Hi Folks,
Is there any SIP FXS ATA with Gigabit ethernet bridge port, in the market
?
--
Isamar Maia
Cel. VIVO SSA: (55) 71-9940-2012
Cel. TIM SSA: (55) 71-9289-5128
Cel. Claro SSA: (55) 71-9146-8575
Fixo: (55) 71-4062-8688
Skype ID: isamar.maia
A vida é muito curta para ser pequena
in advance for any help.
Isamar Maia
+55-71-9146-8575
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This is a PA-1688 chip phone.
Give a look at http://www.aredfox.com/. It has what you need.
Look for Pamtool.
Isamar
On Wed, 14 Feb 2007, Alcides Cremonezi wrote:
Hi! Everyone,
This IP phone came configured for to be used with Soyo VoIP service.
I would like to set it up to work with my
wonder if anybody knows from where we can download ACT firmware updates.
Isamar
On Wed, 25 Oct 2006, joe, at j4computers ([EMAIL PROTECTED]) wrote:
I have a setup with a polycom 601 and an act p160s. All on local segment, no
NAT.
Can call the act p160s, from the polycom, rings, connects
Hil Folks,
I am trying to use latest spandsp(0.0.2) with asteirsk 1.2.9.1
and tiff library 3.7.1 through a SIP channel but the channel is freezing
after answer. It was running ok using 1.0.10.
Any tipo to make it work well?
Thanks in advance,
Isamar
Chris,
I have some boxes in Japan too.
Just set it up as you set for a common T1.
INS64 is BRI. INS1500 is PRI/T1.
I never used Digium for this. GIve preference to Sangoma.
Isamar
On Tue, 18 Apr 2006, Andrew Latham wrote:
J1 is just a T1, the J2 is also very common and likely what you
Buy Sangoma.
Good cards. Good support.
On Fri, 14 Apr 2006, Tony ROBIN wrote:
I am so fed up with Digium cards. My company first owned a TE410P,
I installed it in a Dell server and enjoyed its instability (we
bought it months before Digium warned about the incompatibility
issues). Then we
I am not sure if this debug message is enough information.
Try to do what I told. Switch to another H323 channel driver and see what
happens. Try first chan_oh323.
Michael Mansos(or something like that) and other guys have been done a
good job.
Isamar
Good luck. Try to switch between channel drivers.
Chan_oh323, chan_h323 and ooh323.
and remember to install the *exact* lib versions recommended on the
readmes
May the force be with you...
Isamar
On Sat, 1 Apr 2006, Il Neofita wrote:
Hi,
I installed H323, however when I make a call
Hi Folks,
Is it possible to setup some parameter on Dial command to
hangup a call if the customer press # ?
Thanks,
Isamar
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Looking for a Digium Supplier/Reseller that accepts Paypal.
Thanks,
Isamar
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I think the rule number 1 in the programming world should be:
Why complicate if you can make it simpler?
Isamar
On Wed, 18 Jan 2006, Brent Torrenga wrote:
I think he is getting at something like a Zap channel that passes on it's
own CID info from zapata.conf, as opposed to the calling
Which * version are you using ?
Isamar
On Tue, 17 Jan 2006, news.dalaidily news.dalaidily wrote:
[EMAIL PROTECTED]
i have a problem when hangup an incoming call,
i receive this error:
Incoming call: Got SIP response 503 Server error back from xxx.xxx.xxx.xxx.
and the caller stay
... unless you are younger, much hair in the head and no
white hair :-)
Isamar
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Who doesn't have money to buy anything else but Grandstream, I recommend
PA-1688. Cheaper and better and have IAX2 protocol embeded.
Talk to my friend Jack in China.. http://www.yntx.com/
Who wants to keep defending Grandstream, please send me the address and I
will ship all the broken bunch
and time spent on this will just going to
sink.
Probably it is better to loose time with something else.
Isamar
On Mon, 19 Dec 2005, Luigi Rizzo wrote:
On Mon, Dec 19, 2005 at 03:18:18PM -0800, Kerry Garrison wrote:
I don't know exactly how it works, but since it appears to just be SIP, I
Anybody got already to make Vizufon CIP-4500 working with Asterisk through
SIP?
I got to register by Asterisk send a Notify back and receive a Bad
Request
Isamar
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Try chan_oh323 and if it is not ok, try chan_h323
Both work well in different situations/equipments.
Isamar
On Mon, 5 Dec 2005, Innocent Evil wrote:
Hello,
Would you please share your experience regarding h323 and oh323 in asterisk.
