Kevin,
I've looked at the source of app_queue.c and can see if the logic for
the * key, but nothing related to # in the code. Am I missing something?
Thanks,
Jonathan
Kevin Bockman wrote:
--- Jonathan Tew [EMAIL PROTECTED] wrote:
We've got the app_queue configured to supposedly allow
in to the queue system. Obviously we're doing something wrong to
transfer the call. We hit * to hangup the call. Is there some other
way to transfer the call? I've looked through the source and didn't see it.
Thanks,
Jonathan
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It is currently free, but unfortunately I don't believe it is open source.
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Quoting Hcqm [EMAIL PROTECTED]:
Dan,
is DIAX opensource?
If so where can I find the sources?
Thanks.
Hector
Did you make sure to put the in- in front of your register =
in-login:[EMAIL PROTECTED]
Ernest W. Lessenger wrote:
At 01:32 PM 12/8/2003, you wrote:
Greetings,
I have been experimenting with Asterisk for a few weeks and finally
decided to
take the plunge and purchase a few DIDs for inbound
.
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Quoting Uriel Carrasquilla [EMAIL PROTECTED]:
Excuse my ignorance, how do you turn off qualify?
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf
...
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--
Jonathan Moore
Technology Coordinator
Winfield Public Schools
Office 316-221-5100
Fax 316-221-0508
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,
Jonathan
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better and eventually doesn't annoy me. Any ideas what this
is or where to start trouble shooting?
When we talk SIP to SIP via Asterisk it works beautifully!
Any suggestions or pointers are greatly appreciated.
Jonathan
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rxgain=1.0
callprogress=no
busydetect=no
group = 2
;use with FXO PCI card
signalling = fxo_ls
;channel = 13-24
channel = 1
context = local
group = 1
;use with FXS USB card
signalling = fxs_ls
;callerid = John Doe (710) 555-6200
channel = 13
--
Jonathan Moore
Director of Technology
Winfield
Quoting Steven Critchfield [EMAIL PROTECTED]:
On Thu, 2003-12-04 at 11:49, Jonathan Moore wrote:
I just purchased a T100p from digium and a Carrier Access Access Bank 1
channel
bank (12fxs/12fxo). I have the setup partially working thanks to some help
from
IRC. However I still have
Hajime Lanning [EMAIL PROTECTED]:
Changes are below. Use KewlStart for the FXO channels. (Loopstart +
remote disconnect suppervision) Define all T1 channels. FXS channels
can be loopstart without any issues.
quote who=Jonathan Moore
I just purchased a T100p from digium and a Carrier
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Quoting Steven Critchfield [EMAIL PROTECTED]:
On Thu, 2003-12-04 at 13:42, Jonathan Moore wrote:
Quoting Steven Critchfield [EMAIL PROTECTED]:
On Thu, 2003-12-04 at 11:49, Jonathan
Don't have answers to your main questions but there is a place share war
stories. The Asterisk Wiki
http://www.voip-info.org/wiki-Asterisk
Not that many scenarios posted, but a few.
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Quoting
Ok, I contacted the seller about the ring issue. He has offered to replace the
fxs card in the unit.
1. Is the ring generator on the fxs card or part of the chasis?
2. Can anyone confirm the appropriate jumper settings for connecting analog
phones to CB?
--
Jonathan Moore
Director
to have a channel bank that supports
call sup. I also think there are issues with some CBs and caller ID. If you
want caller ID make sure to ask that it is supported and with which cards.
Adtran seems to be the most recommended vendor on the list. Many refs to TA
750s.
--
Jonathan Moore
Director
bank (or even
the T1 card)
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Quoting Walker Haddock [EMAIL PROTECTED]:
On Thu, Dec 04, 2003 at 06:26:49PM -0600, Paul Oster wrote:
Hi All.
I'm working on an * configuration. We require 8
See below
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Quoting Nick Bachmann [EMAIL PROTECTED]:
Greg Boehnlein wrote:
First and foremost, these Key System installers are big believers
in VoIP and convergence technologies
I have the same problem and haven't found a solution. In fact seems like some
post confirmed that this basically how * works with this hardware setup.
Someone on IRC today said there are some vmail settings for helping to prevent
the phatom vm messages.
--
Jonathan Moore
Director of Technology
Being farely new to the Asterisk scene and searching for
documentation I was wondering why the asterisk.org site didn't run a
wiki. There isn't anything as good as a wiki for collecting
collaborative documentation. Over time someone might want to convert
the knowledge contained in the
Asterisk Users,
Does anyone know the URL for the application API for asterisk? I
haven't been able to find any documentation on it.
