/Snip/
For those of you who may be interest
IAX2 loads are now available for the standard builds...
http://www.aredfox.com/edownloadsiax2.htm
Just a word of caution...
Remember to change the ports over to 4569 from whatever.
And don't forget to grab the palmtool from
This guy found out what I was planning to do as a boxed solution.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of asterisk
h323
Sent: Thursday, December 09, 2004 2:32 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Get rid of H323 problems for 100$
Greg Wrote
/SNIP/
Everyone wins from this, and Sveasoft has a revenue stream that allows
them to keep focused development on improving the firmware. I have over
60 of the WRT54GS units in production and I run Sveasoft firmware on
every single one of them. It is so far ahead of Linksys's internal
/SNIP/
INSERT INTO cdr
(calldate,
clid,
src,
dst,
dcontext,
channel,
dstchannel,
lastapp,
lastdata,
duration,
billsec,
disposition,
amaflags,
accountcode,
uniqueid)
('2004-12-06
13:36:17',
'\"Mohammad\"',
' ',
'9053265824',
'sip',
'SIP/3000-b4f7',
'SIP/9053265824-f4b2',
'Dial',
Tell me which one can get me access to the LinkSys Linux using SSH?
Does Satori has this feature? I am not so concerned with Voice Shaping
and QOS at this time, but more interested in converting this into a
Linux box that is accessible from an ssh client.
I have loaded Satori successfully on
-Original Message---
I just bought two new Linksys WRT54G routers. Sveasoft has loaded Linux
on this router and included a bunch of Linux tools, one of which is
Bandwidth Management.
Do you have the link for the free version of Sveasoft
Thanks for the link.
Tell me which one can get me access to the LinkSys Linux using SSH? Does
Satori has this feature? I am not so concerned with Voice Shaping and
QOS at this time, but more interested in converting this into a Linux
box that is accessible from an ssh client.
Seshu
Tell me which one can get me access to the LinkSys Linux using SSH?
Does Satori has this feature? I am not so concerned with Voice Shaping
and QOS at this time, but more interested in converting this into a
Linux box that is accessible from an ssh client.
Alchemy has ssh access, you need
Thanks Steffen. Please update me if this ever works.
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steffen
Koepf
Sent: Friday, November 26, 2004 3:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
/SNIP/
exten = 555,1,Answer
exten = 555,2,MP3Player(http://www.yourfavradio.com:port/)
exten = 555,3,Hangup
/SNIP/
Can we apply the same format for Video Streaming, if there is a
compatible SIP/MGCP Videophone connected to Asterisk.
exten = 555,1,Answer
exten =
Has this worked finally? Can you send me the configs if they indeed
have been working.
Seshu kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steffen
Koepf
Sent: Friday, November 19, 2004 8:40 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
/SNIP/
My BroadVoice account has been down for over a week with neither an
explanation nor a service credit. Our problems may be a little
different though because I don't remember what happened when I tried to
dial out. I know that I do get a Request Timeout error while trying
to register
/SNIP/
I am using phpmyadmin to view the astcc database and the cdr table is
the only one that i cannot browse
Im sort of new to databases so if I am missing something here please let
me know, as I would like to be able to have logs of all the cdr's for
backup documentation. Thanks!
/SNIP/
That
/SNIP/
Some corrections are needed: 6.3kbps of G723.1 will become around 17kbps
on the IP level without silence suppression because of the additional
overhead imposed by protocols like RTP, IP, etc . If you add the
Ethernet (or WAN protocol overhead) this will increase even more
(although
/SNIP/
Can you legally redistribute the Digium
G.729 code?
/SNIP/
Is that a
question or a Statement? You know that no one can redistributeDigium's
G729 Codec for free.
What is included
here is the Executable that registers your codec, if you buy the license from
Digium and have the
To: Kanuri, Seshu (Company IT)
Subject: Re: [Asterisk-Users] ATCC - Astcc-Admin.cgi File
Hi Seshu,
Also, the .png file for ASTCC is not showing up on my web page and my
presumption is that the ASTCC files I have on my box are a mixture of
Old ASTCC I have installed and some from your recent
Darren,
There is one more important change we need to do - security for the
admin page.
