On Feb 25, 2005, at 10:20 AM, Lee Howard wrote:
In a traditional analog fax you have modulated audio data, that is,
the data stream is converted into an audio representation by the
transmitter, and the receiver demodulates the audio stream to produce
the data stream. A lot of data gets packed
Hi Everyone -
I have webvmail up and running, I can see the messages, forward them,
pretty much everything but listen to them.
Here is what I see in my logs:
192.168.0.1 - - [25/Feb/2005:08:15:40 -0800] GET
/vmail/vmail.cgi?action=audiofolder=INBOXmailbox=2377context=default
Thanks to all your help I have now got it working great.
I have written a quick howto which I plan to add to the wiki if people
approve?
Take a look at:
http://www.burntwires.com/asterisk/Install%20PHP%20Config.htm
(Please excuse the bloated html)
Please leave any feedback and then I will
tim panton wrote:
[..]
Good luck with it.
I think I am lucky! ;-)
We resolved the problem changing the Mother board to one with an Intel chipset.
The first one had Via chipset.
1) At first, we changed the mother board by another with Intel chipset. This one
can set udma2 (or udma3) from BIOS. It
Thanks for the chmod, it definitely needed that!
I didn't have to change the etc.sudoers file though. I'm running Debian, via
the great Xorcom rapid installation.
I didn't change the permit lines either as this is just attesting box and
im not worried about security.
-Original Message-
Hi,
Thanks for the batchfile type, it's a handy one.
I'm not editing over the internet, just local LAN for testing ATM. Protected
via firewall.
Would it not be fairly secure using a combination of the following:
.htaccess file
VPN?
https access?
Limit apache to only allow certain IP's?
And the
Im sorry if this has been asked before as I couldnt seem to find it.
I have an avaya partner ACS r3 system that I want to be able to hook
asterisk into with a x100p card, into and use asterisk to tie into a voip
provider then be able to dial (or connect) to an extension like an
intercom function
Mark,
In the time it took to write all that you could probably have read up
enough about T.38 to realise you were talking complete rubbish :-)
Regards,
Steve
Mark Eissler wrote:
On Feb 25, 2005, at 10:20 AM, Lee Howard wrote:
In a traditional analog fax you have modulated audio data, that is,
Im sorry if this has been asked before as I couldnt seem to find it.
I have an avaya partner ACS r3 system that I want to be able to hook
asterisk into with a x100p card, into and use asterisk to tie into a voip
provider then be able to dial (or connect) to an extension like an
intercom function
El 25/02/2005, a las 12:10, Terje Myhre escribi:
By any web-user (ms explorer) to be able to call from a web-page to a
certain number/extension connected to one specific asterisk.
Almost as a web-based auto-attendant functionality.
Hence:
1. surf to the specific web-site
You 're right, there are some security issues using using sudoers
and system commands.
If the asterisk server is reachable from the outside over http or other
unsecure protocols, it would be really dangerous.
But in a trusty intranet-environment, where firewalls block every attempt
to access the
On Fri, 2005-02-25 at 12:36 +, Julian J. M. wrote:
You could add
exten = 1,2,Goto(context,2,2)
But I don't know what will happen when, after 5 secs, dial SIP/2 is
executed again...
I think you are on the right track. If SIP considers ringing as busy
then you can cascade through your
Hi guys,
I have a weird problem, and I have encountered a few other people with
the same issue. The problem is this:
Whenever I make a call from my IAXy (g711ulaw) to my server, and then my
server transcodes to speex and sends it to a remote asterisk server,
audio is perfectly fine. The same
On Fri, 2005-02-25 at 16:59 +0100, Martijn van Oosterhout wrote:
I'm asking because I'm planning to install multiple machines from the
same image and I need to know what file(s) I need to backup/restore to
make sure I don't lose my licences in the process. The only options I
can think of are:
There is an open source version of the license:
http://www.readytechnology.co.uk/open/g729/INSTALL-041103.txt
You can view the licensing information at the following:
http://www.intel.com/software/products/ipp/
more details can be found on http://www.voip-info.org
Steven Critchfield wrote:
On Fri,
How do I config Asterisk so when the directory cmd is used, the name of
the found entry comes from a pre-record gsm file instead of being spelled
letter by letter?
If the user as recorded is name, this file will be used. When it's not
recorded, * will spell it.
