Fine but don't mix up Swedish Danish beer ...
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED] namens steve szmidt
Verzonden: vr 1/04/2005 16:39
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
CC:
Onderwerp: Re:
Is it possible to modify the caller id on the phone during a call (session) ?
If not does anybody know with which SIP request this could be handled ?
I'm know investigating RFC3311 which seems to offer an answer but if somebody
already has an answer ...
Michael
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 threading stuff, I don't remember) we replaced the
Supermicro board with an intel.
Try the same config on another machine (maybe an
How are things going for echo and so?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp
von Klitzing
Sent: donderdag 3 februari 2005 21:58
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Dialplan
Did you try to boot without lan just the power ...
I've had this same problem to and rebooted the device without lan
connection
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Louis-David Mitterrand
Sent: woensdag 26 januari 2005 11:36
To: 'Asterisk
Does somebody have this new firmware from/for Polycom ?
Thanks
Michael
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Problem solved :
The reason was quite simple ... but annoying :
Interrupts !!! damned !!!
Thank you
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED] namens Florian Overkamp
Verzonden: wo 19/01/2005 10:08
Aan: 'Asterisk Users Mailing List -
Versatel uses CRC4 (in Belgium)
zaptel.conf :
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
alaw=1-31
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED] namens Florian Overkamp
Verzonden: wo 19/01/2005 10:33
Aan: 'Asterisk Users Mailing
Did somebody already configured a Digium card on the network of Versatel in
Belgium or the netherlands, and would like to share his configuration.
(zaptel.conf / zapata.conf)
We have HDLC errors (timings i presume)
___
Asterisk-Users mailing
And thanks Kristian for the link. I'll make sure to give your distro
a shot in case I actually order the box. Just haven't completely made
up my mind yet. Nice pics btw you have on your site, what board is
that?
http://www.pcengines.ch
Regards, Bruno.
Well for example : use SIP on your LAN an use IAX to connect the outside world
...
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED] namens Bryan Mannos
Verzonden: do 25/11/2004 10:04
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
We are located in Belgium and just ordered a PRA line,
the telco asked the following questions :
-
120 or 75 ohm ?
-
Support for CRC4 yes/no
?
-
B channels in 2 way ?
We will buy a digium card but which one should we buy
?
could anybody help me with this ?
Thank you
We are located in Belgium and just ordered a PRA line, the telco asked the
following questions :
- 120 or 75 ohm ?
- Support for CRC4 yes/no ?
- B channels in 2 way ?
We will buy a digium card but which one should we buy ?
could anybody help me with this ?
Thank you
Michael
Sorry for the
http://www.voip-info.org/tiki-index.php?page=Polycom%20Phones
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: zaterdag 6 november 2004 19:48
To: Asterisk-a-users-list
Subject: [Asterisk-Users] Polycom 500 software?
Just unboxed two
They seem to be very good ! but where the hell could we buy them in Europe !?
Michael
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED] namens John Todd
Verzonden: di 17/08/2004 4:56
Aan: [EMAIL PROTECTED]
CC:
Onderwerp: RE:
Use the queue functionality see www.voip-info.org/asterisk
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED] namens Marcin Mazurek
Verzonden: ma 16/08/2004 15:21
Aan: [EMAIL PROTECTED]
CC:
Onderwerp: [Asterisk-Users] randomize Dial()
I have to agree that the hardware is fanatastic ... but the functionality is rather
poor
--- example A ---
receive call A
answer A
Receive call B
answer B
You can't link/transfer these 2 damned calls.
--- example B ---
You first have to push new call before dialing a number.
this seems to be a
] Behalf Of Shaun Ewing
Sent: donderdag 15 juli 2004 11:03
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] CISCO 7960G FIRMWARE
On Thu, 15 Jul 2004 10:08:41 +0200, Michael Devenijn
[EMAIL PROTECTED] wrote:
--- example B ---
You first have to push new call before dialing a number.
this seems
It seems we have a more recent version (but i do not remember where i got it, i think
it is frome a post on this list)
wj000e_img.ftp in wj000e_img.zip
Michael
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Dominique
Kull
Sent: woensdag 14 juli 2004 9:41
In belgium people can receive SMS's on fixed lines, my question is if this feature is
possible on the digium TE410P card in combination with asterisk ?
