On Wed, Sep 23, 2009 at 7:31 PM, Michael Giagnocavo m...@giagnocavo.netwrote:
Right off the bat: there can be tons of cleanup and refactoring, no doubt
about that. Much of the current code is to satisfy my needs in production,
which it does very well.
The current base doesn't have anything
Dear All,
I am in the process of doing IVR development on FreeSWITCH. I am
having doubt in the play_and_get_digits application. I am using Perl
language for handling IVR.
How can I play more than one sound file in play_get_digits application?
For an example,
Make sure that mod_file_string is built and loaded and then try the syntax
that is described here:
http://wiki.freeswitch.org/wiki/Mod_file_string#Examples
Instead of a comma separated list you can use ! and be sure NOT to put a
space after the ! because the function delimits the arguments with
On Wed, Sep 23, 2009 at 11:44 PM, costa.zikal...@gmail.com wrote:
Hi Mathieu
Hi
Does this mean you are able to use email-to-fax?
If yes, would yes would you care to briefly describe how you configured that.
Not yet, but I plan to do so. Il will post my setup in FS wiki.
I first have to
Great - hopefully we'll meet on IRC or the conference sometime on Friday. Email
me when you're on.
A few questions I have:
Clarity - I agree with you there, and thanks!
Testability - is this even remotely practical? Looking at our FS code plugins,
there's simply no way any amount of test
Hello Michael,
Do you still want to follow up on this? I'm having difficulty gathering
the old stuff in an understandable form. Also, it looks like the open
source ACD Spice Telephony by Andrew Thompson can do just what you might
need.
Remko
Van:
Hi All
Has anyone had any experience doing ASR with mod_unimrcp in javascript? In
particular, how do you deal with grammars? A simple piece of demo code would be
massively appreciated - the documentation on mod_unimrcp ASR javascript
bindings is TBD, which I assume means 'to be documented' ...
If I am correct you need to create a sip profile per interface and
hardcode/set the IP address of each interface correctly in the SIP RTP
fields of the profile.
Then you need to set carefully the correct NAT and auth options for each
profile
/aep
--
Stopping junk mailers is good for the
Hi,
Is there any simple way to know:
who is subscribed to certain events via ESL?
check which events i have subscribed during a ESL session?
control which events can one user subscribe?
disable the subscription of certain events and not all at the same time?
/aep
--
Stopping junk mailers is
Hello guys,
lately I've been trying to compile Freeswitch for a MIPS architecture.
With the help of the community I've understood that my target
architecture was wrong because of limitations in the SDK toolchain's.
I'm not writing now to get help but to start (I hope) a discussion.
I would like to
Dear Sir,
I'm looking for A-Z price and quality should be same
http://voicetrading.com. Now i use http://voicetrading.com it's good
quality but very bad support. some time i can payment by credit card,
paypal some time can't i don't know why.
BG
Dome C.
hello,
i'm on latest trunk and for some reason i cannot get timestamps dumped in my
cdrs. I use mod_cdr_csv with default settings plus i enabled to get both a
and b legs dumped.
cdr_csv.conf.xml:
configuration name=cdr_csv.conf description=CDR CSV Format
settings
!-- 'cdr-csv' will
Hi,
Now I seem to reach the webserver. How do i checkout a local copy to run
the builder?
/aep
--
Stopping junk mailers is good for the environment
It seems I had a port forwarded incorrectly for the external access to
the git web interface. here it is again:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
I ported my perl based FS-Logfile-Q931-HexDumps-to-pcap script to C
(linux). It reads FS's logfile (loglevel DEBUG) grabs the Q931 hex dumps
and puts them in a .pcap file which is directly readable and decodeable
by Wireshark/tshark.
The big
This belongs on freeswitch-biz
/b
On Sep 24, 2009, at 5:09 AM, Dome Charoenyost wrote:
Dear Sir,
I'm looking for A-Z price and quality should be same
http://voicetrading.com. Now i use http://voicetrading.com it's good
quality but very bad support. some time i can payment by credit
You can also use phrase macros. (and no its not just for TTS ;) )
/b
On Sep 24, 2009, at 2:02 AM, Michael Collins wrote:
Make sure that mod_file_string is built and loaded and then try the
syntax that is described here:
http://wiki.freeswitch.org/wiki/Mod_file_string#Examples
Instead of
Hi Gabe
Thanks for you response to this question.
