Dave,
'algo' is the file in which u store the recorded conversation.
u can find this file in /var/spool/asterisk/monitor
Girish
From: "Dave Packham" <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: <[EMAIL PROTECTED]>, <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Meetme Recording
Date: T
Hi Jean-Denis,
- Original Message -
From: "Jean-Denis Girard" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 02, 2003 11:16 PM
Subject: Re: [Asterisk-Users] Multilingual version of DIAX
> Dan wrote:
> > Hi all,
> >
> > The multilingual version of DIAX with both IAX
- Original Message -
From: "Paul Lambert" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Cc: "Paul Liew" <[EMAIL PROTECTED]>
Sent: Wednesday, December 03, 2003 4:16 AM
Subject: Re: [Asterisk-Users] incominglimit stuck in app_queue
> I've seen this same thing. But it doesn't happen only for
Hi all,
I just trying to test MSN 4.7 that has SIP.
Because with him i can use a video and voice transmission and * .
But when i try to call someone using the DIALPAD of MSN, when i insert any digit into * the numbers appears twice !!
like this.
channel 456 appears in asterisk 445566
How c
On Tue, 2003-12-02 at 16:35, reginald huey wrote:
> Please
>
> Remove me from the list
There should be a link to unsubscribe yourself at the bottom of this
message. Read ANY other message, and they are there. This is a
self-serve mailing list.
--
Leif Madsen <[EMAIL PROTECTED]>
http://www.ha
The i2004 does not handle SIP natively. There is an addon-software for a PC
that will do SIP with an i2004. But the software is not available publicly.
- Original Message -
From: "Alexander Romanov" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 02, 2003 9:00 PM
Sub
- Original Message -
From: "Gary" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 02, 2003 9:07 PM
Subject: Re: [Asterisk-Users] VoiceGlo
> which would make their "Multimedia Terminal Adapter" an interesting
> device ??
>
Interesting yes, but it does not support IAX
FWIW, my employer manufactured broadcast graphics hardware. Our
training courses run $4500 USD for a 3 day package, + $1000/day
thereafter. That's $6500/wk, including travel and expenses. This covers
training, commisioning and integration...and is about the industry
norm.
Vertical market training
man I feel dumb. I speak spanish (i lived in BA Argentina for a few years) and I
knew about the |filename option. I guess i was thinking that "algo" was the options
"a l g o" . heh
Dave
>>> [EMAIL PROTECTED] 12/2/2003 3:23:02 PM >>>
"algo" is a file where app write a wav data. In spani
The new versions of iaxcomm and DIAX are both now using the iax2 protocol.
So in order to receive incoming calls on either of them in your
extensions.conf file change IAX/clientname to IAX2clientname. Then you
should be able to receive incoming calls on either iaxcomm or DIAX. Also
there is
which would make their "Multimedia Terminal Adapter" an interesting
device ??
On Wed, 3 Dec 2003 10:41:15 +1100, Adam Hart wrote:
>did you even read what I said?
>
>> > but if you look, it's actually using iaxcomm
>
>
>- Original Message -
>From: "Brian West" <[EMAIL PROTECTED]>
>To: <[E
Is anyone successfully using this phone with Asterisk? There is a lot
mentioned about CISCO but nothing about Nortel...
Alex.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Yes.. just letting you know that it was working with * :P
On Wed, 3 Dec 2003, Adam Hart wrote:
> did you even read what I said?
>
> > > but if you look, it's actually using iaxcomm
>
>
> - Original Message -
> From: "Brian West" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Wednes
Hello,
We ordered 100 grandstream 102's from them over a month ago, we got the
first shipment of 40 within a week and a half which was great. We got the
next 40 a few weeks later. And we still have had no communications as to
when the last 20 of the phones that we ordered(and paid for) over a mont
Hi all,
I am setting up * for the first time, every thing is working
fine, but I would like to implement an additional feature:
Thus we have multilingual caller menu – I would like
to play a little sound file to the callees to let them know in which language they
should answer the inco
If you have echo on the X100P's Mark setup chan_zap to pretrain the echo
can, but it had a few issues until today which Mark nailed down the bug
that caused the DTMF to be unreliable.
Ok here is how you would do it:
in your zapata.conf.sample:
; In some cases, the echo canceller doesn't train qui
Is there a way to make the voicemail message announcement include
the callerid. It would be handy to know who called (well, at least
where the call was from) especially if they just hung up.
I know I can get it from msg.txt but for the lay user it would
be much more handy if it was included
This is the expected behaviour when running Asterisk on a system with
graphics also running.
