Hy,
Does anybody knows how to show the digium addons in the freepbx GUI.
The module is available in the GUI but sadly empty!
Everything seems to be correctly installed bute the tables in the
database are totally empty.
Is there any script anywhere to fill those digium tables?
Working with
Hi List,
Is there any way by which we can remove asterisk from machine without
deleting folder manually? I did google and gets various solution by no
success. even after deleted asterisk will be there .
-
Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
--
Maybe you could think about a root command line like that :
yum uninstall asterisk
Le 10/06/2011 11:26, virendra bhati a écrit :
Hi List,
Is there any way by which we can remove asterisk from machine without
deleting folder manually? I did google and gets various solution by no
success.
Hi List,
I want to set my caller ID and name with asterisk. So that when I make
outgoing calls then destination end will see my name with number.
from asterisk end I set both the things into dialplan.
---
--
exten = _X.,n,Set(CALLERID(num)=9172341457)
exten =
What do you mean?
Did you installed from sources or distro packet?
sources: make uninstall
distro: Every distro has its own commands (yum, apt-get ecc)
Alex
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di virendra bhati
Inviato:
On 2011-06-10 07:30, virendra bhati wrote:
Hi John,
Sorry for wrong information. Actually it's J not P option in
ControlPlayBack...
http://www.voip-info.org/wiki/view/Asterisk+cmd+ControlPlayback
That page is correct for asterisk 1.4 but the feature you need is in
1.6.0 and forward.
Have
Hi,
I have 44578900 to 44578999 DID's. and I have extensions(100) for this
DID's. but problem is
callerid Extensions
44578900 100
44578901 101
44578902 102
44578902 103
44578903 104
44578905 200
44578906 275
44578907 277
44578908 354
I need
On Fri, Jun 10, 2011 at 1:48 AM, Hans Witvliet h...@a-domani.nl wrote:
On Thu, 2011-06-09 at 16:32 -0700, Steve Edwards wrote:
On Thu, 9 Jun 2011, Hans Witvliet wrote:
I went originally from a almost working machine running:
asterisk180-1.8.3.2-87.1
To a machine that continuously
On Friday 10 Jun 2011, mahesh katta wrote:
Hi,
I have 44578900 to 44578999 DID's. and I have extensions(100) for this
DID's. but problem is
callerid Extensions
44578900 100
44578901 101
44578902 102
44578902 103
44578903 104
44578905 200
44578906
On Fri, Jun 10, 2011 at 5:35 AM, mahesh katta maheshka...@flexydial.com wrote:
Hi,
I have 44578900 to 44578999 DID's. and I have extensions(100) for this
DID's. but problem is
callerid Extensions
44578900 100
44578901 101
44578902 102
44578902 103
44578903
On Fri, Jun 10, 2011 at 6:27 AM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
On Friday 10 Jun 2011, mahesh katta wrote:
Hi,
I have 44578900 to 44578999 DID's. and I have extensions(100) for this
DID's. but problem is
callerid Extensions
44578900 100
44578901 101
Hai,
Does anybody have problems with a wrong Connected Line ID with asterisk version
1.6
The following bug was for version 1.4, but I cannot make up if this bug is
still in version 1.6
http://forums.digium.com/viewtopic.php?t=7780
In version 1.8 it is possible to change the Connected Line ID,
Arjan Kroon | Mobillion wrote:
Does anybody have problems with a wrong Connected Line ID with asterisk version
1.6
As far as I know, unless you're applying patches yourself, Connected
Line ID is only available for the 1.8 series. I'm running it on 1.4
with patches.
Doug
--
Ben
Hi List,
I don't install from yum repository. I download tar file from asterisk.org
On Fri, Jun 10, 2011 at 3:03 PM, Alexandru Oniciuc
alexandru.onic...@trivenet.it wrote:
What do you mean?
Did you installed from sources or distro packet?
sources: make uninstall
distro: Every distro
Steve Totaro wrote:
For each phone, add callerid=Joe Smith1551212 no quotes in sip.conf
The problem with that solution is that station to station calls will
show the same CID and not the extension.
I'd vote for the database approach.
Doug
--
Ben Franklin quote:
Those who would
On Friday 10 Jun 2011, Steve Totaro wrote:
Why do programmers try to make solution so elegant when an entries for
each phone in sip.conf is all that is needed.
