On Wed, 16 Sep 2009, Danny Nicholas wrote:
> What do you want your message to say? I'd just use busy-pls-hold and the
> caller would eventually get the idea that you weren't going to talk to them.
> You could also consider these
> Off-duty
> Not-auth-pstn
> Not-taking-your-call
> Number-not-answe
On Wed, 16 Sep 2009, Tilghman Lesher wrote:
> On Wednesday 16 September 2009 06:46:13 Khaled W Chehab wrote:
>> What g729 module should I download ?
>
> You should download only the licensed g.729 module from Digium, after paying a
> $10 license per concurrent user. All other modules have variou
Thank you Alex, I'll handle this programatically if there is no other way.
Best regards,
Patrick
On Thu, Sep 17, 2009 at 07:51, Alex Balashov wrote:
> You can set some kind of counter in the dial plan, call an AGI script,
> use func_odbc to make database calls, or otherwise achieve this
> pro
Dear Folks,
Im looking for a way to detect if an analog line is connected to card or not
(Im using Sangoma A200). Im using the dialtone detection when dialing but
need a way to detect the disconnection of the line when it actually happens.
Anyone have any hints or tricks for this?
Regards.
--
Mo
You can set some kind of counter in the dial plan, call an AGI script,
use func_odbc to make database calls, or otherwise achieve this
programatically.
--
Sent from mobile device
On Sep 17, 2009, at 1:16 AM, Patrick
wrote:
> Hello guys,
>
> I've one SIP trunk that support multiple DID. On
Hello guys,
I've one SIP trunk that support multiple DID. Only the trunk is
documented in sip.conf (called DID is taken from the sip-header in
real time).
I would like to limit the number of simultaneous calls on each DID. Is
there a way to achieve this ?
My understanding is that the SIP configura
Hello Ron,
I was thinking also to replace the email sent by the voicemail by a php script.
My questions is simple, how do you manage to get the voicemail
variables from the php script ?
Or, maybe, you get from stdin the content of the "email" that should
be send via sendmail ?
Thanks in advance
On 17/09/09 1:57 PM, Jeff LaCoursiere wrote:
>
> On Wed, 16 Sep 2009, Doug Lytle wrote:
>
>> Matt Riddell wrote:
>>> Basically, the phones are displaying 79 on the screen (the number the
>>> dial for pickup) - as you'd expect, but they'd like to see the CID of
>>> the person who called in.
>>>
>>>
Hi All,
Thanks for all the wonderful contributions, from cell phones right up to
proxies, etc...
Many thanks also to Tony Turner for the great advice.
As for Jared, what can I say...simply legend... :)
I believe this is what I was after.
:)
For all those attending AstriconSee you there!
On Wed, 16 Sep 2009, Doug Lytle wrote:
> Matt Riddell wrote:
>> Basically, the phones are displaying 79 on the screen (the number the
>> dial for pickup) - as you'd expect, but they'd like to see the CID of
>> the person who called in.
>>
>>
> There are patches against 1.4 that allow you to chang
Hey,
Ive installed web meetme and everything is working fine except no records
are being written to the cdr and participants tables, this is because the
cbEnd.php script is not running. Below is the output of the cbEnd.php when I
run in manually. I am running asterisk 1.4.20.1 and web meetme 3.1.0
Matt Riddell wrote:
> Basically, the phones are displaying 79 on the screen (the number the
> dial for pickup) - as you'd expect, but they'd like to see the CID of
> the person who called in.
>
>
There are patches against 1.4 that allow you to change the display to
anything that the phone wil
Using H323 to reach another h323 switch, I have no audio and the following
error:
[Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP
Transmission error of packet 21282 to XXX.XXX.XXX.XXX:6064: Invalid argument
[Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP
Transm
On 17/09/09 10:40 AM, Dan Saul wrote:
> This might be another piece of the puzzle:
>
> It would appear any application using playback functionality exits
> immediately. For example anything involving voicemail or playback. Phone
> calls work with no problem but not if asterisk must play something b
Thanks all. Turns out a package proplem was causing a conflict...I hard to
mess with packages to get all in and happy.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Wednesday, September
This might be another piece of the puzzle:
It would appear any application using playback functionality exits
immediately. For example anything involving voicemail or playback. Phone
calls work with no problem but not if asterisk must play something back.