Deron Wilkerson wrote:
I've setup asterisk on a Dell Poweredge 1300 Server. PIII 600, 500mb
Ram, SCSI HD's. It is running Fedora Core 1. I have 3 X100P cards and 6
Grandstream IP phones.
Everything works
That is correct, & also chan_zap.c rev 1.47 seems to have
resolved as of today
- Original Message -
From:
Softprofit Solutions
To: [EMAIL PROTECTED]
Sent: Tuesday, December 02, 2003 1:14
PM
Subject: Re: [Asterisk-Users] Tone
Detection Problem
I don't think
On Tue, Dec 02, 2003 at 04:28:07PM -0500, Bisker, Scott (7805) wrote:
> Otherwise, the phone seems to work OK except for a slight flickering
> of the LCD (hence my suspicions that this might be a hardware issue.)
Dodgey power supply unit?
David.
___
On Wed, 2003-12-03 at 03:26, Aaron Martin wrote:
> Sorry to everyone on the list, but for some reason this is the only
> reliable way to get hold of John.
>
> John Brown of Chagres Technologies, please contact me! I have been
> trying for weeks now to get hold of you via email and phone after wi
I just talked to him lastnight... He was out of the office for a week or
so. He got back and had to fire a few people for not doing their jobs..
and that he is slowly but surely getting caught up and that QWest
screwed up their number porting. They moved their numbers from QWest to
anohter pr
Deron Wilkerson wrote:
The only way I can get good sound quality with the FXO cards is to
start * while in a terminal window in X windows. Close the terminal
window. Close X. Then reconnect to * using asterisk -gcr from the
command line.
You answered your own question: Do NOT run XWindows
echotraning = yes was fixed for x100p's today. It should work properly
and knock echo off instantly. I nolonger get 5-10 seconds of echo from
SIP -> ZAP now.
w00t
bkw
On Tue, 2 Dec 2003, Softprofit Solutions wrote:
> I don't think so, is it zapata.conf , echotraining = yes
>
> Please confirm
I have a few * boxes spread around at different locations with different
ISP's. I have 1 location with a static IP, the rest
are all dynamic and all are NAT.
I can tell when ever the remotes have a change of IP from looking
at the IAX registrations and now know the new IP.
I was thinking of lett
did you even read what I said?
> > but if you look, it's actually using iaxcomm
- Original Message -
From: "Brian West" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, December 03, 2003 5:21 AM
Subject: Re: [Asterisk-Users] VoiceGlo
>
WROOO
> From: Philipp von Klitzing
> Sent: Tuesday, 02 December, 2003 10:50
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] How to restart * thru phone "when
convenient"
> > You could use "at" to issue the command at a deferred time.
> Yes, sure, but this ain't that nice "asterisk only". :->
Y
I am using a brand new RH9.0 installation. I installed Asterisk afterwards so I am not
sure if Asterisk caused the problem below. The ps doesn't work. It could also be
something else. I also tried installing a some video package. But I thought to ask
here first if someone has seen this before.
- Original Message -
From: "Olle E. Johansson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 02, 2003 7:24 PM
Subject: Re: [Asterisk-Users] IAX2 Ethereal plugin v0.3 is out
> Alastair Maw wrote:
> > On 28/11/03 07:39, Olle E. Johansson wrote:
> >
> >>> The latest ve
On Tue, 2003-12-02 at 12:55, Howard White wrote:
>
>
> The magic question to ask CBeyond is whether the T1 they provide you is
> Primary Rate Interface (PRI) or Basic Rate Interface (BRI). Their web
> site is too heavy on pretty marketing and wy short on technical
> details. PRI gives you
"algo" is a file where app write a wav data. In spanish, "algo" means
"something"... :)
Gus
-= Info about application 'Monitor' =-
[Synopsis]:
Monitor a channel
[Description]:
Monitor
Used to start monitoring a channel. The channel's input and output
voice packets are logged to files until th
- Original Message -
From: "Senad Jordanovic" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 02, 2003 2:14 PM
Subject: RE: [Asterisk-Users] VoiceGlo
> Hi,
>
> Anyone knows what USB phone are they using? Where can one get it from?
> http://www.voiceglo.com/pages/Pr
Hi robert,
I found that the disallow=all and then specify a codec with allow= was required in
sip.conf.
[17471234567]
type=friend
username=17471234567
secret=censored
host=dynamic
nat=yes
disallow=all
allow=ulaw
Jon Hopper
robert ivanc <[EMAIL PROTECTED]> wrote ..
