Interesting. Are you using the included cbend.php script to terminate
conferences?
I occasionally get questions about using WMM with Confbridge, and to date I have
not had an answer .
If you can provide details, even vague ones, about how you did it, I can update
the
WMM package.
Dan
Patrick Lists wrote:
On 16-01-14 21:37, Gergely Kiss wrote:
Dear List,
I'm about to build an Asterisk 11.7 based PBX from scratch for our
company. I'm in the middle of the planning phase and it turned out that
our VoIP provider prefers H.323 protocol for handling voice calls (while
SIP is
Look at Web-MeetMe ( http://sf.net/projects/web-meetme )
If you are on Asterisk 1.6.7 or later you have access to RealTime
MeetMe conference storage, otherwise you need to use a
script and Asterisk application included with the WMM download.
Dan
From: asterisk-users-boun...@lists.digium.com
This list was accurate up to and including Asterisk 11
[0] = Caller #
[1] = Callerid Number
[2] = Callerid Name
[3] = Channel:
[4] = 1 for Admin, NULL for User
[5] = 1 for Monitor, Null otherwise
[6] = 1 for Muted, NULL for UnMuted
[7] = 1 for Resquests Floor, 0
initially muted users in the request to talk queue?
The provision of this parameter in the meet-me source indicates this is
doable... but I am unable to find an appropriate way to do it.
Any hints would be great help.
On Thu, Aug 15, 2013 at 11:03 PM, Dan Austin
dan_aus
Thiago wrote:
I'm trying to limit the number of participants in a conference room
with the realtime option maxusers, but it doesn't work.
Asterisk version?
Any error messages?
Is the conference you are attempting to limit stored in a db (Realtime)?
Dan
--
Rohit Mahajan wrote:
Matt Riddell lists at venturevoip.com writes:
Are you using the latest version of the app_cbmysql?
It looks like it needs to be updated for the latest version.
Alternatively it may say somewhere on their website which version of
Asterisk this works with?
I have
Edwin wrote:
i recently setup an Asterisk system in Hong Kong. their phone
company told me that their T1 PRI switch type is Primary-NTT.
however in chan_dahdi.conf there's no such option. i have it
set to national. it worked fine for a while, but now suddenly
stop working. in coming call
Giuseppe wrote:
Yes, but i think that's better to open an LDAP connection with
extensions user and password. Or not?
Better is not the right way to look at it. You questions is
about early or late binding. Early binding requires a dedicated
username and password to connect to LDAP before it
Daniel wrote:
Hi Group,
is in MeetMe any option to identify the own number (from the view of a
caller)?
I would like to write an option to set on the telephone an request for voice,
if the room is muted. That request should display on our Conference Control
Website and an Admin
Kevin P. Fleming wrote:
This is a valid point, and we'll get this corrected. Our package
repository should have packages for Asterisk 10, but it doesn't.
How likely is it that a Centos 6 repo might be setup at the same time?
--
Tony wrote:
Kevin P. Fleming kpflem...@digium.com wrote:
As I said before... an Ethernet cable will work nearly all the time, and
at a 5m length it's probably fine.
Kevin, under what circumstances would an Ethernet cable potentially not
work with T1/E1? And in those circumstances, what
Eyal Mahalal wrote:
I create Chat room with MEETME and now I have a problem.
I want that the host of the room could identify the participants in the room
by their
speech, so that if a participant uses language the host could kick him from
the room.
Is there a way to do it?
This is one
not find chan_sccp.
Maybe that is the reason why I do not have the sccp.conf file?
So, using the sccp channel, will also face the same problem that the phones
will restarted if I did reload?
Regards
Bilal
--- On Mon, 6/20/11, Dan Austin dan_aus...@phoenix.com wrote:
From: Dan Austin dan_aus
The Asterisk version is 1.8.3.2
The Cisco IP Phone is 7942G and it is running now skinny.
The used TFTP is tftp-server which is installed in fedora.
I placed the following two files (but look like it was not taken from the
TFTP, as
nothing appeared in the messages), but I am able to to
Richard wrote:
No, conference scheduling is not a feature that we have built
directly into ConfBridge, and I'm debating on what it would look
like.
Scheduling isn't built into MeetMe either, but the fact that it
dynamically reads from a database means that you can write external
programs
Kevin wrote:
On 03/21/2011 06:49 PM, Dan Austin wrote:
I just finished a fresh install of 1.8.3.2 at home using the packages
Digium hosts.
After correcting a number of typo/config'o error that had crept in
over the years, I thought I had everything working.
