TFTP SIP loads a few seconds
back to Configuring VLAN the rest of the time.
I am wondering if I have a network problem or could do something to speed
this up.
Thanks,
Tom
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http
on this box).
Thanks alot for any responses/help,
Tom Christensen
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com
Quoting Roger Gulbranson [EMAIL PROTECTED]:
On Mon, 2005-03-21 at 19:03 +1100, Adam Goryachev wrote:
In case you need it, there are X servers available for MS Windows
platforms as well. Used to be one called exceed, but that was about 10
years ago, I just use linux on my desktop now
Quoting Roger Gulbranson [EMAIL PROTECTED]:
On Mon, 2005-03-21 at 08:57 -0700, Tom wrote:
We don't want to have to spend an extra 3 grand for another
server just to take up more space when we have this box that is sitting
here
idle 99% of the time, and as it has worked spectacularly well
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
On Sun, Mar 20, 2005 at 11:12:22PM -0700, Tom wrote:
I have a quick question.
I know that running X on an asterisk server is not officially supported,
Generally it shouldn't cause errors, but will probably degregate
performance, as an X server
Quoting Peter Svensson [EMAIL PROTECTED]:
On Mon, 21 Mar 2005, Roger Gulbranson wrote:
On Mon, 2005-03-21 at 08:57 -0700, Tom wrote:
We don't want to have to spend an extra 3 grand for another
server just to take up more space when we have this box that is sitting
here
idle 99
roll it out to at least
a couple of beta testers to try to stress the system and see if we can use this
solution. Once again thank you for your help and insight
Tom Christensen
Quoting Peter Svensson [EMAIL PROTECTED]:
On Mon, 21 Mar 2005, Tom wrote:
Quoting Peter Svensson [EMAIL PROTECTED
anyone seen poor line quality cause the digium fxo modules to have strange
errors such as these?
Thanks in advance for any replies/ideas/solutions (besides obviously calling the
phone company and telling them they suck)
Tom Christensen
___
Asterisk-Users
don't drop
as they do with the fxo card. how would I go about getting the cards on
different interupts if they are on the same one?
Tom
Quoting Rich Adamson [EMAIL PROTECTED]:
I just installed a new asterisk box with a wctdm with 4 FXO modules. The
lines
in the office have terrible static
diagnosis! I had heard of the shared IRQs
causing bad quality (popping, etc), but not dropped calls as I saw here.
Thanks again!
Tom
Quoting Rich Adamson [EMAIL PROTECTED]:
I just installed a new asterisk box with a wctdm with 4 FXO modules. The
lines
in the office have terrible static
and they are rock solid and just
work.
I believe they are a publicly traded company on the Canadian Stock Exchange
so it probably wouldn't be difficult to find out a company value. Perhaps
they are big enough to buy digium. :)
Tom
I think it'd be a great idea though, if they have the cash for it.. I
bet
Thanks for the informative review Matt. Please tell why you are using RBS
T1 trunks instead of PRIs. Is it the cost or availability issue from the
ILEC/CLEC or is there some other advantage. PRIs and RBS T1s are about the
same price in my part of the world.
Tom
At 09:20 AM 4/7/2005, you
i have had some problems with music on hold. some of the handsets havnt been
able to put people on hold...
When using a grandstream 101 i push hold it puts the other end on hold but
doesnt play the music. although when i do it with a x-lite client it does put
the other end on hold and stats
At 03:36 PM 2/13/2005, you wrote:
On 21:47, Sun 13 Feb 05, Vikram Rangnekar wrote:
+++ Michael Devenijn [13/02/05 18:23 +0100]:
We got also these problems and where searching like fools for solutions
... until the time we changed the main board of the server! (Interrupt
sharing or Hyper
annoying on our end flooding the asterisk
console.
Tom Christensen
VP Product Development
SinglePoint Networks
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
or
something?
