Hi Nick,
You mean if it is possible for Asterisk to use realtime dialplan? If it is,
AFAIK it is possible using a table format for realtime extensions.
Regards,
Ronald
On Mon, Sep 26, 2011 at 1:33 PM, Sam Govind govoi...@gmail.com wrote:
Hmmm..interesting..I haven't came across anything like
One thing I do is to use the mysql command to run simple queries and the
Goto a context with the information... simple and clean
On 09/25/2011 10:33 PM, Sam Govind wrote:
Hmmm..interesting..I haven't came across anything like this so
far..How about making a new table for the insertion of a new
Hi Vladimir,
I tried the steps as you wrote them but i just get the same error :
Channel map:
Channel 01: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01)
Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04)
2 channels to configure.
Changing signalling on channel 1
-Mensaje original-
De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
Suppose I have two IP aliases on one asterisk box.
I have to talk to SIP friend A using IP x.x.x.x and I have to talk to
SIP friend B using IP y.y.y.y.
(In case you're wondering, the reason is
Hi
try to add this
exten = _90Z,1,Set(CALLERPRES()=allowed)
exten = _90Z,n,Set(CALLERID(num)=5631040)
if you get the same result that is means that you don't have permission from
your provider
kind regards
2011/9/26 C F shma...@gmail.com
Confirm with your provider that
Hi list,
My call does not pass through on the first dial, I have to redial again the
number for the call to pass through. I'm not sure if the problem is on my voip
proovider or my asterisk server setup. Any thoughts pls?
Regards,
Malvin
--
Thanks for the response. Although I had already gone thorough a lot of these
types of QAs during my own problem solving, your suggestions did bring a couple
that I had not.
In no particular order...
Earlier, you suspected receiving a signal was causing your problem.
Instead of ignoring
Can you please post the relevant parts of extensions.conf? As well as
a CLI output of when you dial and it fails?
On 9/26/11, Malvin Rito mr...@mail.altcladding.com.ph wrote:
Hi list,
My call does not pass through on the first dial, I have to redial again the
number for the call to pass
Hi,
Is there a way to continue dialplan when a call is abandoned from application
queue()?
If the caller is waiting in a queue, and hang up before timeout, I'd like to
execute an application in dialplan.
I've tested h exten, but it doesn't work for this.
-- Executing
On 09/26/2011 08:52 AM, Marcus Vinicius wrote:
Hi,
Is there a way to continue dialplan when a call is abandoned from
application queue()?
If the caller is waiting in a queue, and hang up before timeout, I'd
like to execute an application in dialplan.
I've tested h exten, but it doesn't work
- Original Message -
From: Remco Barendse aster...@barendse.to
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, September 23, 2011 5:27:27 AM
Subject: [asterisk-users] TDM400 FXO stopped working
Hi list
I have 2 servers
Hi,
are there anybody, who using the chan_misdn included with Asterisk
v1.8? If yes what mISDN version used v1 or v2? Yes, I can read on many
pages for mISDN2 I need to use chan_lcr, but this informations are 2-3
years old, and I can't imagine asterisk v1.8 chan_misdn works only
with linux kernel
On 09/26/2011 11:35 AM, Gergo Csibra wrote:
Hi,
are there anybody, who using the chan_misdn included with Asterisk
v1.8? If yes what mISDN version used v1 or v2? Yes, I can read on many
pages for mISDN2 I need to use chan_lcr, but this informations are 2-3
years old, and I can't imagine
Actually it doesn't say AGI(async:script) it says AGI(async:agi)
and than continues further to setting up an AMI user so the script is
executed through the manager interface?? Than it says
AGI(agi:async).?? Well most importantly it says Cons of async AGI:
It is the most complex method of
On Sat, 24 Sep 2011, Steve Edwards wrote:
Instead of ignoring the signal, how setting up a handler and logging
the reception?
