Cool, looking forward to test this :)
Even André
On 27. mai. 2009, at 04.51, Diego Viola wrote:
> Sorry, this is the link.
>
> http://wiki.freeswitch.org/wiki/Mod_nibblebill#Hangup_the_call_when_the_balance_is_depleted
>
> Diego
>
> On Tue, May 26, 2009 at 9:06 PM, Diego Viola
> wrote:
>> Hi,
Of course, it will be useful to try it.
Artem
> Hi Artem,
>
> Please to see that some of the stuff I wrote is useful to someone..!
>
> I've written an FS module which will send the audio over - it's more
> efficient than using unicast. Let me know if you'd like a copy.
>
> Cheers --
>
> Dave
>
Sorry, this is the link.
http://wiki.freeswitch.org/wiki/Mod_nibblebill#Hangup_the_call_when_the_balance_is_depleted
Diego
On Tue, May 26, 2009 at 9:06 PM, Diego Viola wrote:
> Hi,
>
> Darren just added this today, in case if someone is interested.
>
> http://wiki.freeswitch.org/wiki/Mod_nibble
I just updated it, it was a bug that got fixed already.
22:19 <@bkw__> diegoviola: already fixed
22:19 <@bkw__> update
22:19 <@bkw__> close the }
22:20 <@bkw__> it was a bug we fixed already this morning that catches
unclosed global preprocess vars
Thanks,
Diego
On Tue, May 26, 2009 at 10:16 PM
Anthony - unbelievable! Thank you so much for implementing that! I kept
going through the possibilities of using a FIFO, putting the javascript in a
polling loop, or having everyone enter the conference muted and manually
playing MOH. This feature absolutely makes my code a snap... Yes, I will
Diego Viola wrote:
> Hi, I have downloaded the latest freeswitch trunk, and when I do
> reloadxml I get this.
>
> Error [unterminated ${var}] in line
> /usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xml
> line 12
>
> Any ideas? I haven't edited that file myself.
Have a lo
So your saying instead of:
cd /usr/src/freeswitch.trunk
make clean
svn up
../bootstrap.sh
../configure
make install
do:
cd /usr/src/freeswitch.trunk
make clean
svn up
../configure
make install
?
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@l
Hi, I have downloaded the latest freeswitch trunk, and when I do
reloadxml I get this.
Error [unterminated ${var}] in line
/usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xml
line 12
Any ideas? I haven't edited that file myself.
Thanks,
Diego
_
Michael,
Pardon me for hopping on this thread, but can you explain more about this
new feature? I've been wanting something like this to apply different
behaviors for different conference members. Can this be used to provide a
'moderator' with different behaviors bound to DTMF keys than regular
ca
Brian West wrote:
> Thank you... Now please tell 10 of your friends about FreeSWITCH ;)
Also, if you're a member of a Linux user's group or similar organization, now
might be a good opportunity to raise FreeSWITCH awareness on their mailing
list or at a meeting.
Hi,
Darren just added this today, in case if someone is interested.
http://wiki.freeswitch.org/wiki/Mod_nibblebill#Hangup_the_call_when_cash_is_depleted
Thanks Darren :).
Regards,
Diego
On Sun, May 10, 2009 at 3:06 PM, Diego Viola wrote:
> Darren Schreiber to me
> That won't work. The code i
> Also, in general, do the XML files in config get updated during the ‘make
> install’, or are they left as they were from the previous builds?
>
Running "make install" or "make samples" will not overwrite your existing
configuration files. NOTE: This means that when the default configuration
chan
On Tue, May 26, 2009 at 4:56 PM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:
> the easiest way would be the new feature I added to 13442
>
> in the conference profile add
>
>
>
> to your
>
> and in your dialplan
>
>
>
>
> or
>
>
>
>
> Don't forget the wishlist and donate button on
First off pre8 came prebootstrapped no need to run it. So please try
again.
