Quoting Tony Mountifield [EMAIL PROTECTED]:
Does anyone know of any 4-wire analogue interface cards that could be
made to work with Asterisk? (I'm not averse to hacking channel drivers)
A T1 card to a D4 bank with something like a 4WEM or 4WTO should do the trick.
--Shane
On Tue, Aug 29, 2006 at 02:18:32PM +1000, Devraj Mukherjee wrote:
The simplest way I can think of solving this is using prefixes, so
someone appends a 0 or 1 and the dialplan puts the call through the
selected trunk, where 0 being voip and 1 being PSTN.
Whats wrong with something like this :
Quoting Kevin Savoy [EMAIL PROTECTED]:
Can someone recommend a good text to speech engine that is usable by
Asterisk? I have tried the Festival one and it just doesn't cut it for
commercial applications.
I like Cepstral.
Using the information here:
Quoting Leif Neland [EMAIL PROTECTED]:
According to what I've read somewhere, at least our 911 (112) has an
answering machine, saying Alarm central, one moment and a few seconds
delay, before the call actually is signaled to the dispatcher, to filter out
misdials and crank calls.
So if you
Quoting Ronald Wiplinger [EMAIL PROTECTED]:
Is there a table available, which tells me if a zip code, city and area
code matches?
For now I did it with google, type each info in and found out if it
matches, but it would be easier if there is a table available.
If you subscribe to the LERG,
Quoting Kevin Withnall [EMAIL PROTECTED]:
Does someone have code to do this already ? Ie log alarm stats to a
database and determine when to call out ?
I have a fairly simple system setup that logs things to a database and will
perform some type of
action based on the account, zone, type and
Quoting Pimjai Wesnarat [EMAIL PROTECTED]:
Hi,
I'm using Cepstral as a TTS Engine for Asterisk with Swift application.
It works fine when I have just 1 voice installed. Now I have 2 voices in
the same language installed but I can't seem to find the way to select
which voice to use in
SPONSORED THIS MONTH BY: SOUND CHOICE COMMUNICATIONS LLC
Keep in touch with the World
Hello,
The next Asterisk Users Group meeting has been scheduled for this Saturday
March 11th at 11:30am.
Meetings are held monthly on the second Saturday of each month, excluding
Bill,
I had the same issue over the weekend. Yesterday, there was an announcement
on the list, I think it was from Kevin Fleming, that the svn repositories
where out of sync and they had to be redone from scratch. So, I ended up
clobbering my local working copy and did a checkout of the trunk.
Quoting Andrew Nowrot [EMAIL PROTECTED]:
Hi,
Did anyone try to set up alarmreceiver application over IP network? Which
ATA can I use? I tried to set up it with Linksys PAP-2 but with no luck.
Maybe I did something wrong with alarmreceiver.conf (I tried diverse
settings, but nothing worked).
Quoting Chris Mason (Lists) [EMAIL PROTECTED]:
Lists wrote:
I am hoping the alarm companies adopt quicker to the internet.
I don't see that happening. Internet reliability is not going to be
sufficient for alarms. PSTN lines, for all their issues, don't fail, and
alarm systems can sense
Quoting [EMAIL PROTECTED]:
A secondary issue may be insurance. In a domestic situation, if you receive a
discount for having
an alarm installed, you may find that the insurance discount is only valid if
the alarm is
installed over POTS, and usually by hardwiring.
This is for actuarial
We use Polycom phones at work behind Broadworks.
When we call from one phone to another phone, we see the called name on the
display. This is
because Broadworks sends a remote-party ID back to the calling phone when it
invites the called
phone.
This also seems to work on the Sipura phone.
So
Quoting Mailing List [EMAIL PROTECTED]:
I believe they've done that the entire time. I've never known them to be real
supportive of
competing third party solutions.
They support third-party partners such as Broadsoft.
This
Quoting andrutto [EMAIL PROTECTED]:
I just want to ask if anyone has some experience with Alarmreceiver
application in * 1.2? Is this
application reliable (according to wiki it isn't)?