My wife just complained that she
I just finished a fresh install of 1.8.3.2 at home using the packages
Digium hosts.
After correcting a number of typo/config'o error that had crept in
over the years, I thought I had everything working.
My wife just complained that she cannot call her mother (who is using an
old IAX hardphone I
Manmohan wrote:
I am really not sure if this is related to the meetme in asterisk OR
this is something to do in web-meetme. I tried to google but didnt get
any proper results.
I am facing one issue in Web-meetme on the expiry of any conference
that we create.
I am having Web-meetme 4.0.2
Manmohan wrote:
I am currently on Asterisk 1.6.2.14.
Do you have schedule=yes in meetme.conf? I incorrectly
remembered/thought that all of the Realtime features were
controlled by that option, only a small number, such as
end times and CDR logging
On Sat, Dec 4, 2010 at 12:09 AM, Dan Austin
/libxml2.so.2...(no debugging symbols
found)...done.
Loaded symbols for /usr/lib/libxml2.so.2
Reading symbols from /usr/lib/libz.so.1...(no debugging symbols found)...done.
Loaded symbols for /usr/lib/libz.so.1
Thanks Regards
Manmohan Singh
On Sat, Dec 4, 2010 at 1:12 AM, Dan Austin dan_aus
I'm running 1.6.2.13 and need to record a small number of custom
values use cdr_odbc and cdr_adaptive_odbc, and only the custom
fields.
The good news is that the custom records are being stored in the
database as desired. The bad news is that I get three sets of
warnings/notice about 'SQL Exec
Manmohan wrote:
I commented locale.php in defines.php and it perfectly worked well.
Now i am wondering what is this invite participants for, while adding
conference. wherein it asks for first name, lastname, emailaddress
telephone number..
The 'Invite Others' option is mostly for installs
Manmohan wrote:
I had tried the new version of webmeetme i.e., 4.0.2
The recording works very well.
Great!
I see following php errors whenever i try to add in conference.
[Wed Aug 04 16:29:02 2010] [error] [client 32.93.17.40] PHP Notice:
Undefined variable: order in
Manmohan wrote:
I did added the record option in user options as well.
$Mod_Options = array(array(_(Announce), I), array(_(Record), r));
$User_Options = array(array(_(Announce), I), array(_(Listen Only),
m), array(_(Wait for Leader), w),
array(_(Record), r));
And the gre8 news is, it
Manmohan wrote:
There was on very silly mistake and i missed to check that properly. Really
apologize for that.
Following change was done to get the conf-recording into the proper path:
chown -R asterisk:asterisk /var/lib/asterisk/sounds/conf-recordings
following is the output:
Manmohan wrote:
Following is the output for core set verbose 5,
also i am really not able to get on the admin pin
thing? Do you mean, that with admin pin configured
we cant use recording?
You are actually running a version that has been fixed
to support recording with pin-less or user pins.
Manmohan wrote:
I can see the path does exists but i cant see any recordings
happening inn there. There are no files in it
Following is the output:
/var/lib/asterisk/sounds
drwxrwxrwx 2 asterisk apache 4096 Jun 27 20:54 conf-recordings
I hope m understandly this correctly but m sure m
Jonathan wrote:
I've managed to acquire a few Cisco handsets (7905, 7920)
and would like to use them with Asterisk.
Rather than simply switching to the SIP firmware I thought
I'd use these with chan_skinny - partly because this is
Cisco's primary firmware and therefore the phones might be
Manmohan Singh Jandu wrote:
Excellent!
I finally got it working, it was ODBC drivers issue
actually. Installed the proper compatible version and its working.
I thought that might be the case.
There are still few errors which i see on asterisk console:
[Jul 19 13:58:51] WARNING[30213]:
Manmohan Singh Jandu wrote:
OK, now i added the column members in the table booking manually.
and disabled selinux to have this working.
Now i am struggling with recording option in webmeetme.
Not sure on how to enable it, though m checking the checkbox
while creating the conference. But
Jonathan wrote:
On 26 July 2010 23:50, Dan Austin dan_aus...@phoenix.com wrote:
I'll dig around in my archives to see if I can find my old patches
for either of these.
Many thanks - I'm happy to test patches if I can do so. At least I
can contribute in that way, even if I'm not directly
Manmohan wrote:
Unfortunately m not able to get rid of the below mentioned errors. not sure
on where i am missing now.