The modules are loaded however...
Tsunam
The files used to be "Frederic Chopin – Polonaised Op. 40-2.raw" I have
since replaced the raw files with the original mp3s They are now as follows:
[r...@tsunami musiconhold]# ls -l .
total 13320
-rw-r--r-- 1 asterisk asterisk 5415926 2009-09-16 17:18 hm1.mp3
-rw-r--r-- 1 asterisk asterisk 821797
Hi
how do i set the call-limit on a dahi line - its connected to the pstn
network - shared fax line. How do i tell asterisk not to send more than
1 call there !
Alex
--
"Drug therapies are replacing a lot of medicines as we used to know it."
- George W. Bush
10/18/2000
St. Louis, MO
On Wed, Sep 16, 2009 at 12:24:22PM -0700, Steve Edwards wrote:
> On Wed, 16 Sep 2009, Danny Nicholas wrote:
>
> > I'd try this:
> > - exten => 4000,1,Dial(SIP/4000,20,ikKtT)
> > - exten => s-NOANSWER,1,Dial(SIP/4001,20,ikKtT)
> > - exten => s-NOANSWER,2,Voicemail(4000)
> > - exten => s-BUSY,1,Dial
What are your actual file names (/etc/asterisk/musiconhold/Frederic Chopin
Polonaised Op. 40-2.wav?)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Saul
Sent: Wednesday, September 16, 2009 4:50 PM
To: Asterisk Users M
Hi,
We're using the Pickup app on a customer's site with Linksys phones.
Is there any way to display the callerid of the phone call you've picked
up in 1.4?
I assume (rightly or wrongly) that this is connected line ID.
Basically, the phones are displaying 79 on the screen (the number the
dial
Dan Saul escribió:
> Hi,
>
> I have trouble getting MOH to work after an upgrade from asterisk 1.4
> to 1.6.1.4. The call goes on hold, MOH is started, and then stops
> right away.
>
> Here are the files both of type .raw:
>
> Tsunami*CLI> moh show files
> Class: default
> File: /etc/asterisk
That was a good shot in the dark, but sadly renaming it to something simple
(and removing all non ascii in the process) does not correct this.
On Wed, Sep 16, 2009 at 4:37 PM, Danny Nicholas wrote:
> Just a “shot in the dark” but could MOH be choking on the “long file
> names”? (does it work o
Just a shot in the dark but could MOH be choking on the long file names?
(does it work on fred_chopin_pol_1)?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Saul
Sent: Wednesday, September 16, 2009 4:18 PM
To: aster
Hi,
I have trouble getting MOH to work after an upgrade from asterisk 1.4 to
1.6.1.4. The call goes on hold, MOH is started, and then stops right away.
Here are the files both of type .raw:
Tsunami*CLI> moh show files
Class: default
File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaise
On Wednesday 16 September 2009 15:10:57 Michelle Dupuis wrote:
> On Wednesday, September 16, 2009 4:03 PM, Tilghman Lesher wrote:
> > On Wednesday 16 September 2009 14:34:05 Michelle Dupuis wrote:
> > > I'm trying to enable res_crypto on a 1.4 installation, but menuconfig
> > > says ssl is needed.
On Wed, Sep 16, 2009 at 2:37 PM, Zoa wrote:
>
> What if i send my twin brother to take the exam instead of me... ?
>
> z
>
>
If you think you cannot pass the test yourself, your twin wont be able to
pass it neither, he can be even worst than you
lol
_
My .02 - you're probably going to have to modify
build_tools/menuselect-deps. Tilghman would know this answer better than
me.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Wednesday, Sep
No such package (under fedora 9)...
Should I be lookin in other repo's or do you know what inside that package
it wants? (In case fedora packages it somewhere else)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
On Wed, 16 Sep 2009, Anahi Ludue?a wrote:
Hi People, I want to do the following steps:
- Create a meetme between 2 persons.
- First, 1 person (user1) is entered into the meetme.
- Second, user2 is entered into the meetme. User2 is the marked user and
also he is able to exit the conference by
On Wednesday 16 September 2009 14:34:05 Michelle Dupuis wrote:
> I'm trying to enable res_crypto on a 1.4 installation, but menuconfig says
> ssl is needed. I've installed openssl, openssl-devel, openssl-perl
> but it's still not happy.