Dial to your voicemail and
Anyone able to get to these I am unable to get to them.
-Original Message-
From: Duane [mailto:[EMAIL PROTECTED]
Sent: Friday, February 25, 2005 10:14 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] VoIP/Asterisk
Title: Fax on Asterisk
Is it possible to send an email to Asterisk and have it parse the email or an attachment and send it out as fax?
Thanks,
Wiley
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
By any web-user (ms explorer) to be able to call from a web-page to a
certain number/extension connected to one specific asterisk.
maybe this php script help you (switch caller/called and modify Exten:)
--originate.php--
?php
# configuration
$astip=192.168.0.1;
$astmanager=test;
On February 25, 2005 12:57 pm, Wiley Siler wrote:
Is it possible to send an email to Asterisk and have it parse the email
or an attachment and send it out as fax?
Open up your web browser, go to www.google.com and enter asterisk send fax.
That will get you info on how to send a .tiff file to
Perfect. Thanks! Found lots for incoming and that filled the gap for
out going.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Friday, February 25, 2005 11:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Fri, 2005-02-25 at 09:25 -0800, Mik Cheez wrote:
There is an open source version of the license:
http://www.readytechnology.co.uk/open/g729/INSTALL-041103.txt
You can view the licensing information at the following:
http://www.intel.com/software/products/ipp/
more details can be found
On Fri, Feb 25, 2005 at 04:43:50PM -, C. Tomlinson wrote:
Hi,
Thanks for the batchfile type, it's a handy one.
I'm not editing over the internet, just local LAN for testing ATM. Protected
via firewall.
Would it not be fairly secure using a combination of the following:
.htaccess
Hello all,
Hi I would like to know how to configure a Mediatrix 1102 box to work
with my asterisk box. I have analog phones that i would like to connect
to my Mediatrix box and then connect the Mediatrix box to my asterisk
box. My main problems come from the fact that I have limited experience
I had this issue- it's security on the files. I put a cron job that do
/bin/chmod 777 /var/spool/asterisk/voicemail/default -R
evey 1 minute, but there may be a cleaner solution.
Assaf Benharoosh
MCP, MCSA, MCSE
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
From the docs on this command it should be available in 1.0.1.
I don't find it under the apps directory even after doing an update -d (which I
understand will add missing files or diretories)
I have also downloaded ver 1.0.5 and looked in its apps directory. It isn't
there either.
Any
Title: Video Support Not Working
Hello,
I have a couple of video phones that I am trying to get setup. I have used these phones with sipphone.com and they work great. Now I am trying to get them to work with my * server and I am having problems. The voice portion seems to work fine, but I
Title: Festival - [EMAIL PROTECTED]
Hello All,
I installed [EMAIL PROTECTED] with no problems whatsoever. All features so far work great.
However, I have been trying to setup the festivval weather AGI script and it won't work.
I see the script fire off in the CLI and it completes with
On Fri, Feb 25, 2005 at 11:24:21AM -0600, Steven Critchfield wrote:
It is based on a machine unique key created by querying your hardware.
You will not be able to share your licenses between machines. You will
need to buy licenses for each machine you deploy on.
You misunderstand. Ofcourse I
Title: Festival - [EMAIL PROTECTED]
Does your Festival installation work ok? (run the
tests/example scripts that came with the installation). I installed
Festival, and according to the installation scripts all went well, however none
of the tests/example routines would work - I kept getting
I don't suppose anyone might know why I hear ringing transposed over
itself when I place a call out via PRI?
SIP to SIP is fine
SIP to IAX is fine
SIP to PRI is always transposed
I mean sometimes you don't notice it much because it's lined up right,
but other times you'll hear a really long
Daniel Nystrom wrote
It seems like the Radio discussions is closing in on something I was
interested in.
How about controlling 30 2-way radios via E1 and 30-channel Mux
(channel bank?) with EM signalling?
I think the Mux uses CAS and each channel has Audio out, PTT, Audio
IN, Busy. 6-wire
Hi Assaf -
Already did that - the audio app location shows as a broken link on the
page and plays nothing.
On Fri, 25 Feb 2005 14:06:41 -0500
Assaf Benharoosh [EMAIL PROTECTED] wrote:
I had this issue- it's security on the files. I put a cron job that do
/bin/chmod 777
Title: Festival - [EMAIL PROTECTED]
figured it out. Stopped using the example
festival-weather.script.pl and used the festival-script.pl that is in the
directory already.
