,
Matt Riddell
Director
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subscribing to the bugtraq mailing list. Most days have
security vulnerabilities in most systems. If your box is 100%
inaccessible from the Internet then sure, but if it is accessible then
updates are a bit of a must.
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Director
then it will play ringing tones even if the
phone is busy.
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that was what the question meant.
We're located in New Zealand but install systems worldwide (USA
included). Heh probably don't need the bracketed bit seeing as USA
should now become part of worldwide again :D
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Matt Riddell
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there a boostringer option for modprobe?
Not sure if that's what you're looking for though.
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Matt Riddell
Director
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,
Matt Riddell
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is not given, current context is default
; If family is not given, family of 'extensions' is default
Heh, just read back through your first post.
Think your problem is the family in extconfig.conf
Should be extensions not extensions.conf
I.E. extensions = odbc,asterisk
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.
Use the Local channel:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Local+channels
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accurate :)
According to google's figures (July 2007 est.)
USA: 301,139,947
World: 6,602,224,175
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On 5/11/2008 5:57 a.m., andrea wrote:
Dear List
I'm asking if there is a small hardware already implemented with
software that just do traffic Shaping QoS?
I have found *D-Link DI-102 VoIP QOS Adapter Packet Prioritizer
*But it does not look available in Italy !
Regards Andrea
The Linksys
work for AgentLogin().
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Riddell
Sent: August 30, 2008 6:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Transfers on AgentLogin()
What did
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Matt Riddell
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things was never answered.
How do I do an assisted transfer on AgentLogin()? I don't use zaptel, it's
just pure SIP/VoIP.
Help please.
Thanks,
Mark.
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Matt Riddell
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for mobile | mobile.google.com
Matt Riddell
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asterisk-users mailing
. It is running Zaptel 1.4.11 and Asterisk 1.4.18.
Hi,
I had a similar problem a while back with a 2N box. For some reason
there was a very loud click which caused the echo canceller to lose the
plot.
Have you tried disabling the echo can on that port?
- --
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Matt Riddell
Director
the plantronics (plugged into the handset thing) you still need
to take the phone off the hook.
Unlike the snom 360 where there is a separate socket.
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Matt Riddell
Director
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-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Darrick Hartman wrote:
Matt Riddell wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Paul Hales wrote:
That's a good question - the plantronics are available with
interchangeable ends - which makes them easy to move between different
transmission not supported by channel
At this moment, text is supposed to be 7 bit ASCII in most channels.
The option string many contain the following character:
'j' -- jump to n+101 priority if the channel doesn't support
text transport
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-BEGIN PGP SIGNED MESSAGE-
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Matt Darnell wrote:
Does anyone know of a bandwidth test that tests the upload with the download?
apt-get install iperf
man iperf
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to ulaw/alaw.
Alternatively, make sure your call is in GSM.
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,
Matt Riddell
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-BEGIN PGP SIGNATURE
and no.
I've seen pretty major problems show up in mtr only to be told that the
provider is dropping icmp in times of high load.
We moved our monitoring to sip/iax times, but you only get a point to
point stat in that situation.
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Matt Riddell
Director
recently to get it
right; it appears that, at least in the satellite case, things may have
gotten a little too tight...
If this rings a bell for anyone, any insight would be appreciated.
These calls sip or iax?
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Matt Riddell
Director
of cable, it should help a lot.
The APC UPSes we use come with a warranty for equipment attached to
them, as do the belkin filters.
Check if yours does any you may be able to get the units replaced at no
cost.
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Matt Riddell
Director
.
It checks the DB every few seconds, and updates the credit based on how
long the person has been talking and the rate to that destination.
They can add more money at any time, and the DB value is updated.
When they run out of credit the call is killed via the manager.
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Matt Riddell
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Loic Didelot wrote:
Hello,
I just got my Xorcom BRI bank. Seems to work. But I have some questions.
Is anyone getting good values using zttest?
Is it plugged into the BRI?
Is it the sync master?
i.e. xpp_sync
- --
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Matt Riddell
to see,how i
can verify that i have the gsm bug.
Well, if you have gcc version 4.2.x (you can check with gcc -v)
there's a good chance this is the problem.
