David Quinton wrote:
On Sun, 15 Feb 2009 15:01:42 +, Julian Lyndon-Smith
aster...@dotr.com wrote:
Anyone got any thoughts on this and how it compares to the chan_skype
that's due soon ?
OpenSky is a free service provided by Gizmo5 which allows *any* mobile
phone, web browser or
On Monday, February 16, 2009, Julian Lyndon-Smith wrote:
We also don't yet know the pricing structure of chan_skype ...
I thought it was $99 per channel for corporate licenses or $19 for a
single, personal license ... or have I got the wrong ChanSkype?
http://www.chanskype.com follow the buy
Hi All,
I need to setup asterisk to receive fax.
I'm try Spandsp (opensource) and Attrafax (commercial) both on
asterisk 1.4.23) but the results are disappointing.
with spandsp many times the fax arrives cut.
with Attrafax i have some problem.
Anyone have any idea or solution (Opensource or
Can anyone help?
On Sun, Feb 15, 2009 at 2:51 PM, Jim Boykin boykin...@gmail.com wrote:
It does not work at all even after long time. DNS resolution is not a
problem, because if I load it from command line asterisk -c,
everything works fine.
The problem is when it is configured to be loaded
2009/2/13 Philipp Kempgen philipp.kemp...@amooma.de
Benny Amorsen schrieb:
Top posting is annoying. Gmail is broken; maybe I should just killfile
@gmail.com.
Emails sent through Gmail's *web interface* are broken. :-)
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany
2009/2/16 Fabio Mosti fmo...@gmail.com
Hi All,
I need to setup asterisk to receive fax.
I'm try Spandsp (opensource) and Attrafax (commercial) both on
asterisk 1.4.23) but the results are disappointing.
with spandsp many times the fax arrives cut.
with Attrafax i have some problem.
2009/2/13 Tzafrir Cohen tzafrir.co...@xorcom.com
On Fri, Feb 13, 2009 at 09:59:50AM -0800, John Todd wrote:
I've been involved with getting better data for running Asterisk on
the Amazon EC2 cloud computing system. Here are some calculations
I've made on costs based on current published
On Mon, 16 Feb 2009 22:14:49 Fabio Mosti wrote:
Hi All,
I need to setup asterisk to receive fax.
I'm try Spandsp (opensource) and Attrafax (commercial) both on
asterisk 1.4.23) but the results are disappointing.
with spandsp many times the fax arrives cut.
with Attrafax i have some
Anyone have any idea or solution (Opensource or commercial) to suggest me
?
Best Regards
Try hylafax with IAXmodem. The best results i had it the multitech modems
directly connected to FXS PCI card, you have a nice web interface if you
wish also (avantfax) You can find some nice
2009/2/16 Michael mich...@networkstuff.co.nz
Anyone have any idea or solution (Opensource or commercial) to suggest
me
?
Best Regards
Try hylafax with IAXmodem. The best results i had it the multitech
modems
directly connected to FXS PCI card, you have a nice web
2009/2/16 Grygoriy Dobrovolskyy megaho...@gmail.com
2009/2/16 Michael mich...@networkstuff.co.nz
Anyone have any idea or solution (Opensource or commercial) to suggest
me
?
Best Regards
Try hylafax with IAXmodem. The best results i had it the multitech
modems
directly
Fabio Mosti wrote:
Hi All,
I need to setup asterisk to receive fax.
I'm try Spandsp (opensource) and Attrafax (commercial) both on
asterisk 1.4.23) but the results are disappointing.
with spandsp many times the fax arrives cut.
with Attrafax i have some problem.
Anyone have any idea or
Steve Underwood wrote:
Fabio Mosti wrote:
Hi All,
I need to setup asterisk to receive fax.
I'm try Spandsp (opensource) and Attrafax (commercial) both on
asterisk 1.4.23) but the results are disappointing.
with spandsp many times the fax arrives cut.
with Attrafax i have some problem.