Works good. Is the voice customizable? Does
memory on the box play a part in quality?
Thanks,
Wiely
From: [EMAIL
On Fri, 25 Feb 2005 12:09:16 -0800
Trevor Peirce [EMAIL PROTECTED] wrote:
I don't suppose anyone might know why I hear ringing
transposed over itself when I place a call out via PRI?
SIP to SIP is fine
SIP to IAX is fine
SIP to PRI is always transposed
I mean sometimes you don't notice it much
Looking for zaptel/zapata configuration parameters to successfully
communicate with a Valiant GSM gateway as above.
Surely someone has done this?
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 266.4.0 - Release Date: 02/22/2005
OK,
After checking into this, I have found the following:
I can set it up so either incoming or outgoing sip calls on this trunk
work but NOT both. The sip show registry command shows everything as
it should be.
The section from my sip.conf is as follows:
[Broadvoice]
username = 2x
I am experimenting with my * server to use SIP with my long-distance
providers instead of IAX, so that the media path is from the end user
straight to the provider's gateway (hopefully reducing my bandwidth
consumption). I have it working with VoicePulse Connect but SetCIDNum
doesn't appear to
Here is a copy of my config that works great with broadvoice. I also
have an AGI that I wrote to verify country codes so your users can't
call countries that aren't included in broadvoices plan. If you want
that too, just let me know.
Sip.conf
On Fri, 25 Feb 2005 14:42:09 +
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
OK,
After checking into this, I have found the following:
I can set it up so either incoming or outgoing sip calls
on this trunk work but NOT both. The sip show registry
command shows everything as it should be.
The
Race Vanderdecken wrote:
I am running on an Intel Pentium 3, 1.5 GHz, mother board stuck inside
an old E-machine case and it is very happy... (I only wish I could find
a Okidata B4250 printer driver or a PCL-6 I could understand.)
Nabeel,
Works for me see below.
exten = _1NXXNXX,1,SetCIDNum(55,a)
exten = _1NXXNXX,2,Dial(IAX2/USER:[EMAIL PROTECTED]/${EXTEN},30,Tt)
Kevin
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel Jafferali
Sent: Friday, February 25,
- Original Message -
From: Robert Webb [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com; [EMAIL PROTECTED]
Sent: Friday, February 25, 2005 2:49 PM
Subject: Re: [Asterisk-Users] Asterisk With Broadvoice
On Fri, 25 Feb 2005
Nabeel,
Ignore my last post. Missed the SIP part of your question.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Collins
Sent: Friday, February 25, 2005 4:09 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE:
Great! It works now!! Thanks so much.
Roger Hanson wrote:
- Original Message - From: Robert Webb [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com; [EMAIL PROTECTED]
Sent: Friday, February 25, 2005 2:49 PM
Subject: Re:
He was talking about SIP
On Fri, 25 Feb 2005 16:09:27 -0500, Kevin Collins [EMAIL PROTECTED] wrote:
Nabeel,
Works for me see below.
exten = _1NXXNXX,1,SetCIDNum(55,a)
exten = _1NXXNXX,2,Dial(IAX2/USER:[EMAIL PROTECTED]/${EXTEN},30,Tt)
Kevin
-Original
- Original Message -
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, February 25, 2005 9:20 AM
Subject: Re: [Asterisk-Users] Asterisk With Broadvoice
Great! It works now!! Thanks so much.
What was it
Apparently the combination of the correct registry string and
insecure=very fixed it. Just as you said.
Thanks!
Roger Hanson wrote:
- Original Message - From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday,
-Original Message-
I have an avaya partner ACS r3 system that I want to be able to hook
asterisk into with a x100p card, into and use asterisk to tie into a voip
provider then be able to dial (or connect) to an extension like an
intercom function and be able to dial a number like that.
I have googled, and wikied until blue. Is it possible
to put T1---*Toshiba CTX ? I have a TE405P, with one interface programmed
for the T1, I am not sure how to program the 2nd port to mimick the
T1 to the Toshiba. The Zapata.conf
[channels]
switchtype=national
context=from-pstn
Thanks for the clarification. In that case the following should only be
considered for development.
