Hello,
can you use Progress() in the dialplan for outgoing calls ? For example
just before the Dial()-command ?
Is there a risk involved when using the Progress()-command ?
Kind regards,
Jonas.
--
_
-- Bandwidth and
Hello Everyone,
We are looking for solutions where the transcoding is abstracted away
from our * box (i.e., to the network layer) using some carrier grade
gateway, or router.
The reason for my post is to know about solutions people have used in
the past, and how it fits into their overall
Sangoma has transcoding cards which support access over PCIe or Ethernet.
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
[sym...@gmail.com]
Sent: Friday, April 12, 2013 9:01 AM
To:
Did you already look at transcoding cards?
E.g.: http://www.sangoma.com/media-processing/voice-transcoding-boards/
They also have separate boxes
(http://www.sangoma.com/products/netborder-transcoding-appliance/).
Personally, I prefer to have everything in a single box if there aren't
too many
Hello Gentlemen,
Thank you so much for your response, we have adopted transcoding cards
in our old system, and they do have some limitations, especially when
it comes to concurrent calls. We were looking more into the lines of a
scalable multi server router like a cisco 3745. And loading it with
On 12/4/13 4:38 pm, Nick Khamis wrote:
We were looking more into the lines of a
scalable multi server router like a cisco 3745.
Perhaps it might help to tell the list just how many concurrent calls
you're looking to transcode?
Kind regards,
Chris
--
This email is made from 100% recycled
What do you mean with servers? A simple proxy, or a B2BUA (aka
Asterisk)? Depending on the basic configuration the server might or
might not have to deal with some or all of the RTP-streams.
The already mentioned company Sangoma usually has good documentation
about their products (see
Basic Dial Plan
Why is this plan not engaging the line
exten = 105,n,Dial(SIP/voipvoip.com/1703501)
and dialing the 703 number?
The logs and debug dont show any problems
[incoming]
exten = 44,1,Answer()
exten = 44,n,Wait(1)
exten = 44,n,Playback(beep)
exten =
On Friday 12 April 2013, Thomas Perron wrote:
Basic Dial Plan
Why is this plan not engaging the line
exten = 105,n,Dial(SIP/voipvoip.com/1703501)
and dialing the 703 number?
The logs and debug dont show any problems
[incoming]
exten = 44,1,Answer()
exten =
Sorry for the missing info. Our current architecture is as such:
NAT - SIP/RTP Proxy - *(n)
Our concurrent sessions usually peak at between 700-800 channels. On
average about 450. I will of course look at the documentation to
better understand how a transcoding appliance would fit in our
Thank you!
On Thu, Apr 11, 2013 at 11:47 AM, Asghar Mohammad asghar...@gmail.comwrote:
hi,
it is not difficult in php and mysql i have created a simple billing
system for my wholesale postpay clients without any AGI.
it report ACD ASR all calls ANSWERD calls filter by date by callerid etc.
Hello all,
I need the bootrom.ld file to set up some Polycoms I have
Platform: Model=SoundPoint IP 330, Assembly=2345-12200-001 Rev=A
I've publiched on my FTP files downloaded from
http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html
(3.2.3
http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
From: Daniel - Asterisk earohua...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello Kevin,
Could you please tell me where I can found the 'application' my phones are
looking for?
I've already downloaded spip_ssip_vvx_3_2_3_release_sig combined and split
zips, which lack a bootrom.ld file
Thank you!
Elder
On Fri, Apr 12, 2013 at 12:44 PM, Kevin Larsen
This can be useful too: Application, main: Label=BOOT, Version=3.2.3.0021
29-Mar-07 16:05
It is the log the phones are sending to my FTP
Thanks!
Elder
On Fri, Apr 12, 2013 at 12:50 PM, Daniel - Asterisk earohua...@gmail.comwrote:
Hello Kevin,
Could you please tell me where I can found the
Daniel,
The bootom is not part of the SIP application that you downloaded.
You need to download the appropriate bootrom from the link Kevin
supplied. Before you do any more though, you really need to download the
SoundpointIP Admin guide here:
- Original Message -
From: Jonas Kellens jonas.kell...@telenet.be
can you use Progress() in the dialplan for outgoing calls ? For
example just before the Dial()-command ?
IIRC the application description is very accurate:
This application will request that in-band progress
For anyone else that may be interested in the future, I found a
detailed depiction here:
http://wiki.sangoma.com/ntg-theory-of-operation
Thanks again,
N.
On 4/12/13, Nick Khamis sym...@gmail.com wrote:
Sorry for the missing info. Our current architecture is as such:
NAT - SIP/RTP Proxy -
Thank you both,
Finally I found proper bootrom at:
http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/previous_voip_software.html
As my very version is not available I choose: SoundPoint IP / SoundStation
IP BootROM 3.2.3 Rev B which has been downloaded from my FTP
We are researchers from different parts of the world and conducted a study on
the world’s biggest bogus computer science conference WORLDCOMP
( http://sites.google.com/site/worlddump1 ) organized by Prof. Hamid Arabnia
from University of Georgia, USA.
We submitted a fake paper to WORLDCOMP
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