Darryl Moore darryl at moores.ca writes:
I'll explain.
The g.729 compression algorithm is not protected by copyright, though
specific instances may be. It is protected by a patent.
http://www.sipro.com/G-729.html
An open source version is available here:
Ade-
Does anyone do a low profile PCIe FXO card? I just picked up an HP ProLiant
microserver for $nuppence, which I'd hoped to migrate my Asterisk setup
onto. I currently use an A400P analog card, but the ProLiant only has PCIe
slots, and they're short ones too, so I can't use an A400E card.
AJ-
On Thursday 19 April 2012, samuel wrote:
Just in case it helps:
It turned out that from asterisk version 1.8.4 on, the g729 binaries are
different from the previous versions so it was a version mismatch between
the g729 (1.8.0_3.1.5) and asterisk (1.8.8 and higher).
Thanks to the
Virendra-
After reading these web links. it's pretty clear that
FreeSwitch is batter then Asterisk feature, quality
wise. But asterisk is easy to used.
But the question is still open from my end.
*How* *FreeSwitch can support 1000CC but asterisk not* ?
Can you define your concurrent
Sunny-
I was thinking in Kamailio, but this sip proxy handles only the
SIP signalling traffic, no media processing.
Kamailio + rtpproxy.
-Jeff
On 3 November 2011 17:07, Nick Khamis sym...@gmail.com wrote:
Shouldn't you be using a Proxy?
Nick.
On Thu, Nov 3, 2011 at 1:04 PM, Sunny
All-
My apologies in advance if this is an obvious question and I've missed it on
Asterisk FAQs and how-to's...
Can Asterisk operate with just an FXO card? By that I mean, no network
connection (none, no local network). I want
to build some type of user interface to go off-hook, route FXO
All-
Recently an Asterisk server we host was hacked and used to route some
unauthorized calls. We have since improved our
security measures, including installation of fail2ban.
The interesting thing is the way in which this was discovered. The
unauthorized calls were occurring intermittently
Steve-
On 09/05/2010 04:08 AM, Vikram Ragukumar wrote:
Hello,
We are in the process of debugging a voice quality issue for a client of
ours that is a VoIP services provider. The client uses a softphone that
runs on a pjsip stack.
When placing a call using the softphone, it negotiates
Steve-
We are in the process of debugging a voice quality issue for a client of
ours that is a VoIP services provider. The client uses a softphone that
runs on a pjsip stack.
When placing a call using the softphone, it negotiates the use of G729
codec with the remote endpoint (ptime =
Don-
He is looking for competitive information...what are prospects paying for
Avaya when they could be saving lots of money with Asterisk systems.
Probably a better question for the biz list, but he doesn't deserve the
responses he's getting.
Agree. If it weren't for extreme high cost of
Jeff-
On Sun, 22 Aug 2010, David Backeberg wrote:
On Sat, Aug 21, 2010 at 10:49 PM, Duncan Turnbull dun...@e-simple.co.nz
wrote:
Voice recognition is a pain for people with accents and poor lines and when
Everybody has an accent. Some people live in a place where the people
they
Steve-
On 08/07/2010 03:15 AM, Jeff Brower wrote:
Steve-
El 05/08/10 14:50, Tim Nelson escribió:
- michel freihamich...@gmail.comwrote:
Dear Sir,
I tried to convert ilbc to ulaw and get the same result...Bad Voice
Quality
Regards
Again, iLBC is poor quality to begin
Steve-
On 08/06/2010 05:40 AM, Jeff Brower wrote:
Miguel-
El 05/08/10 14:50, Tim Nelson escribió:
- michel freihamich...@gmail.com wrote:
Dear Sir,
I tried to convert ilbc to ulaw and get the same result...Bad Voice
Quality
Regards
Again, iLBC is poor quality to begin
Steve-
El 05/08/10 14:50, Tim Nelson escribió:
- michel freihamich...@gmail.com wrote:
Dear Sir,
I tried to convert ilbc to ulaw and get the same result...Bad Voice
Quality
Regards
Again, iLBC is poor quality to begin with. You can't take a poor audio
sample and make it better by
Michel-
I tried to convert ilbc to ulaw and get the same
result...Bad Voice Quality
I think you have to be more specific when you say bad voice quality. Like
what? Worse than a cellphone call? Gaps
of audio missing? Robotic or cyborg sound? Static? A background tone or
buzzing?
