Hi Everyone,
Google IS my friend. I found the solution via Google on the second
glance ;-)
It seems that the USB latency was too high and you had to increase a
CAPI-Buffersize in chan_capi.h:
#define CAPI_MAX_B3_BLOCK_SIZE 500
(German instructions:
Hi Marco,
this is not an asterisk issue, it is a UNIX/Linux-issue.
If you overwrite a file, asterisk of course plays the data it reads from
the (at that moment overwritten) file.
I would suggest to delete the file and then write the updated file. On
UNIX if you delete an opened file, the
Hi all!
I'm trying to play some music from asterisk, and when I call to the PBX
from a GSM mobile phone, the more I speak while hearing the music, the
worst is the quality of the music I hear... My audio is at 8Khz,
16bits/sample.
I've tried different codecs for asterisk, but results
On Mon, 25 Apr 2005, Andrew Kohlsmith wrote:
It has absolutely nothing to do with what economically suits them best --
it
has everything to do with the fact that when you buy a clone X100P you DO NOT
KNOW what you're getting. The chipset may be the same but as you can clearly
see from
On Fri, 4 Feb 2005, Dana Olson wrote:
I'm looking at ordering a server from HP. I checked around on Google
and found in the Wiki that the ProLiant DL380 is supposed to be known
to work with *.
HP ProLiant DL380 G4 Server w/ the following options:
Intel Xeon 3.20GHz/1MB
2GB REG PC2-3200 (2
On Sat, 8 Jan 2005, chawki hammoud wrote:
I have Asterisk running on Linux Redhat9 dstr. I
subscribed to a third party sip providers to make LD
calls. Can I initiate a call sessions from asterisk
CLI command prompt after I configure extensions.conf
and iax.conf?
Hi Chawki,
yes you can. You
On Thu, 18 Nov 2004, Rodney Acosta Coya wrote:
my distro is novell linux desktop
yes is an rpm based system
look this
linux:/inst/pbx/asterisk-1.0.0 # - which make
bash: popd: directory stack empty
what can i do??
Is it possible that you are new to linux ? Try typing
which make
On Wed, 17 Nov 2004, Steve Underwood wrote:
It is working pretty well. I think it will be available about the end of the
year. I will not be free. It will be supplied with a commercially licenced
Asterisk.
Hi everybody,
I would like to know a rather basic thing: What do you use SS7 for ?
I
On Thu, 4 Nov 2004, Seth Remington wrote:
The only remaining problem comes from the fact that I have an ADSL
connection at my home and the pppoe changes my IP address every once in
a while. I have set my sip.conf 'host=' command up with a dyndns
hostname and everything works when I start
On Thu, 4 Nov 2004, Andrew Thompson wrote:
Unfortunately not. I have the same problem and the same solution here. You
really have to do a restart of asterisk. I think the reason that asterisk
does not always lookup the IP Adress for the DDNS-Hostname is performance.
But it would be nice
On Fri, 13 Aug 2004, Altus Snyman wrote:
Good day all
I'm using sip protocol and a openline4 card.If I dial out of the pstn
and hangup a answered call it does not disconnect the connection.It
shows there is still a call on the external phone I called but on my
side its says i'm not
On Tue, 10 Aug 2004, Scott Laird wrote:
Why stop there--you can beam pre-recorded messages to phones without a
person or phone line ever being involved. You could send hundreds of
[...]
That's right. Here in Hamburg, Germany one day before our elections my
phone rang and there was a
On Mon, 9 Aug 2004, hank wrote:
can you use .wav files or does it have to be gsm?
I thought about that possibility too. Unfortunately I was not
successful when putting .wav Files instead of gsm Files in
/var/lib/asterisk/sound
Which Formats will * accept and what extensions may be used ? Is
On Sat, 7 Aug 2004, Beierlein Moritz wrote:
Hi
I want to set up a Asterisk system for homeuse with SIP 2 ISDN.
I want to register up to 25 Sip Accounts at my Provider and I want to use up to 10
SIP Phones at Home and one ISDN Phone.
Do you think a Celeron 466 MHz machine with 128MB Ram and
Hi everyone,I wrote some days ago:
I upgraded my source tree from CVS some hours ago because the stable
version does not hangup SIP-Calls correctly.
Unfortunately now I am confronted with nearly random crashes stating
Floating Point Exceptions and many error messages
channel.c:1650
On Wed, 4 Aug 2004, Bastian Schern wrote:
Hi *,
are there already some free German sounds for Asterisk?
Try here:
http://www.voip-info.org/wiki-Asterisk+sound+files+international
Christoph
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Hallo everyone,
at first I would like to say hello to anyone as I am new to this list
and new to asterisk which I find very fascinating.
I am currently using asterisk with the German SIP-Provider sipgate and
with my little ISDN-Line using the Modem-Driver vor I4L.
I upgraded my source tree
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