Do you perhaps have a link to an example (or just further detail) to what
you've descibed below.
I guess one would also use a similar setup to generate dialplans from web
forms.
Thanks again,
Costa
2009/9/24 Gabriel Gunderson g...@gundy.org
On Thu, Sep 24, Alberto Escudero wrote:
Hi,
Now I seem to reach the webserver. How do i checkout a local copy to run
the builder?
If you just want to build then you can put a line like
deb-src http://ppa.launchpad.net/pbxbuntu-drivers/ppa/ubuntu jaunty main
in your sources.list replace
Hello list,
I'm currently testing file transfer within the same SIP domain and
the situation has just got odd! When I send a PDF or MP3 file,
Freeswitch allows its transfer (meaning SIP Traffic is okay:
INVITE, 180 Ringing, 200 OK, SEND Transaction, BYE, 200 OK)
But when I try to send JPEG files,
mime.types file is for http server stuff not SIP
we have never even tried to support file transfer over SIP, it's a feature
request at this point.
On Thu, Sep 24, 2009 at 9:39 AM, David Nembrot david.nemb...@sogeti.comwrote:
Hello list,
I'm currently testing file transfer within the same SIP
On Thu, Sep 24, 2009 at 1:09 AM, Remko Kloosterman r.klooster...@mtel.nlwrote:
Hello Michael,
Do you still want to follow up on this? I’m having difficulty gathering the
old stuff in an understandable form. Also, it looks like the open source ACD
Spice Telephony by Andrew Thompson can do
On Wed, Sep 23, 2009 at 7:34 AM, Svetik VOIP svetikv...@gmail.com wrote:
Brian,
Thank yo very much for your reply.
I have tried to add transfer_ringback action, but it did not solve my
problem.
Destination phone is ringing, but the person who is calling does not hear
ringing tone in hte
Hi Jim,
From conceptual viewpoint, mod_unimrcp is just an alternate implementation of
an abstract ASR/TTS interface FreeSWITCH provides.
Therefore you can use it exactly the same way as other ASR/TTS modules.
See scripts/javascript/ps_pizza.js in FS tree for a working example.
The only thing
This happens with our polycoms as well ... NAT on phone and PBX. Still haven't
had time to look into it so I disabled the sound for new message waiting ...
for now it doesn't keep beeping every few minutes.
On September 23, 2009 08:08:39 pm Brian West wrote:
NO I have never seen it happen what
Use phrase macros as Brian said.
On Thu, Sep 24, 2009 at 1:09 PM, Brian West br...@freeswitch.org wrote:
You can also use phrase macros. (and no its not just for TTS ;) )
/b
On Sep 24, 2009, at 2:02 AM, Michael Collins wrote:
Make sure that mod_file_string is built and loaded and then
Hi Arsen
Thanks for your message - it inspired us to do what we should have done in the
first place, and look at the code. The problem we were having was related to
grammar files not being available locally. Now we have discovered the
builtin: keyword we are up and running :)
Many thanks
Jim
Excellent, thanks!
-MC
On Thu, Sep 24, 2009 at 4:50 AM, Helmut Kuper helmut.ku...@ewetel.dewrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
I ported my perl based FS-Logfile-Q931-HexDumps-to-pcap script to C
(linux). It reads FS's logfile (loglevel DEBUG) grabs the Q931 hex
On Thu, Sep 24, 2009 at 2:14 AM, William King quentus...@gmail.com wrote:
Sure, post it here and I'll add it in the next build in a few hours.
See attached file.
Unfortunately mod_skypiax author did not placed config files
(skypiax.conf.xml, skypiax.X.conf) into conf/autoload_configs, so they
On Wed, Sep 23, 2009 at 10:22 PM, Gabriel Gunderson g...@gundy.org wrote:
On Wed, Sep 23, 2009 at 10:44 PM, Seven Du dujinf...@gmail.com wrote:
It not possible to use 0.0.0.0 for on profile. however, you can create
more
sip profiles for each of your interfaces. Search freeswitch-users
Hmm... That is interesting... swig is needed I believe only for the
mod_perl or the esl modules. I'll find out more information and put it
on the correct package.