We need to incorporate a user name and password for the Administrator
page.
Without this, pretty much anyone who has knowledge of the url can
access the admin page.
Check this link below which is our beta
Importance - Very High
--
One of my clients is looking for the following solution and I need this
fast:
Does anyone have implemented or knows how to implement a
1) Web Based and email based callback
2) Dialout and Message delivery solution
...using Asterisk.
Does anyone in the list have a fully functional ASTCC and
would like to share their CGI, AGI and CONF files/Scripts
and database installation that is customized for:
1) Accepting user input for a Pin for authentication
2) Recognizes the caller id for authentication
3) Has a better GUI to
Not necessarily. The need is a generic calling card app - take user
Input or recognize the ANI, Allow the calls
The users and pins are stored in Mysql database
In order to make the database easy to manage - as the users and pins are
stored in Mysql database,
PHPMysqlAdmin (which is a generic GNU
One of my customers use Grandstream for ASTCC and it suddenly stopped
recognizing
DTMF for my ASTCC Application.
When ASTCC asks to enter destination number, and when the the digits
Are entered, the phone keys does not take any of them. They are dead.
Any suggestions
Seshu Kanuri
G729+ RFC233, but would that matter?.
My guess is that it could be the usual problem with Grandstream that
requires a reset to flush out any bad buffers, once in a while. I will
try that tonight.
/Snip/
What codec and signalling is being used?
I spoke to them a couple of months ago to buy their SIP
Boards so that I can make my own Dialup SIP Phone and I was told that their
residential gateway products are now out of life. Their SIP Implementation has
not worked well and hence they decided to close that
business.
I dont know their
Compression Codecs.
Seshu Kanuri
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 02, 2004 1:44 AM
To: Kanuri, Seshu (Company IT)
Subject: Question--Eezee phone?
Hi
I wanted to ask you a question about this phone, which I have as well.
Do you find
This Phone is based in PA1688 Chip. When IAX2 becomes stable on that
chip, it would be stable on this Phone, probably by end of November.
These phones are great. They work wonderful on SIP and H323 and hence
can extend the life of H323 Termination gateways like AS5300 etc. Full
info on the phone
This Phone is based in PA1688 Chip. When IAX2 becomes stable on that
chip, it would be stable on this Phone, probably by end of November.
These phones are great. They work wonderful on SIP and H323 and hence
can extend the life of H323 Termination gateways like AS5300 etc. Full
info on the phone
The Firmware has IAX2 in it. That is for sure. Whether it is currently
fully working is in question. The title is not fraudulent as when you
open the Phone it has IAX2 for configuration. He is just a reseller of
the phone not the Chip manufacturer.
-Original Message-
From: [EMAIL
What have you just done Steve?
It is like the Soot calling the Charcoal black. Don't you know that you
are the biggest spammer on this board? Have you ever counted how many
mails you send out to this list in a day and most of them mean nothing
to you or anyone on this board.
Geezz dude, grow
The link refers to an expired auction. It is no longer listed as having
IAX2. That claim was withdrawn till IAX2 on it is stabilized by the Chip
manufacturer.
NOTICE: If received in error, please destroy and notify sender. Sender does
My ASTCC is working,in the sense that it gives me a voiceprompt when I dial
the extension configured for the ASTCC by saying "welcome" and "12 digit card
number" but it is not asking me for the promptsto Enter Destination Number etc. Cananyone in the list send me the dialplan in
sequence so
Why are we allowing this A**hole to spam all of us to visit his site
which has nothing to do with Asterisk or VOIP?
-Original Message-
This may sound like a stupid question, and I know it's
off-topic, but
with 8,000+ subscribers, somebody on this list has to know the answer.
On top of what I have posted yesterday about my need to use Asterisk as
a Store and Forward Voicemail server, I have to provide the following
feature to my customer, who wants to save their voicemail message.
1) begin the the message when we hit a key like # Key
2) pause the message when he hits
This is a perfect example of how some (American Slang??) expressions are
perceived differently by different geo-cultural societies, speaking the
same basic language. One thinks that some thing that rocks is rock
solid, where as the other thinks that it is like rocking the boat.
This list very
I have this need to
have my customers store their messages for me indifferent
VoiceMail
foldersby dialing a seperate extension ( one for each such
customer).
