No problems here. 27 min behind according to your post time.
Dean Collins wrote:
List doesn’t seem to be posting out – still active here
http://lists.digium.com/pipermail/asterisk-users/2005-June/date.html but
not being received by email (time warner is the isp but other emails
coming in every
I'm skeptical of any application that relies on "register_globals" to be
"on"that's just sloppy, lazy coding, period.
John Dunham wrote:
Set the global vars for PHP config to on.
edit the /etc/php.ini
We had the same problem and figured it out.
John Dunham
GXC Corp.
-Ori
[EMAIL PROTECTED] wrote:
Hallo,
we have started playing with asterisk about one month ago, and we do like
very much what we are experiencing.
Now we would like to take some step further towards "standardizing"
installed modules, functionalities, tools etc.
The "wall" we are facing now is: choos
Please do yourself and everyone involved a favor and use an open and
widely used management standard such as WBEM. Start at www.openwbem.org
and/or http://sblim.sourceforge.net/ for a jumpstart on WBEM.
I had started on this already, but got sidetracked on other projects and
never got back to
Connect the POTS pair to pins 4 and 5 in the RJ45, and you should be
fine (I say this not having looked at the TDM400 specs, but from the
perspective of standard wiring practice and the assumption that Mark et
al followed same).
Greg
Paul Shiflet wrote:
I just received my TDM400 card from digiu
Since you are not setting up an actual PBX in the true sense of the
term, the hunting has to be done by the telco if you want to take more
than one call at a time on the same number. Cincinnati Bell here locally
calls their standard SMB phone service "Centrex"; Sprint would probably
call theirs
Their repsonsibility ends at the demarc. In a house this is the box
where either exists an RJ11 phone jack, or a pair of screw terminals. If
you have multiple lines coming into a business, you probably will have a
66- or 110-style punchdown block where they terminate their lines (in
some cases they
There should not be any, except for the occasional rekeying.
Greg
On another note... When I observe the activity LEDs for a VPN circuit,
it seems to be going full blast without and activity at either end
point. Can anyone discuss the bandwidth overhead of a VPN circuit?
_
Because SER does not process the RTP stream, it just directs it around.
Greg
Vikram Rangnekar wrote:
Why is SER considered a better SIPserver than asterisk , why is it that SER
can handle more clients than asterisk can. And if this is just cause of say
poor SIP handling code in asterisk then is the
Many people have recommended the Sipura products, and others
have recommended a channel bank connected the a PRI card in asterisk box.
Does anyone know if it is possible to get Caller ID
information into asterisk if using anything other than the TDM WildCard
products (usi
What can I use to find out why packets destined for the outside world (via
65.78.109.2) are not being routed?
Check with your ISP and make sure they have you set up correctly. I have
had issues in the past with that.
Fact is, if you can ping the far end, *and packets are returned*, then
the prob
Many web admin interfaces, of varying featureset and completeness, exist
for Asterisk. See the Wiki for more:
http://www.voip-info.org/wiki-Asterisk+GUI
Greg
German Aracil Boned wrote:
Hello to all
I'm new in the list. My name is German.
I look very pretty asterisk software, but I don't know if
information is from SBC (Southwestern Bell). They claim that if they
bring us one T1 and we want to split it 50/50 voice/data (TDM voice and
IP data) (not necessarily 50/50 but any combination of voice + data
channels), there could be voice quality issues on the TDM voice side
when the data po
gured?
If I use T100P to connect to Adtran where am I going
to connect the T1 comming from my Telco?
Thanks in advance.
robert
--- Gregory Junker <[EMAIL PROTECTED]> wrote:
http://www.voip-info.org/wiki-Asterisk+Channel+Bank
Digium T100P, T1 cable to Adtran T1 port, extensions
to Adtran FXS
http://www.voip-info.org/wiki-Asterisk+Channel+Bank
Digium T100P, T1 cable to Adtran T1 port, extensions to Adtran FXS
interfaces. Follow the instructions on the Wiki for configuring the
T100P both in /etc/asterisk/zapata.conf and /etc/zaptel.conf. Configure
the ports on the Adtran per the Adtra
It would be nice for Mark to comment on this design flaw ...?
Why so quick to assume it's a "flaw"? Perhaps it's a "compromise".
