(with as much verbosity and
debugging enabled as possible) so that all factors involved can be
considered.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out
should have no effect on the existing registration.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
Fax For Asterisk
modules, you'll have to add noload = app_fax.so to your
/etc/asterisk/modules.conf file, so that you don't have two modules
trying to provide the same SendFAX/ReceiveFAX applications.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW
answers and the queue
application connects them together. Issuing an AMI Hangup on the call
leg that is ringing the agent's phone is exactly what you want to do.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype
On 01/09/2011 08:23 AM, mgra...@mstvp.com wrote:
Actually, all of the conference phones are known by the SoundStation
name and the desk phones are SoundPoint.
Sure enough... thanks for the clarification!
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive
the more basic G.722 codec in the
IP335/450/550/560/650/670 models.
The SoundPoint IP6000 and IP7000 conference phones (and maybe the
IP5000, I haven't checked) also support G.722.1 and G.722.1C.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW
the incoming channel as soon as
it hits the dialplan, then wait 3 or 4 seconds, then send the call
onwards to your actual FAX machine. FAX detection is really expected to
be used on calls that would otherwise be answered by a non-FAX endpoint
(IVR, voicemail, user with a phone, etc.)
--
Kevin P
causing unacceptable
audio disturbance, you should be fine using any of these codecs as well.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com
variants.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
. It was never intended to provide a failover or data-sharing
mechanism, and none of the code attempts to take that into account.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check
On 01/03/2011 07:08 PM, Steve Underwood wrote:
On 01/04/2011 04:22 AM, Kevin P. Fleming wrote:
On 01/03/2011 11:26 AM, Tom Rymes wrote:
Hi folks,
I was hoping that someone might be able to help clarify some confusion I
have on DAHDI Fax detection after spending some time searching. My
On 01/03/2011 06:47 PM, Thomas Rymes wrote:
On Jan 3, 2011, at 3:22 PM, Kevin P. Fleming wrote:
On 01/03/2011 11:26 AM, Tom Rymes wrote:
[snip]
1.) Echo cancellation is automatically disabled upon recognition of a
CNG tone, regardless of the faxdetect setting. This can only be disabled
in DAHDI affect that
behavior?
No, none of the Digium HW ECs detect and report CNG tones via the DSP;
CNG tone detection is still done on the host CPU. 'faxdetect' is not set
in DAHDI, it's set in chan_dahdi.conf.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis
On 12/28/2010 05:17 AM, Administrator TOOTAI wrote:
Le 27/12/2010 20:09, Kevin P. Fleming a écrit :
On 12/27/2010 12:37 PM, Administrator TOOTAI wrote:
[...]
d...@myphoneserver:/usr/src$ strings
/usr/src/asterisk-1.4.38/channels/chan_dahdi.so | grep '^DAHDI
Telephony'
DAHDI Telephony w/PRI
On 12/28/2010 07:19 AM, Administrator TOOTAI wrote:
Le 28/12/2010 13:10, Kevin P. Fleming a écrit :
On 12/28/2010 05:17 AM, Administrator TOOTAI wrote:
Le 27/12/2010 20:09, Kevin P. Fleming a écrit :
On 12/27/2010 12:37 PM, Administrator TOOTAI wrote:
[...]
d...@myphoneserver:/usr/src
On 12/28/2010 10:49 AM, Administrator TOOTAI wrote:
Le 28/12/2010 16:53, Kevin P. Fleming a écrit :
[...]
The Hx8 card manual (on the Digium website) clearly states on page 46
that the minimum required version of Asterisk for use with these cards
is Asterisk 1.6.0.1.
Hmmh, reading *before
was the voip-info page that documents exactly what you are looking for.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
is build with
libpri ;-)
d...@myphoneserver:/usr/src$ strings
/usr/src/asterisk-1.4.38/channels/chan_dahdi.so | grep '^DAHDI Telephony'
DAHDI Telephony w/PRI
DAHDI Telephony Driver w/PRI
Asterisk 1.4 has never had BRI support in chan_dahdi.
--
Kevin P. Fleming
Digium, Inc. | Director of Software
On 12/20/2010 07:08 PM, Dovey Forman wrote:
Thanks Kevin.
Did it work with Asterisk 1.2 because it ignored it?