No need for mathematical formulas, AGIs, and databases. You just took
over engineering to a new level.
Because doing it your way
We have two systems one with version 1.6 and one with version 1.8
With 1.8 we don't see the problem
Unfortunately it is not possible to upgrade 1.6 to 1.8.
But are there also pathes for version 1.6
Arjan Kroon
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
-Original Message-
From: Steve Totaro stot...@asteriskhelpdesk.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 10 Jun 2011 06:30:53
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List -
On Fri, Jun 10, 2011 at 4:00 PM, Steve Totaro
stot...@asteriskhelpdesk.comwrote:
On Fri, Jun 10, 2011 at 5:35 AM, mahesh katta maheshka...@flexydial.com
wrote:
Hi,
I have 44578900 to 44578999 DID's. and I have extensions(100) for this
DID's. but problem is
callerid Extensions
On Friday 10 Jun 2011, Steve Totaro wrote:
I never understood hy people who have block of DIDs in a row choose to
make life difficult by not incrementing extensions by one, send caller
ID by prepending the common numbers and only sending four digits.
Well, to be fair, that's what most people
Arjan Kroon | Mobillion wrote:
But are there also pathes for version 1.6
The last patch available for the 1.6 series was for 1.6.0.1:
https://issues.asterisk.org/jira/browse/8824
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
On Fri, Jun 10, 2011 at 5:23 PM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:
On Friday 10 Jun 2011, Steve Totaro wrote:
I never understood hy people who have block of DIDs in a row choose to
make life difficult by not incrementing extensions by one, send caller
ID by prepending the
On Fri, Jun 10, 2011 at 5:23 PM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:
On Friday 10 Jun 2011, Steve Totaro wrote:
I never understood hy people who have block of DIDs in a row choose to
make life difficult by not incrementing extensions by one, send caller
ID by prepending the
Through the AMI how can I tell if a call is on hold or not? I am using 1.4.X
Thanks,
jerry
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar
Good morning gentlemen, is my first post in the list, now I'm starting asterisk
wanted to have your help for some questions.
Well the first function is as follow me. Here
I will demonstrate how this configuration follow me on my
extensions.conf but it is not working, and do not know why, but
Either use ExtensionState or watch for Hold/Unhold events.
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+ExtensionState
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Jerry Geis
Sent:
Many providers do not allow for caller ID name to be sent.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Friday, June 10, 2011 5:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Le 10/06/2011 08:07, Florent THOMAS a écrit :
Hy,
Does anybody knows how to show the digium addons in the freepbx GUI.
The module is available in the GUI but sadly empty!
Everything seems to be correctly installed bute the tables in the
database are totally empty.
Is there any script anywhere
On Fri, 2011-06-10 at 16:31 +0530, virendra bhati wrote:
Hi List,
I don't install from yum repository. I download tar file from
asterisk.org
On Fri, Jun 10, 2011 at 3:03 PM, Alexandru Oniciuc
alexandru.onic...@trivenet.it wrote:
What do you mean?
Did you
I have expanded the EWS calendar functionality within Asterisk 1.8 so it
is now possible to access any calendar within an Exchange 2007 or 2010
server.
I have put the changes onto the reviewboard for astrisk but currently no
one responded.
So if you use the EWS calendar functionality within
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Jerry Geis
Sent: Friday, June 10, 2011 2:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AMI question
Longtime lurker, first time poster. :)
A client of mine is in need of having Asterisk record every call that comes
in from a specific incoming route. I've added the following lines to the
sip_additional.conf file, but no recordings are showing up in the
/var/spool/asterisk/monitor/ folder.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rick Hall
Sent: Friday, June 10, 2011 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Incoming Call Recording
Longtime lurker, first
Queue not sending call to Agent
I am having an issue and i am not sure if it is a bug or a config issue. I
was originally running Asterisk 1.8.1.1 when I noticed this issue. I
upgraded to 1.8.4.2 to see if that would fix it but it didn't.
The issue is that I have a call queue and the
On Fri, 2011-06-10 at 05:52 -0400, Steve Totaro wrote:
On Fri, Jun 10, 2011 at 1:48 AM, Hans Witvliet h...@a-domani.nl wrote:
On Thu, 2011-06-09 at 16:32 -0700, Steve Edwards wrote:
On Thu, 9 Jun 2011, Hans Witvliet wrote:
I went originally from a almost working machine running:
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