>
> Anyone know what else is needed?
Try libopenssl-devel
-
I'm trying to enable res_crypto on a 1.4 installation, but menuconfig says
ssl is needed. I've installed openssl, openssl-devel, openssl-perl
but it's still not happy.
Anyone know what else is needed?
___
-- Bandwidth and Colocation Provided by http://
Hi People, I want to do the following steps:
- Create a meetme between 2 persons.
- First, 1 person (user1) is entered into the meetme.
- Second, user2 is entered into the meetme. User2 is the marked user and also
he is able to exit the conference by pressing #.
- If user2 exited by pressing
On Wed, 16 Sep 2009, Danny Nicholas wrote:
> I'd try this:
> - exten => 4000,1,Dial(SIP/4000,20,ikKtT)
> - exten => s-NOANSWER,1,Dial(SIP/4001,20,ikKtT)
> - exten => s-NOANSWER,2,Voicemail(4000)
> - exten => s-BUSY,1,Dial(SIP/4001,20,iKkTt)
> - exten => s-BUSY,2,Voicemail(4000)
> - exten => h,1,ha
What if i send my twin brother to take the exam instead of me... ?
z
C. Savinovich wrote:
>
> What about if I use the browser from my cellular phone?
>
>
>
> CS
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pas
thank you tilghman...that did the trick.. thanks again!
Tilghman Lesher wrote:
> On Wednesday 16 September 2009 12:38:59 Ron wrote:
>> yup my scripts starts with that line, is there anyway to check on the
>> logs if asterisk voicemail app is executing that command? thanks
>
> Okay, next sanity ch
C. Savinovich wrote:
> What about if I use the browser from my cellular phone?
>
>
>
> CS
>
>
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno
> Sent: Wednesday, September 16, 2009 10:21 PM
> To: Asterisk Users Ma
On Wed, 2009-09-16 at 13:14 +1200, Neeraj Chand wrote:
> Hmm...so by open book, that means access to the internet? Possible to
> get own notes ?
You get access to voip-info.org and searching Google to use as a
reference. We don't allow copying/pasting of config files, or copying
files via the in
On Wed, 2009-09-16 at 13:28 -0400, Steve Totaro wrote:
> Just tunnel your HTTP traffic over an SSH link and go to some dCAP
> brain dump sites.
Yes, there are all kinds of technical ways of trying to cover your
tracks... I've certainly seen a number of them.
That being said, it's pretty ea
On Wednesday 16 September 2009 12:38:59 Ron wrote:
> yup my scripts starts with that line, is there anyway to check on the
> logs if asterisk voicemail app is executing that command? thanks
Okay, next sanity check is that your script is chmod 755 (executable).
--
Tilghman Lesher
Digium, Inc. | S
On Thu, 2009-09-17 at 14:00 +0430, C. Savinovich wrote:
> What about if I use the browser from my cellular phone?
Sorry, cell phone use is not permitted during the testing. We've had
students try to snap pictures of the exam with their cell phone cameras,
so we had to institute a policy against c
What about if I use the browser from my cellular phone?
CS
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno
Sent: Wednesday, September 16, 2009 10:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subjec
I believe the administrator can see what is on your screen with screen with
those screen sharing stuff, this makes it harder a lil bit, and
www.boratproxy.com becomes useless in that case.
On Wed, Sep 16, 2009 at 1:28 PM, Steve Totaro <
stot...@totarotechnologies.com> wrote:
>
>
> On Wed, Sep 16,
Change /usr/bin/sendmail to /usr/bin/sh in voicemail.conf. That will
disable the function in voicemail.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron
Sent: Wednesday, September 16, 2009 12:45 PM
To: Aster
Hi Danny,
if the voicemail function is called then the AGI, wont the vooicemail
function already send an e-mail before going to the AGI? Thanks!
Regards
Ron
Danny Nicholas wrote:
> The ODBC isn't having an effect, otherwise you couldn't run it stand-alone.
