Voxbone works correctly, no problem, the only problem is that you need
to spend a minimum amount of 500€/month to open an account...
Best regards,
Olivier
Le 3/06/11 11:58, randulo a écrit :
On Fri, Jun 3, 2011 at 11:28 AM, devr devrd...@gmx.com wrote:
I am thinking about using numbers
Well,
About sipvicious, just put a kamailio in front of asterisk and just drop
all messages with user agents corrreponding to these messages.
Spivicious first send options messages, read the user agent and drop if
it's corresponding to one of the user agents well known to be used.
In
Neither RTPAUDIOQOS or any of the TRP qos facilities works correctly on
asterisk like channel(...,...), nothing you can trust.
Just forget it.
Olivier
DHAVAL INDRODIYA a crit:
thanks for your reply,
i want to know about meetme recording while i Use SIP as meetme user i
got RTPAUDIOQOS
Hi all,
With Asterisk 1.6.1.6
Trying to have statistic concerning Rtp audio quality, I use
${CHANNEL(rtpqos,audio,all)}
having also tried AUDIORTPQOS and ${CHANNEL(rtpqos,audio,...)}
Sometimes, it works and I have results.
Most of the time I get strange or no results even when the call was
yum search mysql client
yum install 'TheClientYumHasReturnedForYourSystem'
Olivier
Cyprus VoIP a crit:
I think that the missing component is mysqlclient, but when i "yum
update mysql", it does nothing.
Anyone know how to download the RPM? I'm using CentOS 5.3.
Thanks.
No way to authenticate from Ser, it's a proxy...
Just make an entry in your asterisk sip.conf with the IP of SER and no
authentification.
Do the necesssary stuff in your extension.conf to identify and bill your
client.
Olivier
Ishfaq Malik a écrit :
Hi People
We have a client who want to
a register command, is this true?
Ish
hh174 wrote:
No way to authenticate from Ser, it's a proxy...
Just make an entry in your asterisk sip.conf with the IP of SER and no
authentification.
Do the necesssary stuff in your extension.conf to identify and bill your
client.
Olivier
Ishfaq
this is a well-known change. If you use the latest Snapshot of
Asterisk-Java 1.0.0 (http://asterisk-java.org) the issue should be resolved.
=Stefan
Alex Balashov wrote:
Appearances suggest that some part(s) of the AGI protocol changed
between 1.4 and 1.6.
hh174 wrote
events.
hh174 wrote:
In fact, I am not using Asterisk-java.
I use asterisk and fastagi to have a b2bua able to disconnect many
callers if their customer account is around 0.
Is that a bug to be reported?
How to solve that in Java without using Asterisk-java?
Kind regards,
Olivier
Why don't you setup a second server just for mysql and fastagi.
Fastagi can also be installed on your asterisk server with just a small
overhead, as fastagi will keep running in the background, you will not
have the overhead of creating new process at each call.
Olivier
Unni a écrit :
Hello,
I have a problem with fastagi.
In fact I have a fastagi written in Java.
Communcation between asterisk 1.6 and the server works correctly, except
when a 'HANGUP' is sent by asterisk...
In this case, the java server doesn't read the message.
I have tride with PHP, same result.
A ngrep
Fring works perfectly for me.
Tarek Sawah a crit:
Have you tried installing fring? i still like that app .. supports GREAT quality voice over EDGE and GPRS .. plus WIFI and 3G if available..
i tried it with Skype and it's great..
Asterisk and its great
Callcentric VoIP provider and it
off registration and
thus disable VoIP-communications ??
Jonas.
On Wed, 2009-07-01 at 22:11 +0200, hh174 wrote:
Reliably
Transmitting (no NAT)
and you are natted I presume (
Port 5060 is forwarded to the internal IP-address of my
Asterisk-server).
Another Belgian user
Reliably Transmitting (no NAT)
and you are natted I presume (
Port 5060 is forwarded to the internal IP-address of my
Asterisk-server).
Another Belgian user :)
Olivier
jonas kellens a écrit :
Hello List,
I'm having problems with registrating my Asterisk-server to the
Let me also know, I just have a business with Tamiflu :)
I need to contact 6.000.000.000 people to help them.
No spam, I promise.
Olivier
ContactTel Business a crit:
Lol
, simply lol, dont forget the super duper, top secret
patch ,everyone is hiding from you that makes
Not true for me, a restart, if not mandatory, is a good idea, let's say
weekly or more depending on the usage.
My side is asterisk servers as gateways with thousands of calls a day
(at least 10.000 minutes a day).
I have asterisk(s) running for a long time( 4 years and more), with ss7
links
Hello,
Using odbc voicemail and mysql, i have a problem.
After 12 seconds recording, asterisk stop recording and hangup.
I have changed the settings in voicemail.conf to allow 180 seconds but,...
Any hint?
Olivier
___
-- Bandwidth and Colocation
is that it only seems to works on older
Linksys
hardware revisions. How do I make sure I can get those?
Mike
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Sunday, April 05, 2009 15:30
To: oliv...@hh174
Linksys (cisco)WRT54GL and the tomato firmware.
5 minutes setup
Olivier
Mike a crit:
Thanksthe thing is I need many device (one for each of my hosted
customers) and I'd like this process to be as easy for non-techies as
possible, because some of those are technologically-challenged,
Hello all,
Probably a bad news for all...
The Undercompetent Olle E Johansson decided to leave the asterisk team
to create his own Voip server.
The server will be called Minisk (due probably to his poor competence in
Voip).
Following that, Digium decides to stop any development on Asterisk and
Voxbone have many more IPs than that, probably your calls are coming
from another IP.
As these calls are just for your internal calls, just remove this entry
in your sip.conf
Olivier (another Begian)
Johan Dindaine - Asterisk a écrit :
Hi every all,
since a few weeks I came back to asterisk
It's exactly the kind of mail with NO interest at all.
We all know that fax is an old way, for sure, but not all the users
want to go to email, that's why Fax is needed.
If T38 exist, it's to transmit faxes over IP because some users needs
it.
T38, is not at all a voice 'codec' but a way to
Asterisk 1.6 installed with last zaptel...
On cli, when typing zap show channels, I get No such command 'zap show
channels' (type 'help zap show' for other possible commands)
Help doesn't help, of course...
I have a zaptel conf on the /etc/asterisk...
Any Idea?
Olivier
Well,
Installed asterisk, libpri, zaptel,... trunk
Parameters seems ok for asterisk and ss7, linkset is ok
Problem is astersik doesn't matter about the sip messages sent to him,
Ngrep see the messages on port 5060 but astersik doesn't react...
Even sip set debug on doesn't give me any infos...
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