On 03/09/2011 02:57 AM, Dan Journo wrote:
could anybody suggest a usable doorphone and magnetic door opener
hardphone system for me, please? Of course should be connectable to
asterisk. I am in the EU, should be available here.
I would recommend using a normal doorphone, and connecting
The AstLinux Team is happy to announce the release of AstLinux 0.7.5
with options for both Asterisk 1.8.1.1 and Asterisk 1.4.36. More
information about the release is available on our website:
http://www.astlinux.org/content/astlinux-075-release
Direct links to the installation files are
On 12/03/2010 03:30 PM, Doug Lytle wrote:
Napoleón Ernesto López Espinoza wrote:
We're sorry, your call did not go through.
Any clues about this issue?
How about some output from your console when it fails?
It's would also be advised to use a much more recent version. Asterisk
1.4.17 has
Bruce,
AstLinux supports dhcp and dns as well as several vpn options including
openvpn.
You can download a live ISO image to test. http://www.astlinux.org
Darrick
On 11/08/2010 08:34 AM, Bruce B wrote:
Thanks for the input. I am looking to use it as a DHCP server as well.
And I also I want
Are your sound files being transcoded or played back in their native
formats?
On 04/21/2010 12:25 PM, bruce bruce wrote:
Hi Everyone,
I have a weired situation where calls in and out are proceessed all
right but when I dial *97 Asterisk is literally choking when it comes to
announcements
On 07/02/2009 10:14 AM, Tzafrir Cohen wrote:
On Thu, Jul 02, 2009 at 09:53:18AM -0500, JR Richardson wrote:
Hi All,
A couple of customers called complaining that folks were dialing into
their PBX trying to use the Directory to locate users, from a
Blackberry, and getting frustrated due to
Do you have 'canreinvite=no' in your sip.conf entry for this phone? If
not, you should.
On 06/18/2009 07:55 AM, Giorgio Incantalupo wrote:
Hi John,
I already have the ccd dir with the iroute (mandatory for routing to
pc/phone connected to vpn client). During the last test I could register
...there is something in the way the RTP packets are sent/received by
Asterisk and maybe it can be correlated to the missing audio.
Giorgio
Darrick Hartman (lists) wrote:
Do you have 'canreinvite=no' in your sip.conf entry for this phone? If
not, you should.
On 06/18/2009 07:55 AM, Giorgio
Barry L. Kline wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I remember someone wrote a great document concerning Polycom server
provisioning that provided a way to ensure that updates to the firmware
did not overwrite customizations. I'll be damned if I can remember
where I saw
Dean,
I'm using Zabbix to monitor network interfaces, storage, cpu load and a
few other things on several asterisk boxes. I'm just looking at adding
Asterisk specific monitoring. Simple things like sip registration is
pretty easy. Getting the actual status of zap-daddy hardware might be a
Michael Graves wrote:
On Mon, 9 Jun 2008 20:32:29 -0400, Matt Watson wrote:
On June 9, 2008 07:49:13 pm Joseph L. Casale wrote:
What type of PBX hardware do you have on-site? Also what make/models of
phones?
Michael/Darryl,
I do have a local asterisk box, which is why I am baffled. I am
Eve-Ellen Cole wrote:
I am thinking of going with a Dell PowerEdge 1950 ||| for a new
CentOS/Asterisk set up. It will have dual 2.33GHz processors, 16GB
memory, two 500GB hard drives (presumably mirrored). I also plan to get
a Digium TE220B to go with it. (a non-dell server is not an
Andreas van dem Helge wrote:
Anyone have a download link for 3.0 SIP firmware?
If you are going to say ask polycom or ask your vendor don't even
waste your time posting. I've asked the Nazis and they'll probably
take 1 week.
Suggest you get a different vendor then. I got a response from
Jonathan C. Bailey wrote:
We've been using D-Link DES-3028P and DES-3052P switches. They can
supply full power to EACH port unlike the Linksys switches we've
tried. They're also rock solid from our experience.
I echo that recommendation. The Linksys switches are probably the
loudest that I've
Alan Lord wrote:
Lenz wrote:
Hello list,
after spending the best part of an afternoon trying to build Asterisk on
an old EPIA VIA C3, I thought that writing a tutorial would make life
easier for future compilers:
http://astrecipes.net/index.php?n=356
I had never compiled Asterisk for
David Nedved wrote:
--- Darrick Hartman (lists) [EMAIL PROTECTED] wrote:
Do yourself a favor and upgrade a Asterisk 1.4 which has a proper
implementation of DTMF. It's likely your SIP provider upgraded to
something which does not recognize the DTMF tones from Asterisk 1.2.
