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Luis Morales
Consultor de Tecnologia
Cel: +5
Any help will be appreciated. Thanks
> \RR
>
> On Thu, Nov 11, 2010 at 11:22 PM, Luis Morales wrote:
>>
>> I use Nuance, festival, Ibm tts and Loquendo.
>>
>> Now in your case, i suggest use tts on the recommend tts
>> environment. Solaris is not standart sys
languages and
> can one build these independently of the telephony platform being used so I
> could use maybe Asterisk running on Solaris 10 and a cluster/farm of TTS
> servers for TTS processing.
> Thanks
> RR
>
> On Thu, Nov 11, 2010 at 10:21 PM, Luis Morales wrote:
>>
>&g
sterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
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eve
>
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Luis Morales
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Cel: +(58)412
-users mailing list
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Luis Morales
Consultor de Tecnologia
Cel: +(58)412-2352745
-
;t find
> any way without batching it from the /var/log/asterisk/queue
>
>
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Morales
> Sent: Saturday, January 30, 2010 8
ng list
>> To UNSUBSCRIBE or update options visit:
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>>
>>
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Luis Moral
r calls.
>> best
>> --
>> Raimund Sacherer
>> -
>> RunSolutions
>> Open Source It Consulting
>> -
>> Email:�...@runsolutions.com
>> tel: 625 40 32 08
>>
>> Parc Bit - Centro Empresarial Son Espanyol
>> Edificio Estel - Local 3D
&
l:�...@runsolutions.com
> tel: 625 40 32 08
>
> Parc Bit - Centro Empresarial Son Espanyol
> Edificio Estel - Local 3D
> 07121 - Palma de Mallorca
> Baleares
> On Aug 19, 2009, at 12:46 PM, Luis Morales wrote:
>
> Did you try use busytect option enabeled into zaptel.conf file ?
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>
, Aug 18, 2009 at 10:46 AM, Kevin P. Fleming wrote:
> Luis Morales wrote:
>> You rite Kevin,
>>
>> We enabeled sec echo canceller. I'll be test now and let's know the results.
>
> SEC is not a good choice. If you are going to try something other than
> HPE
You rite Kevin,
We enabeled sec echo canceller. I'll be test now and let's know the results.
Regards,
On Tue, Aug 18, 2009 at 8:47 AM, Kevin P. Fleming wrote:
> Luis Morales wrote:
>> The error on system crash is:
>>
>> Digum Board: TD
ND PGP SIGNATURE-
>
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Luis Morales
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Cel: +(58)416-4242091
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Phoenix, Arizona
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m_init)
-
Does any know whats happens ? Any idea or Tips ?
Regards,
--
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4
Try to parser queue_log file in real time and catch the event CONNECT
Regards,
Luis Morales
On Tue, Apr 14, 2009 at 12:44 PM, Ryan M. Colbert
wrote:
> Is there a way in the dialplan to figure out which agent in a ring all queue
> answered a line? I’d like to take specific action based
> And yes i know this is an asterisk list.
>
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Morales
> Sent: April-06-09 2:02 PM
> To: Asterisk Users Mailing List - Non-Commercial
This may be your solution.
Regards,
Luis Morales
On Mon, Apr 6, 2009 at 12:23 PM, ContactTel Business
wrote:
> Why would i want to do that ?
>
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.c
Take a look on xorcom solutions
http://www.youtube.com/watch?v=qt4aPdGIvIQ&feature=player_embedded
Regards,
Luis Morales
On Mon, Apr 6, 2009 at 8:09 AM, ContactTel Business
wrote:
> Any hardware that can do 25-50-100 fxs ports trunked to sip ?
>
>
>
> Example one end a cat
nd do an asterisk cluster based building an locations.
Group extensions by buildings/asterisk servers.
4) Planning and do asterisk network with and distributed dial plan and trunking
5) Try locate an asterisk specialists
6) believe in asterisk!
Regards,
Luis Morales
On Tue, Mar 17, 2009
Look like good. I have an similar server for 100 ext.
Regards,
Luis Morales
On Mon, Mar 9, 2009 at 8:01 AM, Elliot Murdock wrote:
> Hello Everybody!
>
> I am currently setting up an Asterisk server for medium to high load
> (approximately 20-35 concurrent phone lines).
>
>
amp; Asterisk
> http://youtube.com/voiceroute
>
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igo eq $clave) ? 1 : 0;
print "Content-type: text/html\n\n";
print "$i";
--
Now you can use asterisk+curl to send and receive data. If you are
working with .net you can make web services in
Suggest,
Use .net to do an web services and use curl+agi scripts to integrate
your solutions.
Regards,
Luis Morales
On Wed, Feb 25, 2009 at 6:37 PM, Douglas Mortensen
wrote:
> Hello.
>
> I have a software developer creating a .Net / mono program to use as an
> AGI script. W
Does any have experience with E1 telephony support plus asterisk in
costa rica ?
