,
Marco Mouta
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I think I understood what you want:
1- You want when someone dials an extension, do a Lookup in a database
using FWDCIDNAME
2- Then Dial the number that corresponds to this FWDCIDNAME in database
is that?
If it is so, i would recomend you to use AstDB - Asterisk Berkeley DB
(version1) -
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; Number of seconds to wait between
digits when transfering a call
This is timeout after pressing the first digit isn't it?
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PROTECTED]
This works good to retrieve the voicemail pressing message button, but
the Orange light keeps turning on and off all day:(
Any one can help me on this or has experience with this? Could be a
bad interpretation from me about the instructions on wiki.
Thanks,
Marco Mouta
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Error syntax:is Voicemail([EMAIL PROTECTED],bg(10)) ; for busy announce and 10dB record gainOn 10/16/06, Marco Mouta
[EMAIL PROTECTED] wrote:Thanks!But i've solved my problem only using g(#) gain argument from voicemail application! For me was enough.
Voicemail([EMAIL PROTECTED],b,g(10)) ; where
the wav49 version. However, we haven't noticed much difference, still fairly small attachments. Definitely no problems on a LAN or Broadband
connection. From: Marco Mouta [mailto:[EMAIL PROTECTED]] Sent: Tuesday, October 10, 2006 2:18 PM
To: Asterisk Users Mailing
Hi,Is is possible to implement this:Hicom150 --- BRI (QSIG) AsteriskI've been reading Siemens documentation and they say:Digital nailed connectionsCorporate communication networks can be implemented over digital S0 or
S2M nailed connections between several Hicom systems using the CorNet
haven't noticed much
difference, still fairly small attachments. Definitely no problems on a LAN or Broadband connection. From: Marco Mouta [mailto:
[EMAIL PROTECTED]] Sent: Tuesday, October 10, 2006 2:18 PM To: Asterisk Users Mailing List - Non-Commercial
Hi guys,I've been installing Asterisk 1.4 with Asterisk addons, and i could notice that in /usr/lib/asterisk/modules/ doesn't have cdr_addon_mysql.so even after compiling Asterisk Addons!In fact the cdr_addon_mysql.c exists, but it doesn't seems to be compile when i run Asterisk-Addons: make make
Hi allI'm deploying a VoiceMailserver with Asterisk behind a legacy pbx, providing Voicemail to email services for Lecagy PBX extensions.On busy or unanswered calls, Legacy pbx will dial a specific DID (one per extension) to asterisk, and the call is handled by Voicemail application.
I've several
.
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hearing the I'm sorry tone. Anything I'm missing here?l.In data Mon, 02 Oct 2006 00:36:30 +0200, Marco Mouta
[EMAIL PROTECTED] ha scritto: Hi, I've been looking the application dial on my asterisk server 1.2.9, and as far
CLI show application Dial j- Jump to priority n+101 if all of the requested
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.,1,Dial(Zap/G1/${EXTEN})exten= _X.,2,hangupUse it to dial a local extension, i suppose to dial out you are using a prefix
On 10/2/06, bivio [EMAIL PROTECTED] wrote:
2006/10/2, Marco Mouta [EMAIL PROTECTED]:
please post your from-zaptel context in extensions.confThanks to Giordano (immediate=yes) i
/asterisk-users
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My mistake sorry for last postOn 10/2/06, Marco Mouta [EMAIL PROTECTED] wrote:
when you want to dial something via ZAP interface (to PSTN world) you should use dial(ZAP/)On 10/2/06,
Luca Corti
[EMAIL PROTECTED] wrote:Hello,I' using asterisk as a PBX for a dozen of SIP phones of various makes
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on this kind of asterisk solutions.I've googled and read about asterisk at large scale solutions, but still in doubt.
http://www.voip-info.org/wiki-Asterisk+at+large-- Com os melhores cumprimentos,Marco Mouta
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Melcon Moraes
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in extensions.conf and for queries and something else use AGI scripts, or you recomend me to build specific AGIscripts with IVR menus inside (this looks very limited for future WebConfig interface)?
What is your advice, concerning with your experience.-- Best regards,Marco Mouta
asterisk server) offline.Any one has successfull configuration for this?-- Best regards,Marco Mouta
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cumprimentos, Marco Mouta
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warning because I recently upgraded from install-misdn to install-misdn-mqueue but the driver installation manual has not changed. Some comments are due to the fact I'm still making tests to solve the
incoming calls problem I mentioned. Thank you. Giorgio Incantalupo Marco Mouta wrote: Please post
driver does notcomplainthe important is that now it works!!I have only to fix some warning (I hope for an update of the beronet
manual) and it seems all right.Thank you again for help!!Marco Mouta wrote: msns is as far as i know, similar to DIDs but it includes the complete Dialed number
Also your problem could be related with the Answer() you weren't answering the calls on your previous extensions.confPls test both configs with and without answer and reply your results.
