Displaying 20 results from an estimated 1000 matches similar to: "Astricon talk on wideband codecs"
2004 Sep 17
9
Asterisk forum created
I saw several threads requesting an Asterisk forum to complement the
email list. i.e.
http://lists.digium.com/pipermail/asterisk-dev/2004-February/003103.html
I recently created an Asterisk forum within TMC's popular VoIP forums
for everyone to use.
http://voip-forum.tmcnet.com/voip-forum/forum/forum_topics.asp?FID=15
2009 Oct 29
1
Astreicon presentations
Hi Folks,
Are all the astricon presentations up?
I'm especially after the one that tilghman did. I caught the tail end of
the prez when I decided to skip the session I was attending and go for
that one.
:)
2008 Sep 23
5
Extension registration
Hi all,
I have the below extension defined under sip.conf:
[2203]
type=friend
username=2203
secret=123456
host=192.168.0.164
mailbox=2203
context=intern
canreinvite=yes
dtmfmode=rfc2833
When trying to register from a softphone installed on a PC behind a nat with
IP=192.168.0.164, I got 503 FOrbidden...Does anyone have any idea about what
could be the issue?
Regards
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2003 Aug 12
1
New-ish list of hardware phone vendors
This is a fairly large list of hardware phones, features, and URLs,
with some vendors I've never heard of before on it. I suspect that
any phone that doesn't explicitly say "SIP" is an H.323 phone.
http://www.tmcnet.com/it/0403/0403sg.htm#sidebar2
JT
2004 Jul 08
1
Asterisk receives TMC Labs Internet Telephony Innovation Award
Asterisk receives TMC Labs Internet Telephony Innovation Award
http://www.tmcnet.com/it/0704/tmclabs.htm
Jim
James H. Thompson
jht@lava.net
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2003 May 23
5
Who would use Asterisk SS7?
Hello. I am new to this forum. If this issue has been discussed already,
please inform me, but my STFW have indicated that it has not.
If SS7 were available on * would it make commercial sense to use it? There
are case studies on the web (courtesy of the Death Star :->) for ISPs with
10,000+ subscriber lines showing clear cost savings in transitioning from PRI
to SS7 controlled IMT. My
2010 Oct 23
2
Just Take dCAP at Astricon?
Since it is Saturday evening (7PM EST) I am asking this on the list in case
someone who knows sees it and can answer.
Astricon is in my back yard for the first time, and I could hit you with a
rock. I would always like to attend, and spoke at the 2007 Astricon in
Phoenix but don't have the idle cycles.
Question: Can I just go to Astricon and take the dCAP exam only? In and
out? Cost?
I
2008 Dec 22
1
Voicepulse down
Starting around 10:00 AM EST.
All services from them whether I connect by IP or DNS (both east coast
and west). Anyone else?
Fred Posner
fred at teamforrest.com
Main: +1 (212) 937-7844
Direct: +1 (503) 914-0999
www.teamforrest.com
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2004 Mar 17
1
Asterisk in the news
http://www.tmcnet.com/tmcnet/articles/2004/031704rt.htm
Previous article by same author:
http://www.tmcnet.com/it/0104/0104PO.htm
2006 Jun 16
2
MOS Scores and LCR
Is there any tool that can do LCR for Asterisk but also take into
account MOS scores?
Is it possible to automatically generate MOS scores on random "calls"
so as to keep an updated database on a per provider, per destination,
per time-of-day score? Hopefully, with that information we can create
a better LCR module or script?
Thanks,
Daniel
2010 Apr 10
10
Being attacked by an Amazon EC2 ...
Just a "heads-up" ... my home asterisk server is being flooded by someone
from IP 184.73.17.150 which is an Amazon EC2 instance by the looks of it -
they're trying to send SIP subscribes to one account - and they're
flooding the requests in - it's averaging some 600Kbits/sec of incoming
UDP data or about 200 a second )-:
This is much worse than anything else I've
2008 Jun 29
1
Timeout between digits for fxs station
Hi All;
How to increase the waiting time between entering the digits for the analoge phone device that is connected to fxs?
Is it by DigitTimeout? But how it will be apply for analoge station if the user just pickup the handset and dialed the number?
Any help?
Regards
Bilal
2005 Mar 01
2
Important :: Please support the development of a new Jitterbuffer for SIP
Steve Kann has developed a new jitterbuffer for IAX2, that hopefully
will be integrated into Asterisk v1.1 soon, to be part of the 1.2 stable
relase.
Zoa and his bulgarian team is porting this buffer to SIP/RTP, but needs
support in the form of funding in order to take the time to test this
out and complete it in time.
Please paypal your contribution to sponsor@astertest.com today. Every
2003 Nov 29
14
* Party in Paris
I'm coming to Paris Dec 19. I was wondering if there was any interest in
having an Asterisk get together in Paris sometime near there. Any one out
there interested? Anyone in Paris who could help organize something like
that? :)
Mark
2007 Jan 16
3
IAX Trunk timing
I have read that an IAX trunk requires a timing device. What wasn't
clear to me was whether it is like TDM ie 1 timing device for the trunk,
or if each end requires a timing device. I have a zaptel card in one
server; do I have to have one in the second server in order to do an IAX
trunk?
I set up a trunk and so far calls can be made one way, but not the
other. It is probably just not
2009 Nov 11
0
AstriCon Videos and Presentations: First batch is on-line!
This year we recorded quite a few of the AstriCon sessions - 3 out of
the 4 tracks were video taped. The folks at TMC then went through a
fairly painstaking process of synchronizing the video presentations to
the slide decks that each presenter provided, so we have an index-able
and fast-forward-able version of each talk. I'm really excited about
this video presentation method,
2005 Sep 16
1
New version of idefisk softphone released.
We just uploaded the latest and greatest version of the idefisk iax2
softphone, version 1.24
Freely downloadable at: http://www.asteriskguru.com/tools/idefisk_beta.php
Changes since the last release include:
- history panel is working
- receiving messages and urls (sendtext command in asterisk)
- some bugfixes (the annoying hangup bug is finally gone!).
A big thanks to everybody who sent us
2008 Jul 16
4
asterisk + web services
List,
We're working on an upcoming job that may require us to access a web
service (WS). I'm curious to hear peoples thoughts on the best way to
do this with asterisk. We'll be submitting a single number to the WS
and it will return a success or error.
One solution would be to write a simple perl script to interface into
to the WS, and use SYSTEM() from asterisk to call it.
2007 Jan 08
2
G729 license counting
Hello,
How many licenses to buy?? :
From what we understood from digium website, we must buy as many
licenses as the number of maximum simultaneous calls using G729 Codec we
wish to make.
For example, If we want to be able to make a maximum of 10 simultaneous
calls using G729 Codec, we must buy 10 licenses.
Is it right?
Thanks you
2006 Mar 29
6
Asterisk with Vonage
I know Vonage doesn't officially have a "bring your own device" type
program, but they do offer a softphone. Has anyone gotten Asterisk to
connect directly to Vonage? This would be a great help!!
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