Have a happy MayDay!
I cannot see the whole mail now, it's clipped for my mobile, but it
seems the nth bizarry of new alsa config file, that creates an hdmi
device even if you do not have one. Try to edit
/usr/share/alsa/alsa.conf or any other file in /usr/share/alsa dir and
delete any mention of
Anthony Minessale wrote:
Also is there any way to stop uuid_broadcast as I'd
need to stop it somehow if the destination picks up?
break all
"uuid_broadcast phrase::saynumber,1" doesn't set the
'current_application_response' variable in the same way as
"uuid_broadcast playback::filen
Sounds like the MSI Wind :P I had to do the same thing!
/b
On Apr 30, 2009, at 11:16 PM, Nandy Dagondon wrote:
rhino used the dual-core atom mobo d945gclf2 but it requires
downloading/building the linux r8168 LAN driver.
-nandy
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http
rhino used the dual-core atom mobo d945gclf2 but it requires
downloading/building the linux r8168 LAN driver.
-nandy
On Fri, May 1, 2009 at 11:40 AM, Brian West wrote:
> I have two intel atom boxes sitting on a shelf above my desk ... works like
> a charm!
> /b
>
> On Apr 30, 2009, at 10:31 PM
I'm getting Windows Express compiler errors on the latest svn update to trunk
13213.
It looks like the path is wrong to some files.
Instead of folder "Debug", it's looking for files in folder "Debug DLL"
Mark.
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I have two intel atom boxes sitting on a shelf above my desk ... works
like a charm!
/b
On Apr 30, 2009, at 10:31 PM, Mitch Capper wrote:
You may want to look at the Intel Atom combo machines you can get a
1.6 ghz machine probably for around $100-150 USD in a very small
form factor and v
You may want to look at the Intel Atom combo machines you can get a 1.6 ghz
machine probably for around $100-150 USD in a very small form factor and
very powerful.
~Mitch
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hi guys,
i've installed FreeNas using CF-to-IDE adaptor and SanDisk 128MB CF. it's
working fine. but i want to try FS on a 16GB Kingston CF. anyone tried this?
if none, i can also settle down for 8GB. pls mention which brand/size works.
tks,
nandy
On Thu, Apr 30, 2009 at 10:46 PM, Fred-145 wr
Hello,
I am trying to get skypiax working, but I am having trouble with the sound. The
calls fail with CALL FAILUREREASON 7 = Sound I/O error and
I am getting the following error:
ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi
I am running centos 5.3 and have f
the mailing list is not the correct place to report issues.
http://jira.freeswitch.org
We have 8 people conferences that last as long as 12 hours a day every day
and there
is no delay.
Be advised if you open a jira it will require that you download and compile
and retest your issue on SVN trunk.
Have you tried on non-ec2 installs? Maybe some setting on the EC2
instance is messing with it. Also don't hijack threads please! :)
/b
On Apr 30, 2009, at 3:54 PM, Chris Fowler wrote:
I'm using FreeSWITCH (Build 13168M) and we're having intermittent
multi-second delays on conference brid
Somehow 60 sec. interval works on both phones just fine. Appreciate
everybody's input.
Nik Middleton wrote:
> Xlite may be working on the timeout FS is sending.
>
> See the following from the wiki and see if that helps, but I'm not sure
>
>
> In domain, set
>
>
> -Original Message-
> F
I'm using FreeSWITCH (Build 13168M) and we're having intermittent multi-second
delays on conference bridges with more than three participants (this is not a
new issue - just bubbled to the top of the stack to address).
The server is running on Amazon's AWS c1.medium instance, CentOS 5.0 with
Ke
Nik Middleton ha scritto:
> Xlite may be working on the timeout FS is sending.
>
If Xlite is monitoring any user when this user reboot/unregister (then
Xlite get a publish_out event) when
that user come back online then Xlite is never notified until it
re-register/or re-start
Anyway i don't c
Xlite may be working on the timeout FS is sending.
See the following from the wiki and see if that helps, but I'm not sure
In domain, set
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
paul.deg
Can you point out any place we do sub milli second sleeps? The timer
thread should be doing 1ms, I can't think of any that would be less.
