Thanks David. It'll be of a great help to FS newbies like me.
Regards,
-Jingwei
On Mon, Dec 14, 2009 at 9:27 AM, David V. Fansler
dfans...@dv-fansler.comwrote:
Final Count was just over 900 files. At the moment I am putting them in
a logical order – as best I can tell with my limited
, Jingwei Yang jingwei.y...@gmail.comwrote:
I installed libjpeg-7 following this website:
http://www.linuxfromscratch.org/blfs/view/svn/general/libjpeg.html. And
the previous error is replaced by a new one:
gcc -DHAVE_CONFIG_H -I. -I. -I. -I..
-I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG
-fail3 = 1776.7
On Thu, Dec 3, 2009 at 5:29 PM, Jingwei Yang jingwei.y...@gmail.com wrote:
Hello All,
I have a Digium TDM400P pci card with two FXO ports installed on my linux
box. I've connected an external telephone line to the first FXO port. But I
can't either make outgoing calls
Hello All,
I have a Digium TDM400P pci card with two FXO ports installed on my linux
box. I've connected an external telephone line to the first FXO port. But I
can't either make outgoing calls or receive incoming ones. Here are my
setups, please let me know where goes wrong.
*
/etc/zaptel.conf*
Hi Guys,
I got a compilation error of skypiax_protocol.c with the latest version
r15764.
Compiling skypiax_protocol.c...
*cc1: warnings being treated as errors*
skypiax_protocol.c: In function ‘X11_errors_handler’:
skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and
code
) 664-1044 x200
mr...@avgs.ca
On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote:
Hi Guys,
I got a compilation error of skypiax_protocol.c with the latest version
r15764.
Compiling skypiax_protocol.c...
*cc1: warnings being treated as errors*
skypiax_protocol.c: In function â
Not sure whether this error is due to the lack of libjpeg. I just double
checked, this library had been installed.
Package libjpeg-6b-37.i386 already installed and latest version
On Thu, Dec 3, 2009 at 2:33 PM, Jingwei Yang jingwei.y...@gmail.com wrote:
Hi Mathieu, thanks for the promptly
,
That one is on your side. If you changed/updated system libs it might be
worth doing another ./configure
Cheers,
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 3-Dec-09, at 1:33 AM, Jingwei Yang wrote:
Hi Mathieu
Hi Guys,
Is it possible to restrain the call-out to be one-way, meaning the callee
can only listen, but not speak? If so, is it possible to switch off the
constraint dynamically during the call and allow the callee to speak?
Thanks,
-Jingwei
___
.
--matt
Voice Broadcasting http://www.hellohunter.com/voice_broadcast.php
Predictive
Dialer http://www.hellohunter.com
On Mon, Sep 28, 2009 at 3:38 PM, Jingwei Yang jingwei.y...@gmail.comwrote:
Hi Guys,
Is it possible to restrain the call-out to be one-way, meaning the callee
can only
Thanks Brian.
On Mon, Sep 14, 2009 at 8:55 PM, Brian West br...@freeswitch.org wrote:
You have a merge conflict please svn revert sofia.c
/b
On Sep 14, 2009, at 3:46 AM, Jingwei Yang wrote:
Hi Folks,
I've got a compilation error with the latest codes (r14842)
Making all
Hi Folks,
I just tried to install FS with mod_skypiax on a new CentOS 5.2 machine.
After having followed the big help doc from the wiki page (
http://wiki.freeswitch.org/wiki/Skypiax#An_example_of_Skypiax_and_FreeSWITCH_installation_on_CentOS.2C_from_scratch),
I hit an error when running multi.sh
Hi Giovanni,
That's a big relief. Thanks a lot for the reply :)
Regards,
-Jingwei
On Fri, Sep 4, 2009 at 4:25 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote:
Jingwei,
those are normal warnings made by the Skype client (not by
mod_skypiax), you just have to edit /etc/alsa/alsa.conf and
That's efficient :) By the way, do you have any idea about this warning?
ALSA lib pcm_dmix.c:1008:(snd_pcm_dmix_open) unable to open slave
On Fri, Sep 4, 2009 at 5:47 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote:
On Fri, Sep 4, 2009 at 10:56 AM, Giovanni Maruzzelligmar...@celliax.org
Hi Chris, sorry for the late reply. Have been quite busy last few days.
I had shifted 888 from default.xml to public.xml and the dialplan is simply
having an echo action now. I've turned on dl_debug but unfortunately didn't
find anything useful. Logs are attached for your reference.
I don't
client? Will it really hit 888 in your dialplan?
Thanks,
Chris
On Mon, Jul 13, 2009 at 5:27 AM, Jingwei Yang jingwei.y...@gmail.comwrote:
Hi Chris, sorry for the late reply. Have been quite busy last few days.
I had shifted 888 from default.xml to public.xml and the dialplan is
simply
Hi Chris, any thoughts?
Thanks,
-Jingwei
On Fri, Jun 26, 2009 at 11:34 AM, Jingwei Yang jingwei.y...@gmail.comwrote:
Hi Chris, here's the one that confuses me. As far as I understand, the
extension 888 defined in public.xml is for picking up incoming calls. It
should have no influence
working setup behind the NAT router.