> Voicemail.conf states that changing
Hi Tilghman,
yup my scripts starts with that line, is there anyway to check on the
logs if asterisk voicemail app is executing that command? thanks
Regards
Ron
Tilghman Lesher wrote:
> On Wednesday 16 September 2009 11:35:31 Ron wrote:
>> Hi All,
>>
>> I'm trying to use a php script to send voi
On Wed, Sep 16, 2009 at 11:26 AM, Tilghman Lesher wrote:
> On Tuesday 15 September 2009 20:14:32 Neeraj Chand wrote:
> > Hmm...so by open book, that means access to the internet? Possible to
> > get own notes ?
>
> Yes, you have access to the Internet, but your access is proxied, and the
> adminis
What do you want your message to say? I'd just use busy-pls-hold and the
caller would eventually get the idea that you weren't going to talk to them.
You could also consider these
Off-duty
Not-auth-pstn
Not-taking-your-call
Number-not-answering
-Original Message-
From: asterisk-users-bou
On Wednesday 16 September 2009 11:35:31 Ron wrote:
> Hi All,
>
> I'm trying to use a php script to send voicemail e-mail so i can send
> custom e-mail message based on what mailbox.
>
> on my voicemail.conf i have
>
> mailcmd=/var/www/voicemail.php
>
> but when i tried to call an extension and goe
No me encuentro en la oficina,
Volvere el proximo 28 de Septiembre.
Utilice los siguientes contactos:
v...@mildmac.es
rafael.mara...@mildmac.es
edua...@mildmac.es
Muchas Gracias
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com -
The ODBC isn't having an effect, otherwise you couldn't run it stand-alone.
Voicemail.conf states that changing the /usr/bin/sendmail -t is done at your
own risk. You could just do a system or AGI command to run your PHP script
whenever the voicemail application is called, like this:
- exten => s
Hi All,
I'm trying to use a php script to send voicemail e-mail so i can send
custom e-mail message based on what mailbox.
on my voicemail.conf i have
mailcmd=/var/www/voicemail.php
but when i tried to call an extension and goe to voicemail i'm not
receiving the e-mail.
but when i execute "p
Does any have or can point me to /ACR/ Anonymous Call Rejection message
I can download? The one I found was not not too clear.
Thanks, Bart
<>___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Ph
You can pass variables in the Originate Action, see
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate.
Taken from there:
*Variable*: Channels variables to set (max 32). Variables will be set
for both channels (local and connected).
Example(Placing a call from a S
Thanks,
I asked you to execute the GoSub from the Originate action, because I need to
pass some parameters.
First, I created a macro since I could pass the parameters from originate. But
the macro's problem is it doesn't jump to the particular extension (for
example: h extension). So, when you
On Wednesday 16 September 2009 06:46:13 Khaled W Chehab wrote:
> What g729 module should I download ?
You should download only the licensed g.729 module from Digium, after paying a
$10 license per concurrent user. All other modules have various problems (GPL
violation or lack of paying the assoc
On Tuesday 15 September 2009 20:14:32 Neeraj Chand wrote:
> Hmm...so by open book, that means access to the internet? Possible to
> get own notes ?
Yes, you have access to the Internet, but your access is proxied, and the
administrator of the test can see everything that you access. So it's best
it should look something like
exten => 4000,1,Dial(SIP/4000,30,t)
exten => 4000,2,Goto(4001,1)
exten => 4001,1,Dial(SIP/4001,30,t)
If 4000,1 is answered it will never reach 4000,2
if 4000 is busy or not available for another reason it wil goto 4001,1
hope this is useful
Erik de Wild
Tripple-
Hi,
The GoSub() application is intended for use in the dialplan, not to call
it from a Originate Action. What is your specific need? You can
Originate to a extension instead of an application an then if you need
to execute a subroutine, you can use GoSub() and Return() then you need
to on the
It works, thanks a lot, I also change the character for comments.
I am familiar with that page, I had been looking for the information in that
page also in google but noting.
Thanks to all for your help on this, let me continue doing some tests to
complete the task to do.
Best regards
Joh
Olivier escribió:
2009/9/16 Danny Nicholas mailto:da...@debsinc.com>>
Core show channels offers this on releases of asterisk that
use/display bridging (1.4.26 does bridging but does not show
bridged in status).
