hhmmm :))
is there any doc following up mod_opal ?
I really don't want to waste your time :)
T.
On Tue, Sep 1, 2009 at 1:00 AM, Michael Collins m...@freeswitch.org wrote:
On Mon, Aug 31, 2009 at 3:48 PM, Tihomir Culjaga tculj...@gmail.comwrote:
hello,
i'm trying to build mod_opal and
hi, It went well
obviously FS needs v3_6 opal :)
thx.
On Tue, Sep 1, 2009 at 8:09 AM, Tihomir Culjaga tculj...@gmail.com wrote:
hhmmm :))
is there any doc following up mod_opal ?
I really don't want to waste your time :)
T.
On Tue, Sep 1, 2009 at 1:00 AM, Michael Collins
On Mon, Aug 31, 2009 at 11:09 PM, Tihomir Culjaga tculj...@gmail.comwrote:
hhmmm :))
is there any doc following up mod_opal ?
I really don't want to waste your time :)
T.
Last I heard this was it:
http://jira.freeswitch.org/browse/MODOPAL-10
-MC
oh good, on remote router/dsl modem (whatever doing NAT) never use upnp,
never use ALG, just do a simple NAT and it is alway gonna work!
T.
On Tue, Sep 1, 2009 at 2:06 AM, e schmidbauer e.schmidba...@gmail.comwrote:
i cannot reach the remote endpoint. the remote endpoint can reach a
locally
Did the simple make in the libs/esl directory run properly? Just curious.
I'll have to defer to the Ubuntu gurus out there for thoughts on what else
could be wrong.
-MC
On Mon, Aug 31, 2009 at 10:43 PM, Harondel J. Sibble h...@pdscc.com wrote:
Haven't had any responses, anyone have any ideas on
Please look into MODOPAL-10 in jira. You need to apply a patch if you're using
latest opal trunk, ro else you need to use the latest stable version of opal.
However, I'm not sure how automated this is in the build process in Linux. I've
only done this on Windows builds lately.
/Peter
Från:
Hi folk,
First of all, thank you for FS - really strong project.
I have already asked this once in other thread but didn't got any answer.
So, I'll try to re-ask.
We are playing currently with FS under Windows 2008 64bit. So far there are
some issues but I hope we'll solve it in nearest future.
If you want to try FS on Windows only for feature testing etc. then its
okay, however for production deployments (that includes load testing) i
strongly recommend CentOS 5.x.
As far as configuration migration is concerned, you don't need to change any
configuration files, simply copy them to
Hi,
i have a problem using SRTP Encrytion. All intern calls are SRTP encrypted.
Some of my Gateway don´t support SRTP encryption.
In my dialplan I now set the sip_secure_media to false.
action application=set data=sip_secure_media=false/
It works. But is there any chance to encrypt the call
Hello Brian,
I've done this. FS creates the tables sccessfully, but then doesn't fill
them.
isql:
SQL show tables;
+-+
| Tables_in_fs_external |
+-+
| sip_authentication
Sure this works,
you can set rtp or srtp independently to every call leg (if FS is in
media path) and even mix them in a conference.
Best regards
Peter
NOx-WHV schrieb:
Hi,
i have a problem using SRTP Encrytion. All intern calls are SRTP encrypted.
Some of my Gateway don´t support SRTP
I gave up on compiling esl, I got a bunch of errors, there were several
people on the list with problems and apparently no straight solution,
especially for php-esl. I am now using a ruby library, posted here by Diego
Viola I believe.
On Tue, Sep 1, 2009 at 2:33 AM, Michael Collins
How can I see if the FS is in media path?
Or how can i set the FS in media path?
Peter P GMX wrote:
Sure this works,
you can set rtp or srtp independently to every call leg (if FS is in
media path) and even mix them in a conference.
Best regards
Peter
NOx-WHV schrieb:
Hi,
i
Thank you!
--
Best regards,
Dmitry Kadantsev
http://www.doxwox.com - Best web meeting and online collaboration tool.
On Tue, Sep 1, 2009 at 11:00 AM, Muhammad Shahzad
shaherya...@googlemail.com wrote:
If you want to try FS on Windows only for feature testing etc. then its
okay, however for
Do you have any specific notes why production or load testing isn’t recommended
on Windows? Or just lack of data?
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Muhammad
Shahzad
Sent: Tuesday, September 01, 2009 3:00 AM
If you do not explicitely set bypass_media to true, then FS is in the
media path.
Best regards
Peter
NOx-WHV schrieb:
How can I see if the FS is in media path?
Or how can i set the FS in media path?
Peter P GMX wrote:
Sure this works,
you can set rtp or srtp independently to
My basic functionality is this: A calls in, is bridged to B (). I
use bind_meta_app to let B rebridge A to C (). After having been
rebridged to C, C should be able to rebridge A to B *again*, and so
on.
This is the code I have:
context name=public
extension name=ff-ivr
I tried doing a set right before the application is called to make a customer
variable but it doesn't get transferred to the template this way either:
---dialplan snip---
action application=set data=test_var=this is a test/
action application=voicemail data=default $${domain} $1/
gateways do not go into the table... ONLY inbound registrations to the
profile do.