I will also update the mod_skypiax config files in the *.install files.
-William King
Dmitry Bely wrote:
On Thu, Sep 24, 2009 at
if you bind the same profile to more than one ip, all your traffic would
come in one ip and out another and cause tremendous confusion.
To see a working example of this problem see asterisk
https://issues.asterisk.org/view.php?id=2358
(note bkw and mikej contribute to this bug)
Here is me
It beeps every few min cuz you register and we send you a notify again.
/b
On Sep 24, 2009, at 11:43 AM, Chris Burns wrote:
This happens with our polycoms as well ... NAT on phone and PBX.
Still haven't
had time to look into it so I disabled the sound for new message
waiting ...
for
because it's waiting for the other party to answer
if you want to hear ringback or music while you are waiting
see:
http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones
specifically transfer_ringback
On Thu, Sep 24, 2009 at 1:35 PM, Harry Vangberg ha...@vangberg.name wrote:
Hello
My
if you find the time, can you add that to the wiki too?
On Thu, Sep 24, 2009 at 12:07 PM, Jim Page jim.p...@redmatter.com wrote:
Hi Arsen
Thanks for your message – it inspired us to do what we should have done in
the first place, and look at the code. The problem we were having was
Not exactly, as I said, if the original B-leg doesn't hang up, it will
wait 20 second before transfering to the new extension (check the
timestamps!) - but if the original B leg hangs up, it gets transfered
to the extension immediately.
Look at this:
2009-09-24 18:29:48.138326 [DEBUG]
Brian,
I am using the latest firmware. Would a solution be to lower the
registration time so that notifies happen more often? I know that would
increase the amount of traffic on the network but it would keep the
light lit when a user has a message.
DigiLord
On Thu, 2009-09-24 at 13:32
Hi guys!
I'm considering to use SIMPLE protocol for IM in my application, but get a
following error trying to send a message from one registered user to
another:
[ERR] sofia_presence.c:93 Chat proto [sip]
from [1...@xx.xxx.xx.xx]
to [1...@xx.xxx.xx.xx]
11
Invalid Profile xx.xxx.xx.xx
On Wed, Sep 23, 2009 at 8:08 PM, William King quentus...@gmail.com wrote:
Another problem: all music packages in the repository (except 48Khz) are empty.
- Dmitry Bely
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
You're saying the binary sounds files are empty?
And only the music ones?
-William King
Dmitry Bely wrote:
On Wed, Sep 23, 2009 at 8:08 PM, William King quentus...@gmail.com wrote:
Another problem: all music packages in the repository (except 48Khz) are
empty.
- Dmitry Bely
Hello
On Fri, Sep 25, Dmitry Bely wrote:
On Wed, Sep 23, 2009 at 8:08 PM, William King quentus...@gmail.com wrote:
Another problem: all music packages in the repository (except 48Khz) are
empty.
If your speaking of the source package then they should be. If the binary
package
On Thu, Sep 24, 2009 at 08:40:16AM -0700, msc wrote:
On Thu, Sep 24, 2009 at 1:09 AM, Remko Kloosterman
r.klooster...@mtel.nlwrote:
Hello Michael,
Do you still want to follow up on this? I?m having difficulty gathering the
old stuff in an understandable form. Also, it looks like
On Fri, Sep 25, 2009 at 1:00 AM, Frank Carmickle fr...@carmickle.com wrote:
Hello
On Fri, Sep 25, Dmitry Bely wrote:
On Wed, Sep 23, 2009 at 8:08 PM, William King quentus...@gmail.com wrote:
Another problem: all music packages in the repository (except 48Khz) are
empty.
If your speaking
Here is the requested SIP trace that Anthony wanted.
http://pastebin:freeswi...@pastebin.freeswitch.org/10479
This is for ext 1...@192.168.0.2.
On line 168 the phone thinks there are no messages.
On line 206 the phone thinks there are no messages.