1. Here is an example
> core show channels
Channel Locat
Thanks Miguel, It was my mistake.
So, my question is:
if I want to call the GoSub application from
the Originate Action (using AMI), what I need to put in the context
parameter? The GoSub will jump to a special context.
Thanks,
Date: Wed, 16 Sep 2009 09:34:31 -0500
From: mmol...@millenium.c
Hi,
I didn't notice on my first answer, but we are on the -dev list and this
is not related to asterisk code developing. I will answer you on the
-users list, so we can continue the discussion there.
Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
Anahi Ludueña esc
Core show channels only works when a call is active; here's an example:
Channel Location State Application(Data)
DAHDI/2-1(None) Up AppDial((Outgoing Line))
SIP/104-08461bd0 1-d...@macro-trunkdi Up
Dial(DAHDI/R1/w2975000|20|kKtT
2 a
Hi Juan,
1. Please use the semicolon (;) character to comment your dialplan. Your
choice (#) is intended for something else.
2. Probably you have to add the "j" option of Dial application (show
application Dial), like:
exten => 4000,1,Dial(SIP/4000,20,iKkTt*j*)
exten => 4000,102,Dial(SIP/4001,20,
2009/9/16 Danny Nicholas
> Core show channels offers this on releases of asterisk that use/display
> bridging (1.4.26 does bridging but does not show bridged in status).
>
1. Here is an example
> core show channels
Channel Location State Application(Data)
0 active cha
I'd try this:
- exten => 4000,1,Dial(SIP/4000,20,ikKtT)
- exten => s-NOANSWER,1,Dial(SIP/4001,20,ikKtT)
- exten => s-NOANSWER,2,Voicemail(4000)
- exten => s-BUSY,1,Dial(SIP/4001,20,iKkTt)
- exten => s-BUSY,2,Voicemail(4000)
- exten => h,1,hangup
-Original Message-
From: asterisk-users-boun
On Tue, 2009-09-15 at 22:41 -0500, Ian Pilcher wrote:
> Running asterisk-1.6.1-0.23.rc1.fc11.i586 on Fedora 11. I can
> reproducibly crash Asterisk by associating a single voicemail mailbox
> with two SIP extensions. For example:
Please open a report on our issue tracker at http://issues.asterisk.
I comment all the lines in my extensions.conf file to work only with the
lines you provide me Danny:
Extensions.conf
[local-sip]
#exten => _4XXX,1,Dial(SIP/${EXTEN},10,tTr)
#exten => _5XXX,1,Dial(Dahdi/1/${EXTEN})
#exten => 164,1,Dial(Dahdi/1/${EXTEN})
#exten => 0550,1,Dial(Dahdi/1/${EXTEN})
#ex
Hello,
ViciDial has IVR logging(pre-Queue) of IVRs set up through our web
interface(we call them Call Menus), but ViciDial does not use Asterisk
queues at all and it's logging is done entirely in a MySQL database. As a
side note, the logging done by ViciDial (non-IVR of course) is also fully
compa
In regular configuration (extensions.conf) this is one way to do it:
- exten => 4000,1,Dial(SIP/4000,20,iKkTt)
- exten => 4000,s-BUSY,Dial(SIP/4001,20,iKkTt)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan
I have been checking but nothing that clear my idea...
I have the extension 4000 and the idea is when this extension receive a call
and the extension 4000 is busy, the call from PSTN could be send to a second
extension, example: 4001, this need to happen only if the first extension is
busy.
If no
Core show channels offers this on releases of asterisk that use/display
bridging (1.4.26 does bridging but does not show bridged in status).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Tuesday, September 15
Hello Team,
IVR selection of QUEUEMETRICS
As we know queuemetrics had an IVR selection functionality where it can get the
IVR keypress of a caller.
We saw this link
http://forum.queuemetrics.com/index.php?action=printpage;topic=503.0
and upon checking, its only determined the Queue, I want to
I have problemin g729 codec compatibility,I get the g729 module from
http://asterisk.hosting.lv/ and I have Asterisk 1.4.22-3 RPM
What g729 module should I download ?
I already downloaded
codec_g723-ast14-icc-glibc-pentium4.so
[trixbox1.localdomain asterisk]# cat /proc/cpuinfo
proc
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