Steven Critchfield wrote:
On Fri, 2005-02-25 at 09:25 -0800, Mik Cheez wrote:
There is an open source version of the license:
http://www.readytechnology.co.uk/open/g729/INSTALL-041103.txt
You
Right on!
Have a good week-end!
Francois
How do I config Asterisk so when the directory cmd is used, the name of
the found entry comes from a pre-record gsm file instead of being
spelled
letter by letter?
If the user as recorded is name, this file will be used. When it's not
recorded, *
Hi ALL;
When I insert data to Vm table in Realtime config,
I canot see any directory built under : /var/spool/asterisk
1) when Asterisk try to build the mailbox
directoryunder the path : /var/spool/asterisk/...
2) Where is source code that tells Asterisk to
build that directory under:
Hello all, I am
looking at replacing our current Cisco PRI gateway with a new server with a
TE405P card.My primary concern is receiving CallerID Name info on
the D-Channel. Does anyone have any experience terminating a local Qwest
PRI from a 5ES switch into the TE405P or similar? We are
http://www.voip-info.org/wiki-Microappliances+SIP+Active-X+Client
http://www.microappliances.com/site/html/index.php?section=Productspage=clienthowto.php
I havent tried it though, let me
know how it goes.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
I have two Broadvoice lines and there's three people in the office.
Any way to:
1) Pool the connections for trunking, where any one can get a free
line?
2) Prevent more than 1 simultaneous call per line? (So I will not get
hit for 3.9 cents a minute.
I'd like to use the country code AGI.
We're having a problem with Asterisk when we try to pass a call off to a
Lucent PSTN using SIP. This behavior does not exist with SER:
With Asterisk
An ISDN call is started, at the T1 level we receive call proceeding
and immediately we receive a Call in Progress just like the far end
party
1) when Asterisk try to build the mailbox directory under the path :
/var/spool/asterisk/...
Don't know about realtime, but in standard version, the directory is
built the first time you leave a message to this mailbox
hth
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Asterisk-Users mailing
Rich Adamson writes
GMRS, FRS and MURS radios may not be interconnected with the PSTN (47
CFR 95.141). There has been a lot of talk from lobbyists to clarify this
rule, but as it stands you could conceivably connect a *private* network
to GMRS or MURS radios (you can't make any plugins or
I'm still having problems.
Festival works from command line and I can make the speakers talk.
But when I dial my weather extension:
-- Executing Answer(SIP/3000-a844, ) in new stack
-- Executing AGI(SIP/3000-a844, weather.agi) in new stack
-- Launched AGI Script
Ronald Hartmann wrote:
Anyone able to get to these I am unable to get to them.
Seems I have an issue with a dead name server (the box itself) and I'll
be going in and hitting a button in the next 20 mins or so, but there's
3 other name servers so I don't know why dns doesn't just jump to
Title: Festival - [EMAIL PROTECTED]
Wiley, if you follow the instructions as
listed it will work.
Can you post more information about what
actually isnt working? can you post the output of your cli.
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Hello Mark , C. All , Is this device available for sale
in the US ? All the digging I've only found outside US
mentions of sales . Any help appreciated . JimL
On Fri, 25 Feb 2005, Mark Elkins wrote:
On Fri, 2005-02-25 at 13:46 +, C. Tomlinson wrote:
Did you have
GMRS, FRS and MURS radios may not be interconnected with the PSTN (47
CFR 95.141). There has been a lot of talk from lobbyists to clarify this
rule, but as it stands you could conceivably connect a *private* network
to GMRS or MURS radios (you can't make any plugins or modifications to
Turn on debugging (agi debug) and check to see if festival is exiting
with an error? (Maybe)
Ernie
On Feb 25, 2005, at 4:29 PM, James Taylor wrote:
I'm still having problems.
Festival works from command line and I can make the speakers talk.
But when I dial my weather extension:
-- Executing
Still no weather...
AGI Debugging Enabled
-- Executing Answer(SIP/3000-51a3, ) in new stack
-- Executing AGI(SIP/3000-51a3, weather.agi) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/weather.agi
AGI Tx agi_request: weather.agi
AGI Tx agi_channel: SIP/3000-51a3
AGI Tx
On Fri, 25 Feb 2005, James Taylor wrote:
I have two Broadvoice lines and there's three people in the office.