> Hi,
>
> i've just got 2 gr
What they are probably marketing is putting in their own equipment out
there. I install a product that does exactly that. A paradyne jet
fusion. It takes care of the part of which channels are data and which
are voice.
If it's anything like these, the lines will come out on pairs. You will
the
Please
Remove me from the list
ReggieReginald Huey
Do you Yahoo!?
Free Pop-Up Blocker - Get it now
Hi Todd,
Yes Asterisk supports G.729a, you can by licenses
for it at www.digium.com .
I think there is a free to use one, but only for
windows, * your best off with the digium licenses.
Greetings,
Tjardick
- Original Message -
From:
Todd Wallace
To: [EMAIL PROTECTED
Michael,
Where in your extension definition to you dial a channel (SIP, Zap, or other)? You
are missing the dial entry.
-sb
-Original Message-
From: Lists [mailto:[EMAIL PROTECTED]
Sent: Saturday, November 29, 2003 10:53 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Sip Issue
John,
I have 12 7960 phones with 6.0 with no issues. Sounds like a hardware problem to me.
-Original Message-
From: John Todd [mailto:[EMAIL PROTECTED]
Sent: Sunday, November 30, 2003 1:14 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisco 6.0 + Asterisk question
I have sever
On Tue, 2 Dec 2003, tony banks wrote:
> I have 1 IP 7940 with the following Firmware versions
>
> App Load ID:
>
> P00303011201
>
> Boot Load ID:
> PCO303010001
>
> Version
> 3.1(12.1)
Sounds like a Skinny image to me (and an old one, around CCM 3.1, too.)
You can either get a SIP image from ci
Sorry to everyone on the list, but for some reason
this is the only reliable way to get hold of John.
John Brown of Chagres Technologies, please contact
me! I have been trying for weeks now to get hold of you via email and
phone after wire transfering money into your account for the Grands
I've setup asterisk on a Dell Poweredge 1300 Server. PIII 600, 500mb
Ram, SCSI HD's. It is running Fedora Core 1. I have 3 X100P cards and 6
Grandstream IP phones.
Everything works but the sound quality with the FXO cards is poor. Static and
choppiness. It works great between the IP phones.
The
Dan wrote:
Hi all,
The multilingual version of DIAX with both IAX and IAX2 support will be
available for download later today or tommorrow.
If someone interested to help me traslate from English to one of the
following languages (ant not oly) please drop me a direct mail.
- italian
- spanish
- fre
I don't think so, is it zapata.conf , echotraining
= yes
Please confirm and I will try it
Rob
- Original Message -
From:
TC
To: [EMAIL PROTECTED]
Sent: Monday, December 01, 2003 7:01
PM
Subject: Re: [Asterisk-Users] Tone
Detection Problem
is echo train
I would stay away. I have evaluated these units and returned them. I
determined that this
unit or one from them that fits this description was actually a unit
that you put between
your phone and your phone line (1 FXS & 1 FXO claim) and hook to the
ethernet. This
unit would connect the FXS li
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Leif Madsen
> Sent: vrijdag 28 november 2003 5:11
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Asterisk behind NAT << How to do it.
>
>
> Thanks to ww and his patch on bug #104, I have successfully i
Steven Sokol <> wrote:
> Hi,
>
> I seem to be having problems with IAX clients based on the iaxClient
> library. I have been working on my own client (an augmentation to
> the Call Manager I released last week) and it seems to regularly miss
> incoming calls entirely. It also occasionally misses
Title: 'Stop Now', 'Restart' problems
I'm not sure where to start looking for a solution on this. I use use Asterisk::Manager to reload Asterisk with a command like:
$astman->sendcommand( Action ="" 'Command', Command => 'Reload' );
After a while, when I try to do a manual restart or 'st
Philipp von Klitzing <[EMAIL PROTECTED]> said:
>> >exten => 588,6,System(/usr/sbin/asterisk -rx "restart when convenient")
>> >
>> Put an & behind the line?
>
>It does help to get a proper hang up for the client, but there is no
>restart initiated at all... looks like now the system calls gets
>
Title: Message
just
need to buy g729 licences from www.digium.com, install it and off you go.
:)
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Todd
WallaceSent: Tuesday, December 02, 2003 6:37 PMTo:
[EMAIL PROTECTED]Subject: [Asteris
> I am having problems in a couple of installations where I have SIP
> phones (both GS101 and ATA186) connecting to an asterisk box that has a
> public IP address, where the stations are behind NAT.