Hello-
Firstly thanks very much for the work you have put into SpanDSP and the time
you spend to assist people here :-)
I am currently running SpanDSP 0.0.5 with Call Weaver. Is there any or
sufficient gain to be had from upgrading SpanDSP?
Michael
2009/2/16 Steve Underwood ste...@coppice.org:
You don't indicate the kind of setup you are using.
I use asterisk (Spandsp) with a IAX2 trunk (ethernet connection) to
another asterisk (zap).
client-asterisk (Spandsp)-asterisk (zap)-fax
Regards,
Steve
Best Regards,
Fabio
Fabio Mosti wrote:
I use asterisk (Spandsp) with a IAX2 trunk (ethernet connection) to
another asterisk (zap).
client-asterisk (Spandsp)-asterisk (zap)-fax
We have two remote phone systems connected via IAX and 1 fax server at
the corporate offices with a PRI (ZAP). The fax server and
2009/2/13 John Todd jt...@digium.com
On Feb 13, 2009, at 11:19 AM, Philipp von Klitzing wrote:
Hi there,
is gizmo the first user of the Digium Skype solution, or do they use a
different approach/product - any clue?
http://www.gizmo5.com/pc/opensky/
Philipp
I know nothing
Fabio Mosti wrote:
2009/2/16 Steve Underwood ste...@coppice.org:
You don't indicate the kind of setup you are using.
I use asterisk (Spandsp) with a IAX2 trunk (ethernet connection) to
another asterisk (zap).
client-asterisk (Spandsp)-asterisk (zap)-fax
To quote the
Grygoriy Dobrovolskyy wrote:
2009/2/13 Tzafrir Cohen tzafrir.co...@xorcom.com
mailto:tzafrir.co...@xorcom.com
On Fri, Feb 13, 2009 at 09:59:50AM -0800, John Todd wrote:
I've been involved with getting better data for running Asterisk on
the Amazon EC2 cloud computing
The Attrafax software that was mentioned at the beginning of the thread does
support Gateway mode.
Regards,
Marc
-Original Message-
Fabio Mosti wrote:
2009/2/16 Steve Underwood ste...@coppice.org:
You don't indicate the kind of setup you are using.
I use asterisk
Sorry about off-topic, but can you advise the mail client who is
able to organise the web mailing list topic as web interface does ?
(i mean by blocks/topics) I wold be glad to use something else with
the same usability, but dont see any alternative.
Thank you
Just turn on threading
On Mon, 16 Feb 2009 16:05:48 +1300, Matt Riddell wrote:
On 10/02/2009 5:08 a.m., Michael Graves wrote:
I unwittingly started this on Facebook, which I don't user very much.
Here's the gist of it.
A Strange Brew: VoIP/Telephony Crossed With Surround Sound
It couldn't be the puritanical
2009/2/16 SIP s...@arcdiv.com
Grygoriy Dobrovolskyy wrote:
2009/2/13 Tzafrir Cohen tzafrir.co...@xorcom.com
mailto:tzafrir.co...@xorcom.com
On Fri, Feb 13, 2009 at 09:59:50AM -0800, John Todd wrote:
I've been involved with getting better data for running Asterisk on
Hello All
We have started a voice portal for Parents and Students. They can listen
Grades, Attendance status and other relevant information over *phone*.
Please read features below and to listen Demo IVR call at *00 92 42 8315427.
*Initially we are deploying it for a School and planning to spread
Hi All,
I'm looking for a way to filter the AstDB cidname family to show only
those entries with a specified area code in the Asterisk CLI. If this
were a SQL database it would be something like:
SELECT number, name FROM cidname WHERE number LIKE '1234%'
I've tried database show cidname 1234* and
Geoff Lane wrote:
asterisk -rx database show cidname | grep area code
at the Linux shell prompt, but I'm looking for a way to do this
without leaving the Asterisk CLI.