Just do:
export CC=gcc-4.1
export CXX=gcc-4.1
./configure
make
Works for me.
- --
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Matt Riddell
Director
-BEGIN PGP SIGNED MESSAGE-
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Jason Dixon wrote:
On Tue, Jul 08, 2008 at 11:00:43AM -0400, Jason Dixon wrote:
On Tue, Jul 08, 2008 at 12:10:05PM +1200, Matt Riddell wrote:
Action: Command
Command: show queue my_queue_name
ActionID: my_queue_name_12345
This does not appear
, Nicolas Gudino, and I will leave off the fifth as to not leave
anybody out ;)
Me, me, me!
:D
Or, Kevin, Russell, Olle, Josh, Critch (although he's been pretty quiet
lately), I guess the list goes on.
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Matt Riddell
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.
Then they went and spoke at Whitehouse dinners and stuff and kinda
disappeared.
In those days I was heavily into greyhat and IDS systems, but I'm pretty
sure it was common knowledge.
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Matt Riddell
Director
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, it wouldn't ask you to press 1.
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Matt Riddell
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(Matthieu)
exten = s,n,ChanIsAvail(SIP/605,s)
Work, because line 7 exist
I use Asterisk 1.4.18
That's what it's supposed to do - maybe you are referring to the
deprecated +101 jumps?
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Matt Riddell
Director
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yellow and one black
ethernet socket. (WAN/LAN)
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Matt Riddell
Director
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the Authenticate
application?
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Matt Riddell
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Florian Hackenberger wrote:
Hi!
I'm using asterisk 1.4.17 with twinkle and a custom phone based on
iaxclient 2.0.2 and I'm struggling a bit with DTMF and features.conf.
Maybe the feature digit timeout?
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Matt Riddell
acknowledge the call, and log that fact. (Basically an automated soft
phone). I found some info on how to do this here:
Please also attach the call file you are trying to use.
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Matt Riddell
Director
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member won't just hit Ignore on their phone
and send it directly to voicemail.
You'd probably want to look at using the local channel and the followme
application + /etc/asterisk/followme.conf
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Matt Riddell
Director
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-BEGIN PGP SIGNED MESSAGE-
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Tilghman Lesher wrote:
On Thursday 03 July 2008 00:27:00 Matt Riddell wrote:
Tilghman Lesher wrote:
I find that a good number of people are using . in a pattern in
situations that are entirely unnecessary (such as local numbers). The
only place
/asterisk/outgoing
and
date
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Matt Riddell
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.
Bear in mind that if you want to do show queues as opposed to show queue
and you are using a realtime queue which has not been used, you will
need to type show queue my_queue_name first before it will show up in
show queues.
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Matt Riddell
Director
on if they are
available for contracting work.
It is a work in progress and is being completed solely outside of office
hours, so the development pace is somewhat slow, but I'm attempting to
add new features every evening.
Enjoy! :-)
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Matt Riddell
Director
was IAXtel
but that does not seem to be the case.
I think you're referring to FWDOUT.
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Matt Riddell
Director
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the
other 100%?
If so, what echo canceller are you using?
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Matt Riddell
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http
in sight.
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Matt Riddell
Director
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had the same problem on a 410p, and ended up fixing it by changing to
a lower suppression.
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Matt Riddell
Director
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Tilghman Lesher wrote:
On Wednesday 02 July 2008 21:59:02 Matt Riddell wrote:
Tilghman Lesher wrote:
On Tuesday 01 July 2008 14:27:26 bilal ghayyad wrote:
Can anyone advise how to increase the waiting time to consider the
number is dialed
a period in my dialplan is for when I'm dialling an international number, when
I don't know in advance how long the number should be. A little bit of
planning goes a long way.
Here we have variable length cellphones as well as international.
- --
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Matt Riddell
Director
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Tzafrir Cohen wrote:
On Wed, Jun 04, 2008 at 04:06:28PM +1200, Matt Riddell wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Tilghman Lesher wrote:
On Tuesday 03 June 2008 10:12:58 Todd Reese wrote:
Hi All,
I'm stumped on this and I
them?
export CFLAGS=
or will they get overwritten?
