Hello
I was wondering if someone had succesfully configured FS to display caller
ID on a LED like this?
http://usb.brando.com/prod_detail.php?prod_id=00575
That would be a nice alternative to displaying CID information on the user's
PC screen when users need to see who's calling where they're
... or alternatively, on one of those USB digital picture frames?
www.amazon.com/Digital-Spectrum-USB-Photo-Frame/dp/B87BHC
--
View this message in context:
http://old.nabble.com/Displaying-caller-ID-on-LED--tp26280730p26280912.html
Sent from the Freeswitch-users mailing list archive at
Hi anthony,
I was in a need of flushing the events buffer without reading it.I've done
the following ESL(Async) program to flush the events.
First I register for events.
I answered the call and playback some message. Now the events would have
been queued.
I, then send noevents.
After sending
Hi,
yesterday i started to fix pronuce in mod_say_it for numbers, dates and
times. I needed to add some sound files because these was necessary for
a correct italian pronunce.
I've patched these three functions:
- play_group
- it_say_time
- it_say_general_count
I've diff it against revision
$| = 1;
I think that is what you're lookin for.
/b
On Nov 10, 2009, at 4:51 AM, lakshmanan wrote:
I was in a need of flushing the events buffer without reading
it.I've done
the following ESL(Async) program to flush the events.
___
Are non-English sound files available in the SVN version of the code?
I just tried installing the French sound files, but got an error:
Unknown target cd-sounds-fr-install
Unknown target cd-moh-fr-install
make[1]: *** [cd-sounds-fr-install] Error 1
make: *** [cd-sounds-fr-install] Error 2
Can you post your patch to jira.freeswitch.org please.
/b
On Nov 10, 2009, at 7:00 AM, Albano Daniele Salvatore - Lavoro wrote:
Hi,
yesterday i started to fix pronuce in mod_say_it for numbers, dates
and times. I needed to add some sound files because these was
necessary for a correct
Hi,
patch posted to http://jira.freeswitch.org/browse/MODAPP-362
Best Regards,
Daniele
Brian West ha scritto:
Can you post your patch to jira.freeswitch.org please.
/b
attachment: info.vcf___
FreeSWITCH-users mailing list
2009/11/9 João Mesquita jmesqu...@freeswitch.org:
Beat me with a dead cat all you want but I rather the snom m3 than the
Siemens A580IP Siemens has very low volume which makes its call quality
suck despite of being ergonomic and all...
Did you flip hte option in the base station that tells
Patch applied. // comments aren't allowed in .c files in our tree we
try hard to weed them out... anyway its committed now. And thanks for
your contribution.
/b
On Nov 10, 2009, at 8:38 AM, Daniele Salvatore Albano wrote:
Hi,
patch posted to http://jira.freeswitch.org/browse/MODAPP-362
On Tue, Nov 10, 2009 at 6:56 AM, Rupa Schomaker r...@rupa.com wrote:
Oh, and it looks like a new firmware came out for the Siemens today.
Wonder what it fixes (and breaks). Hmm.. wonder where I can find a
list of whats new.
Well, it seems to totally break g722. I haven't had a chance to
Please see the global-intercept example in the default config.
/b
On Nov 10, 2009, at 9:01 AM, Piotr Żurek wrote:
Hello.
Thank You developers for Freeswitch.
I have installed it lately and it's working quite nicely, but I have
one problem:
I need to mimic behavior of my current
Hi
Thanks for the tips. May I ask how to split the file from hadoop to the
shell? Is it like copying the file to certain dir?
I can't find any mod_shell_stream related info from the wiki. Does anyone
know how to use it?
thx,
mark
On Tue, Nov 10, 2009 at 3:29 AM, Andrew Thompson
On Tue, Nov 10, 2009 at 11:29:37PM +0800, mark morreny wrote:
Hi
Thanks for the tips. May I ask how to split the file from hadoop to the
shell? Is it like copying the file to certain dir?
I can't find any mod_shell_stream related info from the wiki. Does anyone
know how to use it?
Add the following:
action application=set data=execute_on_answer=db
delete/${domain_name}-last_dial/${called_party_callgroup}/${uuid}/.
after
action application=db
data=insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}/
in local extensions default example, or change it
I'm close to getting my SPA3102 working - he says hopefully .. . . .
Making and receiving calls seems to be OK, but the SAP3102 doesn't seem to want
to let go of the phone line once it's got it.
Example
I can receive a call, nobody answers and it goes to voicemail - working so far.
FreeSwitch
Good morning everyone. I have a question regarding using MOD XML_CURL and
returning a dial plan.
I have my system setup to respond with the following dialplan.
?xml version=1.0?
document type=freeswitch/xml
section name=dialplan description=Regex/XML Dialplan
context name=default
I believe that French and Spanish sounds are in the works by the community.
The only other sounds I'm aware of are the Russian ones.
-MC
On Tue, Nov 10, 2009 at 6:13 AM, Fred-145 codecompl...@free.fr wrote:
Are non-English sound files available in the SVN version of the code?
I just tried
On Tue, Nov 10, 2009 at 8:55 AM, Mathieu Rene mrene_li...@avgs.ca wrote:
You'll get a single xml curl request, unless you use the transfer
application, which will trigger another one.
Just curious: what about execute_extension? Does that cause a new XML CURL
request also? I didn't see
Hi,
thank you and for your work!
Where i can find coding style rules?