Regards,
Luis Morales
--
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
Done,
There is no problem to do it. Did you pay for this script or you need
support to do it ?
Have an nice day,
Luis Morales.
On Fri, Feb 20, 2009 at 7:24 PM, michel freiha wrote:
> Dear Sir,
>
> I need the followingA customer will dial a specific number like 112,this
> wil
ium.com/mailman/listinfo/asterisk-users
>
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-----
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
-
"Empieza por hacer lo necesario, luego lo que es posible... y de
pronto es
on kernel boot parameters do it:
acpi=off
Regards,
Luis Morales
On Mon, Jan 19, 2009 at 12:25 PM, Plugworld wrote:
> Hi All
>
> I'm having some serious kernel panic while using digium cards.
>
>
>
> It may be related to IRQ shared.
>
> Can this cause a lot of
://lists.digium.com/mailman/listinfo/asterisk-users
>
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-----
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
-
"Empieza por hacer lo necesario, luego lo que es posible... y
Try with ** + iax extension
Regards,
Luis Morales
On Mon, Nov 24, 2008 at 10:20 PM, Bruno Castelo Branco
<[EMAIL PROTECTED]> wrote:
> Hi
>
> Somebody knows if pickup call works with IAX2?
> I enable *8 in features.conf, but doesn't works with IAX2 extensions.
Thnx Philipp!!
On Fri, Nov 21, 2008 at 7:12 PM, Philipp Kempgen
<[EMAIL PROTECTED]> wrote:
> Luis Morales schrieb:
>> Does any know what happens with svn repository on svn.digium.com ?
>
> http://lists.digium.com/pipermail/asterisk-users/2008-November/222147.html
&g
Does any know what happens with svn repository on svn.digium.com ?
--
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
ns with 11.
> Any suggestions?
> -Ed
>
>
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ay such as:
- hanguppolarityswitch = yes (zapata.conf)
Another question, to connect analog line to digium card you make the
rj-11 connect ? there is an posibility that you have incorrect pin
out.
Regards,
Luis Morales
On Fri, Oct 10, 2008 at 5:10 PM, Mike <[EMAIL PROTECTED]> wrote:
> On Fri, Oct 10, 2008
allerid=asreceived
> channel => 3
>
> ; Channel 4: PSTN line
> context=incoming
> group=1
> usecallerid=no
> faxdetect=none
> signalling=fxs_ks
> rxgain=4
> txgain=4
> callerid=asreceived
> channel => 4
&g
Try with fop,
http://www.asternic.org/
Regards,
Luis Morales
On Fri, Oct 10, 2008 at 2:40 AM, Patrick
<[EMAIL PROTECTED]> wrote:
> Hi,
>
> Does anyone have a suggestion how I can analyze the concurrent usage of
> ISDN channels? A client complains about their clients sometime
Mike,
Can you tell us :
- asterisk version
- zaptel version
When you call over this line, when you hangup did you hear an busy
tone ? or any class tone ? To do this test connect your lines to
analog phone and make a call. Let's us know the results.
Regards,
Luis Morales
On Fri, Oct 10,
Try with:
core show channels verbose
or
core show channels concise
Regards,
Luis Morales
On Tue, Sep 30, 2008 at 10:38 AM, Singer Wang <[EMAIL PROTECTED]> wrote:
> Hello,
>
> I have a question. We have a 8 port FXO card in our asterisk server plugged
> into 8 analog lines.
,
Luis Morales
On Wed, Sep 24, 2008 at 4:21 AM, Erik Haider Forsen <[EMAIL PROTECTED]> wrote:
> Hi!
>
> I'm new to this list. I tried to search the list archive for a
> solution on my current setup, but couldn't find any.
>
> We have an asterisk connected directly to
Moy,
How i can do to join asterisk-r2 list ? My congratulations about your
article in digium blog http://blogs.digium.com/page/2/
I will collaborate in your project and give support from Venezuela.
Regards,
Luis Morales
On Sat, Sep 20, 2008 at 7:47 PM, Moises Silva <[EMAIL PROTECTED]>
Not really the unicall setup must be idem. So you can see the unicall
channels ?
It's moises are busy i can give you support too
Regards,
Luis Morales
On Sat, Sep 20, 2008 at 12:15 PM, Dae Yeung Um <[EMAIL PROTECTED]> wrote:
> Hi Luis,
>
> But this E1 has 30 ch
efault
28from-pstn Idle es default
29from-pstn Idle es default
30from-pstn Idle es default
31from-pstn Idle es default
Good luck!
Luis Morales
On Fri
f
Regards,
Luis Morales
On Fri, Sep 19, 2008 at 12:46 PM, Dae Yeung Um <[EMAIL PROTECTED]> wrote:
> All channels 1~15, 17~31 is supposed to be double way. To place and receive
> calls.
