Yes, you may also link (or copy) the .gdbinit file found in the
support-d folder to your home directory.
This is going to enable some GDB macros written for FS.
Once thats done you can do the following commands and include them too:
list_sessions
hash_it_str_x session_manager.session_table swi
Hi Michael,
I checked on wiki, is the following the good way to go (sorry I'm not
very familiar with your debugging tool).
$ gdb bin/freeswitch core.xxx
bt
bt full
thread apply all bt
thread apply all bt full
If I understand well I have to rerun the tests, as I did not start FS
using GDB.
Woof!
On Mon, 02 Mar 2009 19:32:53 -0500, Steve Underwood wrote:
I just had a look through that patent. Its amazing. There is a lot of
> focussed descriptive text, but a patent only really consists of its
> claims. Those claims are astonishingly open-ended, and characterise what
> people had be
I think you need to talk to Brian.
Apparently this is a "new" pocketsphinx which works on a different format from
those found in the pizza demo.
Also, pocketsphinx crashes if it "hears" anything outside the grammar which
apparently is a longstanding bug. Brian mentioned they are working o
Andy Spitzer wrote:
> Woof!
>
> On Sun, 01 Mar 2009 21:28:18 -0500, Brian West wrote:
>
>
>> NO. You want something that people THINK exists and works well...
>> Reliable human/voice detection doesn't exist in ANY form.
>>
>
> I beg to differ. See http://www.freepatentsonline.com/5521967
Hi,
mod_vmd is a bit more sophisticated than that. It looks for the signal
being narrowband energy. However, mod_vmd isn't very reliable, as it
takes a rather high SNR for its narrowband detector to work. So high
that a lossy codec like G.711 can barely manage it.
Regards,
Steve
Anthony Mines
Andy Spitzer wrote:
> Woof!
>
> On Sun, 01 Mar 2009 21:28:18 -0500, Brian West wrote:
>
>
>> NO. You want something that people THINK exists and works well...
>> Reliable human/voice detection doesn't exist in ANY form.
>>
>
> I beg to differ. See http://www.freepatentsonline.com/5521967
Thanks Addison.
The Pizza files are there (as mentionned is it a copy of an already
working system).
In fact freeswitch is complaning about
/usr/local/freeswitch/grammar/model/communicator which he cannot load
So somehow freeswitch is not willing to open the files, but I have no
clue why. So any h
Peter,
You need the grammar files for the pizza demo:
http://wiki.freeswitch.org/wiki/Mod_pocketsphinx#Testing_with_the_Pizza_Demo
has lonks to premade fles for everyhting to get the pizza demo working
with pocketshinx. Those to not come with the source code when you
update from SVN.
Nik
On M
Since you did not describe the exact way you are doing it with enough detail
or any trace I can't begin to tell you
what your problem is. you did not even mention what variable you are using
or show examples.
All I can do is tell you again that if you set the
origination_caller_id_number in the d
On Mon, Mar 2, 2009 at 1:58 PM, Peter P GMX wrote:
> Hello Anthony,
>
> sorry for being tenacious but in some cases it works in a way we need it:
> If I a am not suppressing the cid numer when calling A, the following
> scenario works:
>
> * A receives a Call (originate) with CID '00' (
Hello Anthony,
sorry for being tenacious but in some cases it works in a way we need it:
If I a am not suppressing the cid numer when calling A, the following
scenario works:
* A receives a Call (originate) with CID '00' (default from
switch_caller.c)
* A dials some digits v
In asterisk, with the parameter AMPBADNUMBER = FALSE it is possible to use
"early dial" Grandstream telephones. How do Freeswitch in?
thank you very much.
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Some more info:
the system I am working on is a copy (dd copy) of a system where the
pizza demo works on.
The only thing I changed was to update to the current freeswitch trunk
12293 (it was 10003 before).
Do I need to update the model? I did a make in the model directory, but
no change.
Best reg
On Mon, Mar 2, 2009 at 11:48 AM, Anthony Minessale
wrote:
> i think that's what mod_vmd does
>
I think that's right. It just does the opposite - instead of looking
for differing power levels it looks for the same power level. In other
words it tries to detect distinctly non-human sound. I'll bet y
origination_caller_id number is not ok as a variable unless its in {} as
part of the dial string
it's an exception that is parsed before the channel is even created.
I think you are drawing the wrong conclusion about what works and doesn't
work.
If you can produce a dial string that contains
{orig
i think that's what mod_vmd does
On Mon, Mar 2, 2009 at 11:16 AM, Andy Spitzer wrote:
> Woof!
>
> On Sun, 01 Mar 2009 21:28:18 -0500, Brian West
> wrote:
>
> > NO. You want something that people THINK exists and works well...
> > Reliable human/voice detection doesn't exist in ANY form.
>
> I
On Fri, Feb 27, 2009 at 6:07 AM, Helmut Kuper wrote:
> Hello,
>
> I play around with record_session and would like to have caller and
> callee separated on left and right channel. I found record_stereo is
> used for this. Unfortunately it doesn't work. A and B leg are still
> mixed. Additionally I
Woof!
On Sun, 01 Mar 2009 21:28:18 -0500, Brian West wrote:
> NO. You want something that people THINK exists and works well...
