Thanks, done.
2009/12/7 Michael Jerris m...@jerris.com:
Please report bugs to jira.freeswitch.org.
Mike
On Dec 6, 2009, at 11:45 PM, Seven Du wrote:
Hi,
I know there's some chang on att_xfer, and after upgrade(re-bootstrap)
to trunk code, no sound after att_xfer.
Then I rebuild FS
Have a look at mod_conference
http://wiki.freeswitch.org/wiki/Mod_conference
On Sat, Dec 5, 2009 at 12:47 PM, shehzad p pmh...@gmail.com wrote:
Hello Every one,
I have to design conference, and I need community guidance to efficiently
accomplish that.
I need to create Conference which
Here is what I found...
I tried high-priority scheduling as per your suggestion, reniced the program
explicitly, rewrote timer thread to sleep on cond. variable and activate
only when there are timers and only when the timer actually had to be
clicked, turned off SQL thread and removed polling
I am changing the 3pcc setting because one of my gateways sends INVITEs
without SDP. I will try to update to the latest trunk today and capture
traces as Anthony described. If I can't do it today, it might be at the end
of the week.
Best Regards,
Jerry
_
From: Michael Jerris
Did you do each thing alone too to tell the difference?
-hp alone, disable monotonic alone (i did not see you mention the disable
monotonic)
as for your 4ms thing, yes we require high resolution timing, if we ask to
sleep 1000 microseconds that is what we need it to sleep for or at least as
close
I will certainly shchedule time for the upgrade. Thanks for the answer
On Fri, Dec 4, 2009 at 1:51 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
we changed that message a long time ago so people would not think that
anymore
We are now 3000 rev beyond the version you are at, I
One thing that I forgot to mention, these 2 FreeSWITCH servers are getting
calls with load balancing from another switch. Thus, the traffic type are
pretty much identical and both FSs have exactly the same on configuration. Any
suggestion would be appreciated. Thank you.
One of the properties of -hp is to enable memlockall() which means disable
swapping. This causes all memory used by FS to be resident permanently and
is much more costly in memory usage. -hp also uses a RR scheduler runs the
process at a less nice level and increases a few other process ulimits.
Anthony,
Thank you for your clear response. Based on your recommendation, if I want to
route more calls to the first server, should I take off -hp, or it's better
to run with it. We are running FS for pass-thru traffic with signaling only.
From: Anthony
maybe you can try both ways and see if there is a significant difference?
I think -hp would help more if you were doing media than if you were not but
that does not mean it could not still help performance but really the extra
performance would only show up once you had consumed all the resources
oh and also
use top -H to see which threads are using specific CPU and try to cross
reference them by attaching with gdb and dumping all the thread bt
On Mon, Dec 7, 2009 at 10:16 AM, Michael Jerris m...@jerris.com wrote:
Also I have seen some people reporting that the new tickless timers in
yes if you use the lua odbc sql plugin you should be able to use that for
sqlite, they may also have a native one.
On Sat, Dec 5, 2009 at 9:21 PM, Steve Klein skl...@singular.com wrote:
Greetings. We are attempting to add sqlite access to an IVR application
we are prototyping. We are using
On Sun, Dec 6, 2009 at 11:30 PM, Samuel Abekah-Mensah ab...@greatiam.comwrote:
Pardon me if this has been addressed already.
How does one go about having in the simplest instance 2 servers
registering with each other on startup whereby the users registering
would be able to call each other.
I have downloaded and compiled freeswitch, and it runs fine, can compile
everything without error including spandsp, but can't get esl to compile. My
version is earlier than the snow leopard that is mentioned in the general
install docs, and I have tried it with and without the compiler
Hi!
How can I access the variables that are defined in a users xml file?
For example, say user 1000 has a variable called smsnumber, as defined below:
include
user id=1000 mailbox=1000
params
param name=password value=1000/
/params
variables
variable name=smsnumber
The build system for libesl and everything below that won't work 100%
on the mac just yet. You have to make some changes to how its linked
and you'll have to compile php yourself to get everything in there
properly. The perl one however is much easier to fix.
-SOLINK=-shared -Xlinker -x
Forgive me if I ask the obvious questions...
Did you make in src/libs/esl before doing make phpmod ?
Did you install the php-devel stuff?
-MC
On Mon, Dec 7, 2009 at 9:27 AM, Kendall Stauffer k...@ksac.com wrote:
I have downloaded and compiled freeswitch, and it runs fine, can
compile
On Mon, Dec 7, 2009 at 10:11 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
How can I access the variables that are defined in a users xml file?
For example, say user 1000 has a variable called smsnumber, as defined
below:
include
user id=1000 mailbox=1000
params
I have a Problem with continue_on_fail.