I am confused to choose one.
Thanks,
--
You don't
: Prodding channel
'H323/ip$202.83.196.25:32791/31907' failed
How to solve this problem?
Isamar
On Mon, 5 Dec 2005, David Waugh wrote:
Prely subjective, but I first installed h323 and it worked. Somewhere along the
line something happened and it no longer worked. Recompiling it etc seemed
Ok.. how many channels are you using? More than 100?
Maybe it can be good only for 10 or 20 simultaneous connections...
Isamar
On Tue, 6 Dec 2005, Boris Bakchiev wrote:
I like the chan_ooh323.
I like the idea of selfcontained H323 channel that doesn't rely external
libraries, often
make http://www.voip-info.org your friend..
http://www.voip-info.org/wiki-Asterisk+H323+channels
Isamar
On Mon, 5 Dec 2005, Innocent Evil wrote:
So, we have
h323, oh323 and ooh323
I knew about h323 and oh323 but didn't know about ooh323.
What is URL of ooh323, I want to know more about
Ok. I will give one more shot on that. Last time I had one-way-audio issue
with that. Thanks.
Isamar
On Tue, 6 Dec 2005, Boris Bakchiev wrote:
No, max we used is 30 channels.
But according to voip-info its faster protocol because it offloads media
processing to asterisk (which
be loosing some calls becabuse of this issue?
Thanks in advance for any explanation and/or recommendation.
Isamar Maia
Magiclink / Japan
+81-3-4550-1212
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Actually, exactly now I am trying to do that also...
Isamar
On Fri, 25 Nov 2005, Aaron Anderson wrote:
Are there any kind of patches or experimental libraries that I can use
to pull caller ID info off a japanese pots line?
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This rocks! Use xten or diax.
Isamar
On Tue, 16 Aug 2005, Anton Krall wrote:
Anybody using Plantronics USB headsets? What softphone are you using and
whats your overall experience? Any comments/suggestions?
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Yes. More than 1 port as source and port forward doesn't work.
Isamar
On Mon, 4 Jul 2005, Carlos Alperin wrote:
Eric,
This is the Sip.conf section where you define the port. Do you want to use
more than one port as Source?
;
; SIP Configuration for Asterisk
;
[general]
port = 5060
Yes. That's what I'm trying to sort out with SER.
I just need to forward the packets. Anybody with a sample ser.cfg to do
that?
Isamar
On Tue, 5 Jul 2005, Tzafrir Cohen wrote:
On Tue, Jul 05, 2005 at 08:13:58AM +0900, Isamar Maia wrote:
Yes. More than 1 port as source and port forward
Dear All,
I need to bind two different ports at the same time for SIP.
5060 and another port number.
Is it possible ?
It would be something like
port=5060,5062
Isamar
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It's a little bit hard to compile but Try oh323 first.
Although, There will be some few situations that H323 will work better
than oh323. So, have both.
Isamar
On Sat, 2 Jul 2005, Adeel -31 wrote:
I m new to asterisk n i've got an IP phone that supports h323 protocol
but i dont know
Yes. it's a PA1688. IMHO, it works well for Home users but don't even
think to use it for business applications.
The great thinkg is that it works with IAX2.
Isamar
On Sat, 28 May 2005, Waldo Rubinstein wrote:
I was referred to this URL:
http://www.thevoipconnection.com/store/catalog
with them the model of PA1688
and download the firwmare you want from http://www.aredfox.com
Isamar
On Sat, 28 May 2005, Waldo Rubinstein wrote:
I read about the PA1688 and, yes, it says to support IAX2. However,
reading the PDFs on the Soyo G688, I found no reference to IAX2 at
all. How
Do you have any link? Isn't it PA-1688 Chip?
Isamar
On Fri, 27 May 2005, Waldo Rubinstein wrote:
Has anyone had any experience with the Soyo G688 phone? I'd like to
use it as a agent's phone. Is it reliable? How well does it work with
*? How's the quality? Features?
Thanks,
Waldo
Miranda,
Looks like you have a codec problem. Be sure that all terminals are
talking the same codec and if their settings in sip.conf or whatever
they using has the allow= for the codec in use.
Ex:
If you are using G711u:
allow=ulaw
Um abraco!
Isamar
On 5/25/05
-of-the-day.
The rest doesn't matter.
Isamar
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Good programmer is who makes the things working well *as planned* in the
time-limit planned beforehand, having good results for the *business* in
the end-of-the-day.
