hey * folk,
need to tap the collective wisdom of this list for any details or
pointers to vendors who manufacture/sell SIP or IAX phones with builtin
magnetic stripe readers. these phones will be used in combination with *
in a prepaid application. it would be advantageous if the mag stripe
Hi all,
I have this issue where I need SS7 access. But since SS7 is no good with Asterisk, I
had to look for a converter.
I did not go the SS7-PriISDN route as it costs more than SS7-Voip.
My supplier is asking me if I want SS7-SIP or SS7-H323.
So the question is, what do you guys think? Any
Hi
Most of the threads in the list archive relating to X100P and hangups
are about not detecting hangups. We have got the opposite problem.
We have experienced an increased number of false hangups when
connecting an X100P to an analog port of an ISDN terminal adapter. It
happens more frequently
DigitNetworks (and the like) are selling inferior, driver compatible
hardware, which is ABSOLUTELY NOT the same hardware Digium chose to be
their X100P.
If you can see my previous mails, no user worried about the
hardware whether it is from digium or intel or whether it is having
Hello,
Could anybody tell me if there is a Linux distribution (or Kernel
version) that works better with Asterisk. I am newbie and I dont know if
there is a preferred Linux/kernel version for Asterisk.
Thanks.
___
Asterisk-Users
Well so far as I know there is no preferred
version of Linux for Asterisk.
It will work with the most versions of
linux.
For myself I have it perfectly working
with new hardware on a fedora core 2 version with a 2.6 kernel
But be sure you can not follow the
asterisk installation by
For so far as I know. Asterisk is an open source and allowed with all the
hardware it will make the product even stronger.
For so far as digium they have a great marketing product in hands.
Everybody knows Digium as a great supporter for asterisk and a 100%
compatible hardware supplier.
Digium,
hi,
i'm under the impression that this feature is not available in asterisk,
consider this scenario:
- you are the operator. you answer a call from outside and you want to
transfer it to one of the extensions. after you transfer, if the person
you transferred the call to, doesn't pick up or if
- Original Message -
From: Brian Capouch [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Saturday, September 04, 2004 1:06 AM
Subject: Re: [Asterisk-Users] Commercial CID spoofing system
Jay Milk wrote:
Get an IAX provider
Thanks for your answer Han.
Regards.
-Mensaje original-
De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Johannes van Hulst
Enviado el: sábado, 04 de
septiembre de 2004 13:51
Para: 'Asterisk Users Mailing List
- Non-Commercial Discussion'
Asunto: RE:
Roger,
Roger Schreiter wrote:
there are still some questions to be answered by OpenSS7.com
in order to decide, whether E400P-SS7 is a good choice for
the asterisk SS7 support.
Are you making any progress in the Asterisk SS7 support? I'm eagerly
awaiting status report #3 ;)
Regards,
Dirk-Jan
--
I think what you want is an attended or consultative transfer, this can be
accomplished in different ways depending on your setup, for zap channels see
http://www.voip-info.org/wiki-Asterisk+tips+zap+transfer with SIP this will
normally be implemented on your hard/soft phone. Another alternative
My scenario:
Phones -- PBX
* -- Voip Provider
* --
Hi, I would like to connect a traditional
pbx with asterisk to forward call to a voip provider. I would like to do
this to use a HFC card in NT mode.
Someone experienced with this
Just did a complete cvs checkout (Sep 4, 9:30am cdt). Seems any attempt
to dial out via a tdm04b (fxo) now fails. Does not seize the pstn line.
CLI zap show channels
Chan Extension Context Language MusicOnHold
pseudoinbound-bus-x10 en default
Try this Phone at http://ipphone.eezeephone.com/
This Phone is listed now on ebay for sale at
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemcategory=11908item=5718863004rd=1
- Original Message -
From: Andrew Newton [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, September 02, 2004
Last month I did precisely that. One of my clients, a big travel agency,
has offices in multiple locations in US and India. Since they handle
large number of incoming calls, I decided to get a PRI in one of the
locations in US. No need of Vonage lines in their case. They have a
private route
Sorry. I was not following the thread, but...
What justifies this phone has this price of 99US$ while others are
for retail from 75 to 85US$ .?