/b
On Sep 1, 2009, at 5:16 AM, Peter P GMX wrote:
Hello Brian,
I've done this. FS creates the tables sccessfully, but then doesn't
fill
them.
isql:
SQL show tables;
Try this one.
Outbound
action application=export data=nolocal:sip_secure_media=false/
Inbound
action application=export data=nolocal:sip_secure_media=true/
/b
On Sep 1, 2009, at 4:40 AM, NOx-WHV wrote:
action application=set data=sip_secure_media=false/
Thanks for the input.
You'll have to decide on static vs. dynamic based on your needs. In
either case, once the call is connected to your socket you've got
all sorts of control options. PHP has an ESL abstraction just like
the other languages so there shouldn't be any issue about PHP
Have you gotten past the problems with pthread-win32 on 64 bit? you will need
the trunk version of that library if not because the released version has
problems with 64bit.
There are some other simple compilation problems I assume you may have
already got past? If not see
Dear sir,
I want to say posision in queue to caller but
fifo_chime_list can't say more than one sound file. i try
fifo_chime_list = queue/say1.wav,queue/say2.wav.
Best Regards.
Dome C.
___
FreeSWITCH-users mailing list
I put tomato on the router and still no success. upnp is enabled,
should i disable it? what do you mean by simple NAT?
On Tue, Sep 1, 2009 at 2:32 AM, Tihomir Culjagatculj...@gmail.com wrote:
oh good, on remote router/dsl modem (whatever doing NAT) never use upnp,
never use ALG, just do a
ok, please can you provide a tcpdump/wireshark sniff on before and after
that linksys.
this is something trivial.
T.
On Tue, Sep 1, 2009 at 6:22 PM, e schmidbauer e.schmidba...@gmail.comwrote:
I put tomato on the router and still no success. upnp is enabled,
should i disable it? what do you
Hi Peter,
i did it on linux... it was enough to use
svn co https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/trunkptlib
svn co
https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/opal/branches/v3_6opal
this is something that works well :)
BTW: do you get a correct
We are happy to announce http://www.freeswitch.org/node/201 that the
Friday public FreeSWITCH meetings are returning, starting this Friday,
September 4. Meetings will run from 11am to 5pm CST. The meetings will be
held in the FreeSWITCH public conference, also known as the 888 conference.
get latest trunk and try again.
there was a single character out of place that caused the variables to not
be expanded.
On Tue, Sep 1, 2009 at 8:38 AM, Nick Lemberger nick.lember...@lkfd.netwrote:
I tried doing a set right before the application is called to make a
customer variable but it
you probably don't want to call bridge from bind meta app, try using the
att_xfer app instead
it works like bridge but when you call C you can press # to hangup and
bridge a to c or press 0 to conference call all 3.
On Tue, Sep 1, 2009 at 6:17 AM, Harry Vangberg ha...@vangberg.name wrote:
My
hello
we have got a little problem with the conference application
in our setup we have da system for customers where speakers can dial in
with phonenumber+1 and the listeners dial in with phonenumber
the speakers conference is started with 323963...@conf+flags{waste}
the listeners conference is
I haven't really used waste much myself, but my understanding is that
waste and mute would conflict, since waste says send audio always
and mute says send audio never. I didn't understand why you're using
waste on the listeners... you should be able to get by with waste just
on the speaker (again,
thank you for your response
as a listener waste influences what you hear and mute say's you cannot speak
this is what our customer wanted because the speaker is the only one who is
heard in this conference or meeting room - this rooms are for lectures
we tried to disable waste for the listeners
waste + mute would result in sending audio that was all zeros or generated
silence.
On Tue, Sep 1, 2009 at 4:32 PM, Bradley Brashier bjbrash...@gmail.comwrote:
I haven't really used waste much myself, but my understanding is that
waste and mute would conflict, since waste says send audio
Michael
Yes, from memory the intial make completes successfully, when you go to make
the modules themselves is when it starts barfing.
On 31 Aug 2009 at 23:33, Michael Collins wrote:
Did the simple make in the libs/esl directory run properly? Just
curious. I'll have to defer to the Ubuntu
All of the language modules in ESL require the runtime and the devel
packages for that language for the compile to work.
On Tue, Sep 1, 2009 at 5:33 PM, Harondel J. Sibble h...@pdscc.com wrote:
Michael
Yes, from memory the intial make completes successfully, when you go to
make
the modules
that means something in your path does not support CNG/VAD.
it's perfectly ok to use waste and mute together.
there is no chance that you would not enter the conf muted the way you
describe unless you are
using an older revision of FS that had a bug in the parsing of the
conference flags.
I run into this problem before. Don't remember the exact error but
might be segfault of lame runing in freeswitch-lua.
If you use Linux you would like to try iwatch. It's a perl program
watching your file system and can execute the lame command as soon as
it got the CLOSE_WRITE(or other)
Hi Steven,
Sounds like a very good tip. Do you have any example available to share? I
will be happy to upload it to the wiki when i put it up and running.
/aep
--
Stopping junk mailers is good for the environment
I run into this problem before. Don't remember the exact error but
might be
Tihomir,
Yes as I remember it I did get the correct caller id number. I think you need
to set variable origination_caller_id_number when you originate a call.
/Peter
Från: Tihomir Culjaga tculj...@gmail.com
Skickat: den 1 september 2009 19:20
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