The rest doesn't matter.
Isamar
Sometimes you have to do things in a boring and unelegant way. You want
to do
,
Isamar Maia
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Hi Folks,
Where can I find a list of Predictive dialer solutions for Asterisk?
Thanks,
Isamar
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I guess the prices will go up like a rocket
Isamar
On Wed, 27 Apr 2005, MF Hulber wrote:
Have you seen this story? Cisco definitely wants to own the VoIP
market. I wonder what effect this will have on Sipura products.
http://story.news.yahoo.com/news?tmpl=storyu=/nf/20050427/bs_nf
Anybody would have the japanese voice files for *?
I need now the number's recording at least.
Thanks,
Isamar
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Jochen,
Recently I contact Aculab in UK about that and
They asked me to call Digium Sales.
I called Digium Sales and they told me that nothing is confirmed yet
about a deal between Aculab and Digium.
Maybe something changed
Isamar
On Mon, 11 Apr 2005, Jochen Witte wrote:
Hello
Isamar Maia wrote:
Technically speaking not. But Sangoma's support seems to be pretty much
better.
My understanding is that to an extent when we buy Sangoma we're putting
the dagger to Digium. They're glad to use Asterisk as a selling point
for their hardware, but unwilling to donate
Isamar Maia wrote:
I don't understand this *love* for Digium. Digium is a commercial
institution, period.
Yes, but. They are a commercial institution which took an enormous risk
by giving away for free what is undeniably their most valuable product.
So, if Linus Torvalds had
license to Mark Spencer.
If it's true...This kind of thing Digium *lovers* should take in
consideration.
Isamar
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For a easier comprehension, nowadays, H323 is like english. SIP is like
spanish and IAX is esperanto.
You can IAX. It's wonderful, modern, lot of advantages, pass through any
firewall, blah...blah..blah... but you can find only some strange guys
using that. :-)
Isamar
head, chan_oh323 0.7.1
Have anyone experienced this problem?
Thanks for any help.
Isamar
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Giovani,
Are you using a X100P ?
In my case here for a similar situation, the same happens because
the Zaptel takes sometime to understand the call was hangup.
Try to play with Busydetect/busycount option in zapata.conf
Isamar
On Fri, 11 Mar 2005, Giovanni Miano wrote:
Scenario
PSTN
the call(voice), I would flag that number as bad in the
DB. If it's a voice only answer machine, I would flag that number also as
bad. But if it's a fax or an answer machine with fax, I would flag that
number as valid fax number for future use.
Is that possible?
Thanks a lot,
Isamar Maia
Actually, it was requested to me to build a fax number database.
The real purpose is unknown. I am an IT guy, not marketing guy.
Isamar
On Sun, 20 Feb 2005, Torsten Krueger wrote:
Hello,
On Sun, 20 Feb 2005, Isamar Maia wrote:
For a specific application,
I want to dialout thousands
Ok. I will be burned in fire.. :-)
Or better.. I won't go to the heaven...
Isamar
On Sun, 20 Feb 2005, Andrew Kohlsmith wrote:
On February 20, 2005 08:30 am, Isamar Maia wrote:
I want to dialout thousands of numbers searching for fax machines.
You are an evil, evil man. Worse than
Are you using PC or mac?
Isamar
On Thu, 20 Jan 2005, Wilson Pickett wrote:
I would also suggest that while it is possible to do something, it is
not always wise :) See the significant volumes of reports in the
archives regarding multiple zaptel cards in one system.
I must be lucky: I
Again, good luck!
Isamar
On Wed, 1 Dec 2004, Asterisk users wrote:
Hi folks,
Im totally new to * but I went ahead and told my boss that it was the way
to go for our new telephone system :) now I have a test box and two cisco
phones and a brand new modem card.
Im having plenty
IPTABLES
VideoConference Device -- *(LAN) --- *(WAN) H323 Polycom
chan_h323 chan_iax chan_h323
or chan_oh323 or chan_oh323
Question before spending some time with it... should it work ?
Thanks,
Isamar
Yes. It is possible. But a driver was not implemented for that yet.
Isamar
On Fri, 12 Nov 2004, Kuniyoshi Murata wrote:
Hi,
Does anyone know if it's possible to make Asterisk's Caller ID function to
be compatible with Japan's Number Display system?
TIA
Kuni
--
Kuniyoshi Murata
the Asterisk's IAX connection
SIP doesn't work with firewalls.
Also, next time, post this kind of message in the asterisk-users list.