Any way to:
1) Pool the connections for trunking, where any one can get a free
line?
2) Prevent more than 1 simultaneous call per line? (So I will not get
hit for 3.9 cents a
My perl is not that great, but from your debug output, the STREAM
FILE agi command never executed from the festival-weather-script.pl
script, which should have happened right away.
Does your text2wave work? Is there a tts-.wav file in your
var/lib/asterisk/sounds/tts dir? a tts-???.txt
Hello All,
Has anyone tried Asterisk with SER.?
My main focus is billing and authentication of my endpoints.
I want Asterisk to handle all my endpoints and SER to do billing/accounting
stuff.
Any help will be highly appreciated.
Neel
___
Do yourself a favor and get a Sipura SPA-2100 - much easier to
configure and the quality is better than the Mediatrix unit. First of
all - do you have the Mediatrix Unit Manager software? If not,
configuration will be nearly impossible. Secondly, you will need to
configure the sip ports on the
Can regular analog phones be used and act as extensions, or does an fxs
device need to be put into place. I saw this on voip-info.org. How
would extension setup be possible without the fxo being aware of the
name of the device?
for Analogue Phones connect to Zaptel
has any one implemented open 723 at http://www.readytechnology.co.uk/open/g723.1
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To UNSUBSCRIBE or update options visit:
I just installed SER last night but if you want it ot talk to Asterisk I
found that you should install FREERADIUS Server and RADIUS CLIENT. For it to
function properly
- Original Message -
From: Nitesh Divecha [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial
Here are the official instructions from broadvoice for setup of Asterisk.
Other configurations are not supported.
http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup
Dan
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I have mine set up where you just dial 6 to get out and if it is busy it
rolls over to the next avaliable in the trunk.
- Original Message -
From: Greg Hill [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday,
Hi, all
I am setting up Asterisk for the first time and have some problems.
Setup is very simple -- Astersik box and two Polycom SP300 phones. I will
add bells and whistles as I go, at the moment things are very simple. No
TFTP servers, so phones run with their default configuration.
I set up IP
Christopher -- regarding the country checking: does your AGI also
check for mobile/cell numbers? Checking for just country codes is
trivial and I do it in the dial plan, but knowing which number is
actually a mobile call is tougher as each country does it differently.
So I'd be interested in a
[EMAIL PROTECTED] wrote:
Hello Jim,
thx for the answer..
Im happy I found someone that is using flash :)
It's not perfect, but it can be useful.
Am I right, if I transfer a call with flash, the line will be free
afterwards ?
Yep
Would you mind to past me how you did the flash part
When the call comes from outside on a certain context to play i
invalid extension to an external user is easy just by enclosing in an
incoming context:
exten = i,1,Answer
exten = i,2,Playback(pbx-invalid)
How to play an this context to an internal user, internal user has
access to all contexts.
Rich Adamson wrote:
GMRS, FRS and MURS radios may not be interconnected with the PSTN (47
CFR 95.141). There has been a lot of talk from lobbyists to clarify this
rule, but as it stands you could conceivably connect a *private* network
to GMRS or MURS radios (you can't make any plugins or
I've noticed a growing number of stores using FRS radios. It would make
sense to interface (via soundcard/console driver, with the nessacary
electrical conversion) a VOX FRS radio to asterisk to allow someone in
the office to page/talk with people on the floor or warehouse. You could
throw
Guys
Im having a few issues with Languages.
Ive setup the english language is it came from default:
/var/log/asterisk/sounds
/var/log/asterisk/sounds/phonetic
/var/log/asterisk/sounds/digit
/var/log/asterisk/sounds/letters
and then Spanish as
/var/log/asterisk/sounds/sp
Hi, all
I am trying to connect Polycom 300 to Astersik. I do not want to use FTP
server for now, so I am tryng to set up phone manually.
Network configuration parts is OK, except that it does not ask for SIP
server address. Any ideas where to set this?
Also i have some problems with setting up
On Sat, 26 Feb 2005, Anton Krall wrote:
and then Spanish as
/var/log/asterisk/sounds/sp
/var/log/asterisk/sounds/sp/phonetic
/var/log/asterisk/sounds/sp/digit
/var/log/asterisk/sounds/sp/letters
Now, the normal voices ARE heard in spanish but all digit related voices are
taken from the
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