Displaying 20 results from an estimated 2000 matches similar to: "iax2 trunking known problems?"
2006 Jan 20
5
iDEFISK (mac iax2 softphone) release
]
Hey ho,
A few days ago we released the linux version of the phone, today we are
very happy to have the mac version ready for a little field test.
Freely downloadable from http://www.asteriskguru.com/tools/idefisk_mac.php
At the same time, we also put a newer version of the windows and linux
versions online.
Let us know how you feel about it, a more mac look (brushed metal) is
coming.
2005 Sep 20
4
SUCCESS - 512 Simultaneous Calls with Digital Recording
List users,
Over the last few days we have been working with MCI's development lab
to test our Asterisk setup. We were using a piece of hardware called an
Abacus 5000 that is capable of creating and terminating thousands of SIP
calls. Initially, we could not get past 64 simultaneous digitally
recorded calls without having call quality issues including dropped
calls. We identified an
2006 Feb 03
2
g729 license question
I am wondering how the g729 licenses are done during calls. If I have N
licenses for g729, and N are in use and an additional call comes in that
requests N+1 to be in use, how does asterisk handle that call?
Does it dump it? Does it negotiate another codec automagically?
Basically what happens to that call, obviously it wont (shouldnt) let
you use more licenses than you have available, but
2006 Jan 29
2
Access Codes
Or you can use authenticate() and have it take its 'passwords' form a
text file on your machine.
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
> trixter aka Bret McDanel
> Sent: Sunday, January 29, 2006 5:37 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
2006 Feb 14
2
audio cuts out
has anyone experienced a problem where RTP audio cuts out when doing
30-40 concurrent channels via sip?
The box is freebsd 6, asterisk 1.2.4, voip only (no libpri, no zaptel -
not even a timing source)
The box has plenty of bandwidth, when a call to the same box is iax2 it
works, but when its sip a call gets connected a few frames of audio are
passed and then silence.
When the box is completly
2009 Oct 20
3
High Volume Call Center SIP versus IAX2
I wont say we are extremely high volume (40 concurrent calls) but I get
occasional complaints about quality.
Setup (at same location):
Asterisk 1.4.26.2 FrontEnd
Asterisk 1.4.26.2 Gateway with Sangoma A108D card with 2 PRI and LDT1
Connected via IAX2 trunking on its own VLAN
Is IAX2 the way to go or would SIP trunking be better.
I know its a pretty vague question but I am just trying to
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
2006 Jun 02
20
Prices of g729 codec
Hi, does anyone know the prices for g729 codecs from Digium? I sent an
email a while ago to them but haven't got any response so far.
Prices are per unit or volume?
Thanks,
--
-------------------------------------------
Erick Perez
Linux User 376588
http://counter.li.org/ (Get counted!!!)
Panama, Republic of Panama
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
2005 Sep 19
2
kill a .call file
Any means of killing a .call file that is in progress?
James Taylor
MetroTel
3505 Summerhill Road
Suite 11
Texarkana, Tx 75503
903-793-1956
2004 Jun 17
7
TDMoE Question
Just a Question. I would like to know if TDMoE follows specifiaciones of
TDMoIP RAD protocol that says that there is a compression of 16/1 when
you do TDMoIP.
Manuel Marin Garcia
TRANSTELCO S.A. DE C.V.
Campos Eliseos 9050 B4 – Cd. Juárez, Chih. 32452 - México
Oficina: +52 656 692 11 09 – Fax: +52 656 692 1112 - Celular: 915 727
6141
http://www.transtelco.com.mx
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
2009 Oct 19
2
Astricon talk on wideband codecs
I missed the talk that was given on wideband codecs @ astricon last week.
I tried to lookup the speaker on astricon.net, but that website seems
horribly broken at the moment, showing only a tmcnet video, whatever
page i click on.
Would somebody have the contact details for that speaker ?
Greetings,
Zoa
2005 Jun 02
2
voip provider request
I am looking for a voip provider that provides good rates (below 5
cents/min or unlimited) to UK NCFA numbers. Braodvoice advertises they
do unlimited to NCFA but does not have the ability to actually termiate
those calls as per the CTO Nathan Stratton, and last he said they dont
even have contracts in place to get service provisioned for that. As
such I am looking for another provider to take
2006 Jun 05
2
show channel issue with 1.2.9
has anyone else noticed what appears to be a threading issue in asterisk
1.2.9 (it broke sometime between 1.2.4 and 1.2.9) where if you have
about > 50 calls and do
asterisk -rx show channels
it will display the header but nothing about channels, total calls,
active calls, etc.
--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461 DE +49 801 777 555 3402
2005 Jul 10
6
iax.cc opinion request
I am considering using iax.cc (sixtel) and wondering if anyone had
opinions, good or bad. Are there outages with any regularity? How
responsive are tech support? How is packet loss? I am particularly
interested in termination to the UK, but will accept any comments people
have.
Thanks
--
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605 Germany +49 801 777 555 3402
US
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
2005 Jun 19
4
bluetooth audio and asterisk
Has anyone successfully used a standard bluetooth enabled system to
connect to a standard bluetooth enabled mobile phone (not the bluetooth
to FXS converters) to create an audio path for phone calls with
asterisk, if so is there a writeup on what was done so that others can
replicate this.
What I am thinking is that via alsa/oss/whatever you should be able to
use the bluetooth audio channel as a