This I/O error occurred once. And I am able to communicate google talk
to SIP softphone through FS.
Still does not know why this error happened as I did not change anything!
2009/3/30 Anthony Minessale anthony.miness...@gmail.com:
libdingaling.c:1545 xmpp_connect() io error 2 7
It means
Kulwinder Singh contributed this HOW TO: Freeswitch Skype- OS
Microsoft Windows
Download 118MB HD: http://www.celliax.org/final.avi
Sincerely,
Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell :
Brian West br...@freeswitch.org wrote:
This isn't a buffet where you pull up and demand things be one way or the
other... this is a community where you start helping. I would love to
see more helping and less demanding!
So would I.
I regularly scan the mailing list looking for questions to
hi,
is it just audio or is it that im having broken codecs so cant view any
video?
Regards,
Bipin
Giovanni Maruzzelli-3 wrote:
Kulwinder Singh contributed this HOW TO: Freeswitch Skype- OS
Microsoft Windows
Download 118MB HD: http://www.celliax.org/final.avi
Sincerely,
Giovanni
Hi,
I am doing a killall -HUP freeswitch in order to achieve
cdr-csv log rotation and the process gets killed instead. A few
days ago it worked fine. I also noticed that a few days ago when I
hit Ctrl+C freeswitch did not exit. Now, when I do this the process gets
terminated immediately.
I found out the cause :
When the mod_opal module is loaded the FS process gets killed with a
kill -HUP. I thought it would be good for everyone to know.
Apostolos Pantsiopoulos wrote:
Hi,
I am doing a killall -HUP freeswitch in order to achieve
cdr-csv log rotation and the process
Hi,
Here is what I am trying to accomplish:
//--- completecall.js trigged via api_hangup_hook
use(CURL);
const loglevel='notice';
var uuid = request.getHeader(Core-UUID);
var billmsec = request.getHeader(billmsec);
var urlrequest = UUID= + uuid + billmsec= + billmsec;
xbipin pisze:
hi,
is it just audio or is it that im having broken codecs so cant view any
video?
There are both, video and audio.
Mplayer dump
Otwieram dekoder video: [ffmpeg] FFmpeg's libavcodec codec family
Wybrany kodek video: [ffcamtasia] vfm: ffmpeg (TechSmith Camtasia Screen Codec
if you set the channel variable 'session_in_hangup_hook=true' early in the
call, the session will be present in your script.
2009/3/31 Keith Laaks kei...@voxtelecom.co.za
Hi,
Here is what I am trying to accomplish:
//--- completecall.js trigged via api_hangup_hook
use(CURL);
Hi,
I'm using freeswitch as a glorified answering machine. FS registers with a
VOIP gateway and all calls into the gateway go through an ivr menu and are
allowed to leave a message which gets recorded to a file. The FS box is
behind a NAT firewall. Everything works fine except that
I'm going to guess you're not on SVN trunk? what rev are you on?
/b
On Mar 31, 2009, at 8:04 AM, Andy Ayers wrote:
Hi,
I'm using freeswitch as a glorified answering machine. FS registers
with a VOIP gateway and all calls into the gateway go through an ivr
menu and are allowed to leave a
Hi Brian,
1.03
Thanks
Andy
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: 31 March 2009 14:27
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Calls
Please try SVN trunk.
/b
On Mar 31, 2009, at 8:36 AM, Andy Ayers wrote:
Hi Brian,
1.03
Thanks
Andy
Brian West
br...@freeswitch.org
-- Meet us a ClueCon! http://www.cluecon.com
___
Freeswitch-users mailing list
Hello,
I have found the problem. FS on my local network sends SIP/2.0 200 OK
after an invite and FS on the net through the external profil sends
SIP/2.0 183 Session Progress. But MjSip doesn't know how to deal with
183, so it just ignores the message. For testing I have changed
the 183 header to
On Mar 31, 2009, at 5:59 AM, Anthony Minessale anthony.miness...@gmail.com
wrote:
if you set the channel variable 'session_in_hangup_hook=true' early
in the call, the session will be present in your script.
Very cool. I will get this chan var documented on the wiki right away.