I've upgraded
David Nedved wrote:
Hi All,
Anyone have any idea what could cause incoming calls on a SIP channel
to no longer be able to use DTMF? DTMF on incoming calls on zaptel and
on local SIP softphones and ATAs all work fine. Nothing gets
registered in the CDR or on the console in verbose level
John Faubion wrote:
are plenty of phones on the market which do SIP now - most
modern Nokias do. I use an E90 Communicator, but the E95 is
popular too, so I'm experimenting with using my mobile as my
one phone, via Wi-Fi/SIP when I'm in the home/office and
Out of curiosity, how do
Terry Wilson wrote:
Digium has released version 3.0.3 of its product registration utility.
This is the first version of the registration utility that is compiled
against the uClibc C library. A benefit of this transition is that the
register binary should run more consistently and
Doug Lytle wrote:
Evan Ruff wrote:
Since when is the users list a transport for calendar scheduling?
Since when are humans infallible? Randy made a mistake. He apologized
for it. Let's move on...
--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com
Tilghman Lesher wrote:
On Friday 15 February 2008 23:53:19 [EMAIL PROTECTED]
wrote:
Practically any manufacturer gives similar support including ssh'ing
in the users box.
Really? Which manufacturers, specifically, will allow you to call up, get
remote assistance, and help you get the
I told myself that I was going to stay out of this one, but since you
find this important enough to reply twice to the mailing list with the
same content, it must be worth my time to reply.
If you carefully read the thread, the person who replied from Rhino went
out of his way to NOT try to sell
Ricardo Carvalho wrote:
I had the same problem some time ago...
You got to install also this packages:
net-snmp-devel
newt-devel
lm_sensors-devel
bzip2-devel
That should do it!
Why would this depend on newt? net-snmp and lm-sensor headers and
libraries make sense. newt doesn't make
Carlos Chavez wrote:
I am having a problem with DTMF when sending calls through Teliax
(SIP). In the peer for teliax I defined dtmfmode=rfc2833 and for the
most part it is working. The problem always happens when a user is
trying to call a conference system. They simply cannot get
Tzafrir Cohen wrote:
On Sun, Jan 20, 2008 at 06:33:31PM +1100, Rob Hillis wrote:
PC's age and when they age, things tend to go wrong, particularly when
you upgrade software. Unusual crashes are usually the first sign that
something is going wrong.
And suddenly the same PC has unaged when
bilal ghayyad wrote:
Hi;
Via OpenVPN or port forwarding is known for me, but
via SSH is new for me, how I can do it and what is the
difference by SSH and OpenVPN?
SSH uses tcp. Openvpn, by default uses udp.
--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com
Jared Smith wrote:
On Thu, 2008-01-17 at 17:09 -0800, John Constalgie wrote:
Hence, is my only choice using an SSH tunnel between A and B for the
IAX connection to work? Will it work though with that One-way SSH
factor mentioned before?
It's my understanding that SSH tunneling will only
Vincent wrote:
On Wed, 9 Jan 2008 12:05:34 +0200, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
wcfxo is not needed.
Basically all you need is:
modprobe your_card_s_driver
This also pulls all of its dependencies (e.g: zaptel)
modprobe wctdm
Thanks, but on AstLinux, the modules are not
Vincent wrote:
On Wed, 09 Jan 2008 06:01:32 -0600, Darrick Hartman (lists)
[EMAIL PROTECTED] wrote:
But look in your /etc/rc.conf file for the ZAPMODS variable. You should
have that variable set to:
ZAPMODS=wctdm
Yes indeed:
#ZAPMODS=wctdm
Should I add this module here
Glenn Gillen wrote:
Unfortunately there is only one port, clearly labelled handset
On 31/12/2007, at 11:34 PM, dave cantera wrote:
glenn,
check your handset cord... it might be plugged into the wrong port
in the back of the phone. perhaps the headset jack...
daveC
Push the cord all
randulo wrote:
Well, $9 would pay for up to 500 answers. I also found a free one I'm
looking at now, but you never get anything really good free :)
If $9 can put that survey together in a comprehensible set of questions
and results, I will pay the $9.