On line 309 the phone thinks there are
Alright. I was able to get the freeswitch project officially on
launchpad. So here are the new links:
Nightlies:
https://launchpad.net/~freeswitch-drivers/+archive/freeswitch-nightly-drivers
Official releases plus major bug fixes:
https://launchpad.net/~freeswitch-drivers/+archive/ppa
-William
in that case, it's probably a delay in the media stream where the app is
queued when you press the key
try updating to trunk and add the new i flag to the flags param i.e. 1 b ai
transfer::ff-transfer XML public
On Thu, Sep 24, 2009 at 2:04 PM, Harry Vangberg ha...@vangberg.name wrote:
Not
Thanks William,
This is very helpful.
--Stephen
On Thu, Sep 24, 2009 at 3:58 PM, William King quentus...@gmail.com wrote:
Alright. I was able to get the freeswitch project officially on
launchpad. So here are the new links:
Nightlies:
There are a few other things I can think would be nice additions to
mod_managed. Maybe an event handler that does not require a thread to
be sitting and waiting for events trying in a loop would be nice,
instead something that is triggered each time there is a certain event
class
There are a number of examples out there such as:
http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/intralanman/PHP/fs_curl/
Mike
On Sep 24, 2009, at 7:02 AM, Costa Zikalala wrote:
Hi Gabe
Thanks for you response to this question.
Do you perhaps have a link to an example (or just
If you need to be able to do granular permissions like that you would
either need to extend mod_event_socket or write a proxy that handled
that.
Mike
On Sep 24, 2009, at 5:23 AM, Alberto Escudero wrote:
Hi,
Is there any simple way to know:
who is subscribed to certain events via ESL?
I know of at least one person who has had good luck with small
applications on arm, in fact there are good working instructions for
how to cross for arm on the wiki that are known to work.
Mike
On Sep 24, 2009, at 5:34 AM, Cavalera Claudio Luigi wrote:
Hello guys,
lately I've been trying
I can confirm you should not need the swig dependency at all for
anything.
Mike
On Sep 24, 2009, at 1:49 PM, William King wrote:
Hmm... That is interesting... swig is needed I believe only for the
mod_perl or the esl modules. I'll find out more information and put it
on the correct
Can you get these same values in xml-cdr? I don't think csv was ever
intended to work with different cdrs for a and b leg, it was more
intended as a more familiar interface for those coming over from
asterisk.
Mike
On Sep 24, 2009, at 6:10 AM, Tihomir Culjaga wrote:
hello,
i'm on
I think you need to enable presence as well as have the right profile
aliases in place (they are in the default configuration).
Mike
On Sep 24, 2009, at 4:20 PM, RobertT wrote:
Hi guys!
I'm considering to use SIMPLE protocol for IM in my application, but
get a following error trying to
It has been removed from the dependencies. Thanks go to the reporter for
finding the extra depends.
A new round of builds just went out and built. Let me know if you find
something else. Also mod_skypiax should be available.
-William King
Michael Jerris wrote:
I can confirm you should not need
Dear All,
mod_file_string is working for me. Is there any problem by using
mod_file-string? If it so, how can use phrase macros in perl using ESL::IVR
module?
Please provide your valuable idea...
On Thu, Sep 24, 2009 at 10:25 PM, Diego Viola diego.vi...@gmail.com wrote:
Use phrase
Dear All,
I am doing IVR by using perl ESL libraries. I have used ESL::IVR
module. I get the DTMF by using playAndGetDigits subroutine which is defined
in ESL::IVR.pm. The DTMF digit stored in freeswitch digit variable.
To get the freeswitch variable I used getVar subroutine which is
Or use the socket without async so that it blocks till the action is
complete.
/b
On Sep 25, 2009, at 12:13 AM, velusamy velu wrote:
To get the freeswitch variable I used getVar subroutine which is
defined in ESL::IVR.pm file. When I print that digits, Perl program
prints empty
Either work... file string is new... we use phrase macros in the
default IVR menu... its a nice way to make something internationalized.
/b
On Sep 24, 2009, at 11:56 PM, velusamy velu wrote:
Dear All,
mod_file_string is working for me. Is there any problem by
using mod_file-string?
[Just catching up on this thread.]
William King quentus...@gmail.com wrote:
I would be more than happy to share the code I use.
Here is the git repo:
http://git.home.quentustech.com:9191/cgit.cgi/freeswitch-ubuntu/
When you would like your changes to the Debian build infrastructure in
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