I am trying to compile the asterisk and if fails at
the end on:
make[1]: Entering directory
`/usr/src/asterisk-0.5.0/pbx'gcc -shared -Xlinker -x -o pbx_gtkconsole.so
pbx_gtkconsole.o `gtk-config --libs
Hi all,
Please send all the feedback related to the ActiveX
version of DIAX directly to me, not to the list.
I'll try to handle each request
individually.
Thank you for your understanding,
Dan
P.S. The first version is tested and must works on
Windows 98SE/2000/XP.
Hello
Can asterisk be configured as a PBX with a
third party gateway (cisco router 3640 running Cisco call manager
express). The cisco gateway will only interface the PSTN and asterisk, so the
cisco routerwill handle incoming and outgoing calls. I would like to do
this as we have the
Hi,
For the ones who does not have a web server to test, there is a demo of DIAX
ActiveX available online (for another 4 hours) at:
http://193.231.214.47:25380/dax.htm
If you accept to enable unsigned ActiveX to be downloaded and run on your
PC, then you can play with it (on your own risk..:-))
Hi all,
Clipcomm - looks interesting, you get NAT/Ethernet/Analog line (to SIP)
D-link DVG-1120M/H/S - this is also on the lines of what I'm looking for,
lets you connect standard analog phone directly to it, has NAT and an
Ethernet port - anyone ever tried this with * ?
What I am exactly
On Mon, 2003-12-22 at 06:50, Deepakumar JV wrote:
Hello
Can asterisk be configured as a PBX with a third party gateway (cisco
router 3640 running Cisco call manager express). The cisco gateway
will only interface the PSTN and asterisk, so the cisco router will
handle incoming and outgoing
Hi Ernest,
I have installed as you described, and now it worked.
Seems that installing a minimum system and afterwards installing the
necesary packages with their dependencies seems to not have worked for
me
Thanks for the help all...
David
-Oorspronkelijk bericht-
Van: [EMAIL
Hi list, Is there anyone using the voicetronics openline4 with asterisk.
Does this card work ok for 4 port analoge fxo? Thanks, Tjapko.
---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.551 / Virus Database: 343 - Release Date:
I am after using a web crm system which has a button to then
get asterisk to dial the contact. For this I was looking at call files, which
appear good for the job, I have one small problem with them though.
1/ file is created
2/ external number is called
3/ the external party answers
I'm missing something here. I've put the following in extensions.conf and a
few variations thereof. I've taken the sample configs and added to them, so
when I dial 2200 from netmeeting * answers and runs me through the demo
announcements.
The pots extensions 2200 and 2107 (TDM400) work fine
Hello All,
Another question todayJ
I have just started playing with windows messenger and
asterisk following the little how-to from the asterisk web-site work
well good sound quality but you cannot put people on hold transfer them
or send DTMF (to get asterisk to do the transfer)
Hi,
Why don't you turn the process around:
1/ file is created
2/ internal number is called
3/ internal party answers
4/ internal party hears ringing as the external party is being called.
Ofcourse everything depends on how you have built your dialplan since you'd
need to have access to an
There are 2 ways to place a call in MSN messenger,
either place a voice message or place a call. Place a voice message is like
chat, but you put in your [EMAIL PROTECTED]but if you
use this option, you can't send dtmf digits, place a call drops down a dtmf
keypad. If you don't get the
Hi,
anybody knows a VOIP-PSTN company that doesn't charge monthly for making PSTN calls?
thanks.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
http://www.iconnecthere.com and http://connect.voicepulse.com as long as you don't
need an incoming
phone number.
- Original Message -
From: Hector Q.-datafull [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, December 22, 2003 10:34 AM
Subject: [Asterisk-Users] no monthly fee
Yes,
Check out VoicePulse, they bill by the minute with no monthly fee.
IConnectHere also allows plans that bill by the minute, however there is a
0.65/month fee. Which, considering the ammount you'll save, is very very
tiny.
Regards,
Brent
On Mon, 22 Dec 2003, Hector Q.-datafull wrote:
Hi,
Hi guys. First off, to the folks at Digium: outstanding work. The fact
that Asterisk is open source puts you right at the cusp of what will be
the most important telecom advance since the transatlantic cable.