> Reliable human/voice detection doesn't exist in ANY form.
I beg to differ. See http://www.freepatentsonline.com/5521967.html for one way
to do it. It works rathe
Hello Anthony,
I do this when I orginate the call. This way we suppress the cid when we
call party A and transfer A to an internal extension (our callback
application).
But now comes the part that does not work:
After A enters the target number B (via DTMF), we set the cid variables
via uuid_setva
I think any issues we have are related to pri, the analog doesn't seem
to generate any major bug reports.
Mike
On Mar 2, 2009, at 6:47 AM, Fred wrote:
> Thanks guys for the feedback. So, the OpenZap driver isn't ready for
> production yet?
>
___
Fr
Could you please post this to jira along with a thread apply all bt of
a core file taken from the process with the stuck sessions.
Mike
On Mar 2, 2009, at 2:06 AM, rod wrote:
> Hi All,
>
> I ran some longer tests with FS 1.0.3 acting as an SBC.
> The test machine has the following specs:
>
I have no background in telephony but probably need to use a PBX.
FreeSwitch was recommended by a casual contact so I would like to start
first by setting up a small test.
I have a SPA3102 attached to the box running FS and to a ordinary phone
line. I registered SPA in conf/directory/default/line
Hi.
I have little background in telephony and need to use a PBX but would like
to start first with a small test set-up.
I have a SPA3102 attached to the box running FS and to a ordinary phone
line.
I registered SPA in conf/directory/default/line1.xml and this works to a
point but I can't get cal
yes if you match the job uuid from bgapi to the SWITCH_EVENT_BACKGROUND_JOB
event, you would get the result in that event.
On Mon, Mar 2, 2009 at 8:49 AM, Nik Middleton <
nik.middle...@noblesolutions.co.uk> wrote:
> That’s what I was wondering, however, won’t the response to the bagi (not
> the
That's what I was wondering, however, won't the response to the bagi
(not the initial) give me the info on the call result?
Regards
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
An
The best way would be to add a few custom variables and add a secondary
system that monitors the CDR data and uses the
custom variables to identify what you want to do with the failed calls.
On Sat, Feb 28, 2009 at 4:49 PM, Nik Middleton <
nik.middle...@noblesolutions.co.uk> wrote:
> Hi Guys,
put origination_caller_id_number in the dial string of any call and you can
set the caller id individually for that leg
{origination_caller_id_number=1234}
On Sun, Mar 1, 2009 at 3:38 PM, Peter P GMX wrote:
> Hello,
>
> I have the following problem while providing callback (mod_eventsocket
> i
pardon?
ESL is just a client library for event socket to make it easier to make
event socket apps.
ESL == Event Socket Library
On Mon, Mar 2, 2009 at 3:29 AM, Gopal krishnan wrote:
> Hi,
> Actually what is the difference between ESL in FS 1.0.3 and event socket
> in FS 1.0.2. Is the FS 1.0.3 E
Rajagopal, Sridhar (Sridhar) wrote:
> Hi all,
>
> I am planning to run freeswitch on powerpc MPC8358. Please let me know if any
> changes needs to be done in the code
>
> Regards
> Sridhar
>
It may be easier to say what will currently stop Freeswitch working.
The lack of an MMU is a problem ri
Sridhar,
PIKA's WARP is PowerPC based...AMCC but still Big Endian and PowerPC.
From what I remember the endianness definition was broken in one or
two places, but other than that it was effortless (native compilation).
Thanks,
Wojtek,
On Mar 2, 2009, at 7:11 AM, Giovanni Maruzzelli wrote:
> O
On Mon, Mar 2, 2009 at 12:52 PM, Rajagopal, Sridhar (Sridhar)
wrote:
> I am planning to run freeswitch on powerpc MPC8358. Please let me know if any
> changes needs to be done in the code
Hi Sridhar,
I don't think someone has tried that. It will probably be you that let
us all know which (if an
Hi all,
I am planning to run freeswitch on powerpc MPC8358. Please let me know if any
changes needs to be done in the code
Regards
Sridhar
> -Original Message-
> From: freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On
> Behalf Of
Thanks guys for the feedback. So, the OpenZap driver isn't ready for
production yet?
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Hi Fred,
Yes you can use Sangoma USB FXO with your laptop. You need to install
openzap for this. But for testing you can use this driver. Still there is
some issue with Openzap with FS as for as I used. while installing Sangoma
USB FXO device you need to use beta drivers.
On Sun, Mar 1, 2009 at
Hi,
Actually what is the difference between ESL in FS 1.0.3 and event socket
in FS 1.0.2. Is the FS 1.0.3 ESL superior?
On Fri, Feb 27, 2009 at 6:43 PM, Rex_Alex wrote:
> Hi All, I did what you have all suggested. Now its working perfectly.
> Thanks a lot for all your assistance. Rex.
>
> Raym
Hi Rex,
Please find the attached file for the PHP script. This script has been
executed in FS 1.0.2. put these two scripts in htdocs directory. access the
http://localhost/sample2.php so that two text box will appear. you can able
to give the extension number and mobile number to dial. Try this :
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