Isamar
On Wed, 3 Nov 2004, prasad_s wrote:
Hi all,
I am using asterisk, which is running on one machine having static(global) IP.
I have another
it is
almost gone.
With so long distances, there is nothing better than G.729.
Isamar
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Hi folks,
While upgrading the firmware of an ATCOM AT-323, the power was cut.
Now, it just shows Booting... in the panel and freezes.
Any thoughts?
Isamar
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What I heard is that you can sell pones but you cannot provide VOIP or
termination service over there.
Isamar
On Wed, 20 Oct 2004, Henry Devito wrote:
HI I am in the US and have a customer using * in the US they just acquired
a call center in India. Does anyone know if I can legally sell
It doesn't work. Period.
If want more, tell me your address and I will send a couple to you.
Dialout is not well or totally implemented in asterisk for this board.
Isamar
On Mon, 11 Oct 2004, FRANCISCO PEREZ-LANDAETA wrote:
Hi,
I am in the process of setting up an Asterisk PBX with some
Umar,
I agree with you. The license price for each G729 is very reasonable and I
don't see any reason or merit for this thread.
I think Digium and Asterisk should have other priorities. One of them is
the callerid stuff not working in some countries, specially Japan.
Isamar
On Sat, 25 Sep 2004
that can indicate any
error except some Training errors.
The sender fax thinks that the fax was sent successfully...
Kernel 2.4.21, Slackware, Processor Duron 850.
Any thoughts?
Isamar
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Hi FOlks,
I am trying to setup remotely an AudioCodes Mediant 2000 MG Module 2 to
work with Asterisk through SIP or H323.
But since I don't the product manual, it's being a little hard.
Anybody would the manual in PDF(file or URL) to indicate to me?
Thanks a lot,
Isamar
read that and didn't understand anything
Isamar
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Sorry. I was not following the thread, but...
What justifies this phone has this price of 99US$ while others are
for retail from 75 to 85US$ .?
Isamar
On Sat, 4 Sep 2004, SeshKanuri wrote:
Try this Phone at http://ipphone.eezeephone.com/
This Phone is listed now on ebay for sale at
http
Hi Folks,
I found some old postings about Sangoma card support in *
but nothing indicative if this is supported or not for dialin/dialout.
I found only support indication for VOFR using Sangoma...
Anybody other driver available for Sangoma even not free like
chan_dialogic ?
Thanks,
Isamar
Hi folks,
Anybody making fax-on-demand with * ?
Isamar
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I wanna do it through IVR.
I know how to create the .call files for normal outbound calls but how to
attach the .tiff files ?
Isamar
On Mon, 2 Aug 2004, Brian McManus wrote:
Yes I've implemented a simple web interface that generates a . call file
that faxes generated .tiff files
Me too.
I spent several months to make it working and I figured out it cannot
dialout... only dial-in.
Sell it to someone else and buy and buy a TDM-04b.
Isamar
On Thu, 22 Jul 2004 [EMAIL PROTECTED] wrote:
Nope...I scrapped that idea and just bought a Digium card.
-Original Message
Just for curiosity, Let us know how much time you'll gonna get a RMA of
it.
Isamar
On Thu, 22 Jul 2004, Andres Junge wrote:
I had the same problem, and it was that the power suppply coudn't handle
the new card. My solution (until i get a new power supply) was to unplug
a very big fan
aggressive echo cancel with the new algorithm
This machine has 1 TDM40b and 1 TDM04b and actually I don't know if it's a
problem in one or other. The phones directly connected to the line works
perfectly.
Did anybody have a similar problem?
Thanks,
Isamar
Looks like that the cellphone companies are getting prepared
to any possible competition...
http://www.thefeature.com/article?articleid=100878pos=1ref=1859764
Isamar
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I have already read explanation about that in some places but I don't have
still a clear image about the meaning of Penalty parameter inside of
queues.conf
What means that?
Thanks,
Isamar
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I'm planning to buy a T100P for a project in the company where I work
for but my concern is about the japanese ANI.
Can I get somehow japanese(NTT) ANI working with T100P ?
Feasible? Impossible ?
Thanks,
Isamar Maia
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Hi Folks,
I have the following situation:
I received an inbound call in my extension A and transferred it to the
extension B. But B was busy and I want to capture the call back to my
extension. How should I proceed?
Thanks,
Isamar
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I'm trying to do the following:
exten = i,1,Saydigits(${EXTEN})
My intention is to play the invalid input to the user, but it doesn't
work.