Isamar
On Sat, 4 Sep 2004, SeshKanuri wrote:
Try this Phone at http://ipphone.eezeephone.com/
This Phone is listed now on ebay for sale at
Jason,
We already have this setup running and I can help you with that. You can
also use our termination to all countries in the world.
Please contact me offline at 732-387-4133 and send me a detailed email to
netwebgroup at yahoo.com
Seshu Kanuri
- Original Message -
From: Jason
Steve Totaro wrote:
Voicepulse appears to be down for me right now, but that might be a
result of their frustratingly-common config changes, about which they've
never ever sent me any notice.
can confirm voicepulse is down for me as well.
Well in my case it turned out that they hadn't sent me the
Kris Boutilier wrote:
[foo-context-companya]
exten = _.,1,DigitTimeout(2);2 second pause signifies end of dialing
exten = _.,2,SetVar(target=${EXTEN})
exten = _.,3,SetVar(companyCIDNum=18005551212)
exten = _.,4,SetVar(companyCIDName=companya)
exten = _.,5,Goto(step2${dialed},1}
These will
Title: Message
The piece of plastic is built in to the IP600. Use a
screwdriver or something similar and push the little piece of plastic out from
the top inside the cradle. It will pop out, and you can turn it over and
reinsert it upside down to hold the receiver in place.
From:
Hi all,
I've installed Asterisk on Linux Red Had 9. Now, I was trying to set
up a GUI based system for the PBX.
I downloaded some packages, but I have to have Perl running CGI
scripts through the webserver. It does not allow me to.
I am able to run a basic script that just just prints out html
I have a setup where Vonage box terminates on a asterisk
PBX. Things work great with one call coming on the vonage line into the PBX.
The problem is when that line is busy. The second call to the vonage
number does not get to the pbx at all it goes into vonage voicemail. Am I
missing
Is sendmail or postfix(which ever your distro is using) running or do you
have them disabled? You could also look in the mail log file in your
systems log directory for more clues.
Lyle
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, September 02,
I am not sure about this as I have not actually done it yet. You could park
the call. Hit the # and send it to extension 700. Allison will announce
which slot the call is parked on. You can hang up the phone and pick up the
call from any extension on the pbx by dialing the slot the call is
I had this idea, and after looking for something like this already in
progress, I found another guy who tried to start it... But I was
unable to contact him, and his project seems to be dead. But, I
believe it is possible, and I wanted to know the opinion of the
experienced... So, let's go:
I got
I had the same problem -- make sure these three lines are at the
beginning of each of your .pl's (right after the comment section):
use CGI;
my $q = CGI-new();
print $q-header();
That fixed it for me.
Jim Shilliday
IT Director
Equal Justice Center
1315 Walnut St. Suite 400
Philadelphia PA 19107
Try starting an FTP file transfer of a very large file and then dial that
voice number and see if the ftp transfer drops or pauses. It could be that
the ringing is causing the DSL signal to drop off.
If that is your problem, then replace the filters and then complain to SBC
that ringing is
Has anyone has issues with echo using a Wildcard with a PRI from a
major Canadian Telco? (Bell, Telus, AllStream, Sprint, Group Telecom).
We are using a T1 from GT that is giving use annoying echos whenever a
SIP/IAX2 client calls a
local analog line. Calling cells phones is no issue since
Roland Zagler ([EMAIL PROTECTED]) napisa(a):
Hello!
Is there a way to use AVM Fritz!PCI as a ZAP interface and have it
configured for ZAP channels?
afaik no, but You can use it with isdn4linux or capi. Works just fine;)
br
mazek
--
http://www.marcinmazurek.com/ ::: nic-hdl:
Marconi Rivello [EMAIL PROTECTED] wrote:
I was thinking: we could build an Asterisk network, maybe go even
further and make it P2P like skype (but I believe it's not necessary
at the beginning), and every user would share it's phone line, and be
able to place calls to PSTN through the other
Best bet for such a CoOp would be a give and take relationship. If
they also give you access to something of theirs they are more likely
to treat your stuff with care as well.
But it is risky.
