ignore_early_media=true is not going to do the trick since once the IVR
picks up the call on leg A, the ring tone is stopped and the IVR is going to
play pre-recorded voice menu. And the freeswtich is going to send DTMF to
reach a certain extension number say 101. Then the ring tone is going to
sta
Hi All,
I ran some longer tests with FS 1.0.3 acting as an SBC.
The test machine has the following specs:
- Intel Quad Core Q9550
- 8GB RAM (far too much from what I saw)
After 3 days running SIPP with 750 simultaneous calls (1500 channels) at
20cps mean (50cps max) and call duration of
Hi all,
Anyone interested in contributing to a Qt interface in order to make a
decent softphone using FS please reply to this thread.
(also give your availability so we can have a conference call to
decide stuff)
Thanks,
Math
___
Freeswitch-users m
Usually ringing is done in early media... so the best bet would be to
ignore_early_media=true
/b
On Mar 1, 2009, at 9:05 PM, Henry Huang wrote:
Well, I knew it would be some future fantasy for now..
If not human detection. I guess will try to use Dialplan Tools wait
for silence to wait til
Well, I knew it would be some future fantasy for now..
If not human detection. I guess will try to use Dialplan Tools wait for
silence to wait till the ring tone is finished ,then connect the other leg.
On Sun, Mar 1, 2009 at 6:28 PM, Brian West wrote:
> NO. You want something that people
NO. You want something that people THINK exists and works well...
Reliable human/voice detection doesn't exist in ANY form.
/b
On Mar 1, 2009, at 8:20 PM, Henry Huang wrote:
> Does the freeswitch VAD is able to distinguish ring tone from human
> voice?
> The scenario is to originate a call
Does the freeswitch VAD is able to distinguish ring tone from human voice?
The scenario is to originate a call to a IVR system(don't connect the other
leg here yet) and dial DTMF to get to the designated extension number , once
someone picks up and say hello ( detected by VAD) now release to connec
There are examples on the wiki for this.
Mike
On Mar 1, 2009, at 3:10 PM, Rex_Alex wrote:
>
> Hi Shelby Ramsey,
>
> I would like to do the same in php script.
>
> Please post me a sample.
>
> Thanks,
> Rex.
>
>
> Shelby Ramsey wrote:
>>
>> Rex:
>>
>> The basis for xml_rpc or mod_event is someth
Hello Brian,
thanks for the info. I am a step further, but it cannot load the grammar
files.
I am sending through event_socket:
SendMsg
call-command: execute
execute-app-name: detect_speech
execute-app-arg: pocketsphinx yes no
However I get the message (also when I am using Pizza demo):
2009-03-
Hello,
I have the following problem while providing callback (mod_eventsocket
is used):
1) I want to call a certain destination number A with a suppressed
caller_id_number (this works fine with some vars in the origination string)
2) The destination number A picks up the phone and enters a target
If i'm not mistaken those events will have a member-id in them so you
can tell who they belong to.
/b
On Mar 1, 2009, at 12:01 PM, Cameron Sorlie wrote:
> Using voice activity detection (VAD) in FreeSWITCH, how might I then
> distinguish which side of a call any given TALK or NOTALK event
Using voice activity detection (VAD) in FreeSWITCH, how might I then
distinguish which side of a call any given TALK or NOTALK event relates
to? I am interested not just that there is activity on the call, but
interested also in which party on the call is speaking (or not).
Cam
I, for one, run often FS on an eeepc900 (one year old).
NEver tested max concurrent SIP calls, but for sure is able to run concurrently:
- one FS
- two SIP calls
- two Skypiax calls
- two linux Skype client instances
- two Skype calls
Also, I often use it to generate 6 or 8 concurrent Skype calls
Rex,
I've never actually used PHP for this type of thing ... but you might want
to start by looking here:
http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/php/single_command.php?r=12216
or
http://wiki.freeswitch.org/wiki/PHP_Event_Socket
Good luck. I'm sure some other folks here who u
Hi Shelby Ramsey,
I would like to do the same in php script.
Please post me a sample.
Thanks,
Rex.
Shelby Ramsey wrote:
>
> Rex:
>
> The basis for xml_rpc or mod_event is something along the lines of:
>
> api $command
>
> As an example to originate a call (using xml_rpc or mod_event) you
Rex:
The basis for xml_rpc or mod_event is something along the lines of:
api $command
As an example to originate a call (using xml_rpc or mod_event) you would do:
api originate sofia/external/$some...@$ip:$PORT $EXTENSION xml $context
What language are you trying to do this in?
SDR
__
Hi,
Learned how to enable mod_xml_rpc but didn't find any samples.
Please post me a sample to send requests(like dial) and receive
responses(like uuid) from FreeSWITCH using mod_xml_rpc
Please assist.
Thanks,
Rex.
Ken Rice-2 wrote:
>
> Check out ESL for PHP, Perl etc, or you can use mod_x
Check out ESL for PHP, Perl etc, or you can use mod_xml_rpc to control
things Both methods work well
K
From: Rex_Alex
Reply-To:
Date: Sun, 1 Mar 2009 11:13:34 -0800 (PST)
To:
Subject: [Freeswitch-users] To do telephony functions from web page
Hi All, I am trying to do the telephonic fu
Hi All,
I am trying to do the telephonic functions(like dial, hangup, conference and
etc.) from a webpage (for a customization) rather than doing it from a soft
phone.
What would be the optimal way of doing it?
Please suggest.
Thanks,
Rex.
--
View this message in context:
http://n2.nabble.co
Hello
As an easy way to show a Freeswitch server to prospects, I'm thinking
of buying an Asus notebook along with a Sangom USB FXO gateway.
www.telephonydepot.com/Catalog/Sangoma/Sangoma-USB-FXO-U100-2-Port
If someone's been using those two thingies, I'm curious to know if
they happily run Fre
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