Hi
I am new to asterisk 1.4 can someone tell about how to enable the video
conference in asterisk-gui 2.0.
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videosupport=yes in sip.conf
Regards,
Zohair Raza
On Mon, Apr 9, 2012 at 12:22 PM, p070075 Muhammad Atif Ramzan
p070...@nu.edu.pk wrote:
Hi
I am new to asterisk 1.4 can someone tell about how to enable the video
conference in asterisk-gui 2.0.
--
Actually i want to know that how i configure the asterisk for video
confernce
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On 9/4/12 3:04 am, Takehiro Matsushima wrote:
// I don't know what's difference t and T.
T allows the caller to transfer. t allows the called user to transfer.
You very rarely want Tt - since I doubt you want an incoming caller to
be able to transfer their call all over the place. You
Hi,
Actually asterisk don't provide video conference. In simple terms, the
setting which zohair told just enables two end points to use video codecs
and establish a one-one video session for video capable phones.
For making your asterisk do a video conferencing you may need to look into
Vmukti
After some customer complaints I find myself tcpdumping, gzipping and
transferring large packagedumps over the network to be analyzed.
While this manual process isn't a long-term solution, I'm evaluating
different options. Aside from the manual thing I could see two variants:
- Dump the traffic
Thank you so much.
OK, I understood that to transfer the call t is usually used, is it right?
And I should write so in my last mail.
t and T are described with same sentences in official wiki...
Regards,
Takehiro Matsushima
2012/4/9 Chris Bagnall aster...@lists.minotaur.cc:
On 9/4/12 3:04
Hi
I want to use Call Deflection with DAHDISendCallreroutingFacility Application.
I use Asterisk:1.8.11 libpri:1.4.12 facilityenable=yes transfer=yes
my dialplan is like this:
[Call-Deflection]
exten = 66,n,Proceeding()
exten = 66,1,wait(5)
exten =
Thanks everyone. I was using the Tt flag but in the wrong place in the dial
application.
Cheers
On Mon, Apr 9, 2012 at 4:54 PM, Takehiro Matsushima
takehiro.dream...@gmail.com wrote:
Thank you so much.
OK, I understood that to transfer the call t is usually used, is it
right?
And I should
Hi
What is the best way to combine multiple SIP providers to achieve
1) Higher concurrency (for eg. 2 providers with 50 concurrent calling
limits could be combined to give a limit of 100)
2) Redundancy (use another if one is down)
I have a feeling that this will need some SIP Proxy like
On Mon, Apr 09, 2012 at 10:17:02AM -0400, Mathieu Therrien wrote:
html
head
meta http-equiv=content-type content=text/html; charset=ISO-8859-1
/head
body bgcolor=#FF text=#00
font size=+1Hi every one.nbsp; I download and complie last SVN DAHDI
and VMWI doesen't
Le 09/04/2012 13:42, Johan Wilfer a écrit :
After some customer complaints I find myself tcpdumping, gzipping and
transferring large packagedumps over the network to be analyzed.
While this manual process isn't a long-term solution, I'm evaluating
different options. Aside from the manual thing
On Mon, Apr 9, 2012 at 10:25 AM, Administrator TOOTAI ad...@tootai.netwrote:
At first, if your Asterisk is in a VM install it on the real server, it
solved us on some installations.
We've gone away from VMs altogether.
To monitor the traffic, you can use voipmonitor.org
We purchased the
On 12-04-09 05:58 AM, SamyGo wrote:
Hi,
Actually asterisk don't provide video conference. In simple terms, the
setting which zohair told just enables two end points to use video codecs
and establish a one-one video session for video capable phones.
For making your asterisk do a video
2012-04-09 20:22, Carlos Alvarez skrev:
On Mon, Apr 9, 2012 at 10:25 AM, Administrator TOOTAI
ad...@tootai.net mailto:ad...@tootai.net wrote:
At first, if your Asterisk is in a VM install it on the real
server, it solved us on some installations.
We've gone away from VMs
OpenVZ is not really virtualisation, though for some reason people insist on
throwing it into the same discursive space as Xen, VMware, HyperV, etc.
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax:
I used asterisk with some dialplan customization. It is not difficult. All
client asterisk step in the central asterisk to reach the providers. I have
a central system to monitor calls, call quality, enforce limits and route
versus the best provider. To be sure I have two asterisk servers in
I am not a programmer and I have learned so much from examples and the list.
Perhaps someone could tell me what I am doing wrong in my example below:
I am getting the caller ID and caller name from my local POTS line and I
want to add it into a sql table. I am trying with the following code but
On Mon, Apr 09, 2012 at 06:21:40PM -0400, sean darcy wrote:
I've cut and pasted from the digium fax admin manual:
exten = send,1,NoOp( SENDING FAX )
exten = send,n,Wait(6)
exten = send,n,Set(GLOBAL(FAXCOUNT)=$[ ${GLOBAL(FAXCOUNT)} + 1 ])
exten =
On Mon, 9 Apr 2012, list...@gmail.com wrote:
I am getting the caller ID and caller name from my local POTS line and I
want to add it into a sql table. I am trying with the following code
but the data never gets put into the table.
Can anyone correct my syntax and tell me what I am doing
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of list...@gmail.com
Sent: Monday, April 09, 2012 8:34 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] MYSQL INSERT QUESTION IN DIALPLAN
I am not a
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