Displaying 20 results from an estimated 1000 matches similar to: "Subject: Passing DID to external number?"
2006 Jun 22
4
Passing DID to external number?
Hi,
We run a small switchboard using Asterisk and Free PBX.
We have two main extensions and two ring groups. The first ring group rings
the two internal extensions. If the internal extensions do not pick up the
call after 15 seconds then the second ring group kicks in which should ring
the two internal extensions plus two external numbers.
Firstly, how do I pass the DID number of an incoming
2006 Mar 16
4
How to transmit Video
please tell me that what sip based softphone will beused with Asterisk so that i can trasmit and receive video on my LAN .
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Use Photomail to share photos without annoying attachments.
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2004 Jul 06
0
Samba and SuSE 9.1?
Hi everyone,
I have so many problems with samba on one PC with SuSE 9.1 on.
This PC is on a local area network with another PC with SuSE 9.0 on it
with a working samba connection, and some windows clients.
My 9.1 PC has an ASUS P4S800 motherboard and I use the integrated
Asustek sis900 ethernet lan configured as eth0 with the sis900 module.
Static IP and static gateway (the SuSE 9.0 machine).
2004 Nov 03
1
simulate
are any tool, script, software, that let me simulate network traffic,
like trasmit for 2 hours at 100Mbs?
--
Patricio Bruna http://www.linuxcenterla.com
Ingeniero de Proyectos Canada # 239 Piso 5
Red Hat Certified Engineer Providencia, Santiago - CHILE
Linux Center Latinoamerica Fono: +56 2 2745000, Fax : +56 22747075
2006 Nov 05
1
Call Quality Issues with IAX?
Hey all,
I recently got a message from my provider about IAX:
> We do not recommend the use of IAX. It is a lossy protocol that is
> known to cause crackling, loss of audio and other issues. You can
> use IAX if you want, but we will not assist with any issues you may
> encounter.
Does anyone else know about these "known" problems? I'm not sure
where this provided got
2009 Mar 17
0
iPhone patches question
As of right now, you are out of luck in terms of instream metadata with the
embedded quicktime player that's distributed with the iphone OS. As
Christopher says, there are many ways to get the same effect, but they ALL
require you to build custom iPhone apps. There are a whole mess of radio
apps out there, and they are all pretty similar, and NONE of them use the
embedded quicktime object.
2013 Sep 25
0
error when using ps() function on categorical variables - re propensity score matching
Dear List,
I am having difficulty running the ps() function when variables are stored
as factors and was hoping someone could provide some advice on how to
proceed.
I am running propensity score matching as outlined in:
Greg Ridgeway, Dan McCarey, Andrew Morral, Lane Burgette and Beth Ann
Grin (May 3, 2013) Toolkit for Weighting and Analysis of Nonequivalent
Groups: A tutorial for the twang
2009 Mar 16
2
iPhone patches question
Hi Bob,
I am thinking that metadata could be streamed to the iPhone with some additonal work. There are some possible methods including wrapping a custom interface (either a Web App or iPhone SDK App) to check with a server script to grab metadata and display it as its streaming the Icecast MP3 Format.
For example:
There would be two domains such as icecast.server and metadata.server on the
2006 May 15
3
How to tell if RTP stream is has been reinvited?
Howdy,
How can you tell if RTP traffic has been reinvited/is bypassing an * server?
Sincerely,
Brent A. Torrenga
brent.torrenga@torrenga.com
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771
+1 219 836 8918 x325 Voice
+1 219 836 1138 Facsimile
www.torrenga.com
2009 Mar 17
1
iPhone patches question
When the Tuner2 iPhone Player hits the Apple iTunes Apps Store, this
should solve everyone's player problem:
http://www.indexcom.com/iphone.html
There is a lot of information on this page. It tells an interesting
story about audio quality.
This player is not an Orban project.
-greg.
ORBAN
At 18:16 2009-03-16, oddsock wrote:
>As of right now, you are out of luck in terms of instream
1999 Dec 21
0
" eth1 transmit timeout ... "
Hi all,
I've installed samba server to act as a PDC.