Greg
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Bottom of the page, he just posted the link.
Greg
Darren Wiebe wrote:
I saw that thanks... I was looking for the fields for iax and sip
friends, and extensions. If nobody has a list convenient, I will snoop
some more and see what I
Wow...I am not sure you are ready for the leap you are about to take, if
you have no prior Linux experience. At any rate...
Best thing for you to do is go to the Redhat/Fedora site, download the
FC ISO images, burn them to CD, and install Fedora just like you would
install Windows. Boot from CD
Is it possible to re-route incoming call on Zap channel of TDM400 FXO
card to completely different and external telelephone number via some
magic telephone command or signal? So, the Asterisk Zap channel would be
cleared off of this call?
Like in a scenario when person calls in via PSTN via a
but there
is
some
good info on troubleshooting and error indecations.
I second the post below. Adit 600s are a good choice for Asterisk.
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Quoting Gregory Junker <[EMAIL PROTECTED]>:
Say I
Digium cards need 1000 interupts per card per second due to the lack of
onboard buffer. The buffer was left off of the design to keep the design
simple and therefore inexpensive. All the cards present 8 bits of data
per channel during that interupt and as all telephony is 8000 bits per
channel per
Say I get an ADIT 600 with two FXS8A and one FXO8A from ebay.
> a. Is it good for Asterisk?
A quick look in the Wiki or even a Google search (saving Critch the
trouble here) shows that the Adit 600 (with a T100P) is an excellent
match for Asterisk:
http://www.google.com/search?q=asterisk+adit+60
No, actually he wanted to be able to plug them into a Zap card and have
them work, not convert their protocol to/from SIP and so forth.
*sniff* I miss my AT&T 7406 ;)
Greg
Jim Van Meggelen wrote:
[EMAIL PROTECTED] wrote:
Are there any digital phones that run on asterisk yet? I'm talking
about
Anyone tried using a pocket pc with sjphone or x-ten over a cell phone
connection?
Uhh, good luck. Latency, lack of bandwidth... Nice idea, but I would
stick with the cell phone when you're on the road.
Or wait for WiMax service offering rollouts sometime in 2005.
Greg
__
I have a customer interested in an * system, however she wants to ensure
that the receptionist phone will display who is on the phone and who is
not. It is an office of 10 people, and there are 12 PRI channels available.
She is an older lady and does not want to use a web interface. Any
sugge
How can I even tell if there's been a compilation problem?
The last line in any make-based build will tell you of an error if one
occurred.
At this point, type "asterisk" and then "asterisk -r" at a command
line. The first one starts Asterisk and detaches it as a daemon
(background process)
Not yet. It's under development.
Greg
Alex Brecher wrote:
Is there anything open source out there that has the same or better feature
set than Asterisk PBX Manager ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Clive Carter
Sent: Tuesday, November 30, 20
Ohh noo, now you know that it doesn't take someone being mean or even
mean spirited to get annoyed at the lack of effort some people exhibit.
LOL I know, I could just as easily have phrased it as bkw did. I feel
soiled now :-\
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outlined at the bottom of every message from this list? (Oh gawd, I
sound like Critchfield now :p )
Greg
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If it's person-to-person, why not just use MSN Messenger (or similar)
voice communication, or a gaming communications program like TeamSpeak
or GameVoice? The codec technology is the same.
Jeff Owen wrote:
I’m sure this might not be the correct place to ask and I have done a
Google but I can’t
For this reason alone I find it very hard to even consider a TDM400P card or
two - I always suggest a channel bank (Adit600) and T100P, even if the
density doesn't require it. I'd love to recommend the TDM card, perhaps this
If I could find one with FXO modules I would suggest a used VINA
Inte
They are using a proprietary system, it is not Partner or Merlin, but the
phones will not work with other systems.
If one of the requirements is to continue using their proprietary
phones, you simply are left with getting the old system working.
If they are amenable to different phones, you can f
My question is would you guys setup an anolog system or VOIP for the
phones. There is not a local VOIP provider in our area, so we can not
port the 3 pots lines.
I would use a 3-port FXO card (for the incoming lines) and a 4-port FXS
card (for their existing phones) and just drop-in replace the o
Do I have any other options besides RH 9.0 ?