Kind regards
Michael
DISCLAIMER: The content of this e-mail message does not constitute a commitment of
DKMA bvba This e-mail and any
Yes it is possible, with the chan_CAPI drivers from junghanns.net
i only used the 4BRI cards from Eicon but they are similar to the PRI cards
i didn't have any ISDN knowledge before. but first tried to install the card with CAPI
on a redhat 9 machine
with exactly the description from eicon
I found this tool, but didn't have the time to test it...
http://www.dylogic.com/sito/ArticlesDMD/mirial.html
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of shabanip
Sent: donderdag 24 juni 2004 13:59
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users]
why not use asterisk with QaudBRI and/or E100P ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Miroslav
Nachev
Sent: woensdag 23 juni 2004 16:43
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Call Generator for ISDN (PRI/BRI)
Hi,
I am looking for
for today we only have experience with BRI applications together with asterisk.
is the following scenario possible and stable enough for production?
FYI : We want to build a unified messaging application integrated with SIP.
We have an E1 connection in Belgium with 100 msn's
We would think
i'dd like to but is it stable enough for production (receiving over 500 faxes a day ?)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Klaus-Peter
Junghanns
Sent: vrijdag 18 juni 2004 13:58
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] TE410P / Eicon
Dear,
Is there somebody who have experience with the Siemens Optipoint 400 standard SIP ?
And where can we buy it (i'm from belgium)
We are using for the moment Cisco 7960, 7905, snom 200, Mitel 5055 and in my opinion
the Mitel does his work the best combined with the 7905,
the 7960 is
This is what i got ... and i could not find the problem by my own
/sbin/depmod -a
depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/qozap.o
depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/tor2.o
depmod: *** Unresolved symbols in
(zaptel.o) in your modules ?
Try an :
insmod zaptel
If it does not work, it means it has not been installed.
- Original Message -
From: Michael Devenijn [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, May 19, 2004 10:45 AM
Subject: [Asterisk-Users] Problem compiling zaptel
- I'm not a Linux user i'm trying to get in it ...
- Fedora core 1
- QuadBRI card bought from junghanns.net
- We want to use the card in TE mode to connect to the TELCO
- Downloaded BRISTUFF0.0.2(stable) latest from junghanss.net/asterisk
- followed the instructions on voip-info.org
- compiled
Very good phone especially with the latest firmware ...
Major point :
the soft rubber keys are to hard to push, you can't make a number fast enough
Minor :
the phone is too light
Good :
- Multi national configurations / tones etc...
- Good design
- ... the others told it already
Michael
Here is a sip log from my vegastream 50BRI to my asterisk box and i can't figure out
why the call doesn't go trough ...
sip.conf extract :
[gw001]
type=friend
host=dynamic
defaultip=192.168.0.12
nat=no
dtmfmode=rfc2833
canreinvite=yes
qualify=no
context=tlsgw
extensions.conf extract (from
Go on www.voip-info.org an search for IVR examples ...
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paul Mahler
Sent: Saturday, March 20, 2004 5:35 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Need an example of using the directory command
Does
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Olle E.
Johansson
Sent: Saturday, March 20, 2004 11:05 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Problem with Vegastream 50 BRI
Michael Devenijn wrote:
Here is a sip log from my vegastream 50BRI to my asterisk box and i can't
regards
Michael Devenijn
-Oorspronkelijk bericht-
Van: Craig Waddington [mailto:[EMAIL PROTECTED]
Verzonden: ma 15/03/2004 11:07
Aan: Michael Devenijn
CC:
Onderwerp: FW: IPC5000
FYI
Does anybody have experience with these units ??
http://www.udcsystems.com/
DISCLAIMER: The content of this e-mail message does not constitute a commitment of
DKMA bvba This e-mail and any attachments thereto may contain information which is
confidential and/or protected by intellectual
no i
bought this one
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Craig
WaddingtonSent: Thursday, March 11, 2004 8:58 PMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users]
IPC5000 - Wireless Sip phone
Thanks for the info.