Thanks,
Tom
Quoting Steven Critchfield [EMAIL PROTECTED]:
On Fri, 2005-03-04 at 15:27 -0700, Tom wrote:
Hello,
I have searched and searched, and come up with nothing. I am running
Asterisk
with a wcte110p configured for t1. Our PRI is staying up, and we can make
calls however
to disable DND and it goes away.
Thanks,
Tom
Kurt
__
Do you Yahoo!?
Friends. Fun. Try the all-new Yahoo! Messenger.
http://messenger.yahoo.com/
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com
when bookpool.com has them.
Tom
At 05:51 AM 7/9/2004, you wrote:
Hello,
If anyone is interested in getting a book on asterisk I would
recommend checking out http://www.saww.net/asterisk/
I ordered a copy, but they said it's six weeks or so 'till delivery.
Paul
Paul Mahler
[EMAIL PROTECTED
. The
final 3 are waiting for new wireless POPs.
PRIs usage has not dropped though.
BRIs were never understood and embraced by the US telcos and I am sure they
are on their way out.
Tom
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf
a product sold by
Tigerjet, called the Personal Phone Gateway. I'm purely speculating on
this, but Digium could have used Tigerjet's reference design for their
own board.
Regards,
Tom
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http
Interesting that they use IAX2, but 2.9 cents/minute seems kind of high
for a wholesale rate, especially in the lower 48. I'm shopping for a
good wholesaler right now.
Regards,
Tom
On Thu, 2003-09-18 at 08:44, Peter Pauly wrote:
I don't know if this has been mentioned yet:
Voicepulse
: tom [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Content-Type: text/plain
Organization:
Message-Id: [EMAIL PROTECTED]
Mime-Version: 1.0
X-Mailer: Ximian Evolution 1.2.2 (1.2.2-4)
Date: 14 Oct 2003 12:58:57 -0600
Content-Transfer-Encoding: 7bit
I have a dev kit lite, and I'd like to have asterisk up
now understand where the
profit comes from. Of course we are running it with no maintenance as we
do nearly all of our Cisco gear.
Great gear, crap company.
Tom
At 06:03 PM 3/29/2004, you wrote:
How come you have to repurchase software anyway? It was already bought
and paid for.
Cheers,
Dean
bug in
the file size.
Contact me off-list if you need more info on this and I'll get it to you
tomorrow.
Tom
At 08:25 PM 3/29/2004, you wrote:
Hi,
I am trying to update my CISCO 7940 phones to the SIP firmware.
I have version 3 and 6 of the CISCO sip software.
I have been all over the manuals
On Fri, 2 Apr 2004, Glen Ford wrote:
Does anyone know if avaya voip product is running linux under the hood?
...
Probably not. Linux is GPLed.
More likely a propietary RTOS that they wrote themselves.
Tom
___
Asterisk-Users mailing list
[EMAIL
year contract. The CLEC battle is
heating up here. I can't compete when I have to pay more than that for
VOIP LD calls that terminate on POTS.
Tom
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk
a business SIP connection. There are some
providers out there that can do this for LD.
Thanks
-brian
Tom
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit
?
Thanks,
Tom
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
extension in our office. We are running 6.3 firmware
on the phones.
Any suggestions would be appreciated.
Many thanks,
Tom
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
.
This is sometimes called tandem dialing.
-A.
Tom
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Does anyone have a Polycom SoundStation IP 3000 conference phone working on
an * server? Or the Cisco or 3com version?
I am looking for a high quality conference table phone that is compatible.
Any problems?
Many thanks,
Tom
___
Asterisk-Users
to connect to my
asterisk using SIP? IP Address of cisco is 192.168.0.254
Depends on what you want to do. You can just define your * server as a
SIP proxy in the Cisco config, so any calls that the Cisco answers go to
*.
Tom
___
Asterisk-Users mailing list
out an order
to remove them in February. The tech said they have been removed from all
SBC COs. :(
We are in northern Illinois.