Anyway... a couple newbie questions concerning sound quality - I don't
see any reason
www.nufone.net
On Mon, 22 Dec 2003, Jim Flagg wrote:
http://www.iconnecthere.com and http://connect.voicepulse.com as long as you don't
need an incoming
phone number.
- Original Message -
From: Hector Q.-datafull [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, December 22,
i think nufone and xvoip are based on a per min basis prepaid perhaps
but no monthly fee there is probably others as well
On Mon, 2003-12-22 at 10:34, Hector Q.-datafull wrote:
Hi,
anybody knows a VOIP-PSTN company that doesn't charge monthly for making PSTN calls?
thanks.
try ztmonitor 1 -v
Martin
On Sat, 20 Dec 2003, Daniel Bichara wrote:
Hi,
I am trying to run ZTMonitor to get debug info from my E100P board but I
got the following message:
-bash-2.05b# ./ztmonitor 1
Unable to open /dev/dsp: No such file or directory
Cannot open audio ...
-bash-2.05b#
iTS [EMAIL PROTECTED] wrote:
Hi list, Is there anyone using the voicetronics openline4 with asterisk.
Does this card work ok for 4 port analoge fxo? Thanks, Tjapko.
We are using openline4 for testing purpose, not in production. It seems
to work.
Jorge
You are lucky. I'm getting this:
-- Incorrect password '1334' for user
When I enter 1234. I'm using dtmfmode=rfc2833 and a
GS Budgtone 100 phone. Why do I getr 4x while you get 2x ??
use dtmfmode=info (both in sip.conf and in your GS settings, of course)
search
- Original Message -
From: Sean Adams [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, December 22, 2003 10:50 AM
Subject: [Asterisk-Users] Audio format for announcements
Hi guys. First off, to the folks at Digium: outstanding work. The fact
that Asterisk is open source puts you
Just a note to Mark and others.
In queue.conf, there is a reference to announce-markq that I believe
comes default uncommented.
There is no sample file in /var/lib/asterisk/sounds/announce-markq
If there is no file there and/or you misspell the filename and the
system can't find the announce
Hi!
I'm also curious if anyone else is doing this or if anyone else is using
the Asterisk TDD support.
Excuse my ignorance: What exactly is TDD? Is it US specific?
Philipp
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We have people connecting to an asterisk box over the internet. They're
using the x-lite client behind linksys firewalls. The X-Lite client
discovers the firewall no problem and connects to Asterisk without a
problem. After connecting the agent shows up properly in sip show
peers with the
You need to have HDLC generic support compiled into your kernel ... I
think it's not good to have it compiled in modules ... just embedded in
kernel.
Martin
On Sat, 20 Dec 2003, Daniel Bichara wrote:
Hi All,
I wish to connect * to a Cisco using a E100P board.
When I load the driver I got
Thanks, any special configuration requirement?
Tjapko.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jorge Mendoza
Sent: Lunes, 22 de Diciembre de 2003 05:01 p.m.
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] voicetronics
iTS [EMAIL PROTECTED]
Telecommunications Device for the Deaf
-Original Message-
From: Philipp von Klitzing
[mailto:[EMAIL PROTECTED]
Sent: Monday, December 22, 2003 11:13 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] ToIP (TDD over IP)
Hi!
I'm also curious if anyone else is doing this or if anyone
On Mon, 2003-12-22 at 09:50, Sean Adams wrote:
Hi guys. First off, to the folks at Digium: outstanding work. The fact
that Asterisk is open source puts you right at the cusp of what will be
the most important telecom advance since the transatlantic cable.
Anyway... a couple newbie
Hi mack_jpn
I think problem is CFR 84 sending. In console appears that CFR 84 si
sent but te other fax doesn't receive CRF 84, and then RXFAX is waiting
for the fax but the other fax doesn't send it ever. I try to see source
code.
Regards,
srsergio
-Mensaje original-
De: [EMAIL
TDD is a very simple teletype like unit for Telecommunications for the
Deaf
Which is hooked up to a telephone line with an acousic coupler remember
these ?