I don't know specifically that Asterisk 1.2 ignored Required headers,
but it's certainly possible.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan
a solution for this
or not.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
does not
support it, it cannot process the INVITE request.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
for
a week (or two, sometimes three) before being declared 'ready', so fixes
made before the release date aren't necessarily included. The changelog
included in the release will always indicate what revisions are included
in it, though.
--
Kevin P. Fleming
Digium, Inc. | Director of Software
On 12/10/2010 03:26 PM, sean darcy wrote:
On 12/10/2010 02:57 PM, Kevin P. Fleming wrote:
On 12/10/2010 01:45 PM, sean darcy wrote:
This was supposedly fixed in 1.6.2 on November 22, 2010. So isn't the
fix in 1.6.2.15, released 12/8?
In any event, that bug has been declared fixed, so you
On 12/10/2010 04:18 PM, sean darcy wrote:
On 12/10/2010 05:01 PM, Kevin P. Fleming wrote:
On 12/10/2010 03:26 PM, sean darcy wrote:
On 12/10/2010 02:57 PM, Kevin P. Fleming wrote:
On 12/10/2010 01:45 PM, sean darcy wrote:
This was supposedly fixed in 1.6.2 on November 22, 2010. So isn't
to the Skype network, which is a set of
proprietary (and encrypted) communications protocols.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com
. If 'qualify' is enabled for the
SIP peer and it responds to OPTIONS pings quickly, Asterisk can reduce
the T1 timer value from 500ms down to 100ms, which drops the INVITE
timeout to 6.4 seconds... but it can't be any shorter than that without
violating the RFC requirements.
--
Kevin P. Fleming
Digium
to offer your product to someone in response to a question
they ask on this, list, please contact that person directly.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us
On 12/06/2010 08:12 PM, C F wrote:
Thanks Kevin.
Upto which version fo Dahdi works with 1.4.x?
If I understand your question properly, all versions of DAHDI are
compatible with 1.4.x. All versions of DAHDI are backward compatible.
--
Kevin P. Fleming
Digium, Inc. | Director of Software
know.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
and
all future versions only support DAHDI.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
On 12/03/2010 01:17 PM, Steve Murphy wrote:
On Wed, Dec 1, 2010 at 12:15 PM, Kevin P. Fleming kpflem...@digium.com
mailto:kpflem...@digium.com wrote:
On 12/01/2010 01:05 PM, Steve Murphy wrote:
Hello,
I wonder if anyone else has noticed this.
I see
be better in the
long term than trying to convince an ancient alarm panel's modem to work
over a packet network.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out
see this?
This problem may be in the license file checking code... I've just taken
a quick look at it, and there may be at least one code path that leaks a
pair of pipe file descriptors. I'll enter an internal issue to get this
addressed ASAP. Thanks for the report.
--
Kevin P. Fleming
Digium
direct media
paths.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
itself right now. The changes to support
direct media paths for UDPTL wouldn't be terribly difficult, but nobody
has done the work yet that I know of.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber
channel drivers don't
manipulate the audio content of the channels, they pass it through.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com
!
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
series cards compatible with this kind
of service? Seems like the service would look like a PRI interface, but
I'm not sure. The office is in Singapore.
Yes, you are right. That's an E1 circuit, configured with 10, 20 or 30
active B-channels.
--
Kevin P. Fleming
Digium, Inc. | Director of Software
for the purpose of T1 to E1 conversion?
Yes, the cards in question can handle some ports configured as T1 while
others are configured as E1.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem
failed; aborting.
What can I do to enable it?
What you can do is read the documentation. The built-in help for the
SendFAX application shows you how to enable audio FAX on channels that
support T.38 (where audio FAX mode is normally disabled for reliability
reasons).
--
Kevin P. Fleming
Digium
On 10/20/2010 11:35 AM, VoIP Question wrote:
On Wed, Oct 20, 2010 at 4:25 PM, Kevin P. Fleming kpflem...@digium.com
mailto:kpflem...@digium.com wrote:
This was fixed in Asterisk 1.6.2.12 and later releases, so if you were
running the current version, you wouldn't have experienced
will go a long way
towards helping you be able to resolve these issues on your own.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
be an
application to respond to it, and chan_sip will (rightly) assume that
T.38 cannot be used on this channel so it will respond with a 488.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
they read (and sometimes respond) to this list, so I don't
understand why they don't clarify this issue.