Tom
At 12:49 PM 4/19/2004, you wrote:
For the record, the milliwatt generator, ANI number, etc, is up to each
telco engineering/operations group as to what number
At 01:50 PM 4/20/2004, you wrote:
On Tue, 20 Apr 2004, Tom wrote:
SBC cancels milliwatt tone generators.
--
I called our local SBC CO and asked for a milliwatt tone generator
number. He said that SBC decided they were not needed and put out an
order
Level * = Max Audio Hit )
(RX) (TX)
##*
Thanks,
Tom
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
It doesn't look very hard. FreeBSD supports recursive mutexes. It is
just a matter of getting the appropriate defines. I'm going to look at
this.
Tom
On Tue, 20 Apr 2004, Olle E. Johansson wrote:
The recent addition of recursive mutexes to Asterisk is causing a lot of problems
adjustments
helped to reduce our echo but the thread was never completed. We are not
sure how to test the TX gain adjustment and where on the graph to set the
RX gain when using a milliwatt generator tone.
Tom
Tom
On Apr 22, 2004, at 1:37 PM, Brent Franks wrote:
I do feel the echo cancellation does
On Thu, 22 Apr 2004, Steven Critchfield wrote:
...
VoIP phones have the benefit of linear growth cost. A phone costs $X,
and for the most part will cost $X no matter how many lines you roll
out. So a new extension is just $X increase, and your system is just $X
x N extensions to deploy. Also
, especially if you can install it into an existing router.
Tom
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk
At 02:04 AM 5/12/2004, you wrote:
Aloha,
Does anyone have a good source for Polycom SoundPoint® IP 600/500/300
phones?
http://www.pcnation.com
Everyone sells Cisco 79XX.
Aloha,
Matt
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
as they are wrong.
Tom
Thanks,
James Gardiner
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
to turn off DND.
There should be a better way. An on/off toggle of the soft key that it
creates (to disable DND) would be nice. Has anyone found out a way to do this?
Thanks,
Tom
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http
. Analog phones through our TDM400P do sound much better but the
audio problems on our Cisco SIP phones are echo problems. People are
working on solutions.
Tom
Cheers,
Brian
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com
At 09:18 AM 10/28/2004, you wrote:
Curious, can a WRT54G with this firmware, using Wonder Shaper, act as a
cheap QOS device?
Yes. Join the Sveasoft forum and there is good info on
this. http://www.sveasoft.com/
I personally have not tried the QOS yet but it is supposed to work.
Tom
One of our
. It is a virtual PBX hosted at the CO.
Tom
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Sailer
Sent: Monday, November 01, 2004 1:35 PM
To: Asterisk Users
Subject: [Asterisk-Users] Centrex
OK folks,
I'm trying to help get another remote Asterisk box up
to stay
interested?
Just wondering. :)
Tom
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
is excellent. They bend over backwards to help solve
problems like this and you can even talk to the guy who writes their firmware.
Tom
At 06:19 AM 8/5/2007, you wrote:
I have verified it is EXACTLY 5 hours. At 5 hours, the PRI stops
working until I issue a restart on the wanrouter interface. I have
At 07:02 PM 8/5/2007, you wrote:
I found the firmware files on Sangomas website...but could not find
the upgrade procedure...can you advise on how to do it or provide a
link?
I used this.
http://wiki.sangoma.com/sangoma-hardware
On 8/5/07, Tom [EMAIL PROTECTED] wrote:
I had something
Wow. It shows that there is a lot of ignorance in the DOJ. They
should have thanked BW, not charged him. Thanks for blowing this way
off track Matt.
Tom
At 01:32 PM 10/9/2007, you wrote:
http://www.usdoj.gov/criminal/cybercrime/WestPlea.htmhttp://www.usdoj.gov/criminal/cybercrime
to create a backup D channel for each group in case there is
a failure with the primary D channel.
I am not sure how Digium or Sangoma cards and drivers handle NFAS but
that is probably what you should be looking at and communicating that
to your carrier.