Thanks, I've commented it out.
On Thu, Jun 25, 2009 at 1:31 AM, Jingwei Yang jingwei.y...@gmail.comwrote:
Hi Chris, thanks for your help. Here's my client.xml
include
!-- Client Profile (Original mode) --
!-- to use this profile take the x- away
Hi Chris, here's the one that confuses me. As far as I understand, the
extension 888 defined in public.xml is for picking up incoming calls. It
should have no influence on outgoing calls, right? If not, what is to write
to fit my case? (originate
Hi Guys,
Here's my situation:
The freeswitch server and my machine are behind the same LAN. If I commented
out ext-rtp-ip from client.xml, I'm able to hear the echo (by *originate
dingaling/gmail.com/user...@gmail.com echo*).
However, external calls have no sound at all no matter whether this
...@gmail.com wrote:
search wiki from sth. like disable_rtp_autoajust , I don't remember the
exact var.
On Jun 24, 2009, at 5:40 PM, Jingwei Yang wrote:
Hi Guys,
Here's my situation:
The freeswitch server and my machine are behind the same LAN. If I
commented out ext-rtp-ip from client.xml, I'm
Hi Paul, thanks for your reply. I've give it a try.
On Thu, Jun 25, 2009 at 11:13 AM, paul.d...@gmail.com
paul.d...@gmail.comwrote:
You can use FS socket event interface for that. See free Java lib for
inbound socket event here: http://versafon.com/versafonweb/Software.jsp
Jingwei Yang wrote
-ip
On Jun 25, 2009, at 10:23 AM, Jingwei Yang wrote:
Hi seven, thanks for your reply. I've commented out ext-rtp-ip and put
disable-rtp-auto-adjust inside client.xml. No matter what value this
parameter has (true or false), local IP is able to hear the echo but
external ones still have
with ESL - the event socket library - that can abstract away some of the
grunt work, but there isn't a Java one that I'm aware of.
-MC
On Wed, Jun 24, 2009 at 7:52 PM, Jingwei Yang jingwei.y...@gmail.comwrote:
Hi Folks,
I understand freeSwitch is supporting a couple of languages for call
Hi Guys,
I've configured a gtalk client based on the steps in this url:
http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/.
But i'm not sure how to originate calls to different gtalk users
dynamically. I've tried this:
freeswitch *originate dingaling/gmail.com/user...@gmail.com
once FS is up again..
jmesquita
On Mon, Jun 22, 2009 at 11:33 PM, Jingwei Yang jingwei.y...@gmail.comwrote:
Hi Guys,
I've configured a gtalk client based on the steps in this url:
http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/.
But i'm not sure how to originate calls
Hi Giovanni,
Sorry, pretty busy and fully occupied by other stuff today. Have to delay
the testing and give you the result tomorrow.
Regards,
-Jingwei
On Tue, Jun 16, 2009 at 5:32 PM, Jingwei Yang jingwei.y...@gmail.comwrote:
Sure, I'll append to you the result tomorrow.
Regards,
-Jingwei
bug to jira http://jira.freeswitch.org
http://wiki.freeswitch.org/wiki/Reporting_Bugs
Make sure you attach a backtrace of your issue and file it under skypeiax
so giovanni can track it.
On Wed, Jun 17, 2009 at 4:39 AM, Jingwei Yang jingwei.y...@gmail.comwrote:
Hi Giovanni,
Sorry, pretty
Hi Giovanni,
I've reported it in Jira. Here's the bug url:
http://jira.freeswitch.org/browse/MODSKYPIAX-35
Thanks,
-Jingwei
On Mon, Jun 15, 2009 at 8:16 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote:
Hi Jingwel,
thanks for reporting.
Could you please add a Jira issue with as much
Sure, I'll append to you the result tomorrow.
Regards,
-Jingwei
On Tue, Jun 16, 2009 at 4:42 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote:
Hi Jingwei,
Thanks a lot! I'll take care of as soon as possible.
Btw, before I read the Jira, are you testing in linux?
If you are testing on
Hi Team,
I've been using the record_session feature to record call sessions. Here's
how I prepared the dialplan:
extension name=skypiax
condition field=destination_number expression=^2909/(.*)$
action application=record_session data=/tmp/data.wav/
action
Hi Team,
As the subject indicates, is there a possible way to do that?
I've tried setting up two different skype instances with the same id in
/usr/src/freeswitch/src/mod/endpoints/mod_skypiax/configs/2startskype.sh
/usr/bin/Xvfb :101 -auth
) or use two (or more) different Skype IDs on same
machine.
Thank you.
On Mon, Jun 15, 2009 at 8:29 AM, Jingwei Yang jingwei.y...@gmail.comwrote:
Hi Team,
As the subject indicates, is there a possible way to do that?
I've tried setting up two different skype instances with the same id
Yes, it's the right way to go!
Thanks, man!
On Wed, Jun 10, 2009 at 10:25 PM, dujinfang dujinf...@gmail.com wrote:
On Jun 10, 2009, at 3:33 PM, Jingwei Yang wrote:
Hi All,
I just finished installing freeSwitch and Skypiax. And I'm able to use
skype api directly via the sk command like
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