You always have a choice. Most distros provide some form of download for
their media. RH/FC, regardless of version, is easiest IMO because of
simple ISO image availability.
If you really wanted, you could build up a Linux machine based only on a
ker
See the reply below yours.
I would hazard a guess that Redhat and SuSE, followed by Debian, are
probably the top three (RH and SuSE because of market share, and
"enterprise" server distros thbey have).
Greg
Alex Brecher wrote:
Which Distro is the most commonly used distro with Asterisk please ?
Third, those complaining of low volume in emailed files are usually
using a compressed format. In the uncompressed wav format, the volume is
effectively doubled by shifting the audio data to the left one bit. This
is done at the format level. Of course on playback via asterisk, it
checks to see if
I'm taking a look at the functions involved to see what the issue is, if
anyone cares.
One question thowith all of the sound libraries on the planet, ones
that write to far mroe open formats, such as MP3, which are just as
acceptable as email attachments, and _have_ to be easier to deal wit
I'm not saying that it would compromise *'s 'PBXness'. But you are
comparing products that have DECADES of development and maturity,
building on basic features that * is just now getting stable, and that
use proprietary hardware to accomplish these features.
Kinda my point. I reiterate, if someo
I would like you to name one PBX that does not support this behavior?
Every system from Avaya including their Definity, Merlin Legend, Merlin
Magix, Partner, and their new IP based PBXes support it, as do those
from Mitel, Nortel, InteCom and every other system that I have ever
used. A typical
There are many current projects that perform various levels of
administration assistance (besides at least two current threads in thist
list on the subject ;).
You can also find more at the Asterisk Wiki:
http://www.voip-info.org
Greg
Michael Di Martino wrote:
I am looking for a good Asterisk GU
what exactly do you call modules? is it hardware or software? sorry for
knowing so little :)
> I looked on http://www.digium.com/index.php?menu=wildcard_tdm400p2 but I
> didn't really get it. I have to buy a TDM400 PCI card and then I need
add
> other FXO or FXS cards to this PCI cards? so i ne
Just wondering how difficult it would be for AMP devs to develop a
install wizard or a batch file that can automatically execute the
install and download necessary dependencies... until then, I guess
I'll be continuing to manually config my asteisk files
This requirement is part of the project
I think it's time for the list mod to step, no? End this thread?
Greg
Joe Greco wrote:
You, bloody moron. Is not most email unsolicited.
Are you familiar with the spam problem? Spam is unsolicited bulk e-mail.
It is problematic for any number of reasons. A single unsolicited message
may be unwa
The horse is dead, guys. Let the city workers pick it up now. Thanks.
Greg
Adam Goryachev wrote:
On Wed, 2004-11-24 at 14:17, Kristian Kielhofner wrote:
While I usually refrain from the discussions such as this one, this
comment has left me utterly disgusted.
and while I managed top refrain from
And this is after you did a make clean, at least in the apps directory?
The part about "overriding commands" doesn't make sense to me...
Greg
Eric Hall wrote:
Still getting errors
make[1]: Entering directory `/usr/src/asterisk/apps'
Makefile:103: warning: overriding commands for target `app_rxfax
No. Auto-attendant is a subset of a class of applications that fall
under IVR (interactive voice response).
Greg
Paul Rodan wrote:
Are IVR and "Auto Attendant" interchangeable terms? They both do the "Press
1 for" thing. Sales is asking me how to word it and I've always used both
terms interchang
app_rxfax uses an incorrect structure parameter. Change "callerid" on
line 83 (I think) to "cid".
Greg
Eric Hall wrote:
I did that
[EMAIL PROTECTED] apps]# patch < Makefile.patch
patching file Makefile
Hunk #1 succeeded at 52 with fuzz 2 (offset 11 lines).
Hunk #2 succeeded at 88 with fuzz 2 (off
We are currently working on a WBEM-based management system for Asterisk.
If you are familiar with Novell ZenWorks or Microsoft's MMC or the
like, you know what I mean.
Greg
Jim Van Meggelen wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hank
S
Are you doing this as "root"? You cannot edit anything in /etc unless
you are root (superuser).
Greg
amna saleem wrote:
-- Forwarded message --
From: amna saleem <[EMAIL PROTECTED]>
Date: Thu, 18 Nov 2004 22:26:11 -0800
Subject: changing configuration file
To: [EMAIL PROTECTED]
hi
Greg
Leo Ann Boon wrote:
Gregory Junker wrote:
Is there an open source key system, comparable to *?
