Hi,
A quick look at Digium's Fax for Asterisk datasheet shows a maximum 14.4
kb/s speed which is fine, but I'm wondering if it's possible to achieve
reliable 33.6 kb/s faxing (with a Digium board based Asterisk system) ?
Regards
--
Thanks. I'll look into it.
On Wed, Sep 7, 2011 at 8:54 AM, Sam Govind govoi...@gmail.com wrote:
See this link:
http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL
You'll find similar pages where you can setup to store queue logs/events(as
Alex mentioned) in MySQL DB and further
sometimes when I call the phone is silent
The problem there 'even on calls between internal
Asterisk version 1.8.5
codecs used in the sip-trunk are:
disallow = all
allow = ulaw
allow = alaw
allow = gsm
and users.conf:
allow = ulaw
allow = alaw
allow = gsm
This configuration worked
See absolute timeout. I think yours' a complex thing to achieve I guess
absolute timeout may be the thing that can help. In older versions
absoluteTimeoute(n) could take you to exten T when time n elapsed. now I
guess funtion Timeout() is used as replacement.
here's an excerpt from somewhere:
;
6 sep 2011 kl. 22:30 skrev Leif Madsen:
On 02/09/11 11:27 PM, Joseph wrote:
In asterisk 1.4 I had:
exten = s,n,Answer()
exten = s,n,SetMusicOnHold(default)
But in 1.6 1.8 I think don't need to use: SetMusicOnHold(default)
(beside it is deprecated) as it is default.
In 1.6 and UP I think
On 09/07/2011 02:17 AM, A Dunor wrote:
Hello list, I am a beginner at asterisk. I want to access my asterisk
box from my laptop, on a different network (mobile hotspot). The
asterisk box doesn't have a static ip, how do I connect with it using
ssh or other such programs?
Thanks for your
On Wednesday 07 September 2011, A Dunor wrote:
Hello list. Just another beginner question. I am trying to setup a basic
home phone system. I ordered a TDM410 card, which came with 4 fxo ports.
I want the home phone system to be able to initiate and receive calls.
Can it be done with this card
Hi list,
I want to know that will it be possible that more then 1 AMI is connected
from single Linux machine with different name ?
As we know that default 1st AMI connection will come with 127.0.0.1 and root
information.
My requirement is that I want to handling events for more then one
Hi
This can happen you can create more than 1 AMI connection.
if you need better on access control then you can create new user in
manager.conf with set of privileges that you can offer to each of them
On Wed, Sep 7, 2011 at 15:59, virendra bhati virbh...@gmail.com wrote:
Hi list,
I want
Hi Amit,
My scenario is that, If 3 conference is running in Asterisk then I will play
a sound file with the help of Asterisk AMI then I will get DTMF from all the
users. the same things will be done any all the Konference and all
conference will be play different files.
If you have any alternate
It seems to me that you are overworking AMI to do what could be done with
AGI. You could use an AGI to poll Konference and return a dialplan variable
with the file to use in Playback/Background or even MOH.
From: asterisk-users-boun...@lists.digium.com
I couldn't find the extconfig.conf file in /etc/asterisk and queue_log
doesn't exist either (as a file or as a db table). We're using AsteriskNOW,
so maybe these files/tables were not created.
How should we add them?
Thanks.
On Wed, Sep 7, 2011 at 8:54 AM, Sam Govind govoi...@gmail.com wrote:
Looking at the history of the list I don't expect any answer but lets
try anyway:
Does anybody use overlap dialing from SIP devices to asterisk? Does
anybody have a working example?
--
Daniel Tryba
--
_
-- Bandwidth and
On Tue, Sep 6, 2011 at 10:36 PM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
However you could select/deselect modules using menuselect if you wanted to
automate the process. It's documented over here:
http://ofps.oreilly.com/titles/9780596517342/asterisk-Install.html#Installing_id293439
7 sep 2011 kl. 15:59 skrev Daniel Tryba:
Looking at the history of the list I don't expect any answer but lets
try anyway:
Does anybody use overlap dialing from SIP devices to asterisk? Does
anybody have a working example?
To add to your question: Does anyone have a phone that supports
You have plenty of ways to do this. You can use the room number + user number
to get the conference number. You can use the channel ids to keep a table of
conference members and their statuses.
C. Savinovich
On September 7, 2011 at 9:15 AM Danny Nicholas da...@debsinc.com wrote:
It
On Wednesday, September 7, 2011, Olle E. Johansson wrote:
7 sep 2011 kl. 15:59 skrev Daniel Tryba:
Looking at the history of the list I don't expect any answer but lets
try anyway:
Does anybody use overlap dialing from SIP devices to asterisk? Does
anybody have a working example?
7 sep 2011 kl. 16:20 skrev Andrew Latham:
On Wednesday, September 7, 2011, Olle E. Johansson wrote:
7 sep 2011 kl. 15:59 skrev Daniel Tryba:
Looking at the history of the list I don't expect any answer but lets
try anyway:
Does anybody use overlap dialing from SIP devices to
On Wed, Sep 7, 2011 at 10:28 AM, Olle E. Johansson o...@edvina.net wrote:
7 sep 2011 kl. 16:20 skrev Andrew Latham:
On Wednesday, September 7, 2011, Olle E. Johansson wrote:
7 sep 2011 kl. 15:59 skrev Daniel Tryba:
Looking at the history of the list I don't expect any answer but lets
On Wed, Sep 07, 2011 at 10:02:02AM -0400, David Backeberg wrote:
On Tue, Sep 6, 2011 at 10:36 PM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
However you could select/deselect modules using menuselect if you wanted to
automate the process. It's documented over here:
On Wed, Sep 07, 2011 at 10:20:40AM -0400, Andrew Latham wrote:
To add to your question: Does anyone have a phone that supports this
properly?
Yup, I have a few... http://wiki.snom.com/Settings/overlap_dialing
I'm using a Grandstream (GXP2000) to test.
What I got so far:
Overlap works
On Friday, September 9th, 2011, the Asterisk issue tracker will be
undergoing maintenance (re-indexing to resolve problems with a small
number of open issues).
The issue tracker will be shut down at approximately 03:00 GMT
(Thursday, September 8th, 2011, 22:00 CDT, -0500 GMT).
The
Hi all
i have a very weird problem with curl and utf8 characters
i'm trying to do a cnam lookup from a web-service with curl if the returned
info is English or digits then the callerid name field gets populated with
that but if the returned info is utf8 like Hebrew then the callerid field
remains
On Thu, Sep 8, 2011 at 1:47 AM, Israel Gottlieb isr...@gmail.com wrote:
Hi all
i have a very weird problem with curl and utf8 characters
i'm trying to do a cnam lookup from a web-service with curl if the returned
info is English or digits then the callerid name field gets populated with
you definitely need to create the file extconfig - take sample from
internet. the DB tables need to be created on your own, take help from
internet pages.
On Wed, Sep 7, 2011 at 6:19 PM, Michael voip.quest...@gmail.com wrote:
I couldn't find the extconfig.conf file in /etc/asterisk and
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