Hi,
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?
thanks.
Eyal Mahalal
--
Hello Everyone,
I didn't get a reply to my problem below so I'm posting again just in case
someone who might be able to help missed my previous post.
Thank You...
*
Hello list,
I'm trying to populate my CDR logs
i need to know how can i configure a D 300JCT with asterisk, i want to
connect two PBX where each one have this card on it,i really need your help
as soon as
possible.
i already done some file configuration system.conf and in chan_dahdi.conf
and i have installed the DAHDI and the LibPri modules.
Le 06/12/2011 10:16, Harel Cohen a écrit :
Hello Everyone,
Hi Harel
I didn’t get a reply to my problem below so I’m posting again just in
case someone who might be able to help missed my previous post.
Thank You…
Please take a look at issue ASTERISK-18875
On Sat, 2011-12-03 at 15:36 -0500, Nick Khamis wrote:
Hello Everyone,
Are there any descent generic IVR recordings, that we can
use to quickly get our PBX up and running? It will obviously
not include the company name.
It's easy enough to make your own recordings.
Word of caution though.
Hi All,
I read about the *Hint* in asterisk. I want to implements into my server
for testing purpose. How to use it ? please help me...
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
_
-- Bandwidth
Hello,
On a 4 BRI Euro-ISDN TE/PtmP production system (Asterisk 1.6.1.18, Dahdi
2.5.0, Libpri 1.4.12), I sudently got the console cluttered with messages
like this:
PRI got event: HDLC Abort(6) on Primary D-Channel of span foo
Then a flow of messages (twice per second) like this:
PRI got event:
2011/12/5 Jamie A. Stapleton jstaple...@computer-business.com
I have not ever done what you are talking about.
** **
However, I can tell you that our Openfire XMPP server has similar
functionality because of their Asterisk-IM Plugin.
Are you currently using it ?
With which asterisk
On Mon, 2011-12-05 at 18:51 -0800, Steve Edwards wrote:
snip
Your security needs depends on your environment. At this point in time,
all of the hosts I manage for my clients exist in very limited
environments and have very small attack surfaces. They are racked in
secure data centers. They
Hi All,
I did some google and found some documents on that and finally I got some
response from asterisk . Below is the CLI output of my google.
*haddock8-astrx*CLI core show hint 2218
2218@bhati-subscribe : SIP/2218
State:IdleWatchers 0
1 hint matching
Hello,
AFAIK Hints are used for looking out for a device state before actually
doing anything. Hints can be used from Dial-plan, AMI, or AGI. An example
can be to look for state of a SIP user.
Read these links for better understanding.
http://www.smartvox.co.uk/astfaq_subscribe_hints.htm
Hi All,
Below bold application gives the correct information with asterisk
*HINT*function.
exten = 222,1,NoOp( Call from Gtalk )
*same = n,NoOp(My phone state is currently
${DEVICE_STATE(SIP/2218)})*
same = n,Set(CALLERID(name)=From Google Talk)
same = n,Wait(10)
Hi All,
If you used *DEVICE_STATE *function then there is no need to used *HINT* it
work independently.
It's not become to confusion for me how to when to used *HINT *and
when *DEVICE_STATE
?
*
On Tue, Dec 6, 2011 at 6:20 PM, virendra bhati virbh...@gmail.com wrote:
Hi All,
Below bold
Try this link - I think they describe it better than I would
http://www.voip-info.org/wiki/view/Asterisk+standard+extensions
From: virendra bhati [mailto:virbh...@gmail.com]
Sent: Tuesday, December 06, 2011 7:28 AM
To: Danny Nicholas
Subject: Re: How to use Hints in asterisk
thank
IMO you are trying to circumvent basic Asterisk functionality. It's your
CDR so you can do what you want with it - I think the answer to this is to
populate another DB with the live call data, then update the CDR from that
after the call has ended (perhaps a daemon).
From:
IVR = Idiot Verify and Recognize?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet
Sent: Tuesday, December 06, 2011 4:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
That's too funny! What are some tricks to make it sound professional.
What I mean is, what are some of the typcial things people do to the
recording, to make it sound kind-of robotic? I have no idea how to explain
it. Maybe those of you that have done ivr recordings for corporations could
There are some Allison Smith Speaks blogs out there with good IVR hints.