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Matt Riddell
Director
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get someone who does get it to use one of the phones from the
people who don't get it, do they still get it (i.e. maybe the people who
don't get it just aren't noticing it).
What is the RTP packet size in both situations? Should be 20ms, but may
be 30ms.
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Matt Riddell
Director
-BEGIN PGP SIGNED MESSAGE-
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Have you tried disabling highpriority=yes in asterisk.conf?
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Matt Riddell
Director
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:
timeout
busy
congestion
You can see the gist of how it works - you could add extra cause codes
if you wanted.
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Matt Riddell
Director
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-Addons package.
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Matt Riddell
Director
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starting up zaptel if you have no hardware
installed.
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Matt Riddell
Director
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Razza wrote:
On 10/03/2008, Matt Riddell [EMAIL PROTECTED] wrote:
Has anyone done any integration with this?
All I know so far is that it appears to use some non standard form of SIP.
Any pointers?
If you are looking to use Enterprise Voice
you :D
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Matt Riddell
Director
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Hans Witvliet wrote:
On Mon, 2008-03-10 at 16:05 +1300, Matt Riddell wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Has anyone done any integration with this?
All I know so far is that it appears to use some non standard form of SIP
- rx fax - tiff
The above crashes every time.
If no one else has seen it/has a fix for it I can do a back trace.
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Matt Riddell
Director
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-channel_status('SIP/231'));
I may be off base here as I haven't used the channel status via AGI, but
SIP/200 seems like an account to me rather than a channel.
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Matt Riddell
Director
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-BEGIN PGP SIGNED MESSAGE-
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[EMAIL PROTECTED] wrote:
exten = s,2,BackGround(/var/lib/asterisk/sounds/en/vm-instructions.gsm)
Drop the .gsm at the end of the filename. Asterisk will chose the best
format for the call.
- --
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Matt Riddell
Director
() are not executed if the call was completed. I now handle the
ANSWER status detection in the h extension.
Or use the g option to the dial command:
g- Proceed with dialplan execution at the current extension if the
destination channel hangs up.
- --
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Matt Riddell
Director
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Has anyone done any integration with this?
All I know so far is that it appears to use some non standard form of SIP.
Any pointers?
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Matt Riddell
Director
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second the recommendation on SuperMicro - had nothing but goodness
from them.
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Matt Riddell
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http
.
Bear in mind that any open port will probably get hammered and at some
stage will probably have a security vulnerability.
Don't load modules you don't need (modules.conf)
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Matt Riddell
Director
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./configure
Check that you have gcc et al installed.
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Matt Riddell
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useful
where you can't get a BRI here in New Zealand.
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Matt Riddell
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there is a memory leak?
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Matt Riddell
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but using safe_asterisk instead of the scripts.
The benefit of the make config stuff is that you can then do chkconfig
asterisk on to make Asterisk start up automatically on boot.
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Matt Riddell
Director
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the Asterisk manager if you wanted too.
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Matt Riddell
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much bandwidth or data transferring is taking place in
a call that is network traffic. I want to monitor the asterisk server.
Use your C program to access the Manager - that's what we do for our new
predictive dialer - everything is ANSI C including Manager Parsing.
- --
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Matt
);
} else
- - res = transmit_response_with_sdp(p, 200 OK,
p-initreq, XMIT_CRITICAL);
+ res = transmit_response_with_sdp(p, 200 OK,
p-initreq, XMIT_RELIABLE);
}
ast_mutex_unlock(p-lock);
return res;
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Matt Riddell
Director
machines and not others.
We have mrtg showing number of reigistered peers and uptime/reload time.
It normally happens around 24k of uptime and requires a restart,
although we have other machines which have been up for ~4 years without
problems.
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Matt Riddell
Director
-BEGIN PGP SIGNED MESSAGE-
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Have you tried with AGI Debug on?
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Matt Riddell
Director
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in
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Matt Riddell
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-BEGIN PGP
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Andrea Cristofanini wrote:
yes i see
you have to enable in ADVANCED SETTING
Challenge Response on Phone: = OFF
Regards
Sweet as, cheers man.
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Matt Riddell
Director
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,
password, mailbox and ringtone.
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Matt Riddell
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?