I don't believe it does.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
On Mon, 2009-02-16 at 15:14 +, Geoff Lane wrote:
I'm looking for a way to filter the AstDB cidname family to show only
those entries with a specified area code in the Asterisk CLI.
I don't think this is possible with the current AstDB code.
If this
were a SQL database it would be
On Monday, February 16, 2009, Jared Smith wrote:
If you have that many items in a database and want to do those types
of filters, why not stick them in a SQL database and use func_odbc
to retrieve them from your SQL database inside the dialplan?
Thanks for your suggestion. My Asterisk machine
I am getting a priveldged command error on the manager API.
16-Feb-09 11:51 am asterisk_command() Action: Login
16-Feb-09 11:51 am asterisk_command() Username: XXX
16-Feb-09 11:51 am asterisk_command() Secret:
16-Feb-09 11:51 am asterisk_command() Events: off
16-Feb-09 11:51 am DEBUG:
On Mon, 2009-02-16 at 16:30 +, Geoff Lane wrote:
Thanks for your suggestion. My Asterisk machine has MySQL, and I might
go down that route at some time. However, my query is part of an
almost academic exercise in which I'm trying to find out what AstDB is
capable of. Unfortunately your
On Feb 13, 2009, at 9:59 AM, John Todd wrote:
I've been involved with getting better data for running Asterisk on
the Amazon EC2 cloud computing system. Here are some calculations
I've made on costs based on current published prices on Amazon's
system. Feel free to tell me that I'm wrong
On Monday, February 16, 2009, Jared Smith wrote:
Hopefully that helps make things a bit more clear.
It does - many thanks for your help.
--
Geoff
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
Jerry Geis wrote:
I am getting a priveldged command error on the manager API.
16-Feb-09 11:51 am asterisk_command() Action: Login
16-Feb-09 11:51 am asterisk_command() Username: XXX
16-Feb-09 11:51 am asterisk_command() Secret:
16-Feb-09 11:51 am asterisk_command() Events: off
I need your help: please help test the gender detection module at 575-613-4392.
I wrote a gender detection module and thought I'd try it out. It only takes a
second. I've been showing 90%+ accuracy and I want
to make sure it's working correctly. Rain and significant background noise
seems to
Hi all,
When the Dahdi driver detects DTMF, it seems it's not muting the first 5-15 ms
and sometimes the last 2-10 ms of the DTMF tone. This shows up in recorded
voicemail greetings -- you hear a very short DTMF '#', or sometimes two blips,
at the end of the recording.
I have a Mitel SX-200
Hi!
I wonder what is teh meaning if vmsecret is not defined.
Does this mean that the voicebox can be accessed without PIN code? Or
does it mean that the voicebox can not be accessed at all (of course
except using the s Parameter of VoicemailMain()) ?
thanks
klaus
Michael Smith wrote:
When the Dahdi driver detects DTMF, it seems it's not muting the first 5-15 ms
and sometimes the last 2-10 ms of the DTMF tone. This shows up in recorded
voicemail greetings -- you hear a very short DTMF '#', or sometimes two blips,
at the end of the recording.
The blip
This module detects gender and approximate age range. I'm working on getting
it's accuracy to 80%+ on a consistent basis, after implementing filters to
remove background noise and other artifacts.
It's designed for a number of things. To start, I have several clients
(primarily mobile content
Kevin P. Fleming kpfleming at digium.com writes:
Michael Smith wrote:
When the Dahdi driver detects DTMF, it seems it's not muting the first
5-15 ms and sometimes the last 2-10 ms of the DTMF tone.
The blip at the beginning is expected; the DTMF detector won't trigger
until it has seen
On 17/02/2009 3:05 a.m., Michael Graves wrote:
Phasing tricks using stereo speakers are good as far as generating an
effect. That is, synthesizing the perception of some image width, but
not accurately reproducing an audible scene. There are many fine
commercial examples of this. I own a old
On 16/02/2009 10:23 p.m., Grygoriy Dobrovolskyy wrote:
Sorry about off-topic, but can you advise the mail client who is able to
organise the web mailing list topic as web interface does ? (i mean by
blocks/topics) I wold be glad to use something else with the same usability,
but dont see any
Hi,
When I transfer a call to an extension, the person I call does not have any
idea when that transfer happened so it is a guessing game.