I have setup a hunt group
action application=set data=continue_on_fail=NO_ANSWER,USER_BUSY/
action application=bridge
data=sofia/external/2...@10.11.12.243,sofia/external/2...@10.11.12.234,sofia/external/2...@10.11.12.188,sofia/external/1...@10.11.12.245/
Any direction on where to start would be appreciated. I am trying to get
freepbx working with this, and everything works (I think) except esl
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West
Sent: Monday, December
I did make first, but did not install any extra dev stuff, thinking I already
had them. Is there a way to turn on verbose and finding out exactly what it no
there that is expected?
Thanksmuch!!
From: freeswitch-users-boun...@lists.freeswitch.org
Hi!
That's exactly what I want to do and that was the first thing I tried,
but nothing is passed to the script.
In a case like this, what defines if variable smsnumber is taken from
the A path or B path? (The A path does not have smsnumber defined)
On Tue, Dec 8, 2009 at 5:25 AM, Michael
Maybe, just maybe isse that make target to reconf libtiff?
Regards,
JM
On Thu, Dec 3, 2009 at 6:24 AM, Jingwei Yang jingwei.y...@gmail.com wrote:
I installed libjpeg-7 following this website:
http://www.linuxfromscratch.org/blfs/view/svn/general/libjpeg.html. And
the previous error is
When I got the latest trunk the make gets an error. Should I perhaps
disable the mod_amr?
making all mod_amr
make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'.
Stop
The method I used to get the latest trunk follows:
svn checkout
Michael Collins wrote:
On Sun, Dec 6, 2009 at 11:30 PM, Samuel Abekah-Mensah
ab...@greatiam.com mailto:ab...@greatiam.com wrote:
Pardon me if this has been addressed already.
How does one go about having in the simplest instance 2 servers
registering with each other on startup
Check out
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user
you might just need to set the user so that the vars become available on the
leg you're processing.
-MC
On Mon, Dec 7, 2009 at 10:37 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
That's exactly what I want
Hi all,
I'll slowly pulling my hair out on this one. I had FS successfully
hanging up both legs on a bridge, now today, with nothing changed, I'm
not seeing a hangup of the b leg at all.
FS is behind a PIX, so it might be a weird NAT issue, but A leg calls
hangup just fine. Before when I
Perfect...
action application=set_user data=${dialed_extensi...@${domain}/
works like a charm.
Thanks Mike.
On Tue, Dec 8, 2009 at 5:56 AM, Michael Collins m...@freeswitch.org wrote:
Check out
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user
you might just need to set
Hello,
i now changed the $${domain} name of the server to the domain name the
phones register with.
Now messaging (MWI, notify) works.
Best regards
Peter
Peter P GMX schrieb:
Hello Anthony,
I did some checks today
Here is how the phones are registered:
mysql select sip_host,
try rerunning the ./bootstrap.sh
On Mon, Dec 7, 2009 at 12:49 PM, Jerry Richards
jerry.richa...@teotech.comwrote:
When I got the latest trunk the make gets an error. Should I perhaps
disable the mod_amr?
making all mod_amr
make[5]: *** No rule to make target '/mod_amr.c', needed by
On Mon, Dec 7, 2009 at 11:09 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Perfect...
action application=set_user data=${dialed_extension}@
${domain}/
works like a charm.
Another satisfied customer! :P
___
FreeSWITCH-users
also,
don't use 1.0.4, please us the latest SVN or last svn snapshot at the very
least.
On Mon, Dec 7, 2009 at 12:34 PM, Kendall Stauffer k...@ksac.com wrote:
Any direction on where to start would be appreciated. I am trying to get
freepbx working with this, and everything works (I think)
Thanks for the suggestions. We'll explore.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Tim Uckun
Sent: Sunday, December 06, 2009 5:00 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re:
Thanks. We'll look at that.
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, December 07, 2009 9:35 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] lua+sqlite
Hi All
Ok, so next episode in the saga of getting this monster of the ground :-)
I've gotten the FS up and running pretty much I guess, but I'm missing
something.
It has been set up as per the 'multi-homed' document
(http://wiki.freeswitch.org/wiki/Multi_home_tutorial).
I want to use the
What do you want me to check while running these tests? Sound quality (it's
good now even with original 1.0.4). Or CPU utilization?
It's Debian 4.
Anthony Minessale-2 wrote:
Did you do each thing alone too to tell the difference?
-hp alone, disable monotonic alone (i did not see you
Sorry no, apart from the fact that I was seeing the hangup.
I'm wondering if this a bandwidth congestion issue. Is there anyway on
a bridged call I could trap on dtmf like look for '*' and force a
hangup? I don't seem to able to see this tone on the B leg though.
Regards,
Both,
if it always sounds ok then I guess CPU usage.
On Mon, Dec 7, 2009 at 2:58 PM, eaf erandr-j...@usa.net wrote:
What do you want me to check while running these tests? Sound quality (it's
good now even with original 1.0.4). Or CPU utilization?
It's Debian 4.
Anthony Minessale-2
Hi
Is it possible to trap on DTMF on a bridged call within an LUA script?