Here some informations :
- Linux Version : 2.0.36
- Samba Version : smbd 2.0.0beta3
My Linuxbox has two ethernet interfaces (eth0:200.* and
eth1:192.168.1.1)
After installing samba, everything worked fine during 20 minutes only !
After that, a strange problem occurred with eth1. The following message
appears
on the Linux console :
2006 May 17
2
AAH not getting IP address, likely to be network card?
Hi all,
We use AAH to run our office telecoms registered with two Sipgate accounts.
Today, Sipgate appeared to have had problems with their server with oneway audio on every call. In order to cause the Sipgate message service to pick up in stead of our AAH box, I simply unplugged the network cable.
We now have problems where AAH does not seem to access the network. I plugged the network cable
2005 Mar 11
1
DSP c5x
I'm finding it dificult to get the code to fit within c5x dsp, especially
due to the large heap necessary for the huge allocations in
nb_encoder_init and nb_decoder_init.
Are the people that have this running using additonal external ram?
Has anyone experimented with reducing the internal encoder/decoder stack
sizes
for narrow band... is there a practical minimum that is usable?
Also,
2008 May 28
2
Re: calculating the needed swiotlb in dom0
I just found out that setting swiotlb=128 in dom0 grub.conf allowed me to
run my mythtv domU with additonal options to ivtv:
in /etc/modprobe.d/options
options ivtv enc_mpg_buffers=8
Setting this to any higher value causes a kernel panic. I want to be able
to specify
options ivtv enc_mpg_buffers=16 enc_vbi_buffers=8
How much swiotlb do I need? Is there a way to figure this out or is it just
2004 Sep 10
2
build problems (autoconf/libtool)
Hi.
I want to add some code to flac and am having trouble building it out of cvs.
I've had several problems, most of which were pretty easy to work around or
had already been discussed here. But this one has me stumped (though I am
admittedly new to autoconf, automake, and libtool). Here's the problem.
autoconf does not see any definition for AC_PROG_LIBTOOL. There is such a
2013 Apr 27
1
using metafor for meta-analysis of before-after studies
Hello, Dr. Viechtbauer.
I am trying to perform a meta-analyis on a group of before-after studies using Metafor. I read your webpage including your correspondence with Dr. Dewey (https://stat.ethz.ch/pipermail/r-help/2012-April/308946.html), who also conducted a similar study. These information is very hepful, but I have one additonal question which I wonder if you can give me some instruction.
2006 Jun 08
1
FreePBX 2.1.0: Manually rewriting
do you have selinux enabled? It should not be.
p
p.s. - if it comes to re-installing, you can backup all your settings with the freepbx backup utility and then restore so that you don't have to re-enter everything.
From: "Lachek Butalek" <lachek@gmail.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Date:
2005 Jan 20
2
http/cgi metadata updating
I heard that Icecast 2.2 now supports metadata updating via http for vorbis
streams. We are adding http metadata support into our broadcasting software
for the next release. Can someone point me to a link for the syntax, or is
it identical to the Shoutcast syntax?
http://www.host.com:port/admin.cgi?pass=Password&mode=updinfo&song=Artist_Title&url=http://someurl.com
Has there been
2005 Feb 11
1
[LLVMdev] Function attributes and bytecode
On Thursday 10 February 2005 21:47, Markus F.X.J. Oberhumer wrote:
> In order to get more familiar with the llvm sources I've recently
> decided to try to add support for the always_inline and noline function
> attributes.
I believe it is better to let the compiler decide when or not to inline a
function. Most of the times a developer goes overboard with inlining and ends
up with a
2005 Aug 07
4
Configuring Asterisk@home for Sipgate.
Hi all,
I'm new to the forum. Oh no....newbie question coming, I hear you all cry!
I'm playing around with Asterisk@home and have installed software and fiddled around with sip and extensions files.
I have manage to make out going calls through Sipgate using X-Lite but cannot for some reason receive incoming calls. Incoming calls do not even show up on the switchboard panel.
I've