iLBC
Miguel-
El 05/08/10 14:50, Tim Nelson escribió:
- michel freiha mich...@gmail.com wrote:
Dear Sir,
I tried to convert ilbc to ulaw and get the same result...Bad Voice
Quality
Regards
Again, iLBC is poor quality to begin with. You can't take a poor audio
sample and make it
All-
Vocera badge:
* WLAN b/g
* Talktime 2-2.5 hours, standby 20-27 hours
* headset jack
* OLED display (why don't they ever show this on the pics)
* Linux based
From what I can find on the web, the Vocera badges use a TI DSP (C54xx series)
and therefore run DSP/BIOS, not Linux.
If later
Kyle-
C5441 is a year 2000 DSP chip. If you're considering the hardware/TI DSP chip
path,
C64x or C64x+ series is higher performance (higher channel capacity) and more
relevant in terms of available suppliers, forum tech support, TI support, etc.
-Jeff
Kyle Kienapfel wrote:
For those
Adolphe-
Thank you David. I was thing about the cisco solution
but cost is the issue as I will so many DSP to for this
amount of calls.
If you're not doing G729 or other LBR codec (or encryption, or echo can with
long tail length, or other high level
requirement for RTP processing) then you
Jonathan-
On Tue, May 18, 2010 at 9:38 PM, Adolphe Cher-Aime achera...@gmail.com
wrote:
Hello Everyone,
I must deploy an asterisk system that can support
at least 500 pstn outbound calls.
It's a challenge as it's the first time i'm gonna build such a large
Bruce-
On 05/16/2010 11:22 AM, Jeff Brower wrote:
Bruce-
I'm trying to make an AMI call. I want to call a number, play an
announcement when the call is answered, then call a second number and
connect the two when the second call is answered.
I an able to make a simple call to two
Bruce-
I'm trying to make an AMI call. I want to call a number, play an
announcement when the call is answered, then call a second number and
connect the two when the second call is answered.
I an able to make a simple call to two numbers and connect them using
the manager API but playing
Martin-
checkout new open source voipmonitor.org SIP packet sniffer. I've
developed it for my telco company and I've decided to share it.
Testing and contributions are welcome!
VoIPmonitor is open source live network packet sniffer which analyze
SIP and RTP protocol. It can run as daemon
Iscario-
I'm trying to set up a secure VoIP channel between a Windows softphone
client
and an Asterisk 1.6... server running with OpenBSD. By secure I mean to
prevent any man in the middle to reconstitute any vocal exchange nor
sender/addressee/any header data/ of the VoIP call (in first
Randy-
On Wed, Apr 21, 2010 at 5:33 PM, Steve Murphy m...@parsetree.com wrote:
Assuming that every such spamming/hacking/attack site is funded on a
stolen identity/CC number, it will soon sink into Amazon that they are
getting a bad rep, and losing money on such problems, as all such charges
Pat-
As a podcaster I use Asterisk extensively and often have several people in
a conference room. We'll record the calls via a SIP phone connected to a
sound mixer. Is there an easy way to bump up the audio bitrate for all
callers connected to the Asterisk server and improve the general
Pat-
As a podcaster I use Asterisk extensively and often have several people
in
a conference room. We'll record the calls via a SIP phone connected to a
sound mixer. Is there an easy way to bump up the audio bitrate for all
callers connected to the Asterisk server and improve the general
Bruce-
How can I find out what the source of the problem is guys?
As I said I didn't change anything, except for making few minor changes to
the firewall today and that was at Amazon firewall level and not within
CentOS.
What causes these bad dahdi_test values?
P.S. there is only few
Jay-
Hiya Jeff, thanks for your answer,
2010/4/18 Jeff Brower jbro...@signalogic.com
Jay-
anyone knows an open OMA DM tool that would be able to configure Nokia
phones (mainly the sip-stuff of the e-series) for use with asterisk?