Some hints from my experience
1. The recommended volume adjustment for asterisk is -3 DB (that's -3 if
you look at the wav in Audaciity). This will vary depending on your flavor
of Asterisk and your input (SIP/DAHDI/etc).
What I would like to know is. How could you have possibly known I have
a 90 year old aunt?!?!
Sorry for the Noise!
Merry Christmas/Happy Holidays,
Nick.
On Tue, Dec 6, 2011 at 9:29 AM, Danny Nicholas da...@debsinc.com wrote:
There are some Allison Smith Speaks blogs out there with good IVR
On Tue, Dec 6, 2011 at 5:19 AM, Hans Witvliet aster...@a-domani.nl wrote:
On Mon, 2011-12-05 at 18:51 -0800, Steve Edwards wrote:
snip
Your security needs depends on your environment. At this point in time,
all of the hosts I manage for my clients exist in very limited
environments and have
On Sat, Dec 3, 2011 at 12:59 AM, white hat whitehat...@gmail.com wrote:
When a caller calls my google voice phone number, I must answer, wait and
press one to accept. Sometimes even that does not work.
I just need a little advice on how to write the dial plan. I still have
much to learn
2011/12/5 Olivier oza_4...@yahoo.fr
Hi,
Porting a dialplan from 1.6.1 to and old 1.4 install, it seems Park()
application uses different arguments.
The only doc I could get a hand on is (core show application Park) this
one :
[Synopsis]
Park yourself
[Description]
Park():Used to park
Hello
I have machine running a couple of instances of asterisk. Each
instance create own control pipe (asterisk.ctl). How I can remotely
connect into asterisk which own pipe I know?
I know I can do it if path to pipe specified in asterisk.conf, but I
have not any asterisk.conf accessible, only
Well, that means opening up VPN connections from everywhere. Thats why
I suggested turning off the server completely.
hmmm - I thought that was the point of a vpn
--
_
-- Bandwidth and Colocation Provided by
You don't state your Asterisk version, but this sounds like a task for
chan_skinny perhaps? Or it might just be as simple as hitting an RTP range.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yaroslav
2011/12/6 Danny Nicholas da...@debsinc.com:
You don't state your Asterisk version, but this sounds like a task for
chan_skinny perhaps? Or it might just be as simple as hitting an RTP range.
Asterisk =1.8
No, I want manually connect to asterisk via ssh console. I.e. like:
ssh user@ast-host
Hi All,
I'm using Exchange as our voicemail system. Everything works fine until the
1 week mark when Exchange changes the port number used, then Asterisk 1.8
seg faults and I have no phones (unless I restart the U.M. service before
the 1 week period is up). Since that is a hack, I'm hoping
If I understand correctly, turning off Call Screening in your Google
Voice configuration should directly connect incoming calls and eliminate
the need to press one.
JF
On 12/2/2011 11:59 PM, white hat wrote:
When a caller calls my google voice phone number, I must answer, wait
and press one to
On 12/6/11 9:18 AM, Yaroslav Panych wrote:
Asterisk=1.8
No, I want manually connect to asterisk via ssh console. I.e. like:
ssh user@ast-host /sbin/rasterisk -switch_to_point_asterisk.ctl
path_to_asterisk.ctl
I knowpath_to_asterisk.ctl but did not found any
switch_to_point_asterisk.ctl in
Hey Josh,
I've messed with the google voice account settings extensively.
As of now, in Google voice account settings I have.
Voice tab: forward calls to Google chat checked. Nothing else is checked.
Calls tab: call screening is off. On incoming call, display callers
number. On Caller ID
dwa
As part of the troubleshooting I updated all of the asterisk packages from
the repo with yum. I'm using freepbx distro (centos based) with asterisk
1.8 There were several newer asterisk 1.8 packages available. I'm not
using any custom modules in freepbx. After the updates, I restarted
You could also try putting a Progress() statement between Answer and Wait.
I know there is a latency issue with DAHDI calls; 5 seconds may or may not
be enough for googlevoice.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of white
Hello all,
I am new to the Asterisk IRC users group. I was wondering if it would be
possible to use an IRC client when reading through the posts. If so, can
someone recommend one, and how I should go about configuring the client.
Thank you for your time and assistance.
Peter
Yes, we are using it. Most of the docs on the Internet are for 1.4. However,
we now have it working with 1.8 (after some work).
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Tuesday, December 06, 2011 5:13 AM
To:
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