Hi guys. I just bought and configured a Snom 360 and have noticed that the
LCD is constantly flickering at a rate of ~10-15Hz (that's a guess).
Either way, it's very distracting. Has anyone else encountered this
before? Any solutions?
Cheers,
-- Nick
E: [EMAIL PROTECTED]
P: +61 7 5591 3588
F:
On 11/7/06, Christian [EMAIL PROTECTED] wrote:
Hi,
My messages to the list don't get through. This must be the tenth
message i am trying to send!
Please ignore this test message.
On Wed November 8 2006 13:08, Alex Robar [EMAIL PROTECTED] wrote:
They do get through. Messages you send to
- Original Message -
From: Nick Hoffman [EMAIL PROTECTED]
To: asterisk-users Mailing List asterisk-users@lists.digium.com
Sent: Tuesday, November 07, 2006 10:53 AM
Subject: [asterisk-users] Snom 360 flickering screen
Hi guys. I just bought and configured a Snom 360 and have noticed
On Sat November 4 2006 06:43, Steve Murphy [EMAIL PROTECTED] wrote:
I was encouraged to post this notice on both asterisk-users and
asterisk-dev;
sorry if this is overkill, but it **is** applicable to both communities.
Since the report is fairly large, has a pretty graph, and the whole bit,
On 9/28/06, Simone Ricci [EMAIL PROTECTED] wrote:
Adi Simon ha scritto:
Hi,
Did anyone actually manage setting up a single SER with multiple
Asterisk boxes?
I particulary have a problem of keeping the session alive and by
that I mean directing
all the following sip messages
On Mon September 25 2006 11:05, Bart Fisher [EMAIL PROTECTED] wrote:
Hmm, this must not be installed:
# locate irqbalance
# /usr/src/kernels/2.6.9-34.0.2.EL-smp-i686/include/config/irqbalance.h
How do I install this?
Bart
I'd run `apt-get install irqbalance`, but you'd do something with
On Sat September 23 2006 06:14, Bob Amen [EMAIL PROTECTED] wrote:
snip
which sets the TOS bit on all IAX, SIP and RTP packets. Using iptables
means that we can set up our rules on the router without using ACLs. Our
Cisco Cookbook (http://www.oreilly.com/catalog/ciscockbk/) has a nice
section
On Tue August 29 2006 04:39, Greg Boehnlein [EMAIL PROTECTED] wrote:
On Mon, 28 Aug 2006, Andrew Kohlsmith wrote:
On Monday 28 August 2006 13:02, Greg Boehnlein wrote:
I've pushed over 1,000 concurrent calls this way using the SIPP
program for SIP performance testing. There was some
Joey McDonald wrote:
Have you looked to see if they're being logged to
/var/log/asterisk/full ? That would be much easier to detect.
--joey
Hi Joey. What is /var/log/asterisk/full ? I've never heard of it, and don't
see it in my /var/log/asterisk/ .
Cheers,
-- Nick
e: [EMAIL
Hi guys. I just stumbled upon
http://www.voip-info.org/wiki/index.php?page=Asterisk+G.729+Licensing and
read the section titled Warning. I'm a bit confused now. Are you
violating the GPL (or any other license) if you sell a computer with
Asterisk and a G.729 license installed?
Cheers,
-- Nick
On Fri July 21 2006 18:33, Woodoo People .pGa!
[EMAIL PROTECTED] wrote:
don't forget the following:
if canreinvite=yes, asterisk will NOT stay in mediapath, so, it going to
ask both parties to negotiate codec, and say hello to the stream. (if
both parties supports g729, and can negotiate it,
--
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On Thu June 22 2006 09:25, shadowym [EMAIL PROTECTED] wrote:
Looks like I am going to be doing my first serious commercial install of
FreePBX. I DO mean serious. They are willing to put up with a few
glitches initially which is why I have decided they will be a good first
client. I have
On Tue June 20 2006 08:23, Daniel Salama [EMAIL PROTECTED] wrote:
I have been reading about integrating Asterisk with SER to help
Asterisk deal with large volume of registrations (mainly). I was
planning on fronting Asterisk with SER for that purpose. Not that I
have the traffic at this
Hi guys. I'm trying to disable all debug output, but am not having any
success:
[EMAIL PROTECTED]:~ sudo asterisk -r
Asterisk 1.2.8, Copyright (C) 1999 - 2006 Digium, Inc. and others.
..snip...
certain conditions. Type 'show license' for details.
On Wed June 7 2006 10:12, Eric \ManxPower\ Wieling [EMAIL PROTECTED]
wrote:
Doug Crompton wrote:
Ok well I am not crazy! This seems like such an important issue I am
not sure why it has lasted for so long. DTMF is the backbone of
everything we do here. Without it we would not have calls!!
On Thu May 11 2006 07:16, Jason Adams [EMAIL PROTECTED] wrote:
Hey Everyone,
We are in the process of reviewing headsets for use with our GXP-2000s.
I'm looking for some feedback as to which headsets people are using, the
pros and cons of those headsets, and if they would recommend them to
On Wed April 26 2006 16:31, Jon Farmer [EMAIL PROTECTED] wrote:
JP Carballo wrote:
Yes, certainly, through deadagi.
I just have one question though, why reinvent the wheel?
There are prepaid systems that work with asterisk.
I have yet to find a prepaid system that allows multiple
What does zero-dialed mean with respect to zero-dialed emergency call
routing service?