Tom
At 07:42 AM 1/22/2006, you wrote:
I
,
asterisk-sounds, and zaptel/ztdummy?
(I did not see much on the web regarding this.
Google: uninstall asterisk site:lists.digium.com)
So far, I've done this...
rmmod ztdummy
rmmod zaptel
/etc/rc.d/init.d/asterisk stop
(and verified this with lsmod and ps -ef)
Thanks.
Tom
~~~
[EMAIL
this...
rmmod ztdummy
rmmod zaptel
/etc/rc.d/init.d/asterisk stop
(and verified this with lsmod and ps -ef)
Thanks.
Tom
~~~
[root at localhost ~]# df -k
Filesystem 1K-blocks Used Available
Use% Mounted on
/dev/mapper/VolGroup00-LogVol00
507748111653
I have one that we work with. Digium also does this with Allison.
Contact me off list for more info.
Tom
At 05:19 PM 3/8/2006, you wrote:
Can anyone recommend a company that does professional Asterisk
recordings for things like IVR, greetings, MOH, announcements, etc?
Thanks,
Waldo
Chris,
You might want to rethink your dishonest strategy.
What happens if the legitimate buyer of that card tries to register
his new card and get the HPEC and Digium declines because the sn was
already used?
Tom
At 02:07 PM 4/22/2007, you wrote:
It seems that I need the serial number
to spring for a PRI, they
usually won't complain about the extra cost for better sound quality.
Yes, they support PRIs.
Tom
At 03:37 PM 4/23/2007, you wrote:
Shortly, I'll be purchasing a Sangoma A102. I'm wondering if I should
spring for the hardware echo cancellation circuit or not. Upon
prices have dropped and BRIs have disappeared in favor of
ADSL for data (which doesn't take any copper since it rides existing
POTS) and PRIs for voice (which rides a T1).
Tom
___
--Bandwidth and Colocation Provided by http://www.api-digital.com
Thanks Andrew,
I see the resolved bug report. I'll get the patch fix.
Sorry for the unnecessary mail.
-Tom
On 1/20/07, Andrew Joakimsen [EMAIL PROTECTED] wrote:
http://www.google.com/search?q=423+%22Interval+Too+Brief%22start=0ie=utf-8oe=utf-8client=firefox-arls=org.mozilla:en-US:official
Has anyone found a high quality wireless headset that works well with
Cisco 7960 IP phones on an asterisk system?
I tried the vxxi offering but the sound quality was pretty bad.
Since these are pricey, I don't want to sample blindly.
Experience appreciated.
Thanks,
Tom
FWIW My TNT gateway also connects at 56k on voice calls.
Tom
At 11:55 AM 1/27/2007, you wrote:
Hi All,
We are using MAX TNT to for some T1 PRI interconnects. I'm seeing the
voice channels connect at 56K. Does anyone have the DS0 channels
connecting at 64K for voice, if so what
the Cisco 79xx phones. We offer both to our business
customers. We have a number of choices in our demo room.
The ST2030 looks nice in pictures but my demo hasn't arrived yet.
Tom
At 06:41 AM 2/13/2007, you wrote:
I've just setup about 10 SPA942s. Great phones. Has the look and feel
of a Cisco
that you want to donate.
Otherwise no functional benefits. Only physical.
Tom
Bill
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo
which is well
built and easy to replace. Sangoma A200 analog card. Flash
based. Lots of testing. You provide the support.
That's what we are doing.
Tom
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
A common Cat5 straight through cable will work fine.
T1s use 1 and 2 (rx tip and ring) and 4 and 5 (tx tip and ring) for signals.
A T1 loopback plug would be wired 1 to 4 and 2 to 5.
They come in handy for testing T1 cards or for providing a hard loop
for the telco.
Tom
At 05:42 PM 3/18
What does the SIP debug say when you attempt to dial?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Ruggles
Sent: Saturday, March 28, 2009 4:07 PM
To: 'Asterisk Users Mailing List - Non-Commercial
Since we are talking about security, if I am using * to talk to a cisco
gateway via SIP, is there some sort of encryption you can use? Like a
vpn tunnel?