I'm working on a abstraction toolset for asterisk, some screenshots :
http://www.dkma.be/devenijn/data/tls.jpg
http://www.dkma.be/devenijn/data/tls1.jpg
http://www.dkma.be/devenijn/data/tls2.jpg
http://www.dkma.be/devenijn/data/tls3.jpg
http://www.dkma.be/devenijn/data/tls4.jpg
some technical
Office: (217) 585-0357
Cell: (309) 657-6365
Fax: (309) 213-3500
E-Mail: [EMAIL PROTECTED]
Customer Service: (877) 976-0711
-Original Message-
From: Michael Devenijn [mailto:[EMAIL PROTECTED] On Behalf Of Michael
Devenijn
I'm a bit of a linux zero (at least i'm working to know it but it takes some time)
Do i understand :
The call comes in on a card (in this case zaptel) for exampl,e and if in the dialplan
you have an 'f' extension Hylafax will get the communication from the zaptel card ? or
is it simply a
Did somebody already get a snom 220 phone ??
DISCLAIMER: The content of this e-mail message does not constitute a commitment of
DKMA bvba This e-mail and any attachments thereto may contain information which is
confidential and/or protected by intellectual property rights and are intended for
Does somebody now if there is some opensource software which can handle T.38 SIP and
convert it to a tiff or something ?
Kind regards
Michael
DISCLAIMER: The content of this e-mail message does not constitute a commitment of
DKMA bvba This e-mail and any attachments thereto may contain
does sombody has an example config for a vegastrem 50 BRI and asterisk would help
me a lott
thanks
Michael
DISCLAIMER: The content of this e-mail message does not constitute a commitment of
DKMA bvba This e-mail and any attachments thereto may contain information which is
confidential
Jan,
Where can we get any technical documentation about sccp protocol i've searched with
google and at cisco but i don't find anything useful ...
Michael
Van: Jan Czmok [mailto:[EMAIL PROTECTED]
Verzonden: wo 21/01/2004 15:04
Aan: [EMAIL
May be it's due to a kernel patch ... ??
Try to recompile zaptel, asterisk, ...
Van: Jess Magnaye [mailto:[EMAIL PROTECTED]
Verzonden: wo 7/01/2004 17:36
Aan: [EMAIL PROTECTED]
Onderwerp: [Asterisk-Users] * crashed
I
Use the callmanager their u can use the link event
Michael
Van: Luciano Ramos [mailto:[EMAIL PROTECTED]
Verzonden: di 6/01/2004 13:56
Aan: [EMAIL PROTECTED]
Onderwerp: [Asterisk-Users] AGI Scripting
Hi!.
Is
Code you gave the audio settings of your 7905 and which type and firmware for the snom
Van: tony banks [mailto:[EMAIL PROTECTED]
Verzonden: di 6/01/2004 18:23
Aan: [EMAIL PROTECTED]
Onderwerp: [Asterisk-Users] Got SIP response 482
: [Asterisk-Users] AGI Scripting
Which Call Manager???
Luciano
-Mensaje original-
De: Michael Devenijn [mailto:[EMAIL PROTECTED] nombre de Michael Devenijn
Enviado el: Martes 6 de Enero del 2004 10:10
Para: [EMAIL PROTECTED]
Asunto: RE
it possbile to use drag drop
features ...
Kind Regards
Michael Devenijn
Van:
Michael Devenijn
Verzonden:
ma 24/11/2003 14:38
Aan:
[EMAIL PROTECTED]
Onderwerp:
[Asterisk-Users] SIP to SIP redirect while
ringing
is it possible to transfer
I have some strange question bout the asterisk (gpl license ...) but i'm not an
experienced linux user ...
What happens if for example a big company buys digium , do we have a garantuee that
asterisk stays opensource ???