On 9/6/06, Marco Mouta [EMAIL PROTECTED] wrote:
Hi,Multiple Subscriber Number. This is a telephone number
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It has happened to me, with a x100p card, problems receiving fax that i've solved adjusting the gains.
Don't understand quite well why you say that...
On 9/6/06, Steve Underwood [EMAIL PROTECTED] wrote:
Marco Mouta wrote: Try to increase your rxgain, and check you have echocancel disabled
me.-- Com os melhores cumprimentos,Marco Mouta
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I've solved the problem, but still not understanding very well why do i need it:I've inserted inside [ext-did-custom]exten=h,1,hangupWhy do i need this? this is not usually used to run something after an hangupcall?
thks!On 9/5/06, Marco Mouta [EMAIL PROTECTED] wrote:
Hi all,I think i'm missing
Thank you Very MUCH I really appreciate your explanation, i wasn't getting it!On 9/5/06, Tony Mountifield
[EMAIL PROTECTED] wrote:In article
[EMAIL PROTECTED],Marco Mouta [EMAIL PROTECTED] wrote: I've solved the problem, but still not understanding very well why do i need
it: I've inserted
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Hi all,Do you think it could be an affordable solution using a two fxs ATA device to connect an old legacy pbx (with few users) with a main asterisk server.phonesanalogueSmallOfficeLegacyPBxATA-2FXS-SIP--MainOffice AsteriskServer
This way also I would use ATA device as a Trunk
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Hi Tzafrir,I've made vim /etc/asterisk/extensions_custom.conf then :set syntax=asterisk, and nothing happens. No errors no warnings and also no highlight syntax...Thks,
On 9/2/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Fri, Sep 01, 2006 at 03:03:39PM +0100, Marco Mouta wrote: Hi all, I've just
. This way with only one or two ATA per small office i would be able to connected every one with main office with very lowcost price
I would like to hear from you any suggestions or ideas, is this acceptable for a productions system?-- Com os melhores cumprimentos,Marco Mouta
Hi all,Does any of you knows an Hardphone with VPN client embedded? -- Best regards,Marco Mouta
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, sep 04, 2006, 10:44 +0100, Marco Mouta wrote: I've made vim /etc/asterisk/extensions_custom.conf then :set syntax=asterisk, and nothing happens. No errors no warnings and also no highlight syntax...
Hi!!I have asterisk.vim under /usr/share/vim/vimXX/syntax/ and:set filetype=asterisk:syntax
BTW Could you tell me how to i make it load this option by default everytime?On 9/4/06, Marco Mouta
[EMAIL PROTECTED] wrote:Just Great!What was missing is
:syntax onNow perfect! Thks guys! In fact i couldn't find this basic step any where except here. Ok I'm a newbie, but it will help others
Done,I've created ~/.vimrc file and inside this file:syntax onthks once moreOn 9/4/06, Marco Mouta
[EMAIL PROTECTED] wrote:BTW Could you tell me how to i make it load this option by default everytime?
On 9/4/06, Marco Mouta
[EMAIL PROTECTED] wrote:Just Great!What was missing is
:syntax onNow
options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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:
Marco
Mouta
To:
Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, August 31, 2006 6:07
PM
Subject: Re: [asterisk-users] help
me!!Problem on incoming calls
forgot to mention, it may help if you post your
extensions.confAs you are using from
=
fax,1,Goto(ext-fax,in_fax,1)
;end from-trunk
I tested your example:
The input calls from cellphones execute the [justtotest] section,
the other input
calls not worked fine.
- Original Message -
From:
Marco
Mouta
To:
Asterisk Users Mailing List -
Non-Commercial
the highlight syntax working fine for my asterisk.conf files.Any one can help me?Centos4.2 is my distribuition
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questions for you...you speak Italian
language?
Thanks in advance for answers
- Original Message -
From:
Marco
Mouta
To:
Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Friday, September 01, 2006 3:23
PM
Subject: Re: [asterisk-users] help
me
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or something where we can share this pattern Numbers?Is very hard to discover all the patterns for all the countries without sharing our knowledge...
Any tips?-- Best regards,Marco Mouta
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context for incoming calls from PSTN line...On 8/31/06, Marco Mouta
[EMAIL PROTECTED] wrote:Hi Please Post you Asterisk CLi when incoming is arriving.
On 8/31/06, Patrick
[EMAIL PROTECTED]
wrote:On Thu, 2006-08-31 at 17:27 +0200, Andrea infoteam wrote: Hi,
Please Help me!!! I've installed
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Yeah,Could be a solution! Thanks for your reply.On 8/31/06, Henry J. Cobb [EMAIL PROTECTED] wrote:
Marco Mouta [EMAIL PROTECTED] wrote: Hi all, I'm developing a Click to call Website, but now i'm getting worried with Click to Call fraud Imagine I just create one of this PhoneNumbers
(extra
,-- Com os melhores cumprimentos,Marco Mouta
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...