It transmits with 45 baud / BAUDOT code , but unlike regular modems the
carrier is removed once the key has been released.
TDD is supported by
Hi,
I am trying to set up * and ISDN-PRI (channels 1 - 15) using E100
boards. I installed zaptel and libpri. When I execute modprobe -r
wct1xxp I get an error message:
ZT_CHANCONFIG failed on channel 1: Function not implemented (38)
/lib/modules/2.4.21-SAX/misc/wct1xxp.o: post-install wct1xxp
The registry expires after sime time. You can set the default expirey and
max in sip.conf. It's up to your phone/sip device to reregister after the
registration expires.
Martin
On Mon, 22 Dec 2003, Jonathan Tew wrote:
We have people connecting to an asterisk box over the internet. They're
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean Adams
Sent: Monday, December 22, 2003 10:50 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Audio format for announcements
[...]
2) For my internal SIP phones, I don't care about bandwidth
On Monday 22 December 2003 10:12, Philipp von Klitzing wrote:
Hi!
I'm also curious if anyone else is doing this or if anyone else
is using the Asterisk TDD support.
Excuse my ignorance: What exactly is TDD? Is it US specific?
It's a specification for sending words over a normal telephone,
My Strandstream BT100 is working OK for both inbound and outbound now
except that when you speak into the handset you cannot hear your
own voice in the earpeice. It works OK, the other end can hear the
call but most telephone users have become used to hearing their own
voice.
Is this
Is there a way to set to audio gain for each SIP extension?
I see in the docs this can be done for zaptel but I don't
see it documented for SIP. It would be nice to be able to
make the various kinds of extensions have equal volume.
=
Chris Albertson
Home: 310-376-1029 [EMAIL PROTECTED]
My guess would be that the NAT firewall times out and closes the port.
Reopening it from the inside is no problem, but access from the outside gets
blocked.
In order to keep the path open both ways, the client needs to send some kind
of messages with the proper IP/port in regular intervals.
Their firewall may be timeing them out. Try adding qualify=60 to each
of the entries in sip.conf
On Mon, 2003-12-22 at 10:26, Jonathan Tew wrote:
We have people connecting to an asterisk box over the internet. They're
using the x-lite client behind linksys firewalls. The X-Lite client
Well thats broken.. we have a bug on bugs.digium.com over this.
bkw
On Mon, 22 Dec 2003, Martin Pycko wrote:
try ztmonitor 1 -v
Martin
On Sat, 20 Dec 2003, Daniel Bichara wrote:
Hi,
I am trying to run ZTMonitor to get debug info from my E100P board but I
got the following message:
I am after using a web
crm system which has a button to then get
asterisk to dial the
contact. For this I was looking at call files,
which appear good for
the job, I have one small problem with them though.
1/ file is created
2/ external number is
called
3/ the
I tried running the festival app today with little success. I have a
working festival installation that does TTS to the linux sound output
perfectly.
With * it just produces a sort of hissing sound. The length of hissing is
proportional to the length of text string that it is given to speak.
Sorry for the late reply.
I try port 5060 and it just knocks me back straight
away, I cant see it even try to authenticate in the CLI.
X-lite works both inside the LAN and outside using
SIP.
Messenger version = 4.7
John I will try your suggestion with sip.conf thanks
for the
Hi!
What I would like to get round this is probably the reverse I dont
want the people I am calling to hear ringing. For example as soon as it
Swap the numbers around.
I cannot figure this out - just swap them round?
But if I swap it round
Channel: SIP/User
MaxRetries: 2
Ok here is another problem I have run into.
I have a Sipura 2000 and I have been able to configure line 1 with only
one small problem. But I can't get the line 2 working with asterisk.
Here are samples of my sip.conf and extensions.conf. If I disable line
1 I can then get line 2 working. Is
You have SIP/lcs-sipura1 listed for both extensions in your extensions.conf.
Is this a type-o in your email?
-Original Message-
From: Ariel Batista [mailto:[EMAIL PROTECTED]
Sent: Monday, December 22, 2003 1:11 PM
To: Asterisk User List
Subject: [Asterisk-Users] Sipura 2000 configuration.
exten = 203,1,Dial(SIP/lcs-sipura1)
exten = 204,1,Dial(SIP/lcs-sipura1)
dont you mean:
exten = 203,1,Dial(SIP/lcs-sipura1)
exten = 204,1,Dial(SIP/lcs-sipura2)
?