/b
On May 26, 2009, at 7:03 PM, Lars Zeb wrote:
I followed Michael Collin’s instruction in an earlier email about
building pre8. During the ./bootstrap.h, I encountered the following
error:
Entering directory /
I followed Michael Collin's instruction in an earlier email about building
pre8. During the ./bootstrap.h, I encountered the following error:
Entering directory /usr/src/freeswitch.trunk/libs/libsndfile
Creating aclocal.m4
Running libtoolize...
Putting files in AC_CONFIG_AUX_DIR, `Cfg'.
Cre
the easiest way would be the new feature I added to 13442
in the conference profile add
to your
and in your dialplan
or
Don't forget the wishlist and donate button on the main site
On Tue, May 26, 2009 at 10:46 AM, wrote:
> Ok, that sounds doable... I have no problem bangi
Hi Drew,
When you say that the problem goes away if you don't use start_dtmf, do you
mean that you get one tone recognised per tone or none? If the former, then
you've got DTMF being signalled out of band as well; in that case, why do you
need inband detection?
--Dave
- Original Messag
Hello
I was wondering if someone had successfully ran a PBX on those tiny, $100
devices that run Linux?
www.pogoplug.com
www.plugcomputer.org
I'm thinking of hooking up a Sangoma U100
(http://wiki.sangoma.com/sangoma-wanpipe-usbfxo) to the USB port so it can
handle a couple of analog lines. And
Remember Save the cheerlead, save the world? In this case Digg the
story, get your bugs fixed?
/b
On May 26, 2009, at 4:11 PM, Even André Fiskvik wrote:
...or be followed with 10 yrs of bad luck and hardware failures!
On 26. mai. 2009, at 22.42, Brian West wrote:
Brian West
br...@freeswi
...or be followed with 10 yrs of bad luck and hardware failures!
On 26. mai. 2009, at 22.42, Brian West wrote:
Thank you... Now please tell 10 of your friends about FreeSWITCH ;)
/b
On May 26, 2009, at 3:40 PM, Even André Fiskvik wrote:
Diggedy dug!
On 26. mai. 2009, at 22.03, Diego Viola
Thank you... Now please tell 10 of your friends about FreeSWITCH ;)
/b
On May 26, 2009, at 3:40 PM, Even André Fiskvik wrote:
Diggedy dug!
On 26. mai. 2009, at 22.03, Diego Viola wrote:
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
_
Diggedy dug!
On 26. mai. 2009, at 22.03, Diego Viola wrote:
> Digged.
>
> On Tue, May 26, 2009 at 3:47 PM, Brian West
> wrote:
>> Dear FreeSWITCHers,
>> Now I'm gonna take a moment here to guilt each and everyone of you
>> into
>> checking out the story about Pre8 on Digg. We have all worke
I've got a configuration where I receive inbound calls and dial out to a
pre-determined 800-number based on the DNIS of the call. I set and have everything set up so that DTMF only
comes to me via inband. When I'm providing DTMF data to the IVR, it will
recognize a single keypress as a double-tap.
Digged.
On Tue, May 26, 2009 at 3:47 PM, Brian West wrote:
> Dear FreeSWITCHers,
> Now I'm gonna take a moment here to guilt each and everyone of you into
> checking out the story about Pre8 on Digg. We have all worked long and hard
> to get to 1.0.4 and we still have a little bit to go. So eve
Dear FreeSWITCHers,
Now I'm gonna take a moment here to guilt each and everyone of you
into checking out the story about Pre8 on Digg. We have all worked
long and hard to get to 1.0.4 and we still have a little bit to go.
So everyone out there that asks "What can I do to help the project?
On Tue, May 26, 2009 at 10:06 AM, François Delawarde <
fdelawa...@wirelessmundi.com> wrote:
> Hi Brian,
>
> Is FreeSWITCH going to have Spanish/French/... sounds as well or do those
> need to be?