>
> I'm still testing to make sure I have all the permutations looked at,
> but from what I can
You can buy g729 lic from digium for 10.00 per channel.
bkw
On Tue, 2 Dec 2003, Todd Wallace wrote:
> Does asterisk support G.729a or do you have to add something (is there an open
> source one)
>
>
> Todd Wallace
>
___
Asterisk-Users mailing list
[EM
Steven Critchfield wrote:
On Tue, 2003-12-02 at 07:44, Philipp von Klitzing wrote:
BTW: Where exactly is the difference between Hangup and Softhangup()?
Hangup is something done in the course of the dialplan and works on the
current channel where softhangup is a cli command that works on a name
At 10:37 AM 12/2/2003, you wrote:
Does
asterisk support G.729a or do you have to add something (is there an open
source one)
Yes, Yes, and Maybe (i.e. it's not free, but you can license one through
Digium, and there is a reference source available but absolutely NOT
open-source).
Check out this p
Alastair Maw wrote:
On 28/11/03 07:39, Olle E. Johansson wrote:
The latest version of my Ethereal plugin for IAX2 is now available here:
- http://almaw.com/ethereal-iax2-plugin-0.3.zip
Could you please create a URL that is a bit more non-version-specific?
http://almaw.com/etheral-iax2/
It now,
yes, * supports it.
but is available only by commercial license.
10$ / channel directly from digium.
Matteo.
Il mar, 2003-12-02 alle 19:37, Todd Wallace ha scritto:
> Does asterisk support G.729a or do you have to add something (is there
> an open source one)
>
>
> Todd Wallace
--
Brancaleon
Michael Bielicki wrote:
> You mean on US or EU dialup, but I doubt you will get any success on
> far
> east dialup or african dialup with that. Here you would either need
> speex or g723.1.
>
> Mark Spencer wrote:
>
>> You can definitely do that with GSM and G.729 when running IAX /
>> IAX2.
>>
On Tue, 2 Dec 2003 12:14:24 -0600, "Steven Sokol" <[EMAIL PROTECTED]>
wrote:
>Hi,
>
>I seem to be having problems with IAX clients based on the iaxClient
>library. I have been working on my own client (an augmentation to the
>Call Manager I released last week) and it seems to regularly miss
>inco
I have asterisk boxes in 2 different buildings each connected to the telco
with a PRI. I am now setting up asterisk machines in remote buildings -
dialing out via one of the other 2 machines. These are a snip from each
extension.conf on 1 remote and the 2 machines connected to the PRIs, to
illustra
On Tue, 2003-12-02 at 11:36, Michael Welter wrote:
> Hi,
>
> The PBX at the Colorado Organization for Victims' Assistance fried as a
> result of the building power being cycled. I'm now in the process of
> building an * system to replace the failed PBX. Minimum cost is the
> priority.
>
>
hello boys,
I'm trying to get 30 digital or ADSI phones for asterisk and 2 that serves
as console for receptionist that also works with asterisk,
I've heard of the smartalk ones (SE310 and ST110) for console and ST10
ST20 or ST 30 for desktop phones, but i'd like to know about any others i
may us
I am curious if anyone has tested, or is using any of the ArtDio gateways?
http://www.artdioinc.com/eng/product.htm (flash site)
Do they function as I might be construing them?
The IPS-1101 is listed as having 1 FXS and 1 FXO. Does that mean it will
actually control two seperate calls, one on an
Sorry, I dont know but i want a term added for archive searchablilty.
flex grow flexgrow flex-grow
- Original Message -
From: "Michael Welter" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 02, 2003 12:36 PM
Subject: [Asterisk-Users] Configuring new system for a non-
- Original Message -
From: "Matt Lawson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 02, 2003 11:32 AM
Subject: [Asterisk-Users] IAX port numbers?
> I see that when an Asterisk connects to another one via IAX, it seems to
> use port 4569 for the first one. But if
- Original Message -
From: "Michael Welter" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 02, 2003 12:36 PM
Subject: [Asterisk-Users] Configuring new system for a non-profit organization
> Hi,
>
> The PBX at the Colorado Organization for Victims' Assistance fried
Title: Mensaje
Hi, I want to use
G.723.1 on *, I read it is supported in Pass Through mode, but I don't
understand whats the meaning of that.
I have a GW 5300 and
an ATA 186 and I want to place calls to PSTN.