When you are asking for free help on a mailing list, patience is a
virtue :-) You posted your question approximately four hours ago.
--
Kevin P. Fleming
Digium, Inc. | Director of Software
(resulting in a low signal-to-noise
ratio), and when the listener increases the volume level on their
listening device, the noise level will be increased along with it. For
these sorts of tasks, you really do want the source material recorded at
a fairly high volume level.
--
Kevin P. Fleming
Digium
(if not
millions) of endpoints registering to Asterisk systems all over the
world every day using this mechanism and it works just fine. If it's not
working for you, there is some sort of configuration problem.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW
itself. It will also record this address
and port number as the location of that peer for future INVITE messages
to be sent to it.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
the problem. In general, Asterisk
works just fine with endpoints that are behind NAT devices and never
send their external IP addresses in their SIP messages... there are
probably millions of devices working that way every day.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan
asterisk 1.6.2.13
The way to resolve them is to have whatever device is sending your
system invalid RTP packets stop doing so.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us
'res_fax_digium.so' could not be loaded.
any help will be much appreciated!!
It will be very hard to help you with the information you provided; at a
minimum we need to know what version of Asterisk and of the FAX modules
you tried to use.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
bit machine)
You are trying to use FAX modules for Asterisk 1.4.x with Asterisk
1.6.2.11. Did you use the FAX download selector to get links to the
proper modules to use for your version of Asterisk?
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW
to transport IP over X.25 networking, although I doubt anyone
uses X.25 for that purpose any more.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com
' feature is still marked *experimental*, and that is
primarily because it defeats much of Asterisk's normal behavior; in
addition, there a quite a few normal, working call scenarios for which
it will fail... so it's there, but if you use it, you can expect
difficulties.
--
Kevin P. Fleming
Digium
archives would give you pointers
to the methods you can use today to achieve this.
Asterisk 1.8 was just enhanced to provide some new APIs that will be
necessary for seamless implementation of T.38 gateway mode, and we
expect that work on that will occur in the very near future.
--
Kevin P. Fleming
filename,
instead of using a symbolic link on the filesystem.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
compression :-)
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
solutions out of a forum like that, so a
blanket fee strategy must have been specifically chosen to skew things
in a particular way. Seems like it worked very well.
There is no fee to list free products on AsteriskExchange.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445
*only* be sent to the 's' extension in the target context,
since there is no target number passed over the FXO connection. You'll
have to create an 's' extension to handle incoming calls however you like.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW
loaded.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
.
Alternatively, depending on how you've built your firewall, you can
insert the 'drop all packets from X.X.X.X' *before* any rules that allow
packets from existing connections.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype
slot, though, so yeah... moving the card
to another slot may also be required.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
, as it would
have to remember separate timeouts for each of the originally-dialed
destinations in case they get forwarded elsewhere.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check
in the relevant
modprobe configuration file.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
For last three scenarios Asterisk should work as fax T.38 gateway. Is
it possible?
There is no support for T.38 gateway mode in Asterisk 1.8, although
there is still work on that front. The patches in the issue tracker may
have been updated for Asterisk 1.8 already, though.
--
Kevin P. Fleming
Digium
increase your chances of
success.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
the ExternalIVR protocol to allow
the external process to rewind/fast-forward the file being played back,
and then it could do that based on receiving DTMF input.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming
changes) when changes are deemed necessary.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
for a bridged call so that the media does not have to make
as many hops as the signaling does. The media still moves on the same
ports as the signaling packets, using the same protocol.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL
On 08/13/2010 03:48 PM, Lyle McKarns wrote:
Mostly I was wondering if there are any reasons I cannot
1) Use and AMD board and
2) Run a mixed Intel/AMD enviroment
What is a 'mixed Intel/AMD environment'? It's not possible to have both
Intel and AMD processors in the same system.
--
Kevin P
effect is if you compiled binaries specifically for one family
of processors and used them on the other. As far as how the software
operates, by definition the processor type/family does not matter at all.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW
to an
AMD-based system, because the data *outside* the system was the same.
That was my point. There are many CPU family-specific optimizations that
can be used for various parts of Asterisk, but in the end they don't
affect how Asterisk operates, only the speed at which it does so.
