hi all i have a te110p installed in my system with a lot of Echo..
i decide to install the oslec echo supressor but when y try to add the
module i have this problem.
[EMAIL PROTECTED] zaptel-1.4.7.1]# insmod wct1xxp.ko
insmod: error inserting 'wct1xxp.ko': -1 Unknown symbol in module
[EMAIL
:
On Mon, Jan 28, 2008 at 04:37:25PM -0200, Pablo Allietti wrote:
hi all i have a te110p installed in my system with a lot of Echo..
i decide to install the oslec echo supressor but when y try to add the
module i have this problem.
[EMAIL PROTECTED] zaptel-1.4.7.1]# insmod wct1xxp.ko
insmod
:
On Mon, Jan 28, 2008 at 04:37:25PM -0200, Pablo Allietti wrote:
hi all i have a te110p installed in my system with a lot of Echo..
i decide to install the oslec echo supressor but when y try to add the
module i have this problem.
[EMAIL PROTECTED] zaptel-1.4.7.1]# insmod wct1xxp.ko
insmod
---
--
.-
Pablo Allietti
E-mail: [EMAIL PROTECTED] | LACNIC
Phone : +598 2 604 | http://LACNIC.NET
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hi all i have a TE110P connected to my PBX when i try to call a
extension number in other location 3525 the asterisk give me a error
-- User entered '3525'
-- Executing [EMAIL PROTECTED]:4] GotoIf(Zap/31-1, 0?6:5) in new
stack
-- Goto (lacnicuy,450,5)
-- Executing [EMAIL
hi all, in console mode how i can display the logged users?
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hi all i have a newbrand phone Linksys spa941 and i realize that my
asterisk have ECHO. :(
in the zaptel file i have this parameters
echocancel=128
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
i need to add any other parameter to cancel the echo?
thanks. any tips?
On Fri, Jun 09, 2006 at 04:45:51PM -0400, William Piper wrote:
GSM
and what is the size in KB that gsm spent?
bp
On 6/9/06, Pablo Allietti [EMAIL PROTECTED] wrote:
hi all, i saw in digium that the codec g729 is not free. exist
another
codec with low
On Tue, Jun 13, 2006 at 09:53:36AM +0200, Filip Dr?gowski wrote:
Very nice phones. There is no problem when conected to Asterisk (for
about 6 months now)
any body know this phone? support NAT? and standart codecs of asterisk ?
thank you all!!
-FD
any body know this phone? support NAT? and standart codecs of asterisk ?
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hi all, i saw in digium that the codec g729 is not free. exist another
codec with low bandwith to use in asterisk for free?
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Pablo Allietti
E-mail: [EMAIL PROTECTED] | LACNIC
],
there you need to paste your code to route the call to your meetme
room.
Hope it helps,
Best regards,
Marco Mouta
Ps. Please give me some feeback if it solved.
On 6/7/06, Pablo Allietti [EMAIL PROTECTED] wrote:
hi all i have an asterisk working and i need to add a mettme
hi all i have an asterisk working and i need to add a mettme public
service.
for example i need to download a soft (sjphone) and without any
configuration call to [EMAIL PROTECTED] (meetme) and join a conference but when
i do that i
received an error saying nomber do not exist. but if i call a
is possible to define a parameter to, hangup the line on silent? or ping
dead or something?
because all line have busy after the pc hangup :(
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hi all i have a asterisk configured and working perfectly. but i have a
problem.
if i download a softphone for example sjphone and digit for example
[EMAIL PROTECTED] i receive this call. is possible to block this?
i only want to received calls for login users...
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hi all a dumb question..
how i do to block the 00 for certain sips extensions?
for example i have the extensions 400 to 500
i need to extension higher than 429 can't digit 00
in my extensions.conf i have
exten = 420,1,Dial(SIP/420,20)
exten = 420,2,Hangup
exten = 421,1,Dial(SIP/421,20)
exten
hi all i use asdterisk in my company with Flash Panel Operator to know
who is talking or ringing. But i dont know any web application to know
who is online or offline. any body know any webapp for that ?
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On Wed, Dec 28, 2005 at 09:15:15AM -0600, Kevin P. Fleming wrote:
Pablo Allietti wrote:
hi all i use asdterisk in my company with Flash Panel Operator to know
who is talking or ringing. But i dont know any web application to know
who is online or offline. any body know any webapp
On Thu, Dec 29, 2005 at 02:36:05AM +1100, Adrian Carter wrote:
you need to set the extensions paramters to qualify=yes or
qualify=integer and then FOP (flash operator panel) will reflect the
status of the extensions.