Um, wouldn't they be unavailable if someone just left them a voicemail?
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Matt Riddell
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http
channel.
Can you give a tip, where can I check If I miss something.
What kind of echo cancellation are you using?
* Inbuilt
* HPEC
* Hardware EC
* Oslec
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Matt Riddell
Director
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-BEGIN PGP SIGNED MESSAGE-
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bilal ghayyad wrote:
Hi;
What is the difference between free version and
business version?
Are the differences at voice quality level or it is a
matter of add on features (luxury)?
Just additional features.
- --
Kind Regards,
Matt Riddell
in it rings you as normal.
If a DDI call comes in, it seizes the line, then dials the extension
(last four digits of the DDI).
I was running a site off a GXW4008 gateway, but it had problems (I'm
assuming with the bidirectional ringer voltages).
We changed them over to BRI :)
- --
Kind Regards,
Matt
if that's my problem? I have a 1ghz Celeron and I think I'm
using the 686 build. Would that be the correct match?
Yeah I think so, but give Digium a yell and get them to log in and check
it out.
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Matt Riddell
Director
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binary and not the i686 one.
I called Digium, they logged in, sorted it out, and everything works
fine now.
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Matt Riddell
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to sync
ZX81 heh cool
russellb the public mirror is up to 12800ish ...
russellb it's going to take ... a long time
russellb all night, at least
russellb if you want like ... 1.2.0, that's probably available :)
mvanbaak woot
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Matt Riddell
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to VoiceMail.
If the volume is ok for VoiceMail then the problem is the microphone
volume on your phone.
I normally do ztcfg -v from the command line and module reload
chan_zap.so from the Asterisk console, although it may be the case that
both are not required.
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Matt Riddell
Director
,
Matt Riddell
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-BEGIN PGP SIGNATURE
to the dialplan and
| then test AGISTATUS but it looks like I'm going down the wrong path.
|
| Any suggestions?
Why don't you just set a variable from the AGI and then test for it in
the dialplan
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Matt Riddell
Director
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as the Asterisk-Video mailing list.
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Matt Riddell
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-BEGIN PGP SIGNED MESSAGE-
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Kristian Kielhofner wrote:
On Jan 9, 2008 11:44 PM, Matt Riddell [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Erik Anderson wrote:
On Jan 9, 2008 9:40 PM, Matt Riddell [EMAIL PROTECTED] wrote:
Heh yeah that's what I
passes ---
Best: 0.00 -- Worst: 100.00 -- Average: 100.00
Yeah that's what I thought. Am just trying to remember what caused it
though. Maybe Tzafrir will chime in :)
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Matt Riddell
Director
___
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FQDN's, and would like to see about doing this either over sip or
IAX...
http://www.voip-info.org/wiki-Asterisk+-+dual+servers
- --
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http
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Hi,
Does anyone know of a cheap (very cheap) dual port traffic shaping box
(i.e. sub $100) that can be configured for IAX/SIP?
- --
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com
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Erik Anderson wrote:
On Jan 9, 2008 8:33 PM, Matt Riddell [EMAIL PROTECTED] wrote:
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Hi,
Does anyone know of a cheap (very cheap) dual port traffic shaping box
(i.e. sub $100) that can be configured
with analogue and PRI cards, and
the new FXOtune from mattf looks like it should be really nice!
- --
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News
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Erik Anderson wrote:
On Jan 9, 2008 9:40 PM, Matt Riddell [EMAIL PROTECTED] wrote:
Heh yeah that's what I was thinking of doing. What's the traffic
shaping like? Can I specify max bandwidth etc or use hfsc shaping?
DD-WRT will do both HTB
a smoother
which fixed it, and use that on all sites with IAX phones.
- --
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http
to be some introduced delay
which has meant we had to install OctasicEC for echo can as the on board
hardware one wasn't doing its job.
- --
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http
the brackets on answer and echo but I usually type
that way and then add options. :-)
- --
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html
go.
Oh, you might want to try a downwards expander instead (a noise gate but
with ratio as well as threshold).
- --
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com
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Shane D wrote:
no-ip.org appears to want to charge me money... Is there a free alternative?
Dyndns.org
- --
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end
Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
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