Example:
I have
customer Anne ( Extn - 101) and Customer Jack (Extn
-102)
I want to provide
them two Voicemail Boxes to store
Sorry for the Top Posting but where is this discussion leading to? I do
not want this to turn into a Windows Vs Linux thread, please
Are we not digressing a little too far from Asterisk discussion??
Seshu Kanuri
-- SNIP --
appreciate your further input.
Nevertheless, the persistent
Kevin,
Are you looking for references for an actual implementation by someone
on Asterisk?
I think those who replied to your post do not understand the meaning of
Where can I find the open source G.729 implementation?.
Call me stupid, but I too don't understand the difference. As far as I
Try the two combinations
1) Press the Green Speaker Button Once then 1234 and then press the key
that shows you IP Address. This Will reboot the box.
2) Press *5#5 and a # on the Key pad. This resets the box
If this does not work, download the PALMTOOL from
http://ipphone.eezeephone.com
and
Check the help on WIKI. Wiki is your friend. Google is
your friend.
Also it helps if you are a little more specific as to
what the problem is - OS?Hardware? Software? Drivers?
How are you trying to install? if you are using an RPM,
the install is pretty straight forward.
Seshu Kanuri
Title: Message
Pretty much any SIP Phone can do these tasks in my
Opinion. At least ATCOM's Phones have these features like Auto Answer, Call
Transfers etc. Check their US distributors. These Phones are sold on
Ebay.
Seshu Kanuri
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Is the codec_g729.so file in the asterisk modules folder?
Is it getting loaded at the time of Asterisk load?
When you run show modules on asterisk CLI, do you see a module with a name like
codec_g729.so?
Did you try the command show G729 on Asterisk CLI ?
What does this Show? Does it show
Can we just kill this thread for now as we have discussed enough on this
already and we are not the judges who can rule it for good?.
Just my $0.02 Cents
Seshu Kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roy Sigurd
Karlsbakk
Sent: Thursday,
The humming noise comes from the Power units of these devices and has
nothing to do with the quality of the SIP device. You can reduce this
noise if your power unit has a longer cord or if you are using an
expensive Power unit that has similar input and output.
We found this problem on our
You are mixing oranges and apples here i guess.
G729is a MediaTransmission Protocol Codec the other is a Compressed
AudioFile format.
There are no .g729 audio files as far as I
know.
Seshu Kanuri
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Victor
CartesSent: Monday,
/SNIP/
Have you looked into that open-source implementation of G729? There
was something on the WIKI about 3 different implementations of it. One
being where you paid license per channel fees, one that was
free/open source, and another I can't remember. Check the WIKI.
/SNIP/
There is no such
If you are for bulk deployment of the phones in large numbers, without
losing your skin along with your shirt, I would recommend buying ATCOM
Phones. You can get them at $55.00 a pop in Bulk and $65 to $70 in
retail. These phones have all the basic features.
Try the link below for an OEM version
I tried to setup DIAX and connect to Asterisk for the last few versions.
It never actally been able to connect to Asterisk or my Other SIP
Proxies like VoiceMaster.
What does DIAX do actually?
Is there anyone in this list who has connected to Asterisk and made call
for real?
Seshu Kanuri
Steve,
Thanks for sharing your configuration. I will experiment with this conf
and try to connect.
Context - I agree with you on that and the DNS Entry. I find no need for
this to be user
entered as there must be a way pickup the default context. Also other
problems I found
were where the text
Correct. Line as in Wall Jack not as in Phone. You have to connect
your FXO card with a RJ11 cable between your telephone wall socket and
the RJ11 Port in the FXO card.
(You will connect a Analog Phone if you have a FXS card. If you
connect between wall jack and fxs card. You can potentially
my bad, none of those applications really work for us
and they are too rudimentary of any value.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Altus
SymanSent: Tuesday, October 12, 2004 9:08 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
Why is this thread continuing with postings either for off-list talk
stuff or meant for bizlist?
Let us stop this forthwith.
Seshu Kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Icide
Sent: Tuesday, October 12, 2004 3:38 PM
To: Asterisk
201 - 250 of 250 matches
Mail list logo
