New never used
bought by mistake ( 5V pci slot ) and stored...
pls make offer,
thanks and regards,
Jean-louis Curty
The TE207P is a bundling of our leading TE205P product and our new
VPMOCT064 Octasic DSP-based echo cancellation module. The TE207P is
Digium's and the industry's first two-port
New never used
bought by mistake ( 5V pci slot ) and stored...
pls make offer,
thanks and regards,
Jean-louis Curty
The TE207P is a bundling of our leading TE205P product and our new
VPMOCT064 Octasic DSP-based echo cancellation module. The TE207P is
Digium's and the industry's first two-port
I have googled a lot to find solution to the same exact problem described in
your message but no real solution yet.
here is my config
1 physical network
25 pc windows
25 phones IP330 IP550 SIP 2.1.2 no vlan CDP disabled some with dhcp some
with fixed ip to see if there is a diff
3 switchs
the patton any sip call coming from asterisk is routed to isdn (
outgoing calls )
in patton again , all isdn calls of any ports is routed to the asterisk IP
as a sip call,
jl
2008/1/9, Olivier [EMAIL PROTECTED]:
2008/1/8, Jean-Louis curty [EMAIL PROTECTED]:
Hi,
I succesfully install
exactly
isdn patton - eth/lan sip asterisk
jl
2008/1/10, Olivier [EMAIL PROTECTED]:
2008/1/10, Jean-Louis curty [EMAIL PROTECTED]:
I use 4960 and 4638 gw but it's applicable for any patton gw ( analogue
or isdn ) since it's always the same way of configuring
- define ports
- define
ok sorry for the confusion created,
I mean isdn network , in other word tdm,
so tdm link connected to patton, patton connected in the lan via ethernet
speaking sip,
jl
2008/1/10, Olivier [EMAIL PROTECTED]:
2008/1/10, Jean-Louis curty [EMAIL PROTECTED]:
exactly
isdn patton - eth/lan sip
Hi,
I succesfully install spandsp chan_misdn and digium card. the rxfax works
fine and I get the fax result by email.
I would like to do the same using a Patton gw + zaptel but I can't receive
fax anymore,
the call comes in from ISDN in the Patton gw, patton sends it to asterisk,
asterisk run a
Hi,
I managed to install my patton gateway but i did not succeed to pass the
caller id to the sip phone on incoming calls ...
instead i see call from 105 ( which is the sip client extension of the
patton )
do you know the way to pass the caller id of the caller to the ip phone iso
the gw
be "directed call pickup", not "call pickup".
Could this have been the problem?
Gareth
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of
Jean-Louis curty
Sent: 08 September,
2006 5:25 AM
To: Asterisk Users Mailing List -
Non-C
Hi Dave , long time no ear :-)still progressing on * , I have now my own distribution on a compact flash including bristuff ...i m fighting with this nice 9133i, lastest firmware 1.4 , blf is turned on and working ( light flashing when calls comes in ) but I can not pick it up,
ps I followed the
Hi,I'm trying to get the blf / pickup working properly on the aastra 9133i,I read the wiki voip-info.org for the setup,setup is working fine for the snom, it works also for the aastra ( the light is flashing when a call comes in on another phone ) but I can't pickup the call ... when I press the
sure you can.
From: Jean-Louis curty [mailto:[EMAIL PROTECTED]]
Sent: Wednesday, August 30, 2006 2:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] upgrade problem on IP phone 9133i
good idea! I tried but it doesn't work either...[08/30/06 23:12:33
hi everybody,I bought few units for evaluation but we were not able to upgrade the firmware to 1.4 ,
it's currently set at 1.2,
when we go to the webadmin page,
whether we try to change the IP of the tftp server or the firmware name and set values, the reply is always
Invalid IP address
this upgrade using a 'put' install via tftp clientrather than trying to configure it to 'get' from a tftp server.It's beenawhile so my memory is a bit foggy.I used pumpKIN.
http://kin.klever.net/pumpkin/-Original Message-From: Jean-Louis curty [mailto:[EMAIL PROTECTED]]Sent: Wednesday, August
Hi, we have few cisco's...is there a way to push the queue information to the phone ?thanks in advance,jean-louis2006/8/24, Brodie Macleod
[EMAIL PROTECTED]:I know this isn't answering your question, but what I did for queue
notification was use softkeys on the phones that call a PHP script on
thanks to all of you, I fixed my problem by changing the cable !jl
2006/7/4, stoffell [EMAIL PROTECTED]:
On 5/31/06, Jean-Louis curty [EMAIL PROTECTED] wrote: I does nothing special,
no output, nor error, same.. .:-(you should at least get any output from ztcfg, but aside from that,like Tommaso
here is the qview.pl,
seems that refreshing the information on the screen will not be easy...