Can someone capture packets and re-assemble to make out a conversation?
-Original Message-
From:
I'm looking for sip origination and termination companies. Anyone know of
reliable ones? I am interested in wholesalers also.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To
hi, i have a new install, 1.6 2, 2 extension, but the call doesnt get
thorugh: here is my sip debug outout: thx for ur help!!
asterisk-users@lists.digium.com
--- (13 headers 16 lines) ---
Sending to AA.BBB.CCC.DD : 28127 (NAT)
Using INVITE request as basis request -
is there anyone out there with a running functional system?
thx
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
hi, i
@asterisk
- svn-ed asterisk from digium 1.6
- make install
its running and i can access the CLI
@gui
then i
-svned asterisk-gui from digium
- installed
- repointes apache /var/www/1234 /var/lib/asterisk/static_html
now, i see the login box, but i dont have any credentials. tutorials are
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *tom
*Sent:* Wednesday, July 08, 2009 1:50 PM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] q: install asterisk + asteris-gui
hi, i
@asterisk
- svn-ed asterisk from digium 1.6
- make install
its
/etc/manager.conf:
[admin]
secret = test
read = system,call,log,verbose,command,agent,user,config
write = system,call,log,verbose,command,agent,user,config
- doenst let me log in ;-(
- i tried chown /static_http/config
this is in my apache-logs:
[Wed Jul 08 15:36:23 2009] [error] [client
things
webenabled=yes ; this enables the interaction between the Asterisk web
server and AMI
[tom] ; you can name the user whatever you want
secret = tom
read = system,call,log,verbose,command,agent,user,config
write = system,call,log,verbose,command,agent,user,config
stupid me, i had a ; in front of the [general] line.
thx so far
im logged inand now?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
the ability to have a web-gui, right?
thx tom
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
:8088/asterisk/static/config/index.html
wes my missing link
thx 2 all for ur help
-- Forwarded message --
From: tom tomabr...@gmail.com
Date: Wed, Jul 8, 2009 at 4:19 PM
Subject: Re: [asterisk-users] q: install asterisk + asteris-gui
To: Asterisk Users Mailing List - Non
thx again,
one last question: as i mentioned, i used freepbx before. now i facing only
the section:
- users
my goal right now is to use that asterisk instance just to have intenral
extensions to talk to each other...whats the quickest setup here? i mean i
dont need trunks, dialplans etc, right?
thx danny,
(sorry, bad day today)
one more question: deviceandusers
i had this distinction with freepbx, though i dont know whether this is a
freepbx-thing or an asterisk-setting...
thx
___
-- Bandwidth and Colocation Provided by
[Jul 8 21:23:49] WARNING[4358]: chan_sip.c:10458 check_auth: username
mismatch, have 6001, digest has 1160
[Jul 8 21:23:49] NOTICE[4358]: chan_sip.c:18529 handle_request_register:
Registration from 'sip:6...@192.168.1.4 sip%3a6...@192.168.1.4' failed
for '192.168.1.3' - Username/auth name
hi,
checking my freshly installed astersik-gui, i can see a menu entry called
Users. clicking on that one gives me the pages labeled (on orange) User
Extensions on PBX. if i do make an entry here, it ends up in the user.conf.
file.
so i created a new entry in the sip.conf, reloaded asterisk
hi,
making may way through all this...internal sip registration works,(cant call
yet but anyhow)...
the asterisk box is obvisoulsy behind a router. im not 100% sure if i should
go with port forwarding or NAT and if a or b, what additional setup is
actually correct?
sip_nat.conf # this is when i
We used iax for more than a year and moved to sip about 6 months
ago. The quality from termination providers seems much better now with sip.