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[EMAIL PROTECTED]
1) Is it possible to store the menu sounds in wav
...sure, just put your 8kHz 16 bit mono files named whatever.wav in
/var/lib/asterisk/sounds - asterisk will convert them to what is needed if
needed.
John
This e-mail was scanned and found clean by Monroe-Woodbury's Antivirus.
iTS [EMAIL PROTECTED] wrote:
Thanks, any special configuration requirement?
Nop, but you need to patch channel_vpb.c. Search the archives (oct - nov?)
Jorge
Tjapko.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jorge Mendoza
Sent: Lunes, 22 de
I cracked the concept of how to handle incoming calls and route them to the
right context, apologies for being a little slow on the uptake.
I can now call between pots end points and netmeeting endpoints. Still
having problems with sound despite having set everything to use G711A.
POT to POT
First off, here is what I want to do:
SIP Clients - SER - Asterisk - VoIP provider
Where SER will handle communications between SIP
clients (since I would prefer that my SIP clients not
use all of my bandwidth)
Asterisk will handle calls to a VoIP provider
I have read that people have similar
On Monday 22 of December 2003 17:54, Daniel Bichara wrote:
ZT_CHANCONFIG failed on channel 1: Function not implemented (38)
/lib/modules/2.4.21-SAX/misc/wct1xxp.o: post-install wct1xxp failed
/lib/modules/2.4.21-SAX/misc/wct1xxp.o: insmod wct1xxp failed
your zaptel.conf is wrong
it has to be:
how did u setup your asterisk for this:
I can also start a call through Asterisk to a VoIP
provider, but there is a problem after the first ring:
- Original Message -
From: jerk face [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, December 22, 2003 2:42 PM
Subject:
I installed * to primarily test its voicemail
feature. I installed it on a server WITHOUT any telco board (i.e.,
digium). Installation looks ok, however I am having problems.
MY SETUP:
2xATAs are configured to use * as
GkorProxy
Asterisk is registered to my SER SIP/RTP
Proxy
1.) First
Hi!
I try port 5060 and it just knocks me back straight away, I cant see it
even try to authenticate in the CLI.
You won't see anything unless you type sip debug in the CLI.
Philipp
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On Mon, 22 Dec 2003, Philipp von Klitzing wrote:
Excuse my ignorance: What exactly is TDD? Is it US specific?
TDD - Telecommunications Device for the Deaf (also used by people with
speech problems). Also known as a TTY (Telephone Typewriter) or TDY (not
sure what it means)
I don't know if it
In sip.conf I have the following:
context=OUTGOING
autocreatepeer=yes
[Provider]
type=friend
username=X
secret=X
host=x.FakeProvider.com
So when Asterisk receives a call from SER it will
autocreatepeer and give access to the OUTGOING
context.
--- Jess Magnaye [EMAIL PROTECTED]
Hi,
I have a question regarding the Asterisk Packet Time for SIP Calls. It is
hardcoded at 20ms but when I do an RTP Analysis on a stream it is clear that
these packets are not spaced out at 20ms. In general you see something like:
Packet 50 - Delay 50ms
Packet 51 - Delay 5ms
Packet 52 -
I have a question regarding the Asterisk Packet Time for SIP Calls. It is
hardcoded at 20ms but when I do an RTP Analysis on a stream it is clear that
these packets are not spaced out at 20ms. In general you see something like:
Packet 50 - Delay 50ms
Packet 51 - Delay 5ms
Packet 52 -
Packet 50 - Delay 50ms
Packet 51 - Delay 5ms
Packet 52 - Delay 5ms
Packet 53 - Delay 50ms
Packet 54 - Delay 5ms
Packet 55 - Delay 5ms
The 20 ms is not the inter-packet timing, its the relative content of
what's within the packet. In other words, the packet contains 20ms of
encoded
I might be wrong, but isn't is just saying that the packet has been delayed
x-ms? I'm not sure it's saying that Packet 52 arrived 5ms after packet 51.