I don't see anywhere in the wiki where it says this is unreliable. The wiki
mentions that This
prematurely.
Does anyone have any ideas where the DTMF tones are coming
from? Are there any know problems when there is a cell phone involved?
Thank you,
Michael Young
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,
Michael L. Young
IT Manager
[EMAIL PROTECTED]
Administrative Claim Service, Inc.
888-227-5945
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I have a repeater using app_rpt, it seems to work just fine.
Quoting Mustafa N. Deeb [EMAIL PROTECTED]:
Has anyone been able to compile app_rpt?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips
Sent:
On Thu, Jun 23, 2005 at 11:39:21AM -0500, jltaylor wrote:
I've seen the embedded posts.
Is anyone running Asterisk on the MINI ITX?
Yes, no problems, I have an X100P in the PCI slot, but its only
a single POTS line. I used the MII board, but only because thats
what I had avaliable.
Iain
I can help you I think.
do you have the manuals for the Panasonic?
Quoting Dan Morin [EMAIL PROTECTED]:
If anyone has any experience with a Panasonic KX-TD1232 phone system, I
would really like to talk to you for a few minutes.
I have asterisk connected to a Panasonic system via
We are in the process of installing a PRI line and we
are going to connect it to an Asterisk Box.
Verizon called us today to find out some information. I am
surprised that they have never heard of Asterisk or Digium. But anyways, they needed
some information in order to set up the
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/mgcp/frmwrup
.pdf
Quoting Preston Garrison [EMAIL PROTECTED]:
www.voip-info.org has it
Preston Garrison
direct: 877-748-4142
fax: 310-774-3901
cell: 623-748-4140
-Original Message-
From:
On Wed, May 11, 2005 at 11:15:29PM +0200, Armin Lediger wrote:
I am trying to install asterisk 1.0.7 on a VIA EPIA 5000 board - anyone
of you already managed to do so? I got V1.0.6 running, but 1.0.7 seems
not to compile.
I have the MII-12000 board, and the debian packages work for me,
Hi Mehmet,
On Tue, May 03, 2005 at 11:20:44AM -0400, You wrote:
I tried that and it didn't work. Then I decided to use a different phone
line. I had not thought about this before, it just didn't occur to me.
And everything worked fine. The phone line that doesn't work is my ADSL
line.
On Mon, May 02, 2005 at 01:02:34PM -0400, Mehmet Tolga Avcioglu wrote:
I can't seem to be able to make outgoing calls with X100P card. I can
receive calls fine and it picks up the line and sends the tones, but the
telco doesn't recognize them. While the tones are sent I continue to
hear
Hi Cameron,
You Wrote:
On Thu, Apr 14, 2005 at 08:47:38AM +1200, Cameron Beattie wrote:
Couple of things I noticed:
Dunno why you'd have s extension in [voip-h323] context.
Format of Dial command seems wrong
Try putting the specific number you dial in extensions.conf e.g.
exten =
Hi All,
I'm just starting with Asterisk, so this may be something very
simple. I'm using a X100P, and a softphone (GnomeMeeting) with
OH323 providing the linkage into Asterisk. The (very) simple
setup looks like this:
PSTN---X100PAsterisk---OH323 GateKeeper---GnomeMeeting
(zap channel)
Quoting Rich Adamson [EMAIL PROTECTED]:
It seems to me silly to have a T1/E1 card to connect to a channel bank
when you could just have a 24/30 way FXS card in the slot in the first
place.
Does such a thing exist?
Wouldn't Digium have a lot of customers if
Hi,
I'd like to add to that answer. Try just sending the phone number
without any area code - eg 8617-2169
When calls come in on DID numbers they are always minus areacode so it
would go without saying that they should be sent out the same way.