Tom
At 09:38 PM 8/30/2006, you wrote:
Content-class: urn:content-classes:message
Content-Type: multipart/alternative;
boundary
phones have a higher resolution display. New phones have some
lighted buttons.
Tom
Julian.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com
At 02:30 PM 10/15/2006, you wrote:
Before committing to about 50 of the spa942's, I like to take a last
poll from those on the list to identify any negative issues that
might be associated with the audio, functionality, early failures,
etc, on the spa942.
We have been using Cisco hard phones
and
cheap to make.
http://www.voip-info.org/tiki-index.php?page=Cisco+POE
We are using this with both the 7905g and 7960g phones. We are quite happy
with the phones.
Tom
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http
CallManager and voip enabled PBXs. Polycom will direct all support through
their authorized dealers. At least Cisco will sell a support contract on
some phone equipment. Both companies are feeding grey market sales with
these policies.
Tom
At 07:54 AM 7/13/2005, you wrote:
I have Polycom
At 10:06 AM 7/13/2005, you wrote:
Hello all,
We are looking for some hardware requirements/recommendations to be able
to handle a full DS3's worth of TDM - VoIP traffic. The DS3 would bring
24 calls per T1 x 28 T1s = 672 simultaneous calls. We would then need to
convert those calls into G729
have a pair of TNTs as an * gateway (one for a hot spare)
and they cost a heck of a lot less than a pair of 5400s. Our dialup TNT
has been rock solid for many years other than a power supply failure. Now
if the TNT only ran IOS instead of TAOS
Tom
..o
At 11:47 AM 7/13/2005, you wrote:
Tom wrote:
At 10:06 AM 7/13/2005, you wrote:
Hello all,
We are looking for some hardware requirements/recommendations to be
able to handle a full DS3's worth of TDM - VoIP traffic. The DS3 would
bring 24 calls per T1 x 28 T1s = 672 simultaneous calls. We
.
Tom
At 06:17 PM 7/13/2005, you wrote:
Great points but I think the ease of config on Polycom via FTP along
with the ease up firmware updates is a real winning combination. I have
yet to need the kind diagnostics you refer to while troubleshooting. I
copy a valid config, change the values
At 01:43 PM 7/18/2005, you wrote:
does asterisk support FAX t38 protocol?
No. The bounty sits at $5500 right now.
http://www.voip-info.org/tiki-index.php?page=Asterisk+T.38+Bounty
thanks
___
Asterisk-Users mailing list
Why the big secret? Why not post your solution to the list?
Tom
At 11:03 PM 7/27/2005, you wrote:
Hello everybody, for all of you that have searched for a real fax
solution, look no further. We now have T38 faxing. Please contact me for
more information.
Thanks
Michael D. Schelin
ShellTel
is that it uses IE(active x)
Tom
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
Thank You For Choosing Cache Communications
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
on your relationships.
Tom
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
the EndCall button while an unanswered call is ringing to
achieve the same effect.
The only available menu button is Answer when an inbound call is ringing
on my 7960g.
The menu with EndCall does not come up until I answer the call.
Tom
Sorry for jumping in but I am after the same thing
At 11:49 PM 5/9/2005, you wrote:
They have been down for 3 days now.
What does this mean? Your account is not working? I made all my LD calls
with livevoip today.
Tom
Althought I prize their concept,
perhaps I am wrong in thinking they are close to the only option. Will
anyone know
or floor matt under her chair?
Tom
Mark
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk
-subscribed CLEC, probably not.
Tom
Thanks
Bart
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo
else wants to try it out.
Sangoma A101 T1/E1 PRI AFT Card for Asterisk or VoIP Item number: 5876678236
The auction ends in 3 days.
Tom
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update
Daniel Hazelbaker wrote:
Drat, because the 3Com phones looked pretty good for the price. :)
Good for the price? You can import an atcom AT-320 EE for $40 +pp
(although they are hardly fantastic phones, at least they support IAX2).
They have a few faults (the speed-dial keys aren't really
1 - 100 of 1209 matches
Mail list logo
