forking CDR could help Ricardo.
On 12/15/06, Ricardo Martins [EMAIL PROTECTED] wrote:
Hi John, I´m very interested into this call forwarding capabilities and
I solved this problem filtering on the web-script (in my case, php) the
number the user can intert on the database. (I know it´s not an
number.
After this Hands on I can sucessfully send faxes with Hy-email2fax --
Hylafax---asterisk Sucessfully.
But as i mentioned before i need to get ride of ^M on the subject line.
Any one can help me on this?
Best regards,
Marco Mouta
On 12/13/06, Lee Howard [EMAIL PROTECTED] wrote:
Marco
/etc/asterisk/modules.conf
On 12/13/06, Angel Heart [EMAIL PROTECTED] wrote:
Hi,
In what Asterisk file can I edit for me to temporarily unload such
modules. But later I woudl still be able to load them.
Thanks
Angel
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: JOB 1 DEST 2079^M COMMID 00157
DEVICE '/dev/ttyIAX' FROM 'Marco Mouta [EMAIL PROTECTED]' USER root
Dec 13 11:28:07.51: [ 9242]: STATE CHANGE: RUNNING - SENDING
Dec 13 11:28:07.51: [ 9242]: -- [12:AT+FCLASS=1\r]
Dec 13 11:28:07.51: [ 9242]: -- [2:OK]
Dec 13 11:28:07.51: [ 9242]: MODEM set XON/XOFF
me too, i'm trying to add sip users , i click save, it reports successfully
saved... but there are no sip accounts created...
On 11/29/06, Dumpolid Exeplish [EMAIL PROTECTED] wrote:
i had the same problem. the GUI stopped responding to configuration
changes.
On 11/28/06, James Willing [EMAIL
take a look on Audacity program is opensource and has the option Generate
Beep, then just add some Gain as you want...
On 12/2/06, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote:
We've had a few people complain that the beep before leaving a
voicemail is not loud enough and too short. Does
Hi all,
is there a way I can put a line on hook ? I'd like to keep the line busy
on demand (es. dialing an extension will put on hook line n.1) so the
caller receives busy tone directly from PSTN and not from asterisk.
Thanks.
marco
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do you have created Asterisk views to SER database? Are you using sip
realtime on asterisk?
please post your extensions.conf.
By the way, I'm Portuguese:)
Qualquer coisa manda mail pode ser q consiga ajudar.
On 11/24/06, Ricardo Carvalho [EMAIL PROTECTED] wrote:
Hi Marco,
Ser has IP
,
Marco Mouta
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a scroll on this to display everything? do i need to
resize the buttons?
For sure someone now how to solve this basic question:)
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Hi all,I've 2 * servers with static IP, and i notice that if i set both sip peers with host=server_ip and qualify=yes it presents UNREACHABLE on asterisk CLI.When i changed the host parameter to host=dynamic and set the register string in the [general] of
sip.conf on both servers, the connection
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callwaitingcallerid=yesthreewaycalling=yestransfer=yescanpark=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yesrxgain=0.0txgain=0.0group=1callgroup=1
pickupgroup=1immediate=noThanks Marco-- Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br2006/11/9, Marco Mouta
[EMAIL PROTECTED
Hi guys,
I've been looking on wiki, but i could find it only for chan_capi:
http://www.voip-info.org/wiki/view/Asterisk+PBX+functions
In the CAPI channel
See Asterisk CAPI channels
* Call Deflection (CD) (redirect without answering): Implemented
by chan_capi
How can i do it with my
Silva [EMAIL PROTECTED]:
Finally this works!!! :)
Tanks to Alberto and Marco by your help!
The problems are:
- the cable was connected to the wong card port... :(
- the card config needs to be: ETSI; TE; Point-to-Point (I thought
that was point-to-multipoint).
Best regards,
PS
-users mailing list
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jb
Marco Mouta a écrit :
pls post iax.conf of Both machines , as well as your dial() string on
both servers to connect each other.
That way would be easier to help you.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Hello,
I'm french, so excuse my poor English.
I'm face
specified
and no allow and/or deny restrictions at all. If such an entry is
found, accept the connection. and use the name of the found iax.conf
entry as the connecting username.
Pls give some feedback if you solved the problem.
On 10/27/06, Marco Mouta [EMAIL PROTECTED] wrote:
Hi
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?
Mark
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My mistake:
[kpn-is]
exten= _X.,1,answer
exten= _X.,2,Noop(My telco is sending me this MSN string: ${EXTEN})
exten= _X.,3,wait(1)
exten= _X.,4,Playback(vm-goodbye)
exten= _X.,5,hangup
On 10/27/06, Marco Mouta [EMAIL PROTECTED] wrote:
Plse Read bellow:
On 10/27/06, Mark Hannessen [EMAIL
...
jb
Marco Mouta a écrit :
Hi,
I think i found your problem, look that in your debug you have, -
Accepting UNAUTHENTICATED call from 10.0.0.160:
Take a look on incoming call authentication, and how asterisk handles this:
http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication
,
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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PROTECTED] wrote:Hi Marco,in attachment you can find my
misdn.conf. Consider that I'm still fixingsome warning because I recently upgraded from install-misdn toinstall-misdn-mqueue but the driver installation manual has not changed.Some comments are due to the fact I'm still making tests to solve
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-goodbye)
exten = s,n,HangupEnable an higher debug level for misdn messages in misdn.conf (I think is this the file).Pls post your results Asterisk CLI.On 9/6/06,
Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi Marco,I have not a normal extensions.conf:[outbound_isdn]include = parkedcallsexten = _X.,1
in one asterisk server.
Got it?Best regardsOn 9/6/06, Giorgio Incantalupo
[EMAIL PROTECTED] wrote:Hi Marco,it seems that msns=* is necessaryto make Asterisk work correctly...I
do not why...We have another PBX with a monoBRI but have not this problem, maybe isthe different ISDN telco or the old misdn
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
add this to [justtotest]exten=s,1,Answerexten=s,2,Playback(vm-goodbye)exten=s,3,hangupreply your results and asterisk cli
On 9/1/06, Andrea infoteam [EMAIL PROTECTED] wrote:
Hi Marco,
my from-trunk context in extensions.conf
is:
[from-trunk]
include = from-pstn
[from-pstn]include
the highlight syntax working fine for my asterisk.conf files.Any one can help me?Centos4.2 is my distribuition
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PROTECTED] wrote:
Hi marco,
this is my cli when i receive a call beginning with 0,i have done two
tests:
First test the cli is:
-- Executing Answer(VISDN/visdn1.2/10.I, ) in new
stack -- Executing Playback(VISDN/visdn1.2/10.I,
vm-goodbye) in new stack -- Playing 'vm-goodbye'
(language
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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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using their software...Regards,Patrick___--Bandwidth and Colocation provided by Easynews.com
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-- Com os melhores cumprimentos,Marco
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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http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta
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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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