On Sat, Jul 10, 2010 at 9:41 AM, Manmohan Singh Jandu manmoha...@gmail.com
wrote:
Ahh here is the catch i was still using app_cbmysql for this.
now i had
Manmohan wrote:
My Web-MeetMe_v4.0.1, i followed the instructions in the
README File in the same package.
Good. There are other instruction packages, but since I wrote
the README it is the one I am most familiar with.
Are you using RealTime enabled app_meetme or app_cbmysql
from the WMM
Manmohan wrote:
I was looking for audio conferencing solution where i got Web-meetme.
I had installed Asterisk 1.6.2.9 on Centos 5.4. Its perfecting working
fine. I tried using Meetme even meetme app is working perfectly fine.
I installed Webmeetme 4.0 and integrated with my asterisk. When i
Matt wrote:
On 1/09/09 5:53 PM, Glen wrote:
Matt Riddell wrote:
In the latest readme for WebMeetMe (3.1.0) it states:
* Compile and install CBMySQL
App_cbmysql is now included in the web-meetme package,
located in ./cbmysql. To install just run make; make install
Copy the sample
Make sure you are stripping the 8 on inbound calls to that H323 gateway
under CCM and that it has a valid search space to find your extensions...
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy Ezell
Sent: Tuesday, June 09, 2009
Sean wrote:
Tilghman Lesher wrote:
On Saturday 16 May 2009 08:21:43 sean darcy wrote:
With 1.6.1, I'm trying to set up a test of meetme for creating dynamic
conferences.
Trimmed
I don't want the conference to stay up forever, since I'd like new pin's
each time.
This should be a common
Jimmy wrote:
Second Call out the asterisk console looks like
this-:
-- Executing [92952...@internal:1] Dial(SIP/222-09ab3588,
SIP/Cisco1760/2952210) in new stack
-- Called Cisco1760/2952210
[Apr 22 16:08:58] NOTICE[3450]:
Jimmy wrote:
Dan thank you, yes that seems to help. It looks like the
bridging is happening now and I see the light come on in
the second FXO port, but then I get a busy signal after
that and the call still does not complete. If I set the
second line as priority 1 it completes the first
Shocky wrote:
This is probably outside what Asterisk is intended for, but I'm hoping it can
help.
I need to make and receive calls through a Cisco Call Manager server that I
have no control over. I have to use a Cisco soft phone (Cisco IP
Communicator), which only runs on Windows. But I'm
Gordon wrote:
There are other more advanced things you can do with iptables which I've
been looking at - but the esence is to count/time new connections to a
particular service from each IP address and if more connections per unit
of time happen, then apply a temporary block for a bigger
Steve wrote:
Speaking from T1 PRI and E1 PRI in West Africa, you tell the telco how many
digits to send. Often times, at least in my experience, if not specified, they
will only send the last four providing there are no conflicts.
They should be able to send however many digits you require,
This has been on my ToDo list far too long.
I have a small call-center setup, with basic
time of day/day of week validation before putting
callers in the queues.
With the holidays upon us, I need to add check to
see if 'today' is a holiday so I do not put callers
in unmanned queues. Due to how
Tilghman wrote:
Astdb is a nice idea. Something along the lines of:
GotoIf(0${DB(holiday/${STRFTIME(,,%Y-%m-%d)})}?holiday,s,1)
would work. Holidays are evaluated as 01, which is true.
Anything not in the database would be evaluated as 0, which
is false. This will work both for holidays
Sergey wrote:
Sorry if I'll be very very stupid but really I write to
this conference first. I have problems with configuration
of app_meetme in realtime environment. I use last stable
release of asterisk 1.6.0.3
trimmed db table definition
The issue is not in the database, but a problem
Noah wrote:
I found the maxusers defined in meetme.c, but I'm
not sure how this value is set. Does anybody know
if one can limit the number of users permitted in a
meetme conference? I know there's MeetmeCount(), but
I'd rather avoid the dialplan logic and just set
maxusers instead.
That
Yehavi wrote:
Our university has to upgrade soon its old Nortel PBX's
which holds around 10,000 extensions tied to 5 PBXes. Up
to now we thought about commercial solutions but now
there is a window openning for open source solution.
However, I need examples to convince that this solution
is
Bill wrote:
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?
Can you describe the effect? I am seeing odd behavior
when I have PSTN calls in a conference, oddly
John Todd wrote:
There was discussion recently (on -dev? on -users?
on IRC?) about how there are some shortcomings on RTP
packetization/transcoding. It appears, though I have
not confirmed this, that trying to move a 20ms G.711
stream from a client, though Asterisk, to a remote
gateway
I have just setup a small queue implementation for one
of my branch offices, replacing a 16 year old key system
that had a hacked together pseudo call queuing feature.