On Mon, 26 Sep 2011, Mehmet Avcioglu wrote:
The program is written in a very top to bottom way (himm stateless?
non-oo? not sure what to call it) and would be really
Monday, September 26, 2011, 7:20:10 PM, Kevin wrote:
On 09/26/2011 11:35 AM, Gergo Csibra wrote:
Hi,
are there anybody, who using the chan_misdn included with Asterisk
v1.8? If yes what mISDN version used v1 or v2? Yes, I can read on many
pages for mISDN2 I need to use chan_lcr, but this
hello list,
i have one question if there is any way to know the model of the card diguim
installed
without open the server :)
thanks and Regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
If there is an existing Asterisk install, the contents of /etc/dahdi/modules
should tell you this.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Monday, September 26, 2011 1:16 PM
To: Asterisk Users Mailing
Or
/bin/llsmod|grep dahdi
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Monday, September 26, 2011 1:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] model of
On Mon, Sep 26, 2011 at 06:16:19PM +, salaheddine elharit wrote:
i have one question if there is any way to know the model of the card diguim
installed
without open the server :)
'lspci' should give you the information you want. For example, on
one of my test systems:
# lspci -d
thanks for your response but there is no dahdi becouse i have asterisk 1.4
installed (zaptel)
any help please
2011/9/26 Danny Nicholas da...@debsinc.com
Or
/bin/llsmod|grep dahdi
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
On Mon, 26 Sep 2011, Michael L. Young wrote:
I think the clue is actually right there in the error message.
You say that port 1 is an FXO module? Then your signaling is set wrong. The
signaling should be fxsks.
For port 4, it should be fxoks.
Remember, that in the configuration files,
Give zttool a try...
On Mon, Sep 26, 2011 at 2:32 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
thanks for your response but there is no dahdi becouse i have asterisk 1.4
installed (zaptel)
any help please
2011/9/26 Danny Nicholas da...@debsinc.com
Or
/bin/llsmod|grep
Gergo, why do you want to use mISDN? Use Dahdi. Or do you want to use
a very exotic
isdn card which is only supported by mISDN? tell us more.
My long time experience with mISDN_v1 is, v1 has major echo and fax
problems. because the audio signals are transported very unsynchronic
because of the
On 27/09/11 2:57 AM, Nick Khamis wrote:
That would be amazing! And would allow for more possibilities. A call
is made with the simple insertion of new call data. Some directions on
this please?
Just write a program that polls the database, if it sees a record it
runs an Originate command via
Matt - how dare you tell a man asking for a fish to learn how!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell
Sent: Monday, September 26, 2011 4:48 PM
To: asterisk-users@lists.digium.com
Subject:
On 27/09/11 10:51 AM, Danny Nicholas wrote:
Matt - how dare you tell a man asking for a fish to learn how!
:-) It would have taken way too much time to explain all the steps.
Although, having said that I am doing a tutorial at Astricon on how to
use the Asterisk Manager, so if he's at
Will you be recording this presentation for those of us who can't get to
Astricon?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell
Sent: Monday, September 26, 2011 5:01 PM
To:
Hello Matt,
Thank you so much for your response. The first two are not a problem.
I would rather write the app in C and compile it directly into
asterisk. Are there
any direction on how to work with the dial app? Any types of threads *
may use etc...
Thanks in Advance.
Nick Khamis
Ph.D Computer
Hi there,
I was playing around with jabber / gtalk integration with asterisk and I
would like to know if there is anyway to restrict the users able to add the
account used by asterisk for gtalk by domain.
Right now with the default settings I can see that it is possible to allow
all users or
Nick,
I just created a .NET dialer for a client not too long ago and it didn't
take more than a few hours - it doesn't use a db (thinking that's an awful
lot of empty polling waiting for a call when the call will be records in the
call records anyway), but simply connects/logs in and kicks it
On 27/09/11 11:04 AM, Nick Khamis wrote:
Hello Matt,
Thank you so much for your response. The first two are not a problem.
I would rather write the app in C and compile it directly into
asterisk. Are there
any direction on how to work with the dial app? Any types of threads *
may use etc...
I
On 27/09/11 11:03 AM, Danny Nicholas wrote:
Will you be recording this presentation for those of us who can't get to
Astricon?
Dunno whether they'll be recording - I haven't done an Astricon
presentation for a few years now :-)
--
Cheers,
Matt Riddell
Hello David,
Thank you so much for your response. I am sure it can be easily done
using AGI. The reason I am leaning more
towards storing the call information in a database record, is because
our existing client applications can be easily
modified to write to MySQL. The asterisk cron/thread that
Hello Matt,
Thanks again for your response. This all makes perfect sense. Much
like sip friends, moh, extensions etc..., have ability to process both
existing config files and realtime configurations, the dial app should
be able to do the same, leaving the current functionality untouched. I
have
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