On Sat, 4 Sep 2004 22:11:37 +0100, Kevin Walsh [EMAIL PROTECTED] wrote:
Marconi Rivello [EMAIL
Dinesh Nair wrote:
hey * folk,
need to tap the collective wisdom of this list for any details or
pointers to vendors who manufacture/sell SIP or IAX phones with
builtin magnetic stripe readers. these phones will be used in
combination with * in a prepaid application. it would be advantageous
On Saturday 04 September 2004 16:57, [EMAIL PROTECTED] wrote:
Has anyone has issues with echo using a Wildcard with a PRI from a
major Canadian Telco? (Bell, Telus, AllStream, Sprint, Group Telecom).
I have a PRI with Bell Canada in Listowel, ON (519-291-).
I have echo on some calls but
Asterisk should run well with any Linux distribution.
Mepis, www.mepis.org, is pre-configured for *
and might make your installation faster andeasier.
Paul
Paul
Mahler [EMAIL PROTECTED]
Signate, LLC665 Third
StreetSuite 100San Francisco,
Hi,
I just installed OH323 Plugin and im now tryin to make
simple Configuration to connect Openphone and Xlite to
my Asterisk-Server.
All works fine, i just wanna know if there's a
better way to do it? Is there anything wrong with my
Config?
OH323.conf
[general]
listenAddress=0.0.0.0
Here is my configuration for MEdiatrix 1204, by default the 1204 strips one digit, so it is not necessary to use:
To dial OUTSIDE
EXTENSIONS.CONF
[locales];ignorepat = 9
exten = _9,1,Dial(SIP/[EMAIL PROTECTED])exten = _9,2,Congestionexten = _9,102,Congestion
To receive
hi marconi my name is Mo and i like that adea of everybody conecting their asterisk to the local phones but can you explain me more how to do that.
regards
Mo
-- Original message -- I had this idea, and after looking for something like this already in progress, I found
Kevin:
I see your point, and I agree with a great part of it. But, if you
need to use your phone, and the line is busy because of a bunch of
Brazilian people, you can just use your neighbor's, for example. Or,
if you have only one line, and you and your
brother/sister/wife/whatever wants to use
Marc:
I took a look at those websites, but they look like a big phonebook.
Are you sure that they provide ways of placing calls from SIP to PSTN
for free?
Marconi.
On Sat, 04 Sep 2004 22:22:02 +0200, Marc Storck [EMAIL PROTECTED] wrote:
This already exists, it's called ENUM/E.164 and exists at
Marc
Sure -- you register a free number that reaches your asterisk box, and
configure your box to let incoming callers dial out locally to the PSTN.
Jim
-Original Message-
From: Marconi Rivello [mailto:[EMAIL PROTECTED]
Sent: Saturday, September 04, 2004 9:03 PM
To: Marc Storck
Cc:
Hi,
Without going in depth of my thesis research, this is _no_ utopia and
with a good organisation it can be set very easily also by non techies.
Probably everybody remembers Sipphone.com with there *cute* little
blackboxes where you can put in a landline, ethernet, and a regular phone
(pots).
Marconi Rivello wrote:
In US, local calls are free. So it wouldn't be a problem to make such
a network to get rid of long distance calls. But in other countries
(like here in Brazil) local calls are charged. So there could be some
king of billing (without commercial purposes, just to pay for the
Newbie here. Learning a lot by reading the
lists. Does anyone know If Asterisk will record call if you want it
to. ie. for a small call center. Or would some programming need to
be done on the workstation side if you were creating a softphone of
sorts?
Bill Lohr
check voip-info.org for call recording. There is a dial plan example
using Monitor for that
- Original Message -
From: William C. Lohr Jr. [EMAIL PROTECTED]
Date: Sun, 5 Sep 2004 00:35:57 -0400
Subject: [Asterisk-Users] Call recording
To: [EMAIL PROTECTED]
Newbie here. Learning a lot
Thanks for the help.
Bill
- Original Message -
From: William Suffill [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, September 05, 2004 12:38 AM
Subject: Re: [Asterisk-Users] Call recording
check voip-info.org for call
Greetings,
I'm having a miserable time getting Asterisk working with FWD. All the
samples show something like...
exten = _7.,
How do I get Asterisk to wait until the user is finished dialing instead of
trying as soon as it gets the second digit?
I can use _7XXX, and dial the FWD 3-digit
Good points, Jon.
I know that there are legal issues that must be considered. But the
technology can't be held back because of the ignorant. The police
already has computer technicians to help with digital crimes. And the
internet and voip revolution won't stop because of the illicit ways
they
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