Anything open i could find
Jay-
anyone knows an open OMA DM tool that would be able to configure Nokia
phones (mainly the sip-stuff of the e-series) for use with asterisk?
Anything open i could find was the device manager from funambol, which was
last updated in 2006 :-(
I'm not sure what you're trying to do, but I
Tonty-
This is a solution they proposed, using GSM gateways, but it wont let me
handle 1000 simultaneous calls, the other option was using an E1 but the
cost would be too much to deploy 35 E1s to support that many calls. There
might be a better way of doing it.
Can you explain the multiple
connection.
-Jeff
On Wed, Apr 14, 2010 at 11:56 AM, Jeff Brower jbro...@signalogic.comwrote:
Tonty-
This is a solution they proposed, using GSM gateways, but it wont let me
handle 1000 simultaneous calls, the other option was using an E1 but the
cost would be too much to deploy 35 E1s
On Wed, Apr 14, 2010 at 10:33 AM, Tonty T ton...@gmail.com wrote:
This is a solution they proposed, using GSM gateways, but it wont let me
handle 1000 simultaneous calls, the other option was using an E1 but the
cost would be too much to deploy 35 E1s to support that many calls. There
might
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Fred
Posner
Sent: 12 April 2010 21:57
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Flood of REGISTERs -
Jim-
We have been experimenting with how many licenses are needed
when making calls, recording calls and using chanspy to
listen in on calls when G729 is involved. I can tell you that
way more licenses are needed then I had understood
previously. We are making calls via AMI originate and
Khalid-
:) all users are having the same issue, even those connected to this server
from abroad!
Since you have an identical working system, why are you not able to debug this?
First swap the phone... then swap any
cards in the server, then servers, then check carefully software differences
Jim-
There will be up to 150 phones so there will be 300
channels when they are all on the phone at one time.
I will be using a current 1.4 version.
That's a lot of channels for Asterisk... IIRC the TC400B transcoding card is
rated at up to 96 G729 channels.
Can you clarify your recording
Daryl-
I'm involved in discussions with my carrier right now and am
wondering if anyone has interconnected Asterisk to
Metasphere via SIP?
Yes, we're served by a Metaswitch usng SIP. Works fine.
Metasphere is MetaSwitch's PC/server based system, not to be confused with
their larger,
Jim-
Jim-
There will be up to 150 phones so there will be 300
channels when they are all on the phone at one time.
I will be using a current 1.4 version.
That's a lot of channels for Asterisk... IIRC the TC400B transcoding card is
rated at up to 96 G729 channels.
Can you clarify your
CDR record
for the call.
See also
http://lists.digium.com/pipermail/asterisk-dev/2010-March/043052.html
Yes seems so. Many layers of subtlety :-)
-Jeff
Am 17.03.2010 23:34, schrieb Jeff Brower:
Klaus-
Am 16.03.2010 01:42, schrieb Jeff Brower:
Vikram-
http://www.voip
Steve-
2010/3/17 Vinícius Fontes vinic...@canall.com.br
- Kevin Sandy kevin.sa...@snohio.net escreveu:
We're having an odd issue with codec negotiation from one of our SIP
providers. Here's the basic situation.
We receive an invite from them advertising support for G711, G729, and
Steve-
On Wed, Mar 17, 2010 at 6:02 PM, Jeff Brower jbro...@signalogic.com wrote:
Steve-
2010/3/17 Vin¨ªcius Fontes vinic...@canall.com.br
- Kevin Sandy kevin.sa...@snohio.net escreveu:
We're having an odd issue with codec negotiation from one
Klaus-
Am 16.03.2010 01:42, schrieb Jeff Brower:
Vikram-
http://www.voip-info.org/wiki/view/Asterisk+Letting+SIP+clients+connect+directly
The link above indicates that it is possible to setup RTP streams to
directly flow between endpoints and completely bypass Asterisk. I would
Ioan-
Sounds like this would give a useful measurement regardless of server type,
network config, and other variable issues.
That should be a great tool.
