge runs Ulaw codec by default. If you let people connect
with GSM or other codecs, Asterisk will use CPU power to convert audio
between codecs "
iam using vicidial and meetme for callcenter application. iam geting
Hi
iam using 1.2.17
does any one have information meetme in realtime
and store in mysql i dont see any document
could some one help me
is this possible ?
ram
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I recently installed 1.4.5 and I've noticed a recurrence of a problem that
I thought was solved long ago, namely a very long (2-4 seconds) delay on
meetme calls. That means with two people in the conference room, it takes
2-4 seconds for what one person says to reach the other person.
Is anyone e
hi guys:
recently I want to buinding a meeting confence on asterisk and use the meetme
application.
I have a ztdummy kernel
afteri the lsmod commond:
ztdummy 5768 0
zaptel182660 28 zttranscode,ztdummy
crc_ccitt 3008 1 zaptel
I al
Hello,
I've a monitoring problem with app_meetme,
I'd like to record a zap channel, which goes to a meetme conference
Monitor doesn't record the voice of another members in the conference.
Thanks any help
Tamas
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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
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
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
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
--
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:
> 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!
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
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
-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
> 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:
>> 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
On Mon, 18 May 2009, Jeff LaCoursiere wrote:
>
> I know I should probably post this to the Trixbox forums, but am hoping
> someone might have a quick answer here for me. Client with Trixbox 2.6.2
> with a recompiled asterisk 1.4.23 and various patches for RPID. A meetme
> conference with seve
Hi All,
I am looking for an option in Meetme or similar which will enable to
skip to next priority (a voicemail) if the person in Meetme conference
is alone and if he is there for some time (say 3 minutes)? Any hints on
this? Thanks in advance.
Regards,
--
Kurian Mathew Thayil.
(GPG KeyID: E2323
Hi People, I want to do the following steps:
- Create a meetme between 2 persons.
- First, 1 person (user1) is entered into the meetme.
- Second, user2 is entered into the meetme. User2 is the marked user and also
he is able to exit the conference by pressing #.
- If user2 exited by pressing
I've got hints setup for my MeetMe conferences like so:
exten => _60X,hint,MeetMe:${EXTEN}
and they show up in "core show hints" like so
6...@dialtone: MeetMe:600State:Unavailable
Watchers 1
_...@dialtone: MeetMe:${EXTEN} State:Unavailabl
Hello
In app_meetme.c, there are two configuration file loaded i.e. meetme.conf
and sla.conf..
I want to know that if i removed whole code of sla_* and sla.conf from
app_meetme.c file..
Is this create problem for MeetMe application and register action/event...
--
Regards,
Chandrakant Solanki
ser presses 6 and return to the conference.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of robert
boardman
Sent: Thursday, November 19, 2009 8:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [aster
Hi,
anybody who has a complete list of variables used by meetme conferencing
application in asterisk, plz share.
--
Regards
Rizwan Hisham
Software Engineer
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Hi,
I'm experimenting with the Meetme feature of Asterisk 1.2,
exten => 2095,1,MeetMe(|Ds)
This almost gives me what I want, where each employee can create their own
on-the-fly conferences with a personal Conference Number and PIN. However, as
the PIN is actually set by the first callee, the
Hi! i have an error using the meetme aplication, and just dont work..
my meetme.conf is:
[rooms]
conf = 700
i calling from a sip phone, the extension number is 600. there is the error:
Executing [EMAIL PROTECTED]:1] MeetMe("SIP/600-09111e58",
"700|MI") in new stack
WARNING[20055]: channel.c:302
I am playing around with dynamic meetme conferences, and wanted to have
one person constantly in the conference, with calls "popping in and out".
Is there an option / any way of playing enter / leave sounds to the
person who created the conference only, and not the people leaving /
joining ?
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, Mee
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 th
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
--
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
Pablo Endres wrote:
> Hi people,
>
> I have a user that forgets to hangup his conference calls, so they go
> on forever. Is there a way of limiting the duration of a conf call?
>
> Thanks in advance,
>
> Pablo
Try using ABSOLUTETIMEOUT before starting the conference?
*CLI> help application AbsoluteTimeout
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essage -
From:
Travis Conway
To: [EMAIL PROTECTED]
Sent: Friday, August 06, 2004 3:53
PM
Subject: [Asterisk-Users] meetme
I am trying to get a simple MeetMe running
between a few SIP phones here in our office.
