In musiconhold.conf
[default]
mode=custom
directory=/var/lib/asterisk/mohmp3-empty
application=/etc/asterisk/bin/mohstream.sh
/etc/asterisk/bin/mohstream.sh
--
# BigR Radio Warm Hits
/usr/bin/wget -q -O - http://66.90.121.9:10005 | /usr/local/bin/madplay
-Q -z
One minor comment:
On Sun, Mar 27, 2011 at 02:41:11AM -0400, Alexander Lopez wrote:
In musiconhold.conf
[default]
mode=custom
directory=/var/lib/asterisk/mohmp3-empty
If you don't use 'mode=files', there's no need to set 'directory'.
application=/etc/asterisk/bin/mohstream.sh
[snip]
--
Hi all,
All the examples I've come across seem to suggest configuring
jabber.conf/jingle.conf/gtalk.conf for a real Google account.
What about the scenario where the Asterisk server should connect to an
account on a private Jabber server and using Jingle (voice calling over
Jabber)?
e.g. for
After following changes my D-Channel comes up and its working!!! :)
vi /etc/wanpipe/wanpipe*.conf
TDMV_DCHAN = 0
TDMV_HWEC = NO
@Thanks all of them who helped here...
No beer for others ;)
-S
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Fri, 25
This effort is not intended to replace packaging of Asterisk in the
official Debian or Ubuntu repositories. Our repositories are for
providing access to major versions of Asterisk that are newer than what
is included. We are exploring ways to work as closely as possible with
the Debian and
Here is the dialplan in macro:
exten = s,n,SayNumber($[${ARG1} % 100])
when 662 was passed as ARG1, I had the following at log:
WARNING[15217] pbx.c: We were unable to say the number 62, is it too large?
Do you see any odd in my dialplan?
On Sat, Mar 26, 2011 at 2:44 PM, Sherwood McGowan
When did Dial() with a custom ring cadence replace the default from
indications.conf for subsequent calls?
indications.conf:
ringcadence = 2000,4000
asterisk -rx dahdi show cadences
r1: 667,1333
extensions.conf:
exten= 201,
On Sunday 27 March 2011 14:50:37 Mohammad Khan wrote:
Here is the dialplan in macro:
exten = s,n,SayNumber($[${ARG1} % 100])
when 662 was passed as ARG1, I had the following at log:
WARNING[15217] pbx.c: We were unable to say the number 62, is it too
large?
Do you see any odd in my
On Sun, Mar 27, 2011 at 2:50 PM, Mohammad Khan beepl...@gmail.com wrote:
Here is the dialplan in macro:
exten = s,n,SayNumber($[${ARG1} % 100])
when 662 was passed as ARG1, I had the following at log:
WARNING[15217] pbx.c: We were unable to say the number 62, is it too large?
Do you see
On Sun, 2011-03-27 at 16:14 -0500, Sherwood McGowan wrote:
On Sun, Mar 27, 2011 at 2:50 PM, Mohammad Khan beepl...@gmail.com
wrote:
Here is the dialplan in macro:
exten = s,n,SayNumber($[${ARG1} % 100])
when 662 was passed as ARG1, I had the
Oh crap, you're right, my bad. Yes, I also agree, it's most probably the
language and/or missing files
On Sun, Mar 27, 2011 at 4:30 PM, Jeff LaCoursiere j...@sunfone.com wrote:
On Sun, 2011-03-27 at 16:14 -0500, Sherwood McGowan wrote:
On Sun, Mar 27, 2011 at 2:50 PM, Mohammad Khan
still no luck
i hear it change to t38 but it just doesnt connect
On Sun, Mar 27, 2011 at 5:26 AM, Larry Moore lmo...@starwon.com.au wrote:
Perhaps this will help.
I have a SPA8800 which has 4 x FXS 4 x FXO ports.
It took me some time to produce a working configuration.
In Asterisk I
On 28/03/2011 5:48 AM, Israel Gottlieb wrote:
still no luck
i hear it change to t38 but it just doesnt connect
Do you have two fax devices at your end, even a fax-modem attached to a
computer will do?
You are going to need to provide more information such as your current
configuration
Hello,
I have Asterisk-1.8.3.2, dahdi-linux-complete-2.4.1+2.4.1, and
libpri-1.4.11.5 installed and running on a Ubuntu 10.04 server all built
from source.
Everything is working nicely except one small issue.
The CDR records are stored in the CSV file correctly and complete.
The MySQL
On Sunday 27 March 2011 19:36:45 Eric W. Davenport wrote:
Hello,
I have Asterisk-1.8.3.2, dahdi-linux-complete-2.4.1+2.4.1, and
libpri-1.4.11.5 installed and running on a Ubuntu 10.04 server all built
from source.
Everything is working nicely except one small issue.
The CDR records are
In phone.cfg set the following line to
divert.fwd.1.enabled=0
from:
divert.fwd.1.enabled=1
For more info check page 323:
http://supportdocs.polycom.com/PolycomService/support/global/documents/support/setup_maintenance/products/voice/spip_ssip_vvx_Admin_Guide_SIP_3_2_2_eng.pdf
On Fri, Mar 25,
From the polycom pdf:
divert.fwd.x.enabled
If set to 1, the user will be able to enable universal call
forwarding through the soft key menu.
This sounds like it turns on and turns off the call forwarding feature
on the phone. I can try it out Monday, but I don't see where it has any
relation
Hi
Very thanks for your helps, that's work very goo
Bye
Olivier
2011/3/25 DHAVAL INDRODIYA dhaval.it01...@gmail.com:
Hi Olivier,
here is solutions for your situation , ideally you need to talk with
Provider and they can set SIP URI
for given DID numbre , but that can be solved by
18 matches
Mail list logo