Although even if it was, that doesn't mean that it was sent 5ms after packet
51 either.
-Original Message-
From: Andrew Kohlsmith
Hi list,
I have a asterisk box with an E400P that was running ok until last
week.
The machine just stop responding and after a reboot, the module (tor2)
doesn't load anymore.
anyone could help?
regards
Eduardo
modprobe returns this:
asterix:~#
Packet 50 - Delay 50ms
Packet 51 - Delay 5ms
Packet 52 - Delay 5ms
Packet 53 - Delay 50ms
Packet 54 - Delay 5ms
Packet 55 - Delay 5ms
The 20 ms is not the inter-packet timing, its the relative content of
what's within the packet. In other words, the packet contains 20ms of
jerk face wrote:
In sip.conf I have the following:
context=OUTGOING
autocreatepeer=yes
[Provider]
type=friend
username=X
secret=X
host=x.FakeProvider.com
So when Asterisk receives a call from SER it will
autocreatepeer and give access to the OUTGOING
context.
Could you please explain
I have tried this from the manager console and call files and it doesn't
seem to work the other way round. It will call the sip channel but not
the capi channel - in fact with capi debug this doesn't show anything
getting through
Asterisk monitor comes up twith
Attempting call on sip/nick ofr
On Monday 22 December 2003 15:36, Rich Adamson wrote:
I have a question regarding the Asterisk Packet Time for SIP Calls. It
is hardcoded at 20ms but when I do an RTP Analysis on a stream it is
clear that these packets are not spaced out at 20ms. In general you see
something like:
On Monday 22 December 2003 15:55, Andrew Kohlsmith wrote:
Packet 50 - Delay 50ms
Packet 51 - Delay 5ms
Packet 52 - Delay 5ms
Packet 53 - Delay 50ms
Packet 54 - Delay 5ms
Packet 55 - Delay 5ms
The 20 ms is not the inter-packet timing, its the relative content of
what's
Hi guys,
I posted a somewhat similar question about a month ago and got a
thoughtful resonse from Steven Critchfield, but I've got a quick follow
up question to it.
I'm looking to setup a 16 extension / 10-14 phone line Asterisk install
for a customer who would like to have DID numbers for
On Mon, 2003-12-22 at 15:23, Eduardo Goncalves wrote:
Hi list,
I have a asterisk box with an E400P that was running ok until last
week.
The machine just stop responding and after a reboot, the module (tor2)
doesn't load anymore.
anyone could help?
Supports H323
http://www.viseon.com/prod/c_VisiFone.asp?id=133
E h323 evil.
On Mon, 22 Dec 2003, Steve Totaro wrote:
Supports H323
http://www.viseon.com/prod/c_VisiFone.asp?id=133
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On Monday, December 22, 2003 3:40 PM, john lawler
[SMTP:[EMAIL PROTECTED] wrote:
Hi guys,
I posted a somewhat similar question about a month ago and got a
thoughtful resonse from Steven Critchfield, but I've got a quick
follow
up question to it.
I'm looking to setup a 16 extension / 10-14
autocreatepeer
I just found out about this today from the
Asterisk-Dev mailing list.
The email was from John Bigelow and is as follows:
This will allow any sip user to register with asterisk
with no
authentication.
So if you are lazy or for whatever reason do not want
to create the
peers in the
On Monday 22 December 2003 16:37, Andres wrote:
On Monday 22 December 2003 15:36, Rich Adamson wrote:
I have a question regarding the Asterisk Packet Time for SIP Calls. It
is hardcoded at 20ms but when I do an RTP Analysis on a stream it is
clear that these packets are not spaced out
On Monday 22 December 2003 16:37, Andres wrote:
On Monday 22 December 2003 15:36, Rich Adamson wrote:
I have a question regarding the Asterisk Packet Time for SIP Calls. It
is hardcoded at 20ms but when I do an RTP Analysis on a stream it is
clear that these packets are not spaced
I think we've having some luck with this setting. Of course we had to
crank it up higher so that it didn't consider the clients LAGGED. When
the clients were LAGGED they couldn't receive any calls for some
reason. So like a setting of 200ms seems to work fine for everyone.