If there isn't , I'd be happy to work on developing one. It is clear
that the need still exists for such a user interface paradigm.
Bayonne is supposed to act as a key system, at least
.c file as well, which also is not part
of the tarball. That could be the problem.
Greg
Gregory Junker wrote:
Steve Prior wrote:
I just ran into this last weekend. I believe that you are using a
version
of spandsp which is for an older version of Asterisk.
The patch file is not part of the
Steve Prior wrote:
I just ran into this last weekend. I believe that you are using a version
of spandsp which is for an older version of Asterisk.
The patch file is not part of the tarball; it's a separate download on
the site. I had issues with non-1-0 CVS versions; the v1-0 branch worked
fine.
I was able to patch the apps/Makefile from the v1-0 branch (use "-r
v1-0" on the CVS command line) with Steve's patchfile, without issues. I
included his patchfile for convenience. Which version of the source are
you working with? Worst case, you can just look at the patch file too
see what cha
Asterisk runs on any PC hardware that runs Linux, from that old PII
sitting in the closet gathering dust to that 4-way Xeon blade server in
a rack, and beyond, and all points in between. Digium has a line of PCI
cards that work with Asterisk for T1/E1 lines, ISDN PRIs, analog POTS
lines, etc.
Is there an open source key system, comparable to *?
If there isn't , I'd be happy to work on developing one. It is clear
that the need still exists for such a user interface paradigm.
Greg
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Another strong possibility is that after a while, few operators would be
willing to continue holding their arms in the air to operate a touch screen.
Why would they be holding their arms in the air? You mount the touch
panel in the same place at the same angle as the current console...
Greg
__
Not all over $500 - a quick search finds:
For purposes of replacing a receptionist console with a touch screen
(for example, replacing a 6x9 grid of buttons), that would be too small
as well.
Greg
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You should always design an interface around a human being. A hard
I could not agree more. Usability is my focus in any software
system...including open-source, where it is typically the last thing
considered.
Greg
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[EMA
$400-500 device here. Not very price competitive. I would like to see less
than half that.
What is the price point you are trying to hit? Any piece of a
proprietary telecom system is by nature overpriced to begin with, and
receptionist consoles certainly fit into that category.
I agree that any
Me and another guy are working on LCD drivers etc for Linux. The thing
Including touchscreen?
Ideally someone would tell me how to make something either a) seamlessly
convert serial/parallel/USB port to TCP and back at the other end, or b)
point me to a resource on a simple chip with TCP suppor
Most customers don't want to be in a new era. They want something they are
accustomed to. I don't need any more impediments to making money than I've
already got. So if the customer wants a busy lamp, I am going to do my
best to give it to them.
I agree. This is why engineers do not make good sales
Add to it, my message wasn't a flame but rather a terse comment. Your
Never said it was a flame. I said it was in a tone virutally guaranteed
to make the guy consider you and everyone on the list to be a conceited
jackass.
The difference in your perception of your replies (the one I snipped
inc
olitic. And since I am not as world-weary
as you are, I simply ask that if you cannot be polite, then don't reply
to newb questions. It's really that simple.
Greg
Gregory Junker wrote:
This was addressed in a different thread, as I recall, regarding
"newbie" posters, and it w
4-11-20 at 00:42 -0500, Gregory Junker wrote:
And Steve provides yet another cordial, extremely helpful reply.
Really, friend, does it do *that* much for your ego to step on people in
public? If you can't be friendly, just ignore the damn email, no matter
how many times the question has been a
n
someday.
Instead of being an ass about it, Steve could just as easily said:
"You can find the information you seek on Google.". It's only a few more
words, and far more cordial.
Greg
Matt Riddell wrote:
Gregory Junker wrote:
And Steve provides yet another cordial, extremely help
http://www.openvpn.org
sorry, this should have been
http://openvpn.sourceforge.net
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And Steve provides yet another cordial, extremely helpful reply.
Really, friend, does it do *that* much for your ego to step on people in
public? If you can't be friendly, just ignore the damn email, no matter
how many times the question has been asked.