Regards,
Kimble Young
Adam Goryachev wrote
Quoting John Middleton [EMAIL PROTECTED]:
Hi, I've looked at the Wiki for this, have seen the Swift.agi details,
but has anyone got a current script for Cepstral and an example of
integraton in * please?
It's been a while since I've fiddled around with it, but it should work like
this:
Yes.
Quoting Roger Hanson [EMAIL PROTECTED]:
Is the meeting still on for Saturday 1/8/05?
11:30am at 2375 University Av W STE120, Saint Paul.
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Quoting Jon Bebeau [EMAIL PROTECTED]:
HI all - I know, slightly off list, but.. I'm looking for a NPA NXX database
with City and State.
The North American Numbering Plan Admistrator has some info at
http://nanpa.com/nas/public/assigned_code_query_step1.do?method=resetCodeQueryModel
You can
If the phone has not been converted to SIP, the console may not work. I was
never able to get the
console to work on a skinny phone, but it does work on a SIP phone.
Quoting Paul Brock [EMAIL PROTECTED]:
Randy,
Is it a new unit? The only reason I ask is that hitting the settings button
Quoting Henry Devito [EMAIL PROTECTED]:
I attempted this but I got stuck on one issue. Cisco phones pull data so I
couldn't get them to autoupdate. In other words push data to them.
You can use an http Refresh to keep the screen updating once you've accessed
your XML application.
It's not
Just adding my 2 cents..
I have used PCI cards before that say PCI 2.2 on them (not specifying
whether it is a requirement) on machines that don't support it (old
old). The cards functioned perfectly in all cases.
Note: These were not digium cards.
Regards,
Kimble Young
Lee wrote:
The Digium
with ISDN. You can play
music or a ringing tone to the caller until the exact moment the other
end picks up.
Regards,
Kimble Young
Stuart Elvish wrote:
Hi,
Just wondering whether or not anyone in Australia has been able to get /
create an accurate (or semi-accurate) tone detection system. I need
some of the numbers and capi4hylafax pick up the
others. This way they will cooperate with each other.
Regards,
Kimble Young
http://faxonline.com.au
Carl Sempla wrote:
Hello,
Can I use the Diva Server PRI/E1-30 card with asterisk (with chan_capi or
i4linux) ?
If asterisk detect a fax, how divert
Quoting Matthew Boehm [EMAIL PROTECTED]:
Does anyone have one of these models? Can they confirm that it works with
any other SIP server? How is the PAP2-NA configured? Web? Phone?
The pdf I downloaded from the pap2-na page on shopblt.com says Model: PAP2.
The product manager for this devices
Quoting Jerry Geis [EMAIL PROTECTED]:
Cepstral offers Linux versions.
Just contact them.
http://www.cepstral.com/cgi-bin/downloads?page=voices
Note that you can not download any Linux versions from that page.
They changed something a while back. Released a new TTS engine for Windows and
Good Evening
I found your post about this problem. Did you ever find a fix for it? I'm
experiancing the same
problem.
Thanks.
Quoting Steve Creel [EMAIL PROTECTED]:
I have two Adtran 750's connecting our analog phones to asterisk. On
occasion, I get a channel that gets stuck off
The AVM B1 has firmware which will allow it to work in Euro or USA ISDN
mode.
As for the interface I'm sure you just need some sort of SBUS/S0 to U
Interface connector. Or buy a B1 that's meant for the US market.
It's a great card. Little expensive new though.
www.avm.de
Kimble.
-Original
Shaun
With chan_capi your only real choice is the fritz or an eicon diva server.
If you thought the fritz was expensive then close your eyes for the diva. I
think the 2 channel BRI card is about $1500 here.
You might want to try ebay. You can pickup a fritz for less than 20 Euros
I'm sure.
Quoting Joel Vandal [EMAIL PROTECTED]:
Hi,
I have a client that have currently 400 analog phones (all wired w/ Cat3). I need
multi-ports FXS
interfaces but I only find 24 ports FXS (like Mediatrix 1124) but it's a little bit
expensive to
get 15-16 box (~408 FXS ports).