Do you have any plans to test with Asterisk in 'native bridging' mode? I.e.
with RTP streams not touched in any way
by Asterisk? I assume
Vikram-
http://www.voip-info.org/wiki/view/Asterisk+Letting+SIP+clients+connect+directly
The link above indicates that it is possible to setup RTP streams to
directly flow between endpoints and completely bypass Asterisk. I would
like to know if this configuration would work when,
a) both
Steve-
On 03/13/2010 12:00 AM, Alexandru Oniciuc wrote:
Hello,
I need a hand in choosing a small ATA, even with one FXS port, that
should do only fax with T38.
Ive tried Grandstream (ht286 model) but the faxes go out without ECM,
even if the Fax machine has ECM enabled.
Is
Chris-
Sounds like the Toyota bug has migrated to Asterisk... it's mutated into
runaway ringing :-)
-Jeff
Sorry for my attempt at levity; just couldn't help it plus I'm sure Digium guys
will know how to resolve.
We're having an issue that isn't easily googleable so I thought I might might
Jonathan-
Jeff Brower wrote:
Jonathan-
How did you measure the gaps? Using signal or speech analysis
software to display the recording? If you measure number of
samples between the gaps, does it correspond to multiples of RTP
packet payload length (for example, for 8 kHz G711 multiples
Jonathan-
How did you measure the gaps? Using signal or speech analysis
software to display the recording? If you measure number of samples
between the gaps, does it correspond to multiples of RTP packet
payload length (for example, for 8 kHz G711 multiples of 80 samples
between gaps) ?
Jonathan-
I'm having a problem with conferences both meetme and app_conference,
though I've done most of the testing with meetme.
Essentially, I get little gaps in the audio - usually fewer than a dozen
or so samples, though it does vary. They seem to occur at random, but I
usually get one
Cool Dude-
You keep asking the same question over and over, which is not cool.
voice mail is working when ever call is received, extension 2000 receives it
and if not answered in 20 secs, message
is stored in
voicemail no problem in that. after creating voice mail if some one again
call
Ishfaq-
I'm having a very odd phenomenon happening on our production server
(1.4.17 and using realtime). Sometimes a call will disconnect 30 seconds
after the SIP phone hits the mute button but it doesn't happen all the
time. I've done a sip debug while watching this happen and that doesn't
7 feb 2010 kl. 15.09 skrev Per Jessen:
Thomas Winter wrote:
Hi,
my Asterisk on debian lenny died after 80 days.
server kernel: [7572666.186852] asterisk[3673]:
segfault at 10 ip 7f3b8e90b4aa sp 40bf5f00 error 4 in l
ibpthread-2.7.so[7f3b8e903000+16000]
Anything what can be done to
Sandesh-
I used the converter to convert the sound file to 8KHz, 16 bit PCM encoded
mono wave file which asterisk needs and now the POTS call quality is lot
clear than before but the cell phone is still the same, not much
clear...i think because of its voice codec as you mentioned.
Sandesh-
I tried using some music on hold (music) files, when I test it with normal
SIP phone its clear and good, but when I call from my cell phone or POTS
line it sounds a bit scratchy/static and not clear at all, is there any
software that i need to install in the asterisk system to make
Kingsley-
On Tue, 2010-01-26 at 07:46 -0500, Doug Lytle wrote:
Kingsley Tart wrote:
Thanks for the link. I looked at that page but couldn't see how it
helped with my specific issue, unfortunately, though I admit I'm fairly
new to asterisk so I don't fully understand what's going on.
Allann-
On Wed, Jan 6, 2010 at 8:50 AM, Allann Jones allan...@gmail.com wrote:
But jailbreaking increases the freedom to develop a application and
Oh, I agree with you, but it's probably even better to make a decision
to either buy into the constraints of Apple or find a better, free-er
Daniel-
no I'm using the real commercial once.
I've installed it in November 2009.
Did you have the demo version installed before the commercial version? I.e.
install the commercial over the top of
the demo version?
-Jeff
F6HQZ ha scritto:
Hi Daniel,
Are you using a demo/beta version
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