Search jitter in sample sip.conf. Everything is well documented there.
Regards,
Qasim
On Tue, Apr 23, 2013 at 3:03 PM, Muhammad Yousuf muyous...@gmail.comwrote:
I am using asterisk as SIP/GSM gateway. I have 2 gsm cards installed in
server. I am having some issue in audio quality. I want
Hello;
How I can compare between Asterisk 1.8 and 11 with reference to the following
points:
1) SMS.
2) gtalk and other social media.
3) GUI.
4) Any main difference?
Regards
Bilal
--
_
-- Bandwidth and Colocation Provided by
We're running asterisk 1.8 in the DC on a public IP address.
Connecting to it are about 200 phones behind a LAN in a remote location.
Is there a way to reliably keep asterisk out of the media stream on internal
calls inside that LAN? All phones are Polycom Soundpoint phones.
Asterisk would say
Read up on new features and changelog of asterisk 11 you'll find the
changes there.
Regards,
Qasim
On Thu, Apr 25, 2013 at 3:32 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hello;
How I can compare between Asterisk 1.8 and 11 with reference to the
following points:
1) SMS.
2) gtalk and
You will want to look at the directmedia option. You will want all the
phones on the same lan as the Asterisk server to be directmedia=yes and
the ones on the wan to be directmedia=no. Then, internal calls will send
the media between themselves without involving Asterisk, but ones outside
on
Hello all,
I am looking into building a calendar server (due to business
requierments I can not use public hosted calender like Google), and am
looking for suggestions based on experience with different calendar
applications/servers available for Linux that you have integrated with
Asterisk.
Without knowing requirements, Sugar CRM seems to be the most supported.
Thanks,
Steve Totaro
On Thu, Apr 25, 2013 at 9:22 AM, j...@millican.us j...@millican.us wrote:
Hello all,
I am looking into building a calendar server (due to business requierments
I can not use public hosted calender
Hello,
I've been given the task to study what would a good way to load balance SIP
trafic.
The prospective setup is :
- call centers sending outbound SIP trafic (no inbound) from SIP devices
(with public fixed IP address),
- a couple of outbound SIP trunks to which trafic from call centers is to
Kevin,
Thanks for the info. Clarification. The asterisk server is NOT on the same LAN
as the phones. The asterisk server is in a datacenter only accessible via WAN.
However, all of the phones are in side of the same LAN. Will directmedia still
function that way?
Thanks
David
From: Kevin
David,
you obviously have to test for your situation, but the short answer is
that it should. The connection will start with running through Asterisk,
but very quickly the phones will see that they can talk directly and take
the Asterisk server out of the media path. There are a couple of
You the couple opensips + asterisk will help you. Opensips loadbalance module
is your friend.
Sent from my iPhone
On Apr 25, 2013, at 11:44 AM, Olivier oza_4...@yahoo.fr wrote:
Hello,
I've been given the task to study what would a good way to load balance SIP
trafic.
The prospective
flavor? i do not understand what you mean. please explain more.
thanks
On Wed, Apr 24, 2013 at 8:16 PM, Asghar Mohammad asghar...@gmail.comwrote:
what flavor of h323 you are using?
On Wed, Apr 24, 2013 at 8:50 AM, s m sam.gh1...@gmail.com wrote:
thanks Asghar,
i do it, but no thing
nuFone h323 or openh323?
On Thu, Apr 25, 2013 at 9:33 PM, s m sam.gh1...@gmail.com wrote:
flavor? i do not understand what you mean. please explain more.
thanks
On Wed, Apr 24, 2013 at 8:16 PM, Asghar Mohammad asghar...@gmail.comwrote:
what flavor of h323 you are using?
On Wed, Apr
Might have a look at tine:
http://www.tine20.org/wiki/index.php/Admins/Asterisk_integration
hw
-Original Message-
From: Steve Totaro stot...@totarotechnologies.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing
We have an automated greeting on our Asterisk phone system, that like many,
has the phrase if you know your party's extension, you may dial it at any
time.
Today, I encountered a user dialing in from the outside, attempting to dial
an extension, but he was appending the # key at the end.
This,
Hello,
My health care organization is looking for a way to do appointment
reminders. We currently have staff members who spend part of each day
manually calling patients to remind them of their upcoming appointments,
and we would like to automate this process.
Our electronic health record
16 matches
Mail list logo