I setup this
config:
[general]port =
5060
bindaddr =
xx.xx.xx.xx
context =
Mark,
Following email of a couple of days ago - if you could confirm if you want
me to put things together for you in Paris. As mentioned I am in London from
24th to 30th but otherwise Paris.
Stephen
- Original Message -
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, Dece
Does asterisk support G.729a or do you have to add
something (is there an open source one)
Todd Wallace
WROOGGG
Voiceglo's webphone is IAX and they use GSM. I have my Asterisk server
registered with voiceglo right now.. so I know for a fact its IAX :P
s you didn't hear that from me.
bkw
On Tue, 2 Dec 2003, Adam Hart wrote:
I am having problems in a couple of installations where I have SIP
phones (both GS101 and ATA186) connecting to an asterisk box that has a
public IP address, where the stations are behind NAT.
I'm still testing to make sure I have all the permutations looked at,
but from what I can tell, what i
Hi,
I seem to be having problems with IAX clients based on the iaxClient
library. I have been working on my own client (an augmentation to the
Call Manager I released last week) and it seems to regularly miss
incoming calls entirely. It also occasionally misses the drop signal
when the remote en
On 02/12/03 16:32, Matt Lawson wrote:
I see that when an Asterisk connects to another one via IAX, it seems to
use port 4569 for the first one. But if it has multiple IAX connections
the additional ports seem to be chosen at random.
Is there anyway to predict, or specify which ports or range of
I think we need to look carefully at the situation here. We all know that
training on complex specialist products can often come in at between
2-4,000USD per week per individual (certainly this is my commercial
experience). However normally when I go on a training course there are
around 10 other
No takers? Should I submit a bug report then? I didn't find any open
bugs on stuck
MWI.
Clif Jones wrote:
I have had several cases where the message waiting indicator was stuck
in the on state
with Cisco 7960 SIP phones. Here are the two cases:
1. Single extension that mapped to a single voi
- Original Message -
From: "Mark Spencer" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 02, 2003 3:22 PM
Subject: Re: [Asterisk-Users] VoiceGlo
> You can definitely do that with GSM and G.729 when running IAX / IAX2.
>
> Mark
>
> On Tue, 2 Dec 2003 [EMAIL PROTECTED]
After battling for days trying to figure
out what was wrong with my iax.conf it was determined that I do not have
any inkeys set on the digium server. Now whether that is something new or
just in a few cases I am not sure. Messing around and reading on IRC and
the mailing list I could get
Hi,
The PBX at the Colorado Organization for Victims' Assistance fried as a
result of the building power being cycled. I'm now in the process of
building an * system to replace the failed PBX. Minimum cost is the
priority.
I have a T100P card installed in the new system, and I am about to or
I've seen this same thing. But it doesn't happen only for phones using
the queue I believe it is a bug in the chan_sip driver. What I have
found is that when a phone sip phone is unplugged/not registered and a
call comes in it increments the counter and doesn't reset the counter
when the phone rere
Hi list.
I'm having the next problem.
I have a * with 1 TDM400P (4 ports) and
one X100P, with a working configuration.
Today i add one more X100P card, and i change
the config files as next:
zapatel.conf:
fxsks=1-2
fxoks=3-6
loadzone = us
what do the options "algo" do in the monitor app? I dont see that in the show
application monitor? is this a patch?
Dave
>>> [EMAIL PROTECTED] 12/2/2003 6:56:18 AM >>>
Try something like this:
exten => 2060,1,Answer
exten => 2060,2,Wait,1
exten => 2060,3,Monitor,wav|algo
exten => 2060,4,Meet
Hi!
> > exten => 588,1,Answer
> > exten => 588,2,Wait(1)
> > exten => 588,3,Playback(restart-convenient)
> > exten => 588,4,Wait(1)
> > exten => 588,5,Authenticate(0)
> > exten => 588,6,System(/usr/sbin/asterisk -rx "restart when convenient")
> > exten => 588,7,Hangup
> >
> > The problem: We
Hello,
> But to answer your question, I have a friend that does Checkpoint firewall
> training/consultation and he gets upto $20,000 per week for running training
> classes. Not in the US mind you but abroad, mostly in Europe. He says
> American companies are too cheap.
I wouldn't say anything ab
We are a little Belgian company working for the automotive world (car builders,
leasing companies , ...) and believe me if i make an offer of 2USD for a week i'll
never have to make an offer again !
I also worked in the past (2000 which is considered as the craziest year for Tech
companies
You mean on US or EU dialup, but I doubt you will get any success on far
east dialup or african dialup with that. Here you would either need
speex or g723.1.