Best Regards,
Daniele
Brian West ha scritto:
Patch applied. // comments aren't allowed in .c files in our tree we
try hard to weed them out... anyway its committed now. And thanks for
your contribution.
/b
Hey Hadley,
jump up on irc sometimes.
Regards,
Ognjen
On Mon, Nov 9, 2009 at 9:26 PM, Hadley Rich h...@nice.net.nz wrote:
On Mon, 2009-11-09 at 14:05 -0600, Brian West wrote:
Get an ATA with a Dect handset it works much better... the Snom M3 and
the Aastra are one in the same and they
Hello.
I'm FS newbie and want connect it to SIP provider which does not
require authentication - it make authentication using my IP.
I've searched through FS documentation and didn't find clear answer.
Could you help me or maybe give a link to a doc which can help?
Thanks.
--
Sergey
As easy as:
action application=bridge data=sofia/external/$
{destination_numb...@ip_address_here /
in your dialplan. If you want to make a gateway out of it, you can
enter whatever you want in username and password since they won't be
used. (SIP works using challenge authentication which
just change the dialplan/default.xml as mentioned by brian but i think you
can't use # as the first key 'cuz it normally used as a Send key. you may
change # to * (star key).
On Wed, Nov 11, 2009 at 12:06 AM, Ognjen Seslija osesl...@gmail.com wrote:
Add the following:
action application=set
That depends on the phone... some let you do it.. some don't...
WELCOME TO VOIP!!!
/b
On Nov 10, 2009, at 3:48 PM, Nandy Dagondon wrote:
just change the dialplan/default.xml as mentioned by brian but i
think you can't use # as the first key 'cuz it normally used as a
Send key. you may
I am trying to call into a DID that is pointed to a Conf Bridge on Freeswitch
and when I have 2 people dial in, looks like the Music on Hold never stops.
Here is what my public.xml looks like:
extension name=test !-- your provider or any name you'd like to call
it --
condition
What does your config look like?
/b
On Nov 10, 2009, at 4:39 PM, Ujjval Karihaloo wrote:
I am trying to call into a DID that is pointed to a Conf Bridge on
Freeswitch and when I have 2 people dial in, looks like the Music on
Hold never stops.
Here is what my public.xml looks like:
Hello. I am very new to FreeSwitch, Telephony and IVR.
My goal is to prepare a student assessment IVR system as a college project.
But this IVR is going to be dynamic. So for each student assessment may be
different (number of questions, possible responses, flow of prompts, etc).
Is it
My mistake , it picked the default profile and was waiting for moderator in
the conference.cof.xml file that is provided with the install.
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West
Sent: Tuesday, November
As I opinion, it's not necessary write ivr script for each student. A
static ivr script load question and response dynamic is what you need.
2009/11/11 Malay Thakershi malay.thaker...@continuityhealth.com
Hello. I am very new to FreeSwitch, Telephony and IVR.
My goal is to prepare a
Hi
Sorry to ask again.
I know the command to copy file from hadoop file system to somewhere else.
But how do I make a shell command to output raw audio?
What command is it like? Is it like play()? I am confused.
Thx,
mark
On Tue, Nov 10, 2009 at 11:56 PM, Andrew Thompson
What you will probably want, if you are looking to go 'thicker' with
this would be one of the IVR scripting languages and a database
connection. For instance lua, and the database connection(either mysql
or postgresql or sqlite). ' From there you have users, questions, and
answers mapped in the
I did something like this recently.From the dial plan it is easy to
execute an external application on an incoming call with the caller's info.
At that point if you can just push it down to the LCD panel all the better,
but if your FS server is remote, and has no direct access to the client to
If you donate one to the FsGui project, I can make it happen for you.
Contact me off list if you are interested.
Regards,
JM
On Wed, Nov 11, 2009 at 1:01 AM, Mitch Capper mitch.cap...@gmail.comwrote:
I did something like this recently.From the dial plan it is easy to
execute an external
Here is the required detail.
http://pastebin.freeswitch.org/11049
On Mon, Nov 9, 2009 at 10:04 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
1) install gdb
2) run support_d/fscore_db in the tree from the working directory of the
core.
3) if you are not on svn trunk, make current
You need to install the debug packages so you the symbols because that
backtrace is useless.
/b
On Nov 10, 2009, at 10:10 PM, lakshmanan ganapathy wrote:
Here is the required detail.
http://pastebin.freeswitch.org/11049
___
FreeSWITCH-users
It doesn't look like its trying to look for symbols inside freeswitch
gdb /path/to/freeswitch/here /path/to/core/here
bt
thread apply all bt
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 10-Nov-09, at 8:10 PM,
What is meant by debug packages. Kindly specify where it is available.
On Wed, Nov 11, 2009 at 10:09 AM, Brian West br...@freeswitch.org wrote:
You need to install the debug packages so you the symbols because that
backtrace is useless.
/b
On Nov 10, 2009, at 10:10 PM, lakshmanan ganapathy
That doesn't seems to work for me.
Here is my need.
I'm using Async in the Event socket outbound.
I'll register for events plain all
I'll answer the call.
I'll playback a message.
I'll sleep for 5 seconds.
After that, I'll receive the events.
I don't need the events that are for answer and
On Wed, Nov 11, 2009 at 11:02:10AM +0800, mark morreny wrote:
Hi
Sorry to ask again.
I know the command to copy file from hadoop file system to somewhere else.
But how do I make a shell command to output raw audio?
What command is it like? Is it like play()? I am confused.
I was very
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