--
Kevin P. Fleming
On 08/05/2010 06:25 PM, Roderick A. Anderson wrote:
Kevin P. Fleming wrote:
On 08/05/2010 03:52 PM, Roderick A. Anderson wrote:
I have a Linux-Vserver guest running CentOS 5.5 with Asterisk 1.6
installed from the asterisk.org and digium.com repositories.
I have Asterisk starting (service
and digium.com yum repositories?
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
On 08/02/2010 02:34 AM, Siju George wrote:
Hi,
Is there any Free software that can connect to an Asterisk Server and
Do video Conferencing? or atleast one to one video chat?
One to one video chat is already supported by Asterisk, using SIP or
H.323 video phones.
--
Kevin P. Fleming
Digium
On 07/28/2010 08:20 PM, Landy Landy wrote:
Jeremy,
Thanks a lot that helped and solved the problem. I had it as:
voice=Marta-8kHz before and that didn't work and now changed it to
voice=Marta.
That's because you only have the Marta-16kHz voice installed.
--
Kevin P. Fleming
Digium, Inc
, but there's no way to know that without testing the specific
environment.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
interaction with Asterisk trivially
easy, and handles all the AGI/AMI stuff 'under the covers' for you.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com
a way to reduce the burden on our development team during the beta
testing period.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com
and simplify the configuration a bit.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
, as has already been
pointed out in this thread. No need to reinvent this wheel :-)
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com
is acceptable.
In addition to the suggestions of Zoiper, there is also Blink, although
their primary version is on OSX and the Linux/Windows versions are just
now arriving in early releases.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL
.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
--
_
-- Bandwidth
The file in question is probably part of Flash Operator Panel, in which
case it is readily available in many other places on the Internet already.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem
extractor. Here's
an example:
http://www.shireeninc.com/poe-extractor.html
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
On 07/23/2010 04:40 PM, bruce bruce wrote:
You can also use Ethernet Over Power Lines solution or wireless :-)
His issue wasn't getting the network connection delivered, it was the
power :-)
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW
file does not... so Asterisk will try to play the contents of that
header as alaw data, presumably producing terrible noise.
The best you can do is to use sox to convert them from
alaw-in-WAV-container to raw-alaw.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis
it. There are other Skype gateway solutions that use a similar
method, but they are not free.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com
the regular
Skype client.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
, and then do your logic in the context/extension you
specified before performing the actual dial operation.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out
as if it was
CED, which is the way ANSam was designed to operate on echocan units
that don't have specific ANSam detectors.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check
to automatically add users to the buddy list when they
request it. Instead, manually add users B and C to A's buddy list
(using
a regular Skype client), and those are the only users that will be able
to call A.
--
Kevin P. Fleming
I know that already, it's a matter of convenience.
If I go
are already using 'commercial' Fax for Asterisk (not Asterix).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
to automatically add users to the buddy list when they
request it. Instead, manually add users B and C to A's buddy list (using
a regular Skype client), and those are the only users that will be able
to call A.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW
is a B2BUA UA, so
the two SIP dialogs involved in a 'call' are completely separate.
Asterisk does not have any support for 100rel or PRACK.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem
in recent releases of FFA; there was a bug
previously where the module would cause Asterisk to crash if a document
to be sent could not be queued (for one of many reasons).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype
not reply.
Threads cannot be deleted from the list; once messages are posted, they
appear in the archives (of which there are many) and are delivered to
thousands of subscribers. Sorry.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL
, and when the outbound call
gets delivered to voicemail, since that appears to be 'answered' at the
network level as well.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us
and more effective than trying to put conditional logic and other
programming constructs into the configuration file reader.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us
code; keeping
the decryption keys private would not violate the GPLv2 at all.
How does obtaining a commercial license from Digium provide the poster a
'legitimate' way to secure his configuration files?
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW
On 07/07/2010 03:33 PM, Tilghman Lesher wrote:
On Wednesday 07 July 2010 14:58:05 Kevin P. Fleming wrote:
On 07/07/2010 10:52 AM, Tilghman Lesher wrote:
On Wednesday 07 July 2010 05:24:10 A J Stiles wrote:
On Tuesday 06 Jul 2010, ABBAS SHAKEEL wrote:
Hello Community,
. I am facing
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