Pablo Allietti wrote:
yep. this solve my problem Thanks
.
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---end quoted text---
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Pablo Allietti
LACNIC
.
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LACNIC
Hi all i have some problems with my pbx and asterisk codecs.
if i use g711u or g711a codecs. the line never hangup. and the origin
and destination are connected until i restart my pbx or asterisk
But if i use GSM all work fine.
is possible to solve this problem? or use only gsm codec?
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LACNIC
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hi all. i have my asterisk with a 192.168.0.1 address
which ports i need to forward in my firewall to connect remote xten
clients and make calls?
thsnk
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On Wed, Nov 30, 2005 at 01:17:33PM -0500, Sean Cook wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Looks like your zap channels are droping into the default context...
better to set up a from-pstn context and start there.
hi sean you have a example please?
Pablo Allietti
, span 1did not
receive any number or i have miss configure somenthing in asterisk box?
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LACNIC
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hi all, i have asterisk configured and working but the quality is very
poor. i ear noise and braks in the voice when the people talk to me, and
the people that eared me have the same problem any recommendation?
any files you need to post?
--
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Pablo Allietti
LACNIC
On Thu, Nov 10, 2005 at 12:57:45PM +0800, Dinesh Nair wrote:
On 11/10/05 08:52 Pablo Allietti said the following:
yes but both of them have problem with voice. some skype too anybody can
have this problems in freebsd? i hear cutted conversations`:
perhaps there's contention for your
On Wed, Nov 09, 2005 at 01:20:47PM +0800, Dinesh Nair wrote:
On 11/09/05 07:17 Pablo Allietti said the following:
Hi all
anybody can tell me what sipphone are available for Freebsd?
/usr/ports/net/kphone
/usr/ports/net/linphone
yes but both of them have problem with voice. some skype
Hi all
anybody can tell me what sipphone are available for Freebsd?
i cant find anyone
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Hi all i have a question. is my first time using [EMAIL PROTECTED] and i
need your help
i configure all my asterisk to go outside and work perfect via te110p
but now i need to receive calls. but when in my PBX i digit the number
for example 202 the asterisk receive a s i suppouse. the error
/zaptel.conf):
span=1,1,0,ccs,hdb3
bchan=1-15
dchan=16
bchan=17-31
You have to study the rest of * conf file, but these ones are the important
ones.
Regards,
--hg
- Original Message -
From: Pablo Allietti [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent
, asterisk will dial:
88 + -1234
Hope this helps,
--hg
- Original Message -
From: [EMAIL PROTECTED]
To: Pablo Allietti [EMAIL PROTECTED]
Sent: Tuesday, October 25, 2005 11:52 AM
Subject: Re: Siemens HI-path to ASTERISK
Hi Pablo!
I understood your problem. It is related
anybody can connect a Siemens HI-PATH to ASterisk via e1 ?
i need your help please.
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hi all i have this structure.
Box.(te110p)Pbx(e1)4 analogic lines to outside
is poosible connect asterisk to get outside lines? because i can call
any extension in my pbx with xten but i cant get outside lines. the
asterisk tellme all circuits are busy when i send the
hi all, anybody have a siemens hipath 3500 with a sm2/pri card? because
i need to connect to my box TE110P (e1) and i dont know how is the mode
in the pbx to change it.
thanks
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LACNIC
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problem?
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Pablo Allietti
LACNIC
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On Wed, 2005-09-14 at 11:46 -0300, Pablo Allietti wrote:
hi all, i have a box with a te110p and a pbx siemens... connect both
with a e1.
with a xten soft i can call extensions numbers in my office example 100
102 etc. but when i truy to go outside with the 9 before the call rings
in the first
:
http://lists.digium.com/mailman/listinfo/asterisk-users
---end quoted text---
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LACNIC
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problem.. ring in the extension 100.
I tried to open the file kds again and now it showed me your configuration
:) don't know why it did not show me before
Sander
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Pablo Allietti
Verzonden: woensdag
to ask them to programming. please help me.
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LACNIC
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to change something i can give you the software for
programming siemens pbx if you want
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Pablo Allietti
Verzonden: vrijdag 9 september 2005 16:09
Aan: asterisk-users@lists.digium.com
Onderwerp
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Pablo Allietti
Verzonden: vrijdag 9 september 2005 19:35
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: [Asterisk-Users] Re: siemens pbx what i ask techinician?
On Fri, Sep 09, 2005 at 05:39:22PM +0200, Sander wrote
of crc4 check on the e1 card configuration maybe
you can give me your mail adres so i can make screenshots of the manager e
configuration tool i can't mail pictures to the user list
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Pablo Allietti
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