Mark Spencer wrote:
You can definitely do that with GSM and G.729 when running IAX / IAX2.
Mark
On Tue, 2 Dec 2003 [EMAIL PROTECTED] wr
Hello,
Right now I have app queue working with incominglimit=1, there is no
call waiting signal, but after a while( like couple of hours) some
phones randomly get stuck. The * thinks that they are in use and doesnt
ring them, when they are infact not in use.
sip show inuse, shows that they are
American companies are too cheap? That's laughable. I could get into
economic's but this is not the place. Your typical Cisco firewall PIX
training class avgs. $2499 a week. I can only think he's training 10 people
@ $2,000 a week. If he's making $20k for a week of training, no wonder why
the
I see that when an Asterisk connects to another one via IAX, it seems to
use port 4569 for the first one. But if it has multiple IAX connections
the additional ports seem to be chosen at random.
Is there anyway to predict, or specify which ports or range of ports to
use, for the sake of settin
Hello,
Right now I have app queue working with incominglimit=1, there is no
call waiting signal, but after a while( like couple of hours) some
phones randomly get stuck. The * thinks that they are in use and doesnt
ring them, when they are infact not in use.
sip show inuse, shows that they are
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Low, Adam
> Sent: Tuesday, December 02, 2003 9:47 AM
> To: '[EMAIL PROTECTED]'
> Subject: RE: [Asterisk-Users] Dedicated * voicemail server
>
> > You could add an initial digit based o
maybe it means "United States Dimes" :) $2,000 ain't bad for a week of
training.
But to answer your question, I have a friend that does Checkpoint firewall
training/consultation and he gets upto $20,000 per week for running training
classes. Not in the US mind you but abroad, mostly in Europe. He
the details :
16000USD for the training
+ 10% for administration
+ Travel costs to Belgium (from canada)
+ Hotel costs ...
so 'im not far from 2USD ...( it could be even more )
We have a budget but not such a big one ... and even if i would, i would be a bad
manager to accept such a cos
> > Just to inform the community ... i received an
> > offer last week for 1 week of asterisk training
> > +/-2USD !! We can't aford this !
>
> Is that USD 20.000,- as in twenty thousand US dollars, or is have
> someone played around with the keyboard? If so - who the fuck can afford
At 9:27 AM -0500 12/2/03, Richard Alexander wrote:
> -Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Low, Adam
Sent: Tuesday, December 02, 2003 7:58 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Dedicated * voicemail server
Hey A
Roy Sigurd Karlsbakk wrote:
Just to inform the community ... i received an
offer last week for 1 week of asterisk training
+/-2USD !! We can't aford this !
Is that USD 20.000,- as in twenty thousand US dollars, or is have
someone played around with the keyboard? If so - who the fu
You hear it very well !
As i think i'm polite, so i'm not going to put the name of the company online but
believe me their was a hole in the seiling after i read the email ...
Van: Roy Sigurd Karlsbakk [mailto:[EMAIL PROTECTED]
Verzonden: di
Must have included a week in Amsterdam
Scott M. Stingel
Emerging Voice Technology Inc.
URL:www.evtmedia.com
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Roy Sigurd Karlsbakk
> Sent: Tuesday, December 02, 2003 2:23 PM
> To:
> It is a shame that within a couple of hours they can tell you to remove
> helpfull documentation, but not (seemingly) help answer questions
> regarding there Cisco stuff on this list. I think Cisco must have their
> priorities mixed up!
>
> Just my opinion... which also means I won't support a co
This is also in production use
http://www.tyan.com/products/html/gx15b2723t15.html
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Asterisk-Users mailing list
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I get echo on my X100P for about 15 seconds then it disappears. Believe me,
when I get a chance I'm going to tweak it since my wife nags me about it. I
converted my house over to *.
My setup:
Compaq Deskpro EN SFF P3 500
Dual NIC for real IP and internal IP
X100P Card for PSTN.
Cisco ATA186 (
>I'm a windows developer ready to develop TSP/MSP (TAPI) drivers for
asterisk PBX but i searched a little >more support to develop these
drivers ...
>unfortunatly i have to develop the drivers commercially because i will need
to hire a asterisk "freak" to explain >me in detail how everything w
Hi Gus,
Thanks. It worked
regards...
Girish
From: "CW_ASN - Gus" <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Meetme Recording
Date: Tue, 2 Dec 2003 10:56:18 -0300
Try something like this:
exten => 2060,1,Answer
exten => 2060,2,Wait
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