Eric Wieling
Sorry for the dup post but never got a reply so
I am reposting below:
I am trying to compile the asterisk and if fails
at the end on:
make[1]: Entering directory
`/usr/src/asterisk-0.5.0/pbx'gcc -shared -Xlinker -x -o pbx_gtkconsole.so
pbx_gtkconsole.o `gtk-config
On Monday 22 December 2003 19:58, Rich Adamson wrote:
On Monday 22 December 2003 16:37, Andres wrote:
On Monday 22 December 2003 15:36, Rich Adamson wrote:
I have a question regarding the Asterisk Packet Time for SIP Calls.
It is hardcoded at 20ms but when I do an RTP Analysis on a
Hi,
On Tue, 2003-12-23 at 12:12, [EMAIL PROTECTED] wrote:
[...]
I am trying to compile the asterisk and if fails at the end
on:
make[1]: Entering directory `/usr/src/asterisk-0.5.0/pbx'
gcc -shared -Xlinker -x -o pbx_gtkconsole.so
use this
[3001]
type=friend
;username=3001
;fromuser=Craig1
;secret=secret
host=dynamic
mailbox=3001
context=sip
dtmfmode=info
auth=plaintext
make sure ur MSN version is
4.7.0105.
-B
- Original Message -
From:
Craig
Waddington
To: [EMAIL PROTECTED]
Sent:
Tilghman Lesher wrote:
On Monday 22 December 2003 10:12, Philipp von Klitzing wrote:
Hi!
I'm also curious if anyone else is doing this or if anyone else
is using the Asterisk TDD support.
Excuse my ignorance: What exactly is TDD? Is it US specific?
It's a specification for
Hello,
I am trying to figure out how much bandwidth asterisk requires using IAX between 2
boxes if all available channels are used.
Scenarios:
A. 1 TE410P---Asterisk A --- Internet --- Asterisk B--1 TE410P
B. 2 TE410P---Asterisk A --- Internet --- Asterisk B--2 TE410P
C. 3
Steve Totaro wrote:
Supports H323
http://www.viseon.com/prod/c_VisiFone.asp?id=133
So? Whilst there are still only a few VoIP audio phones available,
almost every computer related manufacturer in Asia has at least one
video phone model like this. There must be dozens of units like this
Hi,
Im conecting to * servers using IAX2 no NAT in my setup. I read Wikis
docs and lists archives but there is no recomendation about what to use.
If I understand I can use friend in both sides but its isnt
recomended; so I should define both a peer and a user on each box ?
cause AFAIR in
They recently announced SIP.
See:
http://www.voip-info.org/wiki-Viseon+VisiFone
Anyone know where to actually buy one?
Jim
James H. Thompson
[EMAIL PROTECTED]
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, December 22, 2003 12:31 PM
Hi All,
i dont what changes i made recently but i am unable
to hold the call for more 10 secs between MSN and GS. PSTN to MSN and PSTN to GS
and vice versa works fine. i am not behind NAT.Also MSN to MSN works fine
too.
my SIP details
[general]port = 5060bindaddr =
0.0.0.0context =
/usr/X11R6/lib/libXext.so.6
.. is part of the XFree86-libs RPM. Find the corresponding tgz, install
it and then try to compile again. It should get past that error.
That did it thanks!
John
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It's a specification for sending words over a normal telephone,
normally used by the deaf. It resembles the old-style modems in
that the handset is interfaced with a microphone and speaker.
This allows TDD to be used with payphones, which do not have
an RJ-11 interface.
Does anyone use
Hello,
I am setting up a VOIP system using * for our remote located broadband
customers. We are bringing in a full voice T1 with 24 channels and going to
use the wildcard T100P. Its going to be another 2 weeks before our voice T1
is installed and I want to take that time to setup our * box. Its
I want wish You all - asterisk-men - Merry Christmas and excellent New Year !
Radoslaw
---
Let's talk about what connecting us together (c) Lafinion [RW]
http://www.lafinion.com mailto:[EMAIL PROTECTED]
: MSN: [EMAIL PROTECTED] Yahoo ID: Lafinion ICQ: 323220515 :
: AIM:
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