>
We have a volunteer who is going to record a set of Spanish prompts. These
would probably be class
This is totally possible. You need to look at the originate command. Also,
if you have an IVR or a dialplan extension that does what you want - plays
prompts, accepts DTMF digits from caller, etc. - then you can use it with
inbound or outbound calls. For an outbound call just route it to the dp
ext
On Mon, May 25, 2009 at 8:19 AM, Dennis wrote:
> hi,
>
> we encounter some small problems withing the past 2 days and we are
> trying to find out more about the problems. for this we downloaded the
> debug logfiles written by fs, but we do not manage to filter all
> log-entries for one single spe
Well, that didn't help.
I've tried to put it into endpoint (conf/directory/default/*.xml) configuration
or profile (conf/sip_profiles/internal.xml) without a success.
Is there any other option?
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitc
It has come to my attention that Polycom hasn't had a business case to
support rport and STUN. If you can please kindly email marek.dutkiew...@polycom.com
and let him know you would like to see STUN and rport support in the
polycom products. Its really one of the last things missing in the
You can set this param on a profile
This should take care of it but really that gateway is broken if it
reboots on a MWI notify.
/b
On May 26, 2009, at 12:19 PM, Andrey Nepomnyaschih wrote:
Hello everyone,
Does anyone know is it possible to disable NOTIFY messages coming
out of FreeS
Hello everyone,
Does anyone know is it possible to disable NOTIFY messages coming out of
FreeSwitch for particular endpoint? The reason I'm asking it is because I have
a gateway (D-Link DVG-7111S) that reboots when it receives such a packet.
Kind Regards,
Andrey Nepomnyaschih
__
At this time we only have english it will take $1200-$2000 to record
each language.
/b
On May 26, 2009, at 12:06 PM, François Delawarde wrote:
Hi Brian,
Is FreeSWITCH going to have Spanish/French/... sounds as well or do
those need to be?
Thanks,
François.
Brian West
br...@freeswitch.
Hi Brian,
Is FreeSWITCH going to have Spanish/French/... sounds as well or do
those need to be?
Thanks,
François.
On Tue, 2009-05-26 at 11:36 -0500, Brian West wrote:
> I'm getting ready for the next sound file order for FreeSWITCH. I
> have a rather large set of files to be recorded for the z
Looks like his outbound call is failing now.
/b
On May 26, 2009, at 6:07 AM, Jason White wrote:
> Your SIP trace might give you a clue as to what happened.
> sofia profile external siptrace on
> (substituting the relevant profile for "external" in the above
> command, as
> required, and repeat
last one set will win!
/b
On May 26, 2009, at 11:49 AM, Larry Marshall wrote:
On inbound calls made to 5551212, which call_timeout will be active,
15 or 20? Is it the last hit?
Is there a URL which describes all the applications, for example,
export in the default.xml?
Thanks for this a
On inbound calls made to 5551212, which call_timeout will be active, 15 or
20? Is it the last hit?
Is there a URL which describes all the applications, for example, export in
the default.xml?
Thanks for this amazing software, Lars
In conf/dialplan/public/00_inbound.xml:
I'm getting ready for the next sound file order for FreeSWITCH. I
have a rather large set of files to be recorded for the zRTP
integration if anyone wants to help out. ;) Please contact me off
list. I would like everyone to update and try out voicemail and
nitpick anything that you feel
I can say, from having met with and talked to the CEO and founder of
Applogic that these guys are really revolutionary in their approach to cloud
computing. I spoke on a panel w/ the founder at an ISPcon several years ago,
and their approach is that of a utility company, treating computing
resource
On May 26, 2009, at 11:20 AM, dujinfang wrote:
Thanks Brain. Got ESL.so, however on my Mac it is #include
instead of .
Actually since we do -framework Ruby it should be ruby/ruby but I
think the line above the -framework Ruby should be removed since
you're doing i tthe Mac way.
/b
Not with FreeSWITCH in our testing. Now if you have stupid defaults
in your virtualization env. it might act funny but I have run FS on
EC2 without a problem.