Here is a clip from my
extensions.conf
Title: Message
Hi
all,
I had
a chance to use some call conferences that had some very neat
functionalities:
- When
you call you are first asked for your name
- When
someone joins the conference a message " is now joining the
conference." is played.
- When
someone leaves the room a messa
hi;
is MeetMe a SIP client? For example if i join a conference room can i
leave the conference room and REFER someone else to join the room?
thanks
M. Smadi
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Hey folks,
Using FreeBSD 5.2.1 and I've got the current zaptel driver installed
from ports (0.8_1) and current ports asterisk (1.0.1). I've set
options HZ=1000 in my kernel config, recompiled and rebooted and as far
as I can tell, I've done everything right but when I try to use the
conference
On Sun, 2 Jan 2005, Serge Schumacher wrote:
> Can it be that the MeetMe application is not installed by default even if
> there is a meetme.conf ?
>
> pbx.c:1280 pbx_extension_helper: No application 'MeetMe' for extension
> (from-sip, 550, 4)
The meetme application is built if the make script fi
Serge Schumacher wrote:
Can it be that the MeetMe application is not installed by default even
if there is a meetme.conf ?
pbx.c:1280 pbx_extension_helper: No application 'MeetMe' for extension
(from-sip, 550, 4)
It is not installed if you haven't got a Zaptel timer. See the Wiki docs
on
Serge Schumacher wrote:
Can it be that the MeetMe application is not installed by default even if
there is a meetme.conf ?
pbx.c:1280 pbx_extension_helper: No application 'MeetMe' for extension
(from-sip, 550, 4)
If you don't have zaptel installed Astrisk won't build Meetme.
Try making sure that the numbers match the one in meetme.conf.
I don't know if this makes a difference but since in my config this is
how it is setup and it works, just give it a try.
On Mon, 3 Jan 2005 00:34:25 +0100, Serge Schumacher <[EMAIL PROTECTED]> wrote:
> Hi,
>
> Can someone see what's
Can someone see what's wrong here please ?
exten => 550,1,Answer
exten => 550,2,Wait(1)
exten => 550,4,MeetMe(18|Md)
exten => 550,5,Hangup
The priority 3 is missing ... 1, 2 then timeout...
--
Joel Vandal
ScopServ Inc.
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As
Serge Schumacher wrote:
exten => 550,1,Answer
exten => 550,2,Wait(1)
exten => 550,4,MeetMe(18|Md)
exten => 550,5,Hangup
and when I call 550 I get this error and the MusicOnHold (exten =>
550,4,MeetMe(18|Md)) also doesn't work:
-- Executing Answer("SIP/ses-0730", "") in new stack
-- Execut
How stupid, thanks a lot
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joel Vandal
Sent: lundi 3 janvier 2005 01:00
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Meetme
> Can someone see what's wr
How stupid, thanks a lot
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
aka ManxPower
Sent: lundi 3 janvier 2005 01:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Meetme
Serge Schumacher wrote
Hi,
I'm trying to detect and or log the duration a a conference (Meetme). I
need it in order to do some billing for theses services.
Any ideas on how to do it?
I googled around but found nothing.
Thanks in advance
epablo
--
Pablo Endres <[EMAIL PROTECTED]>
ComVoz Communications
USA:
Hi there,
I know this question is kind of stupid. But, I don't know anywhere
else to ask. I've received some answers when I asked about the need of having
a zaptel interface to make the meetme application work, that said that it was
better to have a real hardware then the zaptelrtc s
Has anyone testing the maximum limitation for people in a Meetme
conference?
-Brian
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I'm running into a bit of a problem setting up conference calls. The box
I rent at a colo doesn't seem to have USB hardware When I try to load
usb-uhci I receive a "device does not exist" error. Which means I can't
load ztdummy
The system has a rtc clock module, so zaprtc won't work... (
I'm running into a bit of a problem setting up conference calls. The box
I rent at a colo doesn't seem to have USB hardware When I try to load
usb-uhci I receive a "device does not exist" error. Which means I can't
load ztdummy
The system has a rtc clock module, so zaprtc won't work... (
Nitesh Divecha wrote:
Hey All,
Just finished installing Asterisk and configured all the necessary
parameters to start.
I can’t seem to find the Meetme application in my asterisk directory.
I downloaded asterisk from CVS and installed it and all my Snom phones
are working and voicemail too.
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 pr
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: Ill
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: Illegal s
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
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