:
> 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.
>>
>>
>> [E
:
> 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.
>>
>>
>> [E
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 PROTE
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
---end quoted text---
--
.-
Pablo Allietti
E-mail:
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 PROTEC
hi all, in console mode how i can display the logged users?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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
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 ?
--
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/aste
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?
--
.-
Pablo Allietti
E-mail: [EMAIL PROTECTED] | LACNIC
t;context.
>In your extensions.conf look for section called [from-sip-external],
>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.
>
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 e
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 :(
--
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or u
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...
--
.-
__
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)
ext
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= and then FOP (flash operator panel) will reflect the
>status of the extensions.
> Pablo Allietti wrote:
yep. this solve my
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 bod
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 ?
--
thanks.
___
--Bandwidth and Colocation provide
gs have been fixed and asterisk restarted, then
> watch the asterisk CLI to "see" what happens when one phone calls the
> other. If you still have problems, paste the CLI output into a posting
> for us to see. Without that, we can only guess.
>
> Given what you have posted, I don't have a clue what you are trying to
> do with:
> exten => _XXX,1,Dial(${TRUNK}/${EXTEN})
> exten => _XXX,2,Voicemail(u${EXTEN})
> However, when sip extension 402 dials 403, it will match the above _XXX
> and send that call out Zap/g1 (whatever that happens to be).
>
> If you really are working with two asterisk systems tied together with
> Zap channels, then I'd suggest modifying the above to something like
> exten => _5XX,1,Dial(${TRUNK}/${EXTEN})
> exten => _5XX,2,Voicemail(u${EXTEN})
> when the 4XX extensions are on one system and the 5XX extensions on the
> second system.
>
>
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
---end quoted text---
--
.-
Pablo Allietti
LACNIC
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
bly even iax trunking, etc.
>
>
> _______
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
---end quoted text---
--
.-
Pablo Allietti
LACNIC
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
nd Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
---end quoted text---
--
.-
Pablo Allietti
LACNIC
___
--Ban
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?
--
.-
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
--
.-
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To
z+eb1/MaABy2gxUMOcMw1AMwCfYEJI
> VTt9lDiRDMLZhJ2aOL4Qpnw=
> =KqmL
> -END PGP SIGNATURE-
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
&
'Zap/31-1'
-- Accepting call from '' to 's' on channel 0/31, span 1 <<<< did not
receive any number or i have miss configure somenthing in asterisk box?
--
.-
Pablo Allietti
LACNIC
___
--Bandwidth and Colo
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?
--
.-
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`:
>
> per
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 o
Hi all
anybody can tell me what sipphone are available for Freebsd?
i cant find anyone
--
.-
Pablo Allietti
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http
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 mess
. So, if a Asterisk user wants to
> call the outside number -1234, he/she will dial:
> 9 + -1234
> Asterisk with then route this call to HiPath prefixing the trunk access
> code, for example, "88". So, asterisk will dial:
> 88 + -1234
>
> Hope this hel
extensions.conf please please/
>
> 3. At Asterisk, put these lines (/etc/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,
>
anybody can connect a Siemens HI-PATH to ASterisk via e1 ?
i need your help please.
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asteris
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 number
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
--
.-
Pablo Allietti
LACNIC
___
--Bandwidth and Colocation sponsored by
ld work
no uuuaaa the same 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 PROT
aptel.conf]
>span=1,1,0,d4,ami # have tried with 1,0,0 - same problem
>fxsks=1-24
>loadzone = us
>defaultzone=us
>
>[zapata.conf]
>[channels]
>signalling=fxs_ks
>group=0
>;context=incomin
xten => _9XXX.,2,Congestion
> exten => _9XXX.,103,Hangup
>
> Hope this helps,
>
> -Matt
>
> 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 so
problem?
--
.-
Pablo Allietti
LACNIC
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit
anager e
> configuration tool i can't mail pictures to the user list
>
>
>
> -Oorspronkelijk bericht-
> Van: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Namens Pablo Allietti
> Verzonden: vrijdag 9 september 2005 20:29
> Aan: Asterisk Users Mailing List
pronkelijk bericht-
> Van: [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?
>
> O
ou have to ask them 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:
ask them to programming. please help me.
--
.-
Pablo Allietti
LACNIC
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
44 matches
Mail list logo