Kind regards
Michael Devenijn
winmail.dat
Which events did you add ?
Van: Jonathan Tew [mailto:[EMAIL PROTECTED]
Verzonden: di 23/12/2003 16:25
Aan: [EMAIL PROTECTED]
Onderwerp: Re: [Asterisk-Users] Asterisk + CRM
We're starting to integrate * with our
Everythings works great with asterisk exept one feature
with redirect : it doesn't redirect when ringing ...
BTW are their any plans to extend the manager API
??
Michael Devenijn
You're right for pure SIP configurations , but Asterisk acts here as media gateway
and treats all of the media comm. e.g. to eventually communicate with other types
(like PSTN, H323, AIX, ... the voicemail app.)
Michael Devenijn
IT Manager
DKMA
Schaarbeeklei 636
B-1800 Vilvoorde
Tel.: +32 2
to inform the community ... i received an offer last week for 1 week of asterisk
training +/-2USD !! We can't aford this !
Kind regards
Michael Devenijn
DKMA
Van: Harald Baron [mailto:[EMAIL PROTECTED]
Verzonden: di 2/12/2003 9:38
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
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
. Not in the US mind you but abroad, mostly in Europe. He says
American companies are too cheap.
MATT---
-Original Message-
From: Michael Devenijn [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 02, 2003 9:26 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] CTI/TAPI
You hear it very
count me in
Michael devenijn
DKMA
Van: Mark Spencer [mailto:[EMAIL PROTECTED]
Verzonden: zo 30/11/2003 6:28
Aan: [EMAIL PROTECTED]
Onderwerp: [Asterisk-Users] * Party in Paris
I'm coming to Paris Dec 19. I
You have to buy a Cisco contract so you can download the files on their site, but here
is a link with the explanations, because once you have the good firmware ... there is
a way to go :
http://www.loligo.com/asterisk/cisco/79xx/
Michael Devenijn
DKMA
Schaarbeeklei 636
1800 Vilvoorde
Tel
Just turn off the silence suppresion
Van: Jorge Cisneros Flores [mailto:[EMAIL PROTECTED]
Verzonden: vr 28/11/2003 5:57
Aan: [EMAIL PROTECTED]
Onderwerp: [Asterisk-Users] RFC3389 support incomplete
Hi
When
We are working with realspeak and it is a wonderfull product (even in product) it
supports up to 20 languages and has aquired a really good prod. stability !
Van: Steve Underwood [mailto:[EMAIL PROTECTED]
Verzonden: vr 28/11/2003 4:41
Aan:
Michael Devenijn
DKMA bvba
Schaarbeeklei 636
1800 Vilvoorde
Tel.: +32 2 255 10 19
Fax : +32 2 251 03 12
is it possible to transfer a call while it's ringing
??
SIP/cs001 calls SIP/cs002
The SIP/cs002 user transfers the call to SIP/cs003,
where on SIP/cs003 the phone continues to ring ...
inone way or another (trough manager API or
something else, don't care)
i tried redirect with the
I'm testing the Eicon Server BRI/PRI cards for the moment and they are very
satisfactory because they have some interesting features onboard like echo cancelation
() and onboard encoding see www.eicon.com
Van: Daniel ANDRE [mailto:[EMAIL PROTECTED]
Did anybody tried this card with asterisk ?
http://www.digi.com/pdf/prd_mca_datafirequad.pdf
.
Also i've you think it's quite impossible to do it
we can discuss our idea's
Michael Devenijn
DKMA bvba
This one helped me a lot :
http://www.loligo.com/asterisk/Cisco/79xx/
kind regards
Michael Devenijn
www.dkma.be
Van: Micke Andersson [mailto:[EMAIL PROTECTED]
Verzonden: vr 24/10/2003 9:33
Aan: [EMAIL PROTECTED
it.) but strange enough their is a speaker ...
Michael Devenijn
IT DKMA
I bought a couple of 7905G phones with a Callmanager license
but i found on the site these phones can have a SIP image (which i downloaded)
but before i upload the image i want to know if anybody tested them
?
Michael
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