Greg
Steven Critchfield wrote:
On Fri, 200
Linux 2.6 kernel includes IPSec directly, and ipsec-tools can be used to
create a secure point-to-point link. OpenSWAN makes use of the kernel
IPSec in 2.6, and makes it available in 2.2 and 2.4 kernels. IPSec can
use shared keys or x509 certificates within or without a PKI for
authentication.
"I'll stop doing it when Walsh stops posting about it:
> http://www.faqs.org/rfcs/rfc1855.html
>
(from the RFC)
"...Don't wander off-topic, don't ramble and don't send mail or post
messages solely to point out other people's errors in typing
or spelling. These, more than any other beha
Your actual question then is "can the zaptel driver be connected with to
a faxgetty?" faxgetty expects a serial port, if I am not mistaken. So,
"can zaptel give me a pseudo-serial port I can use with faxgetty?"
Not having tried it myself, my expectation would be that it can not.
Greg
Eric Hall w
I have my SPA-3000 taking a PSTN line inbound and forwarding it to my
Asterisk server after a few rings. I don't hear any "dial tone" when I
do that kind of forwarding. I do it via the dial plan and I also tried
it via "CFwd SelX Caller/Dest". How are you attempting to do it?
I am just starting
Ditto. There's another very clear advantage to OpenVPN over IPsec,
and that's the fact that many firewalls are hard to run IPsec through,
but OpenVPN, using a single ephemeral UDP link, will work just fine.
I believe that the original poster is not concerned with getting it
through a Linksys rout
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gregory Junker
Sent: Thursday, November 18, 2004 10:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] "Lobotomized" Sipura SPA-3000
configurationneeded
Good deal, I looked at
on my SPA-3k's. I have
been considering upgrading but haven't had a chance yet.
-Jeff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gregory Junker
Sent: Thursday, November 18, 2004 9:23 PM
To: Asterisk Users Mailing List - Non-Commercial Di
Jeff Owen wrote:
I have mine working so that all incoming calls are passed directly to * and
no user heard any dial-tone or digits, even when the call goes back to the
SPA-3k for the Line1 user.
Share some config tips?
Greg
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I want even less than that
All I want to do is have the SPA-3000 configured so that it offers up
its FXO and FXS ports to Asterisk -- nothing more, I want Asterisk to be
the brains.
1) The SPA should hand incoming calls on the FXO to Asterisk.
That's all I want. For an interim measure I would li
Does something like this exist?
Dozens of different efforts are underway along these lines.
http://www.voip-info.org/wiki-Asterisk+gui
Greg
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Perfect Forward Security? Yes, OpenVPN can easily be configured for
dynamic re-keying at any specified interval and provides all the ciphers
that the openssl library supports. I use and highly recommend it.
Cool, I will definitely look into it; it wasn't too technically
difficult getting OpenSWA
s phones.
> Thank you again...
>
> Nahuel Ramos.
>
>
> On Thu, 18 Nov 2004 14:09:24 -0500, Gregory Junker
> <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote:
> > Nahuel
> >
> > Pick up one or more Carrier Access ADIT 600 channel b
with Xlite and ATAs, but I
need a lot of internals phones.
Thank you again...
Nahuel Ramos.
On Thu, 18 Nov 2004 14:09:24 -0500, Gregory Junker
<[EMAIL PROTECTED]> wrote:
Nahuel
Pick up one or more Carrier Access ADIT 600 channel banks off of eBay
(they always come loaded down with FXS ch
Nahuel
Pick up one or more Carrier Access ADIT 600 channel banks off of eBay
(they always come loaded down with FXS channels), grab a quad-span
Digium TE410P (or TE405P, depends on your PCI capabilities) -- 4 T1/E1
ports, that's 96/120 voice channels theoretical max -- and go to town. :)
http:/
I use an OpenVPN tunnel as well, and IAX over that, and it works dandy.
I highly recommend it. It's definately the easiest to configure,
understand, and to get across diverse links. It is NAT-friendly, all
UDP, etc. In my opinion, OpenVPN is to IPSEC as IAX is to SIP or H323.
Does OpenVPN supp
Use iptables to secure your * box and allow traffic only from known
servers/hosts. I would say that step one. When you do that then you can
use a VPN to make sniffing more difficult.
What link do you have between the offices? Is it public internet ?