The 'agents' are not dedicated to the queues and want to
be able to logon and get one call only from the queue.
I know this is
); long enough for a call to be delivered
exten = 133,n,RemoveQueueMember($member)
I was hoping that someone might have a more elegant solution.
Dan Austin wrote:
I have just setup a small queue implementation for one
of my branch offices, replacing a 16 year old key system
that had a hacked
This release primarily focuses on security.
A number of problems involving SQL injection
and XSS were identified and reported by Jean-Michel
Besnard.
Jean-Michel was kind enough to help with the testing
as each vulnerability was addressed.
The new release is available in the downloads section
John wrote:
Thanks Steve for your suggestions.
In China you will generally get either MFC/R2 or EuroISDN. MFC/R2 is
much more common.
This is exactly my current problem.
NETCOM in Shanghai just told my local contact it is an E1 and that's it.
I have no idea whether it is MFC/R2 or
is installed, but has a problem
Set verbose to 5 and try *CLI load app_cbmysql.so
The output will tell us if the module does not exist, or
why it cannot be loaded.
Dan
Asterisk 1.4 and Meetme is the latest version 3.0,
ztdummy is working fine.
Thanks
On Thu, Jun 26, 2008 at 6:48 PM, Dan Austin
Ali wrote:
I followed this howto
http://www.voip-info.org/wiki/view/MeetMe-Web-Control
and
http://asteriskpr.blogspot.com/2005/09/guide-to-install-web-meetm_112614171575673316.html
to install web meetme with asterisk, I know the meetme
module is included however I need to be able to ban
Aaron wrote:
Okay, another Cisco related issue (sorry!).
Single Asterisk box at location 1.
Single Cisco box at location 2, however the Cisco is
also the PBX for location 3 (same physical machine, calls
routed via VoIP).
Trying to have Asterisk be able to call EITHER Call Manager
Mojo wrote:
[EMAIL PROTECTED] wrote:
I am planning to write a module to find if a Special Information was
detected or not.
Can anyone please help me to figure out the below fields?
1. The Frequency of a frame
2. Length of frame in milliseconds
Aren't all the frames in asterisk 20ms long,
Tony wrote:
Has anyone here any experience in getting an Asterisk
box to talk to a Cisco Unity system? I have a
potential customer who would like to add a conference
bridge to their existing Cisco Unity setup.
The digging I have done so far suggests that it should
be possible to talk SIP
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 meetme
Tzafrir wrote:
On Sun, Dec 09, 2007 at 03:30:13PM -0500, Michelle Dupuis wrote:
Well, we can already integrate to major platforms via SMTP. The real
value
is in deep integration to the most popular email platform in
business:
Exchange.
I would love to see smart Exchange integration, where
Paul wrote:
I have six cisco 7911g connected on asterisk over
chan_skinny. Four of them are working OK. two of
them even the screen on the phone is indicating that
is registered and has number loose connection to
asterisk . On asterisk the message is Skinny Client
was lost,
Paul wrote:
Thank you for your answer. I am using asterisk
1.4.13 and keepalive has a value of 120 in
skinny.conf.
You can try reducing the keepAlive. The phone
will still loose registration, but will re-register
faster. Other than that, I would look at the
health of your network,
Jordi wrote:
Any one have experience with this CISCO Wireless
IP phone running with Asterisk??
It doesn't support SIP protocol I believe, so I need
to know if the skinny channel can work with the 7921.
The 7921 works fine with SVN trunk, and I think the
trivial changes required to support
Lacy's response in the thread 'Why does
everyone seem to dislike *now?', has a small
bit that caught my eye.
Chan_Skinny made a lot of progress between 1.2 and
1.4, and even more in the later 1.4.X releases.
I am curious as to which features/functions that
chan_skinny might be lacking compared
Shane wrote:
I don't think that Asterisk currently sends a
remote-party-id to the called party. That would
proably have to be added to the sip channel.
It *does* work with Broadworks, another SIP based
phone system.
On a phone registered to Broadworks:
Your phone invites the
Replying to myself:
Shane wrote:
I don't think that Asterisk currently sends a
remote-party-id to the called party. That would
proably have to be added to the sip channel.
It *does* work with Broadworks, another SIP based
phone system.