/b
On May 26, 2009, at 1:05 AM, John Nicholson wrote:
> Virtualization has issues with timing in my experiance.
>
> Sent from my iPh
I know this but you don't have to have it in x-lite or eyebeam
directly. You just need the zfone application along with Eyebeam or X-
Lite right now.
/b
On May 26, 2009, at 12:52 AM, Jim Burke wrote:
>
> FYI...According to Counterpath ZRTP is not added to the retail
> versions of Eyebeam or
I'm attempting to replicate the behavior of an Asterisk system with
FreeSwitch and I need a feature that, I'm surprised to say, doesn't seem to
be supported (easily).
Ok, so I've setup my dialplan so that when a specific extension is hit, it
calls out to some javascript which acts like an IVR to
First off, I apologize if this has been sent multiple times, the mailing
list won't cooperate with me... Hopefully that is resolved now.
I'm attempting to replicate the behavior of an Asterisk conferencing
system and I need a feature that, I'm surprised to say, doesn't seem to
be supported (ea
Thanks Brain. Got ESL.so, however on my Mac it is #include
instead of .
But it can't find the ESL when I require 'ESL' in ruby. Even I put
ESL.so in one of the dir of $:
Any clue for me?
se...@du-sevens-mac-pro:~/workspace/test/freeswitch/oldtrunk/libs/esl$
make rubymod
cc -I/Users/sev
I've been running a production FS app on EC2 since December. It's been
really stable. Same server/instance since day1. We've haven't had any
complaints
Erik
On Tue, May 26, 2009 at 8:11 AM, Kristian Kielhofner <
kristian.kielhof...@gmail.com> wrote:
> On Tue, May 26, 2009 at 10:31 AM, Bri
Hi,
I am trying to setup ASR in FreeSwitch using Nuance ASR server and MRCP.
Both FreeSwitch and Nuance installed on Windows Server 2003. FreeSwitch
version is 1.0.3 (12567M)
I found an example in Perl at
http://www.softivr.com/wiki/index.php/FreeSWITCH_MRCP_in_Perl and decided to
do the same in
There are a few examples in the wiki, showing how to configure IVR for
inbound calls. My question lies in whether it is possible to write a
dialplan in xml or scripts to configure IVR for outbound calls. Here is a
typical scenario (of usage):
(1) Call any extension or external endpoints (maybe PST
Hi,
Following the wiki: http://wiki.freeswitch.org/wiki/Event_Socket_Library
On MacOSX 10.5, I can't get ESL for ruby work. make throws error:
sevens-mac-pro:~/workspace/test/freeswitch/trunk/libs/esl$ make rubymod
make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/Users/
seven/wor
I didn't see an error line maybe you have permission problems butyou
just pasted only the INFO line. You do know if you disable the sql
you loose the ability to show calls and channels?
/b
On May 25, 2009, at 9:57 PM, mashudi wrote:
> Dear Muhammad Shahzad ,
> it is work and the err after h
On May 26, 2009, at 10:11 AM, Kristian Kielhofner wrote:
Hey Brian,
FreeSWITCH in EC2 is a bit of a mystery to me...
Call me old fashioned but in my mind VoIP and geography are linked
in %99 of scenarios. Having VoIP services in a pure "cloud"
environment just doesn't sound like a good ide
On Tue, May 26, 2009 at 10:31 AM, Brian West wrote:
> Not with FreeSWITCH in our testing. Now if you have stupid defaults
> in your virtualization env. it might act funny but I have run FS on
> EC2 without a problem.
>
> /b
Hey Brian,
FreeSWITCH in EC2 is a bit of a mystery to me...
Call m
Thanks Brian.
2009/5/26 Brian West :
>
> On May 26, 2009, at 9:10 AM, Niall Crosby wrote:
>
>> Dear FS users,
>>
>> If I am getting Sangoma hardware to connect Freeswitch to E1, should I
>> get a card with echo cancellation or not?