MIke
On Apr 30, 2009, at 2:28 PM, Paweł Pierścionek wrote:
> Hi,
>
> With really old kernels (100Hz) if You do sleep(1ms) You sleep for
> 10ms on average
Then it would be recommended to not do tickless clock :P
/b
On Apr 30, 2009, at 1:28 PM, Paweł Pierścionek wrote:
Hi,
With really old kernels (100Hz) if You do sleep(1ms) You sleep for
10ms on average.
With enterprise kernels (250Hz) Your sleep resolution increases by a
factor of 4.
With
Hi,
With really old kernels (100Hz) if You do sleep(1ms) You sleep for
10ms on average.
With enterprise kernels (250Hz) Your sleep resolution increases by a
factor of 4.
With fresh kernels (1000Hz) You get real 1ms timer resolution -
10fold increase compared to old kernels.
With ti
PHP works with ESL which not entirely unlike AGI
On Thu, Apr 30, 2009 at 10:20 AM, Ross McKillop wrote:
>
>
> Brian West wrote:
> >
> > Ross,
> > I can see no reason to have the conference module do that for you
> > when there are so many ways to do that externally with javascript, lua
>
Sorry in my eagerness I seem to have sent my original email with a few
words missing. Too much cut and paste!
Q1 - I have looked on the wiki and was unable to find a list of RFC's
that FS is intended to comply with.
The page http://wiki.freeswitch.org/wiki/Specsheet#Protocols has a list
of SIP
I can confirm that we comply with rfc 5411 in that we agree that is a
list of sip specs that we may or may not honor, and that we may or may
not have ever seen or read. Joking aside, the sofia list is pretty
good, there are some things noted as it would be implemented in the
application.
Brian West wrote:
>
> Ross,
> I can see no reason to have the conference module do that for you
> when there are so many ways to do that externally with javascript, lua
> or any other the other languages then you can inject the sound file
> into the conference on demand before you dr
Richard Lamkin ha scritto:
Q1 -- I have looked on the wiki and was unable to find a list of RFC's
that FS is intended to comply with.
The page http://wiki.freeswitch.org/wiki/Specsheet#Protocols has a
list of SIP protocols by name but these have no RFC number against them.
Have I just miss
Correction: 2 min. registration timeout does not work for either
Grandstream 386 nor for X-Lite.
Will try 1 min., but I am skeptical. Grandstream has other weird issues
btw, like not getting dial tone from first attempt or sometimes giving
buzzing noise instead of one.
My fw is fairly old Netge
Thanks guys for the links on CF-to-IDE adaptors.
--
View this message in context:
http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23317579.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.
___
Freeswitch-users mai
Q1 - I have looked on the wiki and was unable to find a list of RFC's
that FS is intended to comply with.
The page http://wiki.freeswitch.org/wiki/Specsheet#Protocols has a list
of SIP protocols by name but these have no RFC number against them.
Have I just missed the page?, if not is there any p
Worked for Grandstream, but not for X-Lite.
Nik Middleton wrote:
> Don't know where the setting is in FS, but force them to register every
> 120 seconds and see if that helps
>
> Regards,
>
> -Original Message-
> From: freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-user
Ross,
I can see no reason to have the conference module do that for you
when there are so many ways to do that externally with javascript, lua
or any other the other languages then you can inject the sound file
into the conference on demand before you drop the participant in. I
like simp
As a former user of the app_confcall (http://www.freeswitch.org/node/100)
Asterisk module produced by FreeSWITCH I'm in the process of moving a number
of Asterisk-based services to FreeSWITCH and am trying to find the
equivalent of the "record name before enter" feature of app_confcall.
It seems
As a former user of the app_confcall (http://www.freeswitch.org/node/
100) Asterisk module produced by FreeSWITCH I'm in the process of
moving a number of Asterisk-based services to FreeSWITCH and am trying
to find the equivalent of the "record name before enter" feature of
app_confcall.
It
> After 6 months of discussions with Attractel, today we finally got a new
> version of Zoiper Bizz, which works with TLS and SRTP (previous versions
> only supported TLS).
> I have added the info, how to set it up, in the wiki
> http://wiki.freeswitch.org/wiki/Interop_List#Zoiper_Bizz_2.10_and_TLS
Paweł Pierścionek
writes:
G'day Paweł.
> boot Your kernel with "divider=10 nohz=off" options :)
>
> Recent kernels are tickless which basically causes all freeswitch
> timers/sleeps to fire at requested microsecond intervals. With nohz
> kernels You get hundred times more system calls with free
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