I've tried setting the gateway to use inband, but no joy. It looks like
I could use start_dtmf, but I can't see how to launch this within LUA
Regards,
___
session:execute(start_dtmf);
/b
On Dec 7, 2009, at 4:02 PM, Nik Middleton wrote:
Hi
Is it possible to trap on DTMF on a bridged call within an LUA
script? I’ve tried setting the gateway to use inband, but no joy.
It looks like I could use start_dtmf, but I can’t see how to launch
session:execute(start_dtmf);
this app captures inband audio tone dtmf and interprets them aka calls your
callback etc.
On Mon, Dec 7, 2009 at 4:02 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Hi
Is it possible to trap on DTMF on a bridged call within an LUA script?
I’ve
On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Hi
Is it possible to trap on DTMF on a bridged call within an LUA script?
I’ve tried setting the gateway to use inband, but no joy. It looks like I
could use start_dtmf, but I can’t see how to launch
Can this be done in an lua script?
Regards,
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 07 December 2009 22:18
To: freeswitch-users@lists.freeswitch.org
Once the call is bridged, while I can see an inband DTMF event being
generated, it doesn't call my hook unfortuneately
function onInput(session, type, obj)
if type == dtmf and obj['digit'] == '*' then
session:hangup();
return true;
end
Hello,
i now changed the $${domain} of the server to the domain name the
phones register with.
Now messaging (MWI, notify) works. Thanks to all for your support.
Best regards
Peter
Peter P GMX schrieb:
Hello Anthony,
I did some checks today
Here is how the phones are registered:
mysql
It can. I use it like:
session:execute(bind_meta_app, 1 b s execute_extension::dx XML
features);
session:execute(bind_meta_app, 2 b s
record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strft
ime(%Y-%m-%d-%H-%M-%S)}.wav);
session:execute(bind_meta_app, 3 b s
did you set the inputcallback too?
On Mon, Dec 7, 2009 at 4:59 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Can this be done in an lua script?
Regards,
--
*From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
Yes I did, is it possible mod_vmd is interering? It's stopped before I
call the start_dtmf function
session:setHangupHook(myHangupHook, blah)
session:setInputCallback(onInput);
session:execute(vmd,start);
if (session:ready() == false) then
freeswitch.consoleLog(info,
Skype have opened their beta program up to all comers.
http://www.skype.com/business/products/pbx-systems/sip/support-faqs/#paddedContent
Three lines in a sip_profile make FreeSWITCH talk nicely; but using the
PCMU codec.
Any progress on SILK native support? Last I saw was discussion back
Brian West br...@freeswitch.org wrote:
They have yet to type make on a 64bit box and build us a binary
that is 64bit. Chances are they mucked it up like the BroadVoice
codecs were and it just won't work on 64bit just yet... if they
would just give us the src we could be done in under two days
We can ONLY hope someone will do this and BSD/MIT the library and NOT
GPL it... if they GPL it then we'll have to have someone write it all
over again... love the Open Source oil and water.
/b
On Dec 7, 2009, at 7:39 PM, Jason White wrote:
it I suspect.
Given that they released the codec
Question --
I'm not sure if it's a genuine problem,as I can see it, it just
complains that I haven't created any sip_profiles in /lan, but is that
necessary?
Response: --
Since you moved the internal profile
We have FreeSWITCH Version 1.0.4 (exported) running at a high volume traffic.
I normally check the concurrent calls by looking at the number of sessions from
status command. However, the number of concurrent calls in FS is normally
higher than it's supposed to be after we ran traffic for
I also have this problem on a trunk version more than 1000 revisions
behind, so I think the best way is to upgrade to trunk and report this
again if still have problem.
2009/12/8 DJB djbin...@yahoo.com:
We have FreeSWITCH Version 1.0.4 (exported) running at a high volume
traffic. I normally
Version 1.0.5 pre 8 is due out any minute. Definitely upgrade to trunk
or at least pre8 when it's available.
-MC
Sent from my iPhone
On Dec 7, 2009, at 6:29 PM, DJB djbin...@yahoo.com wrote:
We have FreeSWITCH Version 1.0.4 (exported) running at a high volume
traffic. I normally check the
For starters, try using the latest svn snapshot. Your version is 6 months
old and several thousand revs old.
On Dec 7, 2009 8:34 PM, DJB djbin...@yahoo.com wrote:
We have FreeSWITCH Version 1.0.4 (exported) running at a high volume
traffic. I normally check the concurrent calls by looking at
Hi Mark
Ok, thanks.
Yes I have a gateway placed in external called musimi.dk (or should it be in
public?), and I'll just create the empty XML's in lan to get rid of that error.
I'll remove the second part of the dialplan, my idea was that it was needed for
calls between sip phones hooked up
Hello all,
*debug voip rtp session named-event*s shows that it receives and
understands the DTMFs, but it does not send them to the PSTN (sends only
those received via INFO). I haveto find some time and go to the remote site
to update to the latest IOS... I will update after this has been done.
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