-MC
like i said:
maybe that phone does not support early media
try adding the answer application to your dialplan
early media == 183
answer = 200
it depends on your dialplan in FS
On Tue, Mar 31, 2009 at 9:06 AM, can_...@gmx.de wrote:
Hello,
I have found the problem. FS on my local network
Hi.
I'm trying to compile FS on dragonfly BSD 1.10.1-RELEASE. I am
compiling the release source from
http://files.freeswitch.org/freeswitch-1.0.3.tar.gz
I had to write a uname wrapper script to fake being FreeBSD to get
configure to complete successfully. I also added -D__FreeBSD__ to
I am a Yate user and I can tell their mailing list suffer the same problem.
My solution? I often ask for help but, as a personal policy, I always write
an article or add to an existing one on the project wiki explaining and
documenting what people explained to me. This creates a triple value:
Got a few more questions about running LUA scripts, please forgive me, I'm
an absolute newbie with LUA.
If I want to subscribe to a custom event, and I use
con = freeswitch.EventConsumer(CUSTOM my::event);
I get an error. Is this because I must subscribe to the CUSTOM (only) event,
and then
as replied earlier, if your doing nothing but consuming events, you
can just block instead of sleep:
con:pop(1)
there is also a msleep function that you can call the same way you do
console_log, it takes milli seconds as its arg. Note this should NOT
be used when you have a script
I know before I asked about blocking for an event, and maybe I should have
created a new topic..
but now I want to actually sleep (rather than block) for a set time
frame...this app will not be consuming events.
can I get an example of how to use msleep in a lua script? This lua script
will be
http://wiki.freeswitch.org/wiki/Lua#freeswitch.consoleLog
On Mar 31, 2009, at 11:50 AM, Matthew Fong wrote:
I know before I asked about blocking for an event, and maybe I
should have created a new topic..
but now I want to actually sleep (rather than block) for a set time
frame...this app
Lua has no sleep or pause ... if you read thru the lua wiki they show
you various ways to accomplish that.
On Mar 31, 2009, at 10:50 AM, Matthew Fong wrote:
I know before I asked about blocking for an event, and maybe I
should have created a new topic..
but now I want to actually sleep
Thanks, the freeswitch.msleep(5000) works!
Any comment about the first Q...
con = freeswitch.EventConsumer(CUSTOM my::event);
I get an error. Is this because I must subscribe to the CUSTOM (only) event,
and then filter out the events using the Event-Subclass myself? Or am I
missing something in
http://www.freeswitch.org/docs/class_event_consumer.html#a0
It takes 2 args, not one to specify the subclass
Mike
On Mar 31, 2009, at 12:11 PM, Matthew Fong wrote:
Thanks, the freeswitch.msleep(5000) works!
Any comment about the first Q...
con = freeswitch.EventConsumer(CUSTOM my::event);
FYI, I've documented the msleep method here:
http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.msleep
I will work on better and more organized API documentation. If anyone out
there has time/energy/knowledge of the scripting APIs and is willing to help
out please email me off list.
-MC
2009/3/31 Raffaele P. Guidi raffaele.p.gu...@gmail.com
I am a Yate user and I can tell their mailing list suffer the same problem.
My solution? I often ask for help but, as a personal policy, I always write
an article or add to an existing one on the project wiki explaining and
documenting
This part is interesting, and the subject of a discussion we had
recently. A number of systems try that second re-invite after a 488, but
the SIP specs say the call is pretty much dead after the 488 message is
exchanged. Are they just hoping that maybe the other end will be
non-compliant
correct, we *do not* proxy re-invites except when bypass_media or
proxy_media is set.
On Tue, Mar 31, 2009 at 11:59 AM, Gabriel Kuri gk...@ieee.org wrote:
This part is interesting, and the subject of a discussion we had
recently. A number of systems try that second re-invite after a 488,
Ciao Bipin,
there is both video and audio.
Use vlc (http://www.videolan.org/vlc/) or mplayer
(http://www.mplayerhq.hu/design7/dload.html), and you'll be ok :-).
Sincerely,
Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
I recommend you reformat this file to be in some format that will play on
windows and mac and anything else without having to get anything or nobody
will watch it =D
2009/3/31 Giovanni Maruzzelli gmar...@celliax.org
Ciao Bipin,
there is both video and audio.