Let me see if I can put what you ask
Tim Reimers wrote:
I have a single phone line (happens to be Charter Communications VOIP,
but I have their ATA and they’ve connected to red/green pair in the
house wiring)
Ok. so they've installed an ATA which connects your analog phones to
their VoIP (perhaps SIP) service.
What I’d like
Jason Lixfeld wrote:
I guess what I'm asking is if there is a recipe anyone has used to
allow a voicemail message (once recorded by asterisk) to playback on
iPhone when said recorded voicemail is received as an email
attachment. I understand you can convert using sox, so I guess that's
Kevin Smith wrote:
Hi Robert,
While I'm not sure how our network compares with yours, we run about
twenty 601 phones along with our office workstations (some stations are
without a phone). Each station with a phone is connected with the other
Ethernet port on the phone so we have one
Brandon Black wrote:
Hi,
I'm relatively new to Asterisk, and I'm looking to build a tiny
system for home use.
Welcome.
What I'd like to do is set up an Asterisk box with 1x FXS (to the
cordless phone base station) and 1x FXO (to the Vonage
pseudo-PSTN-line), and also have it act as a
Thomas Kenyon wrote:
Tony Mountifield wrote:
I have a client who wants a Meetme box with 12 FXO ports, to connect
to Analogue lines coming from an Ericsson PBX.
It looks like I could do this with four different hardware configurations:
a) three TDM04B cards (based on TDM400P)
b) one TDM04B
Doug wrote:
http://www.atacomm.com/
ATACOMM
Dear Atacomm Customers,
We apologize, but as of 6:00pm CST Friday, September 21st, Atacomm
and its parent company Ataractic Corporation has ceased
operations. We appreciate the 7 years of loyalty and support from
our customers. We
Matt Watson wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott Moseman
My understanding was that it's not required for pass-through.
PSTN Phone - g729 Gateway - Asterisk - g729 Phone
Does this not equate to pass-through? Maybe I
Peder @ NetworkOblivion wrote:
Is there a way to decrease the volume on the native files version of MOH
in 1.4? I've had several people complain that it is too loud.
run the files through sox
--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com
BJ Weschke wrote:
On 8/31/07, Joe Acquisto [EMAIL PROTECTED] wrote:
Any great disadvantage to using polycom 300/500/600 vs the 301/501/601?
I recall reading in the release notes of the latest release of the
firmware (2.2+) that I believe they've finally stopped supporting the
earlier
Dermot Bradley wrote:
Darrick Hartman wrote:
Just because someone is using an old kernel or doesn't know what they
are doing doesn't mean the hardware is bad. I've had very good
success
with dozens of different VIA boards (from the original mini-itx board
up
to current C7 models, the
Zeeshan Zakaria wrote:
Darrick, can you tell which mini-itx board you have and what processor
it has on it? I don't them with Pentium processors, instead they have
some VIA C3 and C7 processors, which are completely new to me and I have
no idea how will they perform with Asterisk.
I have a
Zeeshan Zakaria wrote:
I want my freedom to setup and configure PBX hardware and software how i
want, not how Digium or anybody else wants, so not interested in
Asterisk Appliances.
So anybody with experience with Supply Logics computers. Or any other
recommendations for asterisk pbx
[EMAIL PROTECTED] wrote:
Hi John,
Try ...
carriers.icall.com - No minimum, unlimited concurrent calls, great
price, some areas US 0,009. Only USA
voipjet.com
teliax.com - Not so cheap, and they do one-minute rounding ... not good
at all. But they hold a very good quality
Teliax
Gang Chen wrote:
On 6/22/07, Gary Chen [EMAIL PROTECTED] wrote:
We are using Level 3. At this point, changing carrier is not an option.
Gary,
I use Level(3) with G729a and RFC2833. No problems, no requirement
for inband G729.
--
Kristian Kielhofner
I can connect to Asterisk IVR
Joshua Colp wrote:
Andres wrote:
I can connect to Asterisk IVR using a SIP phone and send RFC2833 with g729.
It works fine. But when test call from PSTN to Asterisk, if I set dtmf=auto
with g729, I got warning saying something like * does not support inband
for g729 and sutomaticlly
Kristian Kielhofner wrote:
On 6/20/07, Steven [EMAIL PROTECTED] wrote:
I could understand if it couldn't register to an ITSP or similar.
But, (I had this happen today) asterisk takes forever to start up and SIP
phones can not register to it.
DNS should not need to be used for anything in
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