>
>
> The line is supposed be E1-MFC/R2 and works perfect with a Panasonic PBX,
> actually
Ok,
in your E1 setup:
1-15: to outgoing calls
16-30: for incomming calls
?
Now for make calls your telephone company must be provide MFC-R2
signaling. In your case the logs files show an invalid signal on make
call.
Regards,
Luis Morales
On Fri, Sep 19, 2008 at 10:10 AM, Dae Yeung Um
Dae,
Activate debug full:
asterisk -vr
in other console do:
tail -vf /var/log/asterisk/full
Try to put call and send us more details about your logs
Regards,
Luis Morales
On Thu, Sep 18, 2008 at 11:49 PM, Dae Yeung Um <[EMAIL PROTECTED]> wrote:
> In fact I se
Check on this link,
http://www.moythreads.com/wordpress/
I have working asterisk+mfc/r2 with 4 E1 on Venezuela and work fine!
Regards,
Luis Morales
On Thu, Sep 18, 2008 at 1:57 AM, Dae Yeung Um <[EMAIL PROTECTED]> wrote:
> Hello
>
>
>
> I'm new in this list,
ital.com --
>
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I agree with Karl,
We have working an Digium TDM card and grand stream ata. On ata
device we connect an old fax machine. All work fine, you can send or
receive fax form old fax machine using an zaptel device.
Regards,
Luis Morales
On Thu, Sep 4, 2008 at 2:41 PM, Karl Fife
<[EMAIL PROTEC
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Check on dial plan rules, remember if you need dial to +number your
rule must be +|number this submit the number on your dialout plan
without +.
Regards,
Luis Morales
On Fri, Aug 22, 2008 at 3:40 AM, ronald <[EMAIL PROTECTED]> wrote:
> Hi Sir,
>
> I actually have a plus sign
Excelent!!
but may be better if you send to the list the zaptel.conf and zapata.conf
Regards,
Luis Morales
On Thu, Aug 21, 2008 at 10:19 PM, Lee, John (Sydney)
<[EMAIL PROTECTED]> wrote:
> I am really grateful to all the experts on the mailing list who gave me
> some very good ad
Thxs!!
On Wed, 2007-09-26 at 10:26 -0400, Doug Lytle wrote:
> Luis Morales wrote:
> > That's an good tips. Where i find information or help to provisioning
> >
>
> http://www.voip-info.org/wiki-Polycom+Phones
>
> Doug
>
_
That's an good tips. Where i find information or help to provisioning
the phones with ftp ? In my case the setup was made on each phone using
polycom web interface.
Regards,
Luis Morales
On Wed, 2007-09-26 at 09:23 -0400, Doug Lytle wrote:
> Luis Morales wrote:
> > Doug,
&g
Doug,
Where is located sip.cfg file ?
Regards,
Luis Morales
On Wed, 2007-09-26 at 08:32 -0400, Doug Lytle wrote:
> Luis Morales wrote:
> > Hi,
> >
> > Does any know adjust the volume for polycom ip soun point ? I adjust by
> > the phone on the current call, but when
Hi,
Does any know adjust the volume for polycom ip soun point ? I adjust by
the phone on the current call, but when hangup the volume lost the
volume configuration. There are any way to set phone volume by
default ?
Regards,
Luis Morales
Try using Unique call id, for example to pass this parameter into agi
script you can use:
exten => s,n,agi,myscript.agi|${UNIQUEID}
Regards,
Luis Morales
On Wed, 2007-06-13 at 19:35 +0530, Jaswinder Singh wrote:
> ption to execute an AGI script after the Dial (I need to track the
yep!
On Wed, 2007-06-06 at 16:51 +0200, Roger Schreiter wrote:
> Ed Nuñez schrieb:
> > Is anyone else having trouble going into voip-info.org today?
>
> Yes. Me.
>
>
> Roger.
>
>
>
>
>
> ___
> --Bandwidth and Colocation provided by Easynews.com
Hi Adi,
My be better if you send us the code about how did you do to catch and
retrive the data from asterisk.
Regards,
Luis Morales
On Fri, 2007-06-01 at 01:21 +0300, Adi Simon wrote:
> Hi Martin,
>
> Thanks for your reply. Maybe I wasn't clear enough. I am already
ID(name)}|
${EXT204})
Regards,
Luis Morales
On Thu, 2007-05-31 at 01:48 +0200, Vincent wrote:
> On Tue, 29 May 2007 07:39:40 -0400, in
> gmane.comp.telephony.pbx.asterisk.user Luis Morales wrote:
> ># send the result over callerid ;-)
> >$AGI->exec('SetCallerId',
Hi Gilles,
I think that you must be take the incomming call control using AGI perl
scripting. Take a look on this script:
;extensions.conf
[internal]
exten => group,1,LookupCIDName
exten => group,n,AGI(web.agi|${CALLERID(num)}|${CALLERID(name)}|
${EXT204})
= web.agi=
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