Use vlc
Hi Guys,
I know this sounds an odd question, but I need to inject audio into an
outbound call. The reason for this is that for a pre-paid billing app,
I need to let the call initiator know they are running out of credit.
Is there a facility to do this? Ideally I only want to let the
Worked for me, just needed to add the missing codec for media player
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Giovanni Maruzzelli
Sent: 31 March 2009 21:09
To:
You shouldn't have to go get anything :P If you have to spend time to
get something to watch the video it sometimes isn't a good thing...
have you tried YouTUBE?
http://www.perian.org/
/b
On Mar 31, 2009, at 3:45 PM, Nik Middleton wrote:
Worked for me, just needed to add the missing
uuid_displace
/b
On Mar 31, 2009, at 3:44 PM, Nik Middleton wrote:
Hi Guys,
I know this sounds an odd question, but I need to inject audio into
an outbound call. The reason for this is that for a pre-paid
billing app, I need to let the call initiator know they are running
out of
or, probably better for this situation, uuid_broadcast
2009/3/31 Brian West br...@freeswitch.org
uuid_displace
/b
On Mar 31, 2009, at 3:44 PM, Nik Middleton wrote:
Hi Guys,
I know this sounds an odd question, but I need to inject audio into an
outbound call. The reason for this is that
I know that on a sofia profile you can set it to be greedy or generous
in codec negotiation on the inbound side.
How does freeswitch negotate codecs on the outbound side and what ways
are available to affect how this takes place?
Is there a way to make the codec preferences of the inbound
Hello,Friends, can someone please help me understand what hardware and/or software (like FreeSwitch) do i need, to have a working setup like below:DSL from
telco-switch+NAT--ubuntu server |__voip phone |___normal PSTN phoneQ. what voip related software will i need to run
Michael Collins m...@freeswitch.org wrote:
I like it... The Wiki Tax
It's an excellent suggestion.
As an aside, would it be possible for the wiki administrator to modify the
configuration so that there is a means of subscribing without having to deal
with a captcha?
For people who can't see
http://wiki.freeswitch.org/wiki/Codec_negotiation
http://lmgtfy.com/?q=freeswitch+late+negotiation
On Tue, Mar 31, 2009 at 17:04, Josh Forman jfor...@wcgltd.com wrote:
I know that on a sofia profile you can set it to be greedy or generous
in codec negotiation on the inbound side.
How does
On Mar 31, 2009, at 6:36 PM, Jason White wrote:
It's an excellent suggestion.
As an aside, would it be possible for the wiki administrator to
modify the
configuration so that there is a means of subscribing without having
to deal
with a captcha?
Not sure we can do that but I suspect we
badeguruji badegur...@yahoo.com wrote:
Q. what voip related software will i need to run above setup?
FreeSWITCH
Q. what hardware (like phone card etc.) do i need in my PC?
A digium or similar card, and a SIP phone of course.
Q. will i be able to use my existing phone number? (i do not have
First off please no HTML emails We have a large community of blind
users that might have a hard time reading your messages or replying
which they do all the time! ;) Thanks guys.
On Mar 31, 2009, at 6:27 PM, badeguruji wrote:
Hello,
Friends, can someone please help me understand
Dear FreeSWITCH Community:
As you know, FreeSWITCH has been growing leaps and bounds and it's going to
keep growing as the word spreads. The core development team of Anthony,
Mike, and Brian are very appreciative of the community's help and
involvement in the project. Simply put: the community is
Hello,
You can't make more than one outbound call if you have only one phone
line from bsnl.
Regards,
Mitul Limbani,
Founder CEO,
Enterux Solutions Pvt Ltd,
The Enterprise Linux Company(r),
http://www.enterux.com/
On 01-Apr-09, at 5:19, Jason White ja...@jasonjgw.net wrote:
badeguruji
Oooops,
I was not aware you cannot see the video on Windows (I use mplayer and
vlc on windows, and never bother to start windows media player :-) ).
I agree that the best would be youtube or so.
I don't know how to upload video on youtube, and I'll be not in my
office for a week.
Can one of
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