On a phone registered to Broadworks:
Your
Shawn wrote:
I'm having a wierd problem with a Cisco 7960 (sccp2)
and asterisk (1.4.2)
If the call that I'm trying to make goes through,
everything works fine. But if there's any sort of
error (like me messing around in my extensions.conf,
etc). I can't get the connection to drop. ie:
Jason wrote:
Dan Austin wrote:
Shawn wrote:
I'm having a wierd problem with a Cisco 7960 (sccp2)
and asterisk (1.4.2)
If the call that I'm trying to make goes through,
everything works fine. But if there's any sort of
error (like me messing around in my extensions.conf,
etc). I can't
Dovid wrote:
Does anyone know if it is possible to change the
packetization time in Asterisk ? I was told by a client
of mine that adjusting this with using G729 can greatly
lower the amount of bandwidth used.
Your client is correct. Configurable packetization was added
introduced
Tom Wrote:
Hi all,
We have recently implemented an Asterisk system using Trixbox
(asterisk v1.4.4 at the moment, yet to move to 1.4.9) but are
getting pressure to switch back to our old key system unless
we fix two major issues. So please help me avoid switching back!
Have you tried
Something changed in the final linking of the channel, and it now
produces libchan_h323.1.0.1 instead of libchan_h323.so.1.0.1
Either edit the Makefile to copy libchan_h323.1.0.1, or manually
copy that file...
Dan
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Keshav
David Wrote:
On Thu, 2007-07-05 at 16:42 -0600, Stephen Bosch wrote:
David Boyd wrote:
I seem to remember that the wan Pipeline units supported BRI, and
also
provided a couple of analog phone jacks. I will dig around in the
basement and try to find the one that I had, if I find it, who
Greg wrote:
So, if you ever use a Cisco SIP Phone with an Asterisk
server, it's not possible to localize menus, soft
keys, and so on ?
Not unless someone wants to add support for it in the SIP
channel, which I doubt. I would be more than willing to
provide the SIP messages that a
is lacking.
Is it as simple as add the SQL table and placing a meetme family in the
extconfig.conf?
It also looks like Dan Austin at phoenix dot com was working on a
scheduler for this. Any news on that?
___
--Bandwidth and Colocation provided by Easynews.com
Alex wrote:
I tried with the ping ... all of the phones respond
in ca. 0.3ms, so network seems to be OK. More than
90% of CPU on * box is idle even in peak times, so
this shouldn't cause echoes either, right? Hmmm, so
handset could be an issue, but did anyone ever
experience any handset
Ondrej wrote:
I finally got some time to test the SVN branches and
here are my comments:
Cool.
One thing that does not work for sure - I had some problems to
terminate the running conference from within the web page - I
just clicked the button and nothing happened.
This is likely a
Yehavi wrote:
I am trying to apply the called party identification
patch (patch 8824) and managed to make it work with a
static data. Where do I take the name of the called person
(the equivalent of CALLERID, but the other way...)?
Short answer is that you cannot.
Longer answer is that it
Chris wrote:
It seems that more and more phones these days are
coming with XML mini-browsers. I'd like to have a
go at developing something useful to use on them,
but in all honesty, most of our customers use their
phones to make and take calls and very little else.
So I'm open to
Basic bug fix releases
Both have updates to app_cbmysql to be thread-safe,
reconnect to the database in case of timeout and to
detect missing/mis-configured conference app/conference
participant counting apps.
The last one has caused Asterisk to crash. Now
If it does not find MeetMe or
Alex wrote:
It seems to me that what you are really talking about
is manipulating the display features of the phone.
Caller ID is unlikely to have this effect as the phone
does not consider the From: URI in the SIP header unless
the call is of an incoming nature.
This feature is often
Andrew wrote:
On Saturday 28 April 2007 11:22 am, Chris Bagnall wrote:
Thanks to all who replied to my thread a few days ago SIP devices
with
packet loss tolerance. One of the suggestions that came out of that
thread
was to replace routers at users' premises with ones that support QoS.
The latest zaptel release has a bug that can cause
segfaults. Did you upgrade zaptel at the same time?
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thomas
Kenyon
Sent: Wednesday, April 25, 2007 9:58 AM
To: Asterisk Users Mailing List -
Bill Wrote:
On Wed, 25 Apr 2007 12:18:10 -0500, The Asterisk Development Team
wrote
The Asterisk.org development team has released Asterisk-addons
version
1.4.1.
When I run make install I get:
[EMAIL PROTECTED] asterisk-ooh323c]# make install
cp .libs/libchan_h323.so.1.0.1
Grigoriy wrote:
I'm trying to connect Asterisk 1.4 and Cisco CallManager 5
using SIP Trunk without MTP (media termination point).