You can get 40
Young
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Robert Barnes
Sent: Tuesday, August 03, 2004 8:25 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Called ID in Australia
Hello All,
Can any Australians who have any info or current patches relating
Quoting Marty Mastera [EMAIL PROTECTED]:
Hello everyone
Searching the archives and google always comes up with entries regarding
the dyn dns option in the 7960, but I can't find answers to my
specific question
It's a way to specify a DNS via config file which has priority over
,
Kimble Young
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of dkwok
Sent: Tuesday, July 13, 2004 10:57 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Oz ISDN
In Australia, Telstra, the local telco provides isdn modem for isdn
connection. The modem has 2
PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of dkwok
Sent: Tuesday, July 13, 2004 1:10 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] OZ ISDN
Kimble Young wrote:
If you go the analogue route:
* You'll get poor audio compared to ISDN which is crystal.
* Each number will act like a seperate line
any card supporting this standard
would work.
Regards,
Kimble Young
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Clint Tevlin
Sent: Thursday, June 17, 2004 6:49 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Anyone have experience with chan-capi
Lars,
Find this line in the makefile:
# if you want to compile against latest (non-stable) asterisk cvs
#CFLAGS+=-DUNSTABLE_CVS
uncomment the second line.
Regards,
Kimble Young
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Lars Roland
Sent: Monday
Jason,
Lucky I had the solution in front of me.
Read here:
http://lists.digium.com/pipermail/asterisk-users/2004-April/044125.html
You basically need to run a patch against chan capi.
Regards,
Kimble Young
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf
Quoting Rich Adamson [EMAIL PROTECTED]:
The cisco v6.x sip releases also include the ability to auto-answer a
call (required for phone paging), however some folks tend to suggest that
is a security problem as anyone can call that autoanswer extn number
and listen in on whatever is going on
Arggh I don't like the virus :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Friday, 23 April 2004 3:25 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] :)
Argh, i don't like the plaintext :)
archive password:
Eric,
Use tiffcp to merge multiple tiff files.
tiffcp src1.tif src2.tif srcX.tif destination.tif
If you have tiffinfo installed then tiffcp should be available as well.
Hope that helps.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Eric Wieling
Quoting Eric Wieling [EMAIL PROTECTED]:
Anything that says CallManager is NOT SIP. You want SIP.
These are the the part numbers. Your pricing will vary.
++--+--+
| part | description
A large one!!
Sorry I couldn't resist that.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tony
Mountifield
Sent: Wednesday, 3 March 2004 8:40 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Size of PC for conferencing?
Can anyone advise from
I am using digium h/w.
When I was in musiconhold , sound is strange .
Pls give a recommand !
young
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Thanks your kind reply
I am using radhat 7.3 .
And asterisk 0.7.1 latest version.
I use default file
/var/lib/asterisk/mohmp3
Would you explain more detailly to me ?
I spent about 1 week .
Thanks a lot
Young
- Original Message -
From: "David Liu" <[EMAIL PROTECTED]
pe=friend
secret=young
dtmfmode=inband
host=61.36.179.239
threewaycall = yes
callgroup=1
pickupgroup=1
context=from-sip
[hst220]
type=friend
host=61.36.179.220
callgroup=1
pickupgroup=1
threewaycall = yes
context=from-sip
[hst238]
type=friend
host=61.36.179.238
dtmfmode=inband
callgroup=1
pickupgrou
Hi,
I'm after some ISDN card resellers in Australia. I've noticed the AVM cards
should work well here, however I've found the reseller here has no website,
is listed in Yellow pages as management consultants (??) and when I called
quoted almost $400.
Does anyone know where I can get the AVMs in
Our company can offer VoIP to premises and domestic users and bill the
premises as a whole. We need something to enable the hotel owner to bill
each guest in a hotel in real time. What solutions do exist presently?
(PS: Our radius (and every telephony equipment outside the hotel) does not
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