Is there a way to send a beep to the caller just before transferring the call?
Preferably by setting something in FreePbx?
Sincerely,
robert
Ken D'Ambrosio wrote:
Hi, all. My subject line says it all: is there a WiFi SIP phone with VPN
abilities? Failing that, a WiFi phone that runs Linux? I already know
one phone that does meet my requirements -- the iPhone. The new software
comes with a Cisco VPN client, and a SIP client can
The DADHI function is probably intended for more generalized use. Maybe for
recording voicemail greetings it should not be used and a different function
used instead. There is no reason why it isn't possible to backup in the
recorded message and erase the blip. The detection time should
What about receiving Skype calls on Gizmo or other SIP device?
Looking into the website I don't see anything regarding that.
On Mon, Feb 16, 2009 at 8:05 AM, Olivier oza-4...@myamail.com wrote:
2009/2/13 John Todd jt...@digium.com
On Feb 13, 2009, at 11:19 AM, Philipp von Klitzing wrote:
Wilton Helm whelm at compuserve.com writes:
The DADHI function is probably intended for more
generalized use. Maybe for recording voicemail greetings it should not be
used and a different function used instead. There is no reason why it
isn't possible to backup in the recorded message
Hi all,
I am trying to make a scenario when someone dial *10*, the mp3player()
function would act and play a list of MP3 files.
However, I have no idea how to randomize the function (mpg123 is
capable of shuffling the MP3 files, buat how to implement it in
extensions.conf?)
Perhaps any of you
I'm thinking of starting a partyline, where people call in and talk to other
people. For record keeping and billing purposes, I'd like to go by the callers
telephone number.
This method works fine as long as the caller doesn't have callerid blocked, but
what are my options if they do block
The old classic is to say something like ' your callerid is blocked,
please get out your credit card'
PaulH
Alfred Monticello wrote:
I'm thinking of starting a partyline, where people call in and talk to
other people. For record keeping and billing purposes, I'd like to go
by the callers
I have written this configuration script which uses OpenSky to make Skype
calls directly from Asterisk devices using my companies SIP to Skype
gateway. Users can dial skype_anyskypeusername or manually add names or
extensions which can get mapped to the correct dialing sequence. The right
sequence
Looks like my provider is not passing dtmf correctly .. Had a serious
laugh, system kept asking me if I was ready., ended up finding myself
talking to the IVR .
On Mon, Feb 16, 2009 at 11:45 AM, Asterisk Asterisk
nt_aster...@yahoo.comwrote:
This module detects gender and
Hello All
We have started a voice portal for Parents and Students. They can listen
Grades, Attendance status and other relevant information over *phone*.
Please read features below and to listen Demo IVR call at *00 92 42 8315427.
*Initially we are deploying it for a School and planning to
On Fri, Feb 6, 2009 at 17:11, Jeff LaCoursiere j...@jeff.net wrote:
Anyone have much luck with these on ATA's? I have a few sites that use
them succesfully with multi-port Audiocodes boxes, but just connected ten
machines to Linksys 2102s and they are very flaky. Using u-law on a 100Mb
Hi,
How can I stress test an asterisk IVR? I am looking for some kind of
sip phone which can be programmed to send out digits after specified
time to simulate users pressing menu items. If it can originate large
number of calls simultaneously then it's great!
Does any one have any
Hi
[Feb 17 07:59:45] WARNING[21539]: channel.c:3477 ast_request: No channel type
registered for 'USTM'
[Feb 17 07:59:45] WARNING[21539]: app_dial.c:1502 dial_exec_full: Unable to
create channel of type 'USTM' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time
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