Howerver, Cisco 79xx phones do not support RFC2833, they
always notify CCM5 via SKINNY channel no matter where they
send RTP to.
If you are running the phone
Grigoriy wrote:
Dan Austin wrote:
If you are running the phone loads that shipped with CCM5,
then your skinny phones have 'support' for RFC2833. CCM
figures out during the call if the call will traverse a
SIP trunk and instruct the phone to use RFC2833 for DTMF
I have a CCM5-Asterisk trunk
Ondrej wrote:
What version of Asterisk are you using? I've had recording
working with SVN before 1.4, the 1.4 betas and currently 1.4.1.
*** Update ***
Recordings are tied to a moderator joining the conference at this
time. I may need to change that based on feedback/requests to
do so.
Ondrej wrote:
Ok, I understand that now as well - you click that button
and thunderbird should popup with the mail composer open,
right?
Yes.
Does not happen to me - most likely problem w/ my firefox
settings.
Browser security settings most likely
Now it all make a sense, sorry for
Ondrej wrote:
Ok I had a chance to test web-meetme 3.0.1 and I have few
comments here -
the Makefile for CBmysql lacks procedure that verifies existence of
/var/lib/asterisk/sounds/conf-recordings directory where the
conference
records should reside.
You are right that this should be
Ondrej wrote:
Ok I had a chance to test web-meetme 3.0.1 and I have few
comments here -
the Makefile for CBmysql lacks procedure that verifies existence
of /var/lib/asterisk/sounds/conf-recordings directory where the
conference records should reside.
You are right that this should be
Wooi wrote:
I have the similar problem on 1.4.1. I don't remember
having it in 1.4.0, I could be wrong. I have a SIP
provider, when calls come in, it play MOH while waiting
for to be picked up. ztdummy is loaded.
Another interesting thing I notice,
exten =
Matt wrote:
To my knowledge, Asterisk's packetization rate is hard
coded at 30ms. If I wanted to, where in the code could
I go to change it to 20ms. Is there anything bad that
might happen if I change it (asterisk related)?
You don't mention what version you are using, but 1.4 does
Minor bug-fix release, no new functionality.
Bugs fixed:
* app_cbmysql would fail to load
* Incorrect handling of recurring conferences that
spanned a DST transition
Minor cleanup:
* A couple image files were duplicated with
both upper
Ray wrote:
Could anybody give me an authoritative answer on whether
Asterisk can support T.38 pass-through when the clients
are behind NAT? We have Asterisk servicing clients behind
NAT (with nat=route, canreinvite=no) and would love to get
T.38 going but have had no luck so far. The
Ma write:
HI, today I download Web-MeetMe-3.0.0 for asterisk
1.4.0 but when I call the extension which invoke
cbmysql, a warning appears:
What version of Asterisk? I ask because I have had
Reports of problems against svn trunk and svn branches
after 1.4.0 was released.
WARNING[20225]
Ma wrote:
WARNING[20225] pbx.c: No application 'CBMysql'
for extension (default, 1995, 3)
I check the application, it didn't registered
CLI core show application CBMySQL
Your application(s) is (are) not registered
But I can see it use show module
I made a small mess of
Pavel wrote:
I prefer h323 included in asterisk tree,
I have caller id issues with ooh323 and nobody
answer to bugreports oh323 from inaccessible
network is unmaintained/death project, incompatible
with asterisk 1.4.
PJ
Response to ooh323c bugs is very slow, and patches can
take some time
Buki wrote:
Sorry I forgot to change the subject line in my last posting!
I have been using the Web Meet-Me 2.1.0 on Asterisk 1.2.12.1
for many months now and I am a big fan and I have been very
happy with it.
I'm glad it's working well for you, positive feedback is always
welcome.
I
Rob wrote:
On 1/5/07, Dan Austin [EMAIL PROTECTED] wrote:
Trunk has already moved on and code compatible with 1.4, may have
problems on it. For a sanity check, I wiped out my test system
and rebuilt it with fresh components for 1.4 (libpri, zaptel,
asterisk,
asterisk-addons), and I
*CLI core set verbose 10
Verbosity was 0 and is now 10
*CLI module unload app_cbmysql.so
Unable to unload resource app_cbmysql.so
Command 'module unload app_cbmysql.so' failed.
*CLI [Jan 5 11:09:04] WARNING[30610]: loader.c:465
ast_unload_resource:Firm unload failed for app_cbmysql.so
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