Thanks,
Jonathan
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Found some info on the Web that may help some
of the ADSI programmers out there.
The following guide is for a WebSphere implementation
but the average developer type should be able to pull
enough out of it to help writing ADSI scripts for
Asterisk. Seemed to have good overview of ADSI
Book costs $49.50
PDF Download - free
--- Jonathan Biggs [EMAIL PROTECTED] wrote:
Found some info on the Web that may help some
of the ADSI programmers out there.
The following guide is for a WebSphere
implementation
but the average developer type should be able to
pull
enough out
Late Sunday night, getting
cvs update asterisk
? asterisk/doc/api
cvs server: Updating asterisk
M asterisk/app.c
cvs [server aborted]: missing expected branches in
/usr/cvsroot/asterisk/asterisk-ng-doxygen,v
[EMAIL PROTECTED] src]#
Checkout does same thing
What did I mess up?
New to ADSI but my guess would be...
If you want Asterisk to put the call on hold
you could just program the soft keys to send the DTMF
tones that a regular phone would use to put the call
on hold.
If you look at the example that can with Asterisk.
(/etc/asterisk/asterisk.adsi ..I have been
Hello all,
Have a few questions.
New to asterisk , just getting setup with
1 X100P and 2 TDM400p. Redhat 9
Hope I sent this to correct list
Setting up some Aastra/Vista Powertouch 350 phones.
Things work outofbox on ADSI programming, vmail
downloads, menus etc.
Question.
* not that it's too hard. If I choose to have a simpler
system - or more importantly choose for all the users at an
installation to have a simpler system - I can't do that.
Jonathan
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already replied to this, but I'd add that I started out
with an article from O'Reilly that explained pretty well how to get asterisk
going:
http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=1
Jonathan
--
Jonathan Hogg
Director, Technology
Seventh Wave Systems Ltd.
4-14
of a mess of mangled attempts to make
it work.
Any help gratefully appreciated.
Jonathan
--
Jonathan Hogg
Director, Technology
Seventh Wave Systems Ltd.
4-14 Tabernacle Street
London EC2A 4LU
Telephone: +44 20 7074 0423
http://www.seventh-wave-systems.com
--server' in a
config by the 'festival_server' script? I'm not in front of festival at the
moment, but you might want to try tracing through that script and see what
it's doing.
Jonathan
--
Jonathan Hogg
Director, Technology
Seventh Wave Systems Ltd.
4-14 Tabernacle Street
London EC2A 4LU
the
questions, but not the answers.
Thanks for the link and the help.
Jonathan
--
Jonathan Hogg
Director, Technology
Seventh Wave Systems Ltd.
4-14 Tabernacle Street
London EC2A 4LU
Telephone: +44 20 7074 0423
http://www.seventh-wave-systems.com
extensions, but I need a PSTN gateway service that
can offer numbers in London and NY.
I'm talking to a UK provider, but they only do SIP at the moment. I'm
working with one of their tech guys to see if they can support IAX via an
Asterisk installation at their end.
Jonathan
--
Jonathan Hogg
can only seem to ring myself (1 700 873 7731). I just tried
you and got nothing.
Jonathan
--
Jonathan Hogg
Director, Technology
Seventh Wave Systems Ltd.
4-14 Tabernacle Street
London EC2A 4LU
Telephone: +44 20 7074 0423
http://www.seventh-wave-systems.com
:
./configure make
* cd into festival:
patch -p1 .../festival-1.4.3.diff
./configure make
* Add $PWD/bin to PATH
* Run festival_server
Seems to work fine.
Jonathan
--
Jonathan Hogg
Director, Technology
Seventh Wave Systems Ltd.
4-14 Tabernacle Street
London EC2A 4LU
Telephone
voicemails as audio/wav attachments to mails in a
maildir appeals to me for running an IMAP server on top of it and providing
easy remote access to voicemail.
Jonathan
--
Jonathan Hogg
Director, Technology
Seventh Wave Systems Ltd.
4-14 Tabernacle Street
London EC2A 4LU
Telephone: +44 20 7074 0423
, but is
still a pain. Ideally I'd like to be able to configure the phone via DHCP
extensions. That would be ideal as I can configure the lease time to manage
how frequently the phones update and I can centralise the configuration with
the IP details.
Jonathan
--
Jonathan Hogg
Director
not
recognise which room in the hotel initiated the international (VoIP) call,
so that's the main problem - Only the hotel's PMS knows which guest phoned,
so with our current setup we cannot bill the individual guest, but the hotel
as a whole)
Jonathan
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