Quoting Mike Diehl mdiehlena...@gmail.com:
Is there a list somewhere?
There is a list by state here:
http://www.call811.com/state-specific.aspx
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Quoting Tim Nelson tnel...@rockbochs.com:
Do you have any sort of site/mailing list/etc setup to facilitate
this group? I'd be interested in attending such a meetup in the
future.
http://www.tcaug.net/
Quoting Thczv F. Thczv thczv.th...@gmail.com:
When I set up my Asterisk box at home I didn't want to have to dial 9
to dial off premises, so I gave all my local phones three digit
extensions with this format: 1[1,0]*. My thought is that there are no
area codes that start with 0 or 1, so if I
Quoting Fred Posner f...@teamforrest.com:
Starting around 10:00 AM EST.
All services from them whether I connect by IP or DNS (both east coast
and west). Anyone else?
Yes, I'm experiancing the same problem.
Their www.voicepulse.com and connect.voicepulse.com seem to be offline
as well.
International numbers are variable length, so the timeout applies for those.
North American National numbers are a fixed length.
Generally, the phone company will collect 7, 10 or 11 digits for North
American numbers.
For example, I live in Minneapolis, MN.
My number is 612-xxx-.
I have
Quoting Gustavo A Gonzalez [EMAIL PROTECTED]:
I am looking for a sample sip configuration of a SIP provider that runs
Broadsoft VoIP switch.
This is what I use:
register = 3115552368:abcdefghijklmnop:[EMAIL PROTECTED]/3115552368
[broadworks]
type=peer
host=1.2.3.5
dtmfmode=rfc2833
Quoting Doug Lytle [EMAIL PROTECTED]:
C F wrote:
Then there is basicly no way to do this besides for cracking it? I
Not that I am aware of, no. This subject went around several years
back. They also talk about brute forcing the password as well. As far
as I recall, nobody came back
It would be possible if Asterisk sent a remote-party-id back to the
calling phone.
Polycom and Sipura phones (possibly Cisco phones) Support this with
SIP on Broadworks and it works great.
--Shane
Quoting Eric \ManxPower\ Wieling [EMAIL PROTECTED]:
It is not possible to do this the way
. I never investigated further.
--Shane
Quoting Eric \ManxPower\ Wieling [EMAIL PROTECTED]:
Have you ever actually done this with Asterisk?
Shane Young wrote:
It would be possible if Asterisk sent a remote-party-id back to the
calling phone.
Polycom and Sipura phones (possibly Cisco
It's all priced by quantity of each feature you license, number of
users, number of concurrent calls, things like that.
Previously it only ran on Solaris. It now also runs on Linux.
I wasn't involved with our initial purchase, but I couldn't imagine
you could have a working system for less
Quoting C F [EMAIL PROTECTED]:
Anybody here having any problems with nufone?
Calls are not going thru, when trying to call their customer service
number it doesn't go thru.
When trying to resolve www.nufone.net I get (sourec:
http://www.dnsstuff.com/tools/lookup.ch?name=nufone.nettype=A ):
Quoting Curt Shaffer [EMAIL PROTECTED]:
I wanted to see if there was anything reasonable in price out there yet that
performed an RF to IP bridge via asterisk. What I mean by this is callers
from PSTN can be patched to a UHF/VHF radio and vis-à-vis. I know there is
an option available for the
Quoting Savoy, Kevin - Williston, ND [EMAIL PROTECTED]:
Not sure if this can be done or not, but I can't seem to find it
anywhere on the Wiki. When dialing interoffice with Asterisk 1.4.2, I
would like to have the caller id of the person I am dialing displayed
and not the number I just dialed.
is strictly prohibited. If you are not the intended recipient (or
authorized to receive for the recipient), please notify the sender by reply
e-mail and delete, or destroy all copies of this message immediately.
-Original Message-
From: Shane Young [mailto:[EMAIL PROTECTED]
Sent: Monday, February 19
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
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
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
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:
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
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:
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
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.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
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
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
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
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
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
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
43 matches
Mail list logo