I gathered that he was mostly concerned with ma
IPSec, especially with PFS, should be all you need.
The 2.6 kernel comes with IPSec as part of the kernel, and suites such
as OpenSWAN make it quite simple to set up secured links between two
endpoints. Given that OpenSWAN is free, I don't see how it gets much
more affordable. ;)
Keep in mind t
The phone has to be registered with Asterisk first. What is your setup
in sip.conf for this phone?
Greg
Michele wrote:
Hello, this is my first message on thi ML;
I'hava a problem: I have a voismart 101 phone and at the moment of registration
or when I make a call,in the asterisk's consolle i can
sorry wasn't paying too close attention too the model number, the other
current reply addresses that.
Just pay the $7 or $10 for the firmware license already, sheesh.
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*puts on flame suit*
(bah ignore the other prematurely-sent reply)
Seriously, though. It's not a scam, it's their business model (which is
shared by many many companies). The software license is separate from
the hardware. Always has been. You probably should have known that
before buying the p
*puts on flame suit*
Bob Willock wrote:
I just bought a couple of these Cisco 7970G phones and it seems that they
require a SIP image binary file to load when the phone boots and this file
updates the firmware of the phone to run in SIP mode. The only problem is
that Cisco seems to want to profit f
So, could we just agree to read around our idiosyncrasies and go back to
paying attention to the CONTENT of a message, not its FORMAT?
Discarding messages because they're in the wrong format is equal to
discriminating against another human being based on outward appearance;
be it skin-color, religi
I'll stop doing it when Walsh stops posting about it:
> http://www.faqs.org/rfcs/rfc1855.html
>
(from the RFC)
"...Don't wander off-topic, don't ramble and don't send mail or post
messages solely to point out other people's errors in typing
or spelling. These, more than any other behav
folder. No supplier gets a purchase if their people are not properly
trained in e-mail communication. My employer spends quite a bit as
You are kidding, right? "Properly trained"? By whose standards? What
international commerce committee on email standards published the
training regimen of whic
Also interesting comment about Skype possibly being interfaced directly
into a gaming solution for online game chats, does anyone know if
Asterisk has been licensed to offer something similar? Sounds like an
area that could be worth investing in.
Sort of overkill considering the popularity of pr
How remote are the remote offices? Miles? States? Countries? Best of my
knowledge, the days of exchanges based on proximity to a particular CO
are over, and those numbers (assuming they are in the same area code)
often can be routed anywhere. You could also look into having a company
like Voice
router. Will he be able to start downloading/uploading on that bandwidth
even though its hooked directly to the Asterisk server? If so, how can I
prevent the bandwidth usage but still allow VoIP calls?
If you do not route IP traffic over the T1 then there is no way anyone
can upload or download.
If I want to address individual points in turn I happily trim and
inline. To say that "top posting is unprofessional" is simply a
meaningless blanket statement; in your opinion it may be, but I doubt
it's your main criteria for assessing whether you want to do business
with someone or not.
Gre
Dude...you seriously need either to relax, or remove yourself from this
and all mailing lists if it is that bothersome to you.
I consciously changed my Thunderbird formatting to insert replies at the
top. I prefer it. So do many others.
Get over it, and yourself. Jesus...
Greg
Kevin Walsh wrote
How are the two offices connected?
In terms of an Asterisk solution, at a high level you are looking at an
Asterisk machine on each end, each of which is connected to the existing
office phone system or the local PSTN via TDM cards (or T1/E1 with channel
banks, etc). Without more details it's hard
I've registered channel #acd on Freenode if anyone wants to pop in.
Greg
JAMES BOTHAM wrote:
Hi there,
I agree with Greg and also with the documentation
group, we are all great at bitching about * (I know I
have done a lot of it but thats because UK and support
for us is minimal or so it feels) we
That option certainly is feasible.
Greg
Stefan de Konink wrote:
Gregory Junker wrote:
So, if anyone is interested, I am suggesting particularly a
standalone, cross-platform project that is simple to install,
configure, operate and manage. It should operate with or without a
database. It can
The Web Service interface is possible, if the architecture finds it useful.
At this point implementation details are less important than featureset
and usabilty details.
Greg
Michael Giagnocavo wrote:
Why yet another project proposal? Because the majority of those I have
seen so far are web (PHP
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