On Tue, 13 Oct 2009, Dan Journo wrote:
> To avoid the problem of deleting/copying calls that are still being
> recorded, I could record the call into a temp directory. Then using the
> dial plan, I could copy the temp recording into the ftp root directory
> once the call has ended.
True, but i
On 10/12/09 13:29, Ivan Stepaniuk wrote:
>You are using the ATAs to access from one Asterisk to the other one?
>Wouldn't make sense to connect those two asterisk through SIP or IAX via
>Internet instead of calling via PSTN anyway?
>
>Anyway, In this case it seems that this is not asterisk related,
Hi,
To avoid the problem of deleting/copying calls that are still being
recorded, I could record the call into a temp directory.
Then using the dial plan, I could copy the temp recording into the ftp
root directory once the call has ended.
Dan
Thank you for contacting Kesher Communications Lt
> On Behalf Of Ivan Stepaniuk
>The script is very simple and far from complete, it just moves the
> content into the mounted FTP directory. It has some verbose output as it
> is run from inside another script that redirects the output to a log
> file.
What happens if the script is run whil
Hi
I look after a site which is using asterisk and a vsp for its primary
telco needs, so I am on holiday for a week and of course some jack arse
has decided to reboot the server and something has gone wrong with the
remote access. Now they don't have any internet and i can't fix it
remotely. Ba
Thanks for that.
I really appreciate it!
Dan
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ivan Stepaniuk
Sent: 12 October 2009 22:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
Dan Journo wrote:
> Thank you for replying. I hadn't thought about the problem of simultaneous
> calls. It would be a problem if a number of calls ended at the same time.
>
> If you can post it, the script would really be helpful as I'm only a beginner
> with Linux
The script is very simple a
Hi,
I'm trying to get voicemail to record video as well as audio. So far,
only audio is recorded. I'm using files instead of the ODBC or IMAP.
I'm running 1.6.1 beta code (I've tried several versions here). I've
also tried running the video branch code and get similar results.
I can run
On Mon, Oct 12, 2009 at 05:59:16PM +0200, Eckhard Jokisch wrote:
> Hi,
> I set up my asterisk so it does dial out on DAHDI/1 which is on an FXS modul.
> But the phone that is attached to the line does nothing at all.
> asterisk-CLI shows a lot of
> -- DAHDI/1-1 is ringing
> -- DAHDI/1-1 is ringi
Hello!
I need to send a digit to a channel of an established call, from "outside"
of Asterisk, I suppose it must be from the AMI.
I want to send a * for example, but in addition to reproducing the sound of
that digit (I dont care thatl), I need that the digit sent actually performs
an action.
Fo
On 10/12/09 13:29, Ivan Stepaniuk wrote:
>You are using the ATAs to access from one Asterisk to the other one?
>Wouldn't make sense to connect those two asterisk through SIP or IAX via
>Internet instead of calling via PSTN anyway?
>
>Anyway, In this case it seems that this is not asterisk related,
Luis Silva wrote:
> I have an asterisk in 1.2 version with 30 g729 licenses. I what to
> upgrade it to 1.4 and also upgrade the O.S. from fedora 8 to centos 5.3.
>
> For what I understand I can make the backup of the license files in
> /var/lib/asterisk/licenses/ if I need to reinstall the oper
Speaking blindly here, you should be able to unless there is some kind of
server architecture involved (probably not with /v/l/a path.)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Silva
Sent: Monday, October 12, 200
Hi,
I need some help.
I have an asterisk in 1.2 version with 30 g729 licenses. I what to upgrade
it to 1.4 and also upgrade the O.S. from fedora 8 to centos 5.3.
For what I understand I can make the backup of the license files in
/var/lib/asterisk/licenses/ if I need to reinstall the operatin
- "Eckhard Jokisch" wrote:
> Hi,
> I set up my asterisk so it does dial out on DAHDI/1 which is on an FXS
> modul.
> But the phone that is attached to the line does nothing at all.
> asterisk-CLI shows a lot of
> -- DAHDI/1-1 is ringing
> -- DAHDI/1-1 is ringing
> -- DAHDI/1-1 is ringing
>
Hi,
I set up my asterisk so it does dial out on DAHDI/1 which is on an FXS modul.
But the phone that is attached to the line does nothing at all.
asterisk-CLI shows a lot of
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 is ringing
Even when I lift up the handset during (and no a
Hi to all, is it possible to setup a live audio streaming in Asterisk
using for source monitor, mixmonitor or chanspy?
Thanks
--
/*/
nik600
http://www.kumbe.it
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Astr
On Mon, Oct 12, 2009 at 4:51 AM, Olivier wrote:
> Hi,
>
> With 1.6.1.7-rc2, doc says:
> select*CLI>
> -= Info about function 'SPRINTF' =-
>
> [Syntax]
> SPRINTF(,[,...])
>
> [Synopsis]
> Format a variable according to a format string
>
> [Description]
> Parses the format string specified and re
Hei!
I'm trying to send special characters out to ss7 link, but libss7 seems
to convert them to zeroes. The challenge is that our service provider
demands some of the regional numbers to be sent in format D0+number.
When I use D in front of the number in dialplan, libss7 replaces it with
00, So
On Monday 12 October 2009 05:51:31 Olivier wrote:
> [Description]
> Parses the format string specified and returns a string matching that
> format.
> Supports most options supported by sprintf(3). Returns a shortened string
> if
> a format specifier is not recognized.
>
>
>
> I'm trying use sprin
Dear All,
Can I mix realtime conf and static configuration files?
Thanks for responses.
Rgds,
Robor
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.ast
Dear All,
I read somewhere that app_meetme in 1.2 has a single lock therefore
supporting 10+ conferencing is problematic... is this still the case with
version 1.6?
What is the functional difference between meetme and confbridge??
Thanks in advance for responses.
Rgds,
Robor
__
Hi Iván,
Thank you for replying. I hadn't thought about the problem of simultaneous
calls. It would be a problem if a number of calls ended at the same time.
If you can post it, the script would really be helpful as I'm only a beginner
with Linux.
Many thanks
Dan Journo
-Original Message
Dan Journo wrote:
> I'm working on a call recording solution. I would like recordings to
> either be automatically uploaded via FTP, or posted to a URL for
> processing by our main server.
> Is Asterisk capable of doing this or will I have to create a separate
> application that monitors a temp dir
Joseph wrote:
> I just double checked the setting of the remote asterisk and it has the same
> setting as mine.
> Sip.conf has in Global:
> dtmfmode = rfc2833
> individual extension has no dtmf setting at all, so global setting take
> precedence.
>
> All units Linksys, Sipura have
> DTMF Tx Meth
Hi,
With 1.6.1.7-rc2, doc says:
select*CLI>
-= Info about function 'SPRINTF' =-
[Syntax]
SPRINTF(,[,...])
[Synopsis]
Format a variable according to a format string
[Description]
Parses the format string specified and returns a string matching that
format.
Supports most options supported by sp
You can try using NFS. Also you can pay some one to write script that would
move the files over on hang up.
- Original Message -
From: Dan Journo
To: asterisk-users@lists.digium.com
Sent: Monday, October 12, 2009 01:15
Subject: [asterisk-users] Call Recording and Posting
H
27 matches
Mail list logo