-- Nick
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If you receive this email by mistake, please notify us and do not make any
use of the email. We do not waive any privilege, confidentiality
On Tue March 28 2006 10:33, Kevin P. Fleming [EMAIL PROTECTED]
wrote:
Kerry Garrison wrote:
Does anyone know if a TDM2400 will fit into a Dell 2850?
It will fit, but you will need to solve the power supply problem if you
intend to use FXS ports on it :-)
Why is that? Do FXS ports draw
On Sat March 25 2006 18:06, Nick Hoffman [EMAIL PROTECTED]
wrote:
Hi guys, I've been using a Polycom IP 301 for a couple of weeks now
and find that it's extremely slow for configuring. For instance, it
takes several minutes to boot up, apply any changes via the web
interface takes
Hi guys, I've been using a Polycom IP 301 for a couple of weeks now and
find that it's extremely slow for configuring. For instance, it takes
several minutes to boot up, apply any changes via the web interface takes
at least a minute, etc. Is this normal behaviour? Is there anything that
can
Hi guys, is the method described at
http://www.voip-info.org/wiki/view/Asterisk+bounty
the only way to create an Asterisk bounty? If not, please let me know what
other ways there are for creating a bounty.
Cheers!
-- Nick
e: [EMAIL PROTECTED]
p: +61 7 5591 3588
f: +61 7 5591 6588
If you
On Wed March 15 2006 17:31, Martin Joseph [EMAIL PROTECTED] wrote:
Great, now your your ulaw users can hear shitty audio all the time.
Martin, let's keep this mailing list polite as free of negativity as
possible. There's no need to bite at people. You could easily have said
the same thing a
On Tue March 14 2006 10:15, Gabriel Afana [EMAIL PROTECTED] wrote:
Haha, Buuurn.
I have the book on my desk too. I am going to go step-by-step to setup
DUDNi. If I can get it working, I'll post step-by-step details for you
guys on how to do it yourself.
- Gabe
Hi Gabe. Are you referring
On Thu March 9 2006 02:14, Ron McCarthy [EMAIL PROTECTED] wrote:
I havent got any mails since 2:42 this morning..usually i get at least
the normal 10-15 a hour, if someone gets this can they reply?
Thanks!
Ron
Hi Ron, I've received many emails from the mailing list over the past 24
hours.
On Thu March 9 2006 03:43, Warren Burstein [EMAIL PROTECTED] wrote:
I have a Linksys PAP2. Identical setups for the two channels in both
the unit and in Asterisk. In particular, both channels enable g729 and
set it as the preferred codec, and have disallow=all and allow=g729 in
sip.conf.
On Thu March 9 2006 08:52, [EMAIL PROTECTED] wrote:
Hello,
You can use ser as an outbound sip proxy and asterisk
as a register server .
Your sip agents will get MWI, ...
Harry
Hi guys. With that solution, remember that Asterisk can handle a fraction
of the number of registrations that
Hi guys. Without having a FWD account, can Asterisk redirect calls to FWD?
For instance, an extension behind Asterisk dials 99751234, and Asterisk
says that starts with 99. let's strip off the 99 and call 751234 at FWD,
IE: sip:[EMAIL PROTECTED]:5060.
Is that possible, or would services such
On Thu March 2 2006 19:22, Stefan-Michael. Guenther (in-put GbR)
[EMAIL PROTECTED] wrote:
Hi,
what about the Asterisk PBX Manager:
http://www.thirdlane.com/opensource.htm#manager
It's based on webmin and well documented.
Stefan
Hi Stefan. What documentation have you found for
On Thu March 2 2006 19:32, John Joseph [EMAIL PROTECTED] wrote:
thanks for this info, I have some doubts
If I had already installed AMP , but I want to have
PBX Manger installed , so that I can use both of them
and compare each other
will it cause problem if I install PBX manager
,
On Tue February 21 2006 18:53, Dinesh [EMAIL PROTECTED] wrote:
Hello all,
I want to sniff all these info to test a sip ip phone talking to a
asterisk server. I have used tcpdump, but It just shows the
UDP, length: 602
Anyway to see the sip uri. Host info?
Regards,
Dinesh.
Hi Dinesh.
Hey guys. If my Asterisk box connects to the PSTN using SIP and IP over
ethernet and doesn't require any authentication, what sort of a trunk
would need to be created?
Thanks,
-- Nick
e: [EMAIL PROTECTED]
p: +61 7 5591 3588
f: +61 7 5591 6588
If you receive this email by mistake, please notify
Hi guys. I've found a fair bit of information with regard to how to setup
Asterisk to send and receive calls through SER (SIP Express Router),
however I can't figure out what information is up-to-date. Many of the
examples suggest using insecure=very for the SER entry in sip.conf, but I
On Mon February 13 2006 02:15, [EMAIL PROTECTED] wrote:
Hello,
I have an IAX2 trunk like this running well with IAX2 and SIP users
mixed at each side.
Runing like a charm :-)
Don't forget to add username definition from this example.
To avoid too much load for your CPUs with transcoding,
When using Asterisk and SER together, should SER place calls to the PSTN,
and Asterisk only deal with special features such as voicemail, queues,
autoattendants, etc? Or should SER be used ONLY as a proxy/registrar, and
all calls be routed to Asterisk so that Asterisk places the calls to the
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