On Sat, 6 Sep 2008, hugolivude wrote:
OS = CentOS 5
Asterisk = 1.4.21
Router = WhiteRussian 0.9
Not sure whether I have a problem w/ Asterisk or White Russian config,
so I'm posting to both lists.
I have 2 Asterisk servers running behind a Linux router w/ White
Russian. I'm having a lot
On Sun, Sep 7, 2008 at 7:47 AM, Tim Panton [EMAIL PROTECTED] wrote:
On 7 Sep 2008, at 08:38, Gordon Henderson wrote:
On Sat, 6 Sep 2008, hugolivude wrote:
OS = CentOS 5
Asterisk = 1.4.21
Router = WhiteRussian 0.9
Not sure whether I have a problem w/ Asterisk or White Russian
config,
so
On 7 Sep 2008, at 08:38, Gordon Henderson wrote:
On Sat, 6 Sep 2008, hugolivude wrote:
OS = CentOS 5
Asterisk = 1.4.21
Router = WhiteRussian 0.9
Not sure whether I have a problem w/ Asterisk or White Russian
config,
so I'm posting to both lists.
I have 2 Asterisk servers running
Mark Hamilton wrote:
a) How can I make it so #2 doesn't have to be exceptionally fast, and maybe
get a second of delay in there permitted?
;featuredigittimeout = 500 ; Max time (ms) between digits for
; feature activation (default is 500 ms)
--
Ben
Hi. I am getting the following error while trying to compile the
dahdi-tools-trunk from svn this morning.
gcc -g -O2 -I. -O2 -g -fPIC -Wall -DBUILDING_TONEZONE -MD -MT
sethdlc.o -MF .sethdlc.o.d -MP -c -o sethdlc.o sethdlc.c
sethdlc.c: In function 'set_iface':
sethdlc.c:205: error: 'union
On 08:24, Sun 07 Sep 08, Steve Totaro wrote:
On Sun, Sep 7, 2008 at 7:47 AM, Tim Panton [EMAIL PROTECTED] wrote:
This is one of those cases where it is almost certainly simpler to
use IAX2 not SIP.
You will need zero config on the router and it will 'just work'
- assuming your provider
On Saturday 06 September 2008 21:47, Brian wrote:
Hi Thomas,
The queue definitions and its member list will be reloaded each time a
caller joins the queue. So you don't need to reload it manually.
Hi,
is not work for periodic-announce-frequency and periodic-announce.
An reload is
I recently installed 1.4.21.2 on Debian 2.6.18-6 and since then, I am
experiencing occassional garbled voicemail messages. Specifically, what
happens is that the first 15-20 seconds of the message is fine, but
sometimes after that the sound starts to break up and the end of the
message is
On 7 Sep 2008, at 15:06, Bruce Komito wrote:
I recently installed 1.4.21.2 on Debian 2.6.18-6 and since then, I am
experiencing occassional garbled voicemail messages. Specifically,
what
happens is that the first 15-20 seconds of the message is fine, but
sometimes after that the sound
On Sun, Sep 07, 2008 at 09:22:57AM -0400, John covici wrote:
Hi. I am getting the following error while trying to compile the
dahdi-tools-trunk from svn this morning.
gcc -g -O2 -I. -O2 -g -fPIC -Wall -DBUILDING_TONEZONE -MD -MT
sethdlc.o -MF .sethdlc.o.d -MP -c -o sethdlc.o sethdlc.c
on Sunday 09/07/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote
On Sun, Sep 07, 2008 at 09:22:57AM -0400, John covici wrote:
Hi. I am getting the following error while trying to compile the
dahdi-tools-trunk from svn this morning.
gcc -g -O2 -I. -O2 -g -fPIC -Wall -DBUILDING_TONEZONE
Hi,
On Sat, Sep 06, 2008 at 09:52:45PM -0400, hugolivude wrote:
OS = CentOS 5
Asterisk = 1.4.21
Router = WhiteRussian 0.9
Not sure whether I have a problem w/ Asterisk or White Russian config,
so I'm posting to both lists.
I _think_ I have the ports forwarded correctly on my router. I
On Sun, Sep 7, 2008 at 4:56 PM, Thomas Winter [EMAIL PROTECTED] wrote:
On Saturday 06 September 2008 21:47, Brian wrote:
Hi Thomas,
The queue definitions and its member list will be reloaded each time a
caller joins the queue. So you don't need to reload it manually.
Hi,
is not work for
Hello,
I have been testing a trunk IAX and another SIP, using sipp to
generate SIP calls to a Asterisk box.
The testing dialplan just connects to another Asterisk box, who
answers the call and playback some files.
I noticed that the cpu load is higher when I use an IAX, about 90% for
25
Hi all,
In my modules.conf I have the autoload=yes, and there is one
codec_iLBC.so module in the modules folder.
However, when I do show translation, I see no translation to/from iLBC
nor G.729, and I'm not able to establish call to channels using these
codecs.
I read that there are some
Dear Gurus,
I've got a serious problem with my inherited Asterisk. Sometimes the
DTMF digits lost, and the
client could not step in the IVR. I use mISDN and i could find my
presses in the mISDN log
but not in the Asterisk inside. The situation is the same when I use my
Linksys SPA 922.
My
I may be wrong about this, but * understads that these codecs exsist, but
without a codec_XXX.so, it cant do translation on the codec.
In this case, * can do pass throught (eg: g729 - g729) but cannot do
translation (eg: g729 - gsm).
You need to install the codec before you can do a translation.
Make sure there is no noload = codec_ilbc.so in the module folder
You can also try to manually load the codec from the cli try load
codec_ilbc.so
For g729 you need to buy a licence from Digium.
--
Stelios S. Koroneos
Digital OPSiS - Embedded Intelligence
Tel +30 210 9858296 Ext 100
Fax +30
Hey Guys,
I am trying stream live music via icecast streaming server into a
conference room, this will allow persons joining the conference to hear the
music.
I have been googling and i have come across a few tutorials, that give
instructions as to how to get it done. But they all mention the
On Sunday 07 September 2008 21:49, Atis Lezdins wrote:
On Sun, Sep 7, 2008 at 4:56 PM, Thomas Winter [EMAIL PROTECTED]
wrote:
is not work for periodic-announce-frequency and periodic-announce.
An reload is necessary.
Asterisk is 1.4.21.1
It shouldn't be necessary. However you can try
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