rgds
jl
2006/6/1, Brent Torrenga [EMAIL PROTECTED]:
Might want to take a look at http://web.csma.biz/apps/xml_xmldir.php
for astarting point.Sincerely,Brent A. TorrengaTorrenga Engineering, Inc.907
I hope with the sip firrmware it's the same :-)
jl
2006/6/2, Michiel van Baak [EMAIL PROTECTED]:
On 00:15, Fri 02 Jun 06, Jean-Louis curty wrote: here is the qview.pl, seems that refreshing the information on the screen will not be easy...
It is.at least with the SCCP image it's one line
Hi everybody
I hope that somebody can help me with the following
I have
2 quadbri cards
2 - 1t0 cards
1 pabx alcatel 4200
I would like to connect my asterisk to the alcatel ,
I installed bristuff 0.3.0-1p ,
loaded the zaphfc driver in NT mode
configured zaptel and zapata , it works great.
I'm still looking for this missing feature of a good asterisk ACD,
I have a qview.pl which display the queue state but it's on the pc not on the phone display,
shall we start a bounty , is there any developper here who knows how to do it , where to start ?
thanks
jl
2004/12/7, Brian Roy [EMAIL
I does nothing special,
no output, nor error,
same.. .:-(
2006/5/31, Henk [EMAIL PROTECTED]:
Try to do ztcfg –s before you run ztcfg -vv
Henk
From:
[EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED]] On Behalf Of Jean-Louis curtySent: woensdag 31 mei 2006 12:52
To: Asterisk Users Mailing
thanks a million,
got the same issue on cisco 7912 and thanks your posts it's fixed!
jl
2005/1/24, Oswaldo Arratia [EMAIL PROTECTED]:
Yes, I raised both values.I have ForwardToVMDelay:120andSigTimer:0x03C00064 so the phone itself does not send the caller to VM,and
I take care of the voicemail
Hi,
Anyone already configured a powerconnect 34xx to prioritize voip traffic over the lan ?
I just bought a 3424 for the lan and I like to give priority to voice ,
thanks in advance,
jean-louis
___
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avez vous trouvé une solution ?
jean-louis curty
2006/1/6, maingault [EMAIL PROTECTED]:
Hi,
I'm a new user of Aterisk, and I have to configure a VoIP Gateway.
I have an Alcatel PBX with an E1 card, connected, for the moment, to a local carrier.
I would like work with a french VoIP provider
Hi,
I successfully connected 2 servers via IAX but I'm pulling my hair to
connect 2 extra servers , Anyone connected 3 or 4 servers together ? is
it possible ?
I d like to share the dialplan so _2 goes to server A _3 goes to serverB _4x goes to server C etc from the 4 servers
any
will try that,
thanks bob !
jl2006/2/16, Bob Goddard [EMAIL PROTECTED]:
On Thursday 16 Feb 2006 22:20, Jean-Louis curty wrote: hi, My question is may be a bit out of scope but I don't know where to turn, I have a 1760 with a ccme 24 user licences 1 bri card.
I want to configure a bri card
hi,
My question is may be a bit out of scope but I don't know where to turn,
I have a 1760 with a ccme 24 user licences 1 bri card.
I want to configure a bri card in a cisco 1760 on port 0/0,
the card is new, seen by the router, show isdn status gives layer 1 desactived , layer not activated,
anybody succeeded with this issue ? I mean writing xml code to display
queue info on the Cisco screen
jl
2004/12/7, Brian Roy [EMAIL PROTECTED]:
On Sat, 4 Dec 2004 13:08:19 -0600, Joe Dennick [EMAIL PROTECTED] wrote:
I, too would be very interested in this application.
