Displaying 20 results from an estimated 7000 matches similar to: "Converting an audio file to a ".gsm" format"
2007 Aug 04
1
Connecting two Asterisk servers with a framerelay connection
What modules do you want on it?
Yours,
Michael Munger, dCAP
404-438-2128
michael at highpoweredhelp.com
________________________________
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of MOSBAH
ABDELKADER
Sent: Saturday, August 04, 2007 3:16 PM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Connecting
2007 Aug 09
8
How to use OpenVPN with Asterisk
Hello,
I want to create a VPN between two Asterisk servers using OpenVPN.
How to configure Asterisk and OpenVPN to do that.
Thanks.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070809/ddaed76b/attachment.htm
2010 Feb 11
13
SIP tunnel
Hello,
I have the following situation: A firewall is blocking all SIP and RTP
traffic in the side of some of my clients. My clients cannot change settings
of the firewall.
I need to solve this problem and I need some help from you.
I have this idea: implement a SIP user agent which does not use well known
SIP ports (uses http port 80 for example) and use other ports that are not
blocked
2009 Jul 03
7
Asterisk capacity
Hello,
What is the maximum number of simultaneous calls supported by asterisk.
thks
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090703/0794c554/attachment.htm
2010 Mar 08
3
Calculating R Factor and MOS metrics for VoIP
Hello All,
MOS and R factor are the two QoS parameters used to estimate VoIP call
quality.
I have found that they are calculated from other metrics like jitter,
latency, packet loss,...etc. But, haven't found any formula or arithmetic
rule to calculate them.
Do you have an idea about their formulas or an open source that calculates
them. Is it possible to interpret them from wireshark.
2007 Aug 10
0
asterisk-users Digest, Vol 37, Issue 46
I've found OpenVPN to be easy to configure and very robust. It has a
zillion options, but they are just that - options. I haven't used it for
VoIP, but I've put it to good use doing layer 2 bridging which has
eliminated many problems with certain programs traversing NAT and
load-balancing routers. I can't think of any reason why it would not
work well with Asterisk.
> On
2010 Jul 22
3
My Switch is being attacked using sip scanner tool (Service Abuse Attack)
An attacker is scanning my Asterisk Switch to gain illegitimate access to
VoIP call functionality.
Using a sip scanning tool, *it* sends REGISTERs with random identities. And
when it discovers one identity subscribed in my switch, it tries to
authenticate with random passwords using this user name.
For the moment, I have replaced this account. And also blocked the IP it has
used but each time
2007 Aug 10
2
Locating Asterisk documentation after installation
Hello all,
After installing Asterisk, i have installed the docs by "make progdocs".
But i don't know where to locate this documentation.
please Help.
Thanks.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070810/ceb95948/attachment.htm
2010 Aug 01
3
fail2ban does not work for my asterisk installation
The failregex statement in my jail.conf file is:
*
failregex* = NOTICE.* .*: Registration from '.*' failed for '<HOST>' - Wrong
password
NOTICE.* .*: Registration from '.*' failed for '<HOST>' - No
matching peer found
NOTICE.* .*: Registration from '.*' failed for '<HOST>' -
Username/auth name mismatch
2009 Jul 24
4
Asterisk on OpenWRT
Hello,
Did anyone succeeded in installing Asterisk on OpenWRT system.
pls help.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090724/aee7ee12/attachment.htm
2009 Apr 08
2
Asterisk Trunk billing
Hello,
I have a problem with Asterisk trunk billing. I have bought some number of
trunks from a VoIP provider with his own rates. I am planning to sell some
of these trunks to my clients with my own rates. The problem is: how to
process this trunk, Can I process it as a normal SIP/IAX client (if yes how)
and apply my billing rates to it.
Thanks.
-------------- next part --------------
An HTML
2004 Jun 14
4
<<< GSM Audio Files >>>
Hello:
Thanks for the input so far.
Heres the issue--
This is a production environment-- where many people "touch" the files.
ie-- The audio engineer is a freelancer who wants to master the files at the
highest quality TO HIS EAR and experience-- He knows NADA, Not a thing about
SOX-- but is a ProTools GURU.
The SOX resampled files work on our asterisk box-- but I gotta put someone
2007 May 27
2
Asterisk 1.2.18 problem
hello,
I have installed asterisk 1.2.18 in suse 10.2. After typing asterisk in the
terminal command line (i don't think that asterisk runs when doing this) i
type "asterisk -r" but the response" is "Unable to connect to remote
asterisk (does /var/run/asterisk.ctl exist?)".
how to solve this.
thanks.
-------------- next part --------------
An HTML attachment was
2010 Apr 16
2
SS7 over an FXO interface
Hello,
Is it possible to transfer ss7 signaling over an FXO interface.
I need to setup an ss7 test system composed by two Asterisk based IP-PBX
systems with anlog interfaces only (FXO and FXS). I want to know if it is
possible to connect the two IP-PBX as following:
- FXS interface in PBX1 -----------------> connected to
-----------------> FXO interface in PBX2 =============>
2018 Oct 21
4
Configure Ubuntu Server 16.04 for icecast2
Hi,
Thank you so much for your reply,
I've a dedicated server in OVH, where I have done speed test for the server
:
*bkf at xxxxx ~> speedtest-cli Retrieving speedtest.net <http://speedtest.net>
configuration...Retrieving speedtest.net <http://speedtest.net> server
list...Testing from OVH SAS (x.x.x.x)...Selecting best server based on
latency...Hosted by fdcservers.net
2005 Jul 06
4
converting windows .wav to .gsm
HI ALL;
I have problem converting a windows .wav file to .gsm format by Sox.
Could anyone help.
Cheers,
Mohammad
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050706/3408bfd5/attachment.htm
2005 Feb 04
2
gsm audio files
Hello, anyone knows if exist the audio files in spanish??
or how can i record the voice in gsm extension???
can i play for some announce a random file??
TIA
Edgar
2010 Mar 24
5
Asterisk 1.6 and OpenVPN RTP problem
Hello All,
I have installed Asterisk 1.6 with openVPN in the same machine. I have set
up a VPN connection between 2 SIP clients and Asterisk using x-lite.
The 2 clients connects to Asterisk. SIP signaling goes ok over the vpn
tunnel.
When attempting to make a call between the clients, the siganling part of
the call goes well. But, when the call is set up, some RTP packets are
exchanged at
2004 Jun 14
3
<<< GSM AUDIOFiles >>>
Hello:
I would like to produce some GSM Prompt audio files for a Telephone
Directory Project-- and have hired a freelance audio engineer to record, and
edit the actual files--
However the GSM files he gives me to upload into asterisk DO NOT work when
played back throgh "Stream File" or "Get Data" in my agi. It seems that
there may be more than one GSM file type (with
2003 Jun 15
5
.gsm files
Hi guys,
Being a true Linux geek, I've never been too much into sounds or sound
files other than a few .mp3 songs I got. My question is pretty
straightforward and simple. I see that the music format of choice for
asterisk is .gsm. What can I use to listen to files in .gsm format and
what is the most effective way of recording files into .gsm format?
The last part of the question is