>
> I would use hardware echo cancel if at all possible.
>
>> Also
On May 26, 2009, at 9:10 AM, Niall Crosby wrote:
> Dear FS users,
>
> If I am getting Sangoma hardware to connect Freeswitch to E1, should I
> get a card with echo cancellation or not?
I would use hardware echo cancel if at all possible.
> Also, is there any advantage to PCIe over PCI? (I under
Ok, that sounds doable... I have no problem banging around with some
code. Thanks for the advice...
I'm new to FIFOs and FreeSwitch in general, so please check my logic
here...
- When a user enters their conference number, I check to see if that
conference exists (because the conference shou
The makefile will have to be changed to work with OS X since the
linking is done differently.
It would be very similar to this one http://www.bkw.org/esl.imac.diff
Below will get it to compile:
imac:esl brian$ svn diff
Index: ruby/Makefile
=
There is a cloud computing company named 3Tera (AppLogic) that does have
an international presence and will keep your FreeSWITCH instance running
on a dedicated server using Xen and HA. I spoke with one of their
senior engineers about 1 month ago in regards to actually setting up an
LCR scenar
Jason,
There are many ways to accomplish this using FreeSWITCH. All of
which will require you to do a little bit of coding in js, lua or some
other language.
1. Park all callers into a fifo.. (see mod_fifo)
2. When leader auths in your script then you uuid_transfer them all
into the
Hi Artem,
Please to see that some of the stuff I wrote is useful to someone..!
I've written an FS module which will send the audio over - it's more efficient
than using unicast. Let me know if you'd like a copy.
Cheers --
Dave
- Original Message -
From: ?
To: free
Dear FS users,
If I am getting Sangoma hardware to connect Freeswitch to E1, should I
get a card with echo cancellation or not?
Also, is there any advantage to PCIe over PCI? (I understand the
difference between these and am gonig with PCIe, just checking encase
there are legacy issues that make
This one happens to every new guy trying to make FS into a dialer app using
JS.
for every sessionX you create in js with the new Session constructor
sessionX.setAutoHangup(0);
This allows the channels to remain alive outside the script once they are
hungup/transferred etc.
On Mon, May 25, 2009
Brad Tuan wrote:
> But,the response message change from "407 Proxy Authentication Required" to
> "480 Temporarily Unavailable" today.
>
> Anybody can tell me what happen??
Your SIP trace might give you a clue as to what happened.
sofia profile external siptrace on
(substituting the rel
I am sure the Auth-call is closed and the acl is closed.
And i didn't change any setting.
But,the response message change from "407 Proxy Authentication Required" to
"480 Temporarily Unavailable" today.
Anybody can tell me what happen??
2009-05-26 17:37:50 [INFO] sofia_presence.c:617
Hei!
Hvilken speditør?
Per Westrøm
Ing. Per Westrøm, Leiv Eirikssonsgt. 7, 0271 Oslo,
Norway Tlf. + 47 22444550, Fax + 47 22554964,
Mail p...@le7.no [mailto:p...@le7.no
-Opprinnelig melding-
Fra: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswi
Ahh right now I get what you are saying, I thought from the wiki that
I would have to set the feature on and then tell it what cause codes I
wanted to trap. Will fix up my dialplan cause I don't want it to trap
other causes for this scenario.
Thanks!
On Tue, May 26, 2009 at 6:09 PM, Jason White
Jim Burke wrote:
> If I understand your comment correctly, I did not have both of the
> above snippets in the dialplan at the same time. The dialplan was
> modified continually to get the correct vars that worked for my
> situation and then reloadxml to get them working.
Right, you can't have
If I understand your comment correctly, I did not have both of the
above snippets in the dialplan at the same time. The dialplan was
modified continually to get the correct vars that worked for my
situation and then reloadxml to get them working.
Regards,
On Tue, May 26, 2009 at 4:49 PM, Jason
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