MaxRetries: number Number of retries before failingThis way i get two GSM calls to the same mobile while the first one is sucessfully running... I only figured out to use now MaxRetries:0Any guess why does this happens?
-- Com os melhores cumprimentos,Marco Mouta
in
extensions.conf, that's your dialplan.Hope it helps,Ps. Plse give me some feedbackOn 8/16/06, Juan Luis Moyano
[EMAIL PROTECTED] wrote:Marco Mouta escribió: Hi , Please post here your
extensions.conf in your central server only with that i can figured out or at least try to help u. Best regards, Marco Mouta
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Hi,
Another question. With latest version of asterisk softwares am I ableusing rxfax? I had read some remarks about incompatibility between TDMcard and rxfax. Is it still exist?
I've been using rx for fax reception with TE110P as well as X100P (this only for tests and learning) with very
in the same LAN subnet with theAsterisk box.I can send fax out using txfax in call file, but I did have to lower
the rxgain and txgain.This is what I'm trying to do:Fax machine --- SIP ATA--LAN--Asterisk --PRI-- PSTNHave you tried this?Do you have to disable Echo canneler?Thanks.
AndyOn 8/15/06, Marco Mouta
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those files
jingle.conf and jabber.conf, i mean who is who, and their goals.-- Com os melhores cumprimentos,Marco Mouta
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for format_mp3.soWhat could be wrong?I've made already several asterisk installl and never got this problem...
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Hi,Your Consultant has developed it with PHP scripts, so you must check those files in /var/lib/asterisk/agi-binYour application logic is there.Hope it helps,Best regards,Marco Mouta
On 8/3/06, Randy Paries [EMAIL PROTECTED] wrote:
On 8/2/06, Time Bandit [EMAIL PROTECTED] wrote: The problem
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,
Marco Mouta
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Erik,
What a great and detailled explanation! Thank you very much!
Ps. If you know anything about legal issues asked abouta g729 please
post it here:)
Best regards,
Marco Mouta
On 7/26/06, Erik [EMAIL PROTECTED] wrote:
Marco Mouta wrote:
By the way could any one tell me wich
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my mistake you post it! could you pos it in file.conf format?
On 7/25/06, Marco Mouta [EMAIL PROTECTED] wrote:
It seems you didn't post any thing about you [general] sip.conf
neither allowed codecs
On 7/25/06, Carlos Alberto Bernat Orozco [EMAIL PROTECTED] wrote:
Hi group
Thanks Marty
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Chattanooga, and Montgomery.
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I must say that for mailing lists Gmail seems to me just perfect! I
apreciate your integration into Forum. But Gmail seems to me even more
friendly!
Best regards,
Marco Mouta
On 7/21/06, zoa [EMAIL PROTECTED] wrote:
There are some others out there, we did something similar at
http
that just
monitors your G729 licences, and keeps on track which codec is going
to be used: Ulaw or G729.
Don't know if this is a good idea, just a suggestion.
Best regards,
Marco Mouta
On 7/21/06, Woodoo People .pGa! [EMAIL PROTECTED] wrote:
No, we aren't intending to check for available g729
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Could you post your sip.conf?
On 7/21/06, Jose Limeres [EMAIL PROTECTED] wrote:
Yes, of course. SIP, RTP and IAX ports are port forwarded to the * box.
On 21/07/06, Marco Mouta [EMAIL PROTECTED] wrote:
Did you port forwar in your router RTP ports ? 1-2 to your *Box ?
On 7/21/06
Hi,
I think i found your error. you are missing a context for your peer
PeopleCall , this way no context for incoming calls!
Am I wrong?
Hope it helps,
Marco Mouta
On 7/21/06, Jose Limeres [EMAIL PROTECTED] wrote:
Here is my SIP.conf. (just replaced psswds with *)
Thanks.
[general]
port
-users
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already have set language to 'se' in indications.conf.
Next question. If asterisk where to play a digit - does it look in
/sounds/se/digits or /sounds/digits/se ?
Regards,
Jan
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Marco Mouta
Skickat: den 19 juli
Try to active callwaiting in those unreachable extensions. You just
need to dial *70 from every SIP extension.
Be aware that *70 (call waiting ) may be disabled in your freepbx.
Hope it helps,
Marco Mouta
Please give me some feedback
On 7/17/06, Tim P [EMAIL PROTECTED] wrote:
Not sure where
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Com os melhores cumprimentos,
Marco Mouta
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advice me on this settings ? or is this something worst?
Best regards,
Marco Mouta
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- Call party A - Call duration into my database
- then call party B and bridge it with A and keep CDR of the call
duration between A and B.
Does any of you has experience with this?
Best regards,
Marco Mouta
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,
Ganbaa
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Marco Mouta
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