We are also
hi,
I know this topic has been covered several times and I read everthing
I found on the subject but I restarted one of my cisco phones and it
suddenly gave the famous message :protocol application invalid
how can I use ethereal to see what's going on ? anybody could give
me the command
7910 works fine wiz asterisk but you can not transfer calls, for that
reason I will sell mine if somebody is interested...
jl
2005/7/16, Javier Chia [EMAIL PROTECTED]:
[EMAIL PROTECTED] takes only 15 minutes to install in a
Xeon 2.8. However downloading 700mb ISO file could
take all night.
thanks I 'll try ... :-)
jl
2005/7/4, Jean-Louis curty [EMAIL PROTECTED]:
Hi,
I'd like to understand what should i do to use G729 codec in a legal way,
how do I order licences ? to whom ? how do I install them on asterisk etc ?
thanks in advance ,
jl
Hi,
I'd like to understand what should i do to use G729 codec in a legal way,
how do I order licences ? to whom ? how do I install them on asterisk etc ?
thanks in advance ,
jl
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Hi,
I'd like to realize the following setup traditional phones -- pbx
(non ip ) -- asterisk --- ISDN network
is it possible to set up an asterisk server in such way that it acts
as gateway between a traditionnal pbx and an isdn network ( I'm in
France ). this to let people continue with
Anybody has experirence with Festival + french voice ?
I read 2.1.3 .--The French language
Festival does not (at the moment) implement yet the French language
but it's not difficult to make it work with the Mbrola synthesizer
which offers high quality French voices.
from the website
Your approach is really great, I love it,
questions:
do you have a roadmap ( sort of ) ?
which exiting features can we expect ?
when do you think you will reach 1.00 ?
success,
jl
On Wed, 16 Feb 2005 10:59:29 -0800 (PST), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
New features include
Hi everybody,
I'm testing [EMAIL PROTECTED] 0.4,
looks great so far
I was working when I have been alerted by a bip comming from the * pc...
I connected a screen to it and saw that there was a message which looked like :
Message from [EMAIL PROTECTED] at Thu Feb 10 09:01:00 2005 ...
worried about the security. How did they
connect to
your system? Through telnet or what?
Since I've disabled all such services.
Best,
Christian
- Original Message -
From: Karl H. Putz [EMAIL PROTECTED]
To: Jean-Louis curty [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non
Stewart [EMAIL PROTECTED] wrote:
On 10/02/05 15:10 +0100, Jean-Louis curty wrote:
so I stopped asterisk, type mail and got a strange mail saying that
user [EMAIL PROTECTED] could not be reached and body was like if it was
the result of commands ifconfig etc
unfortunally I don't have
Hi,
Anybody managed to enable capi on [EMAIL PROTECTED] centos ?
i did make menuconfig, enabled it ( I think ) created a bzImage but
don't know what to do next to boot from it...
any tip or how to welcome,
thanks
jl
___
Asterisk-Users mailing list
Hi,
Anybody managed to enable capi on [EMAIL PROTECTED] centos ?
i did make menuconfig, enabled it ( I think ) created a bzImage but
don't know what to do next to boot from it...
any tip or how to welcome,
thanks
jl
___
Asterisk-Users mailing list
Hi I use RH 9 + asterisk v1.0 stable + 2 PCI fritz cards + chan_capi
3.5 and it works fine,
Since my users want fax fonctionnality and customers know 1 of the msm
as fax number I wanted to try the chan_capi-0.3.5 patch
if I patch chan_capi and run make, I get an error message , as you
can read
Hi everybody,
I use RH 9 + asterisk v1.0 stable + 2 PCI fritz cards + chan_capi
3.5 and it works fine,
Since my users want fax fonctionnality and customers know 1 of the msm
as fax number I wanted to try the chan_capi-0.3.5 patch
if I patch chan_capi and run make, I get an error message , as
Hi
I use RH 9 + asterisk v1.0 stable + 2 PCI fritz cards + chan_capi
3.5 and it works fine,
Since my users want fax fonctionnality and customers know 1 of the msm
as fax number I wanted to try the chan_capi-0.3.5 patch
if I patch chan_capi and run make, I get an error message , as you
can read
Hi everybody,
I'd like to know if anybody tried to write a xml doc to monitor the
number of calls in Q, when working with an ACD it's convenient to see
how many calls are waiting so the agent can speed up the conversation
when it gets too busy :-)
I was wondering if it was poss to display this
Hi everybody,
I'd like to know if anybody tried to write a xml doc to monitor the
number of calls in Q, when working with an ACD it's convenient to see
how many calls are waiting so the agent can speed up the conversation
when it gets too busy :-)
I was wondering if it was poss to display
test
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Jean-Louis curty to Asterisk
More options 4:38pm (7 minutes ago)
Hi everybody,
I'd like to know if anybody tried to write a xml doc to monitor the
number of calls in Q, when working with an ACD it's convenient to see
how many calls are waiting so the agent can speed up the conversation
Hi everybody,
Anybody could give me a little hint to apply the patch described below and
how to enable sfftobmp ? reading the post below, fax.php seems to be used to
mail the result but was not able to find it, do I have to write it ?
Thanks in advance,
jl
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