Any suggestions?
Thanks,
Isamar
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just want to confirm if it would work with the current
chan_dialogic.
Thanks,
Isamar
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terrible echos happen.
Is there any workaround for this case?
Thanks,
Isamar
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allow=ulaw
Why don't you remove this?
Isamar
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?
Which kernel version are you using? Can you send me your .config ?
Thanks again,
Isamar
On Thu, 24 Jun 2004, Martin List-Petersen wrote:
Is your kernel ACPI enabled ?
The motherboard in the PE400SC is basically the Dimension 8300, which i
use for my development box with 1 X101P, 1 TDM400P
Nakano San,
Have you tried to make * only to route the connection and
they just talk point-to-point without * bridging?
Isamar
On Thu, 24 Jun 2004, Masakazu Nakano wrote:
I tryed it.
but callee cannot answering with video in SIP.
# surely videosupport=yes in sip.conf
H.323 is works
connection and the PSTN running in the asterisk.
What I am asking to myself now if it is technically possible
without transcoding or having G729 licenses.
Isamar
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It's already possible to use VideoPhone with Asterisk.
I'm planning to buy 2 of them. Anybody using any Video SIP phone with
asterisk?
Isamar
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time is 5 min. Firmware version 1.5.0.0
Asterisk version is 7.2
Anyone has any clue?
Isamar
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Hi,
There is already any CGI script to show the active online extensions
through the web?
Thanks,
Isamar Maia
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I tried some combinations of setup seen in some postings
and didn't get success on this yet.
I have grandstream phones outside the network trying to
call an * server inside my network through NAT/Iptables.
The problem that I'm facing is one-way audio.
Any suggestion?
Thanks,
Isamar
This isn't directly related to your question, but I just recently committed
some more documentation on IRQ sharing to the http://www.asteriskdocs.org
book. Feel free to check it out, it may be of benefit.
Finally, TDM400P also has IRQ issues, right?
Isamar
AFAIK.. it shows up a crazy error...
The G.729 crying for more licenses...
Isamar
On Tue, 1 Jun 2004, Mike Heininger wrote:
Hi,
if the G.729 codec runs out of licenses does * fallback to another
codec?
TIA,
Mike
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I had a little nightmare playing with X100Ps and IRQs and I
decided to buy TDMP400P/FXO and FXS.
The question is, can I put multiple boards in the same motherboard
without worrying about IRQS? TDM400P shares IRQs with other boards?
Isamar
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From where can I download the portuguese sounds?
Thanks,
Isamar
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Yes. Give away your LJs to some university for research...
They are not for business... and don't buy X100P. Buy TDM400P.
It has the same price of a LJ and have 4 FXOs instead of only one.
Isamar
On Wed, 19 May 2004, Jer wrote:
Dear all
I read on the list back in 2003 that * does
a lot. I'm trying to find a motherboard with more IRQs until today.
If I don't get it, I'm gonna order the TDM400P(4FXO), otherwise I would
need to have two machines what'd be a big mess.
Isamar
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to accomplish that ?
Currently, I'm playing with an ASUS A7V600.
Thanks for any tip,
Isamar
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-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Isamar Maia wrote:
| Hi folks,
|
| I'm trying to make an * PBX for a customer using 4 X100Ps
| and 1 TDM400p(4FXS).
| The problem I'm facing is to make one unique IRQ for each
| PCI slot/board since shared IRQs create all kind of weird
to Digium one time about that, but I've heard that it's not a
priority. One year is passed.
Isamar
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on this list in the
past 7 days.
I didn't get this yet. The helicopter noise still sounds
I'm changing the cables today to eliminate any interference possibility.
Isamar
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echo cancel.
Both lines are coming from an ISDN line,channels A and B respectively.
Should it be cable problem or another issue, in this case with ISDN lines?
Isamar
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.
Isamar
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Nagoya/Japan
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think it was because of MARK Spencer... burn him! :-)
Isamar
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-archive.com/[EMAIL PROTECTED]/msg30982.html
?
or something like
http://www.mail-archive.com/[EMAIL PROTECTED]/msg02388.html
?
Thanks,
Isamar
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Me too.
Isamar
On Mon, 29 Mar 2004, Paul Mahler wrote:
Where and when is the rollout meeting? I'd love to attend.
Thanks!
Paul
Paul Mahler
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I have two X100Ps for testing.
I want to receive a call from one and dial out through the other
available.
When I make a call, it try to dial to the channel in use.
How to solve this?
Isamar
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