Hi,
I would like to use Freeswitch to provide a Video/Voice mail service that is
integrated with an email service.
I would like to have the ability to email the Video/Voice messages as well
as the SIP users being able to collect their Video messages using their
video soft-phones.
Has anyone
Note I was saying your caller id problem, how did you see the
undesired caller id when you got CALL Rejected?
On Jun 18, 2009, at 1:10 PM, Edmar Cruz wrote:
Not working... CALL Rejected
dujinfang wrote:
comment lines in the user directory do the trick:
variable
I'm not quite sure if this is the expected behaviour, I just wanted to make
sure.
I've developed a simple IVR application using event socket. I dial in to the
dialplan and park the call, and then I let the IVR application do whatever it's
supposed to. I basically listen for DTMF events and
Yes, I removed the tags but with no effect. I think the problem is that the
webserver doesn't look in the directory where featuers.xml is deposited on the
freeswitchserver (/opt/freeswitch/conf/dialplan/).
The issue is that FS finds the context when dialplan is the directory
can you try uuid_record uuid stop filename before playback?
On Jun 18, 2009, at 3:07 PM, Peter Olsson wrote:
I’m not quite sure if this is the expected behaviour, I just wanted
to make sure.
I’ve developed a simple IVR application using event socket. I dial
in to the dialplan and park
Call me crazy but your DB DSN appears to be zenoss, but you have a db dsn of
tcapi still in the config file.
If your test was:
# isql zenoss edmar edmar
Then zenoss should be your db_dsn:
param name=db_dsn value=zenoss/
Not
param name=db_dsn value=tcapi/
You should be seeing
Yes I guess this would probably solve the issue :) But since I stumbled across
this weird behaviour I just wanted to make sure if this was expected or not, or
if it might be a bug...
I thought playback was just sending the audio to the caller, but in this case
it seems that playback sends it
Hi is there any possible free sites ip that i can connect so I can could to
any mobiles phones?
I know some several ip sites has the capability to call for free Ip to
Voip... I know freeswitch can do this
Can you give me an example site?
--
View this message in context:
There are gateways that allow you to set your own caller ID? I thought it'd
always use the number of the SIM.
Jan
On Thu, Jun 18, 2009 at 12:28 AM, jay binks jaybi...@gmail.com wrote:
Ive used these in the past.
Ok thanks a lot for that. Sorry my mistake..
Darren Schreiber wrote:
Call me crazy but your DB DSN appears to be zenoss, but you have a db dsn
of
tcapi still in the config file.
If your test was:
# isql zenoss edmar edmar
Then zenoss should be your db_dsn:
Not
Hi,
I have setup FS for both inbound and outbound.It is working fine.
Now I would like to configure Automatic Call Distribution(ACD).How to
configure it in Freeswitch?
Sam
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Hi,
I have been trying to pick up TALK and NOTALK events but with no
success, I have enabled VAD for both in my config and the rtp is
stopping and starting as expected however when I hook up to the event
socket and request event talk notalk nothing is ever fired, any
thoughts on where I am
I suspect you're going for TALK and NOTALK as the event names?
its CUSTOM conference:: maintenance
/b
On Jun 18, 2009, at 8:00 AM, Steven Brown wrote:
Hi,
I have been trying to pick up TALK and NOTALK events but with no
success, I have enabled VAD for both in my config and the rtp is
If you're donating you can send it to my paypal br...@freeswitch.org,
I also received the sound order for the zrtp sound files and a few
odds and ends we needed. The order was 650 dollars and thus far I
have only received a 50 dollar donation to help pay for it. So if you
wanna pitch in
Wow, I apologize for the duplicate posts.
The mailing list didn't want to cooperate with me last night...
j3flight wrote:
I haven't gone to the trouble (yet) of making this work, but I believe you
could use execute_application from the conference controls to do just about
anything with
What I did last night was to go ahead and modify mod_conference.c to include
a new count conference control. I've got it getting to the right place,
and spitting debug messages with the right data about which member and what
the count is, but for some reason the text-to-speech isn't working.
I am not aware of anyone who will give you free access to any kind of PSTN
network. If you do find someone please let us in on the secret. :)
-MC
On Thu, Jun 18, 2009 at 2:58 AM, Edmar Cruz darklio...@yahoo.com wrote:
Hi is there any possible free sites ip that i can connect so I can could to
Where is the pastebin with all of your configuration files?
-MC
On Thu, Jun 18, 2009 at 2:17 AM, Rudolf Denert rden...@tng.de wrote:
Yes, I removed the tags but with no effect. I think the problem is that the
webserver doesn't look in the directory where featuers.xml is deposited on
the
On Thu, Jun 18, 2009 at 1:03 AM, Bilbo christian.bour...@gmail.com wrote:
Hi,
I would like to use Freeswitch to provide a Video/Voice mail service that
is
integrated with an email service.
I would like to have the ability to email the Video/Voice messages as well
as the SIP users being
Hello!
We are seeking possibilities to use CTI features with Freeswitch.
This features are:
- click-to-dial
- call popup
- answer call,hangup
- call transfer
Does Freeswitch support any cti standarts (SIP CTI aka TR/87, TAPI, CSTA..)
or there is already written module or third-party software?
On Thu, Jun 18, 2009 at 5:06 AM, selva kumar panse...@gmail.com wrote:
Hi,
I have setup FS for both inbound and outbound.It is working fine.
Now I would like to configure Automatic Call Distribution(ACD).How to
configure it in Freeswitch?
Start with this:
Hello FreeSWITCHers out there! I have it on good authority that the
FreeSWITCH developers have all convened in an undisclosed location. Rumors
that they are plotting to take over the world are not yet confirmed but I
will keep you updated as information becomes available. :)
It would be great for
Thanks for the suggestions guys, I think I will go with PORTech for now.
@João Mesquita: Let me know when mod_khomp is done, I might consider
getting some khomps in the future when the module is ready.
Regards,
Diego
On Thu, Jun 18, 2009 at 4:26 AM, Jan Kubrjan.k...@gmail.com wrote:
There are
Thanks Brian,
Yes I had been looking for TALK and NOTALK, CUSTOM conference::maintenance
works great.
Steve
Message: 4
Date: Thu, 18 Jun 2009 08:16:58 -0500
From: Brian West br...@freeswitch.org
Subject: Re: [Freeswitch-users] VAD, TALK and NOTALK events
To:
Hi all!
I'm having some troubles with call quality using conferences. The
scenario is like this:
An agent makes a call to freeswitch and enters in a conference room
waiting for outbound calls; on the other side there is an application
generating outbound calls and when one is answered it is
Please post bugs to http://jira.freeswitch.org
/b
On Jun 18, 2009, at 11:20 AM, Victor Toofic wrote:
Hi all!
I'm having some troubles with call quality using conferences. The
scenario is like this:
An agent makes a call to freeswitch and enters in a conference room
waiting for outbound
Done :)
Guten Appetit
On Thu, Jun 18, 2009 at 5:19 PM, Michael Collins m...@freeswitch.org wrote:
Hello FreeSWITCHers out there! I have it on good authority that the
FreeSWITCH developers have all convened in an undisclosed location. Rumors
that they are plotting to take over the world are
Thank you so much! The devs are really loving this.
-MC
On Thu, Jun 18, 2009 at 11:40 AM, Saeed Ahmad saeedahmad1...@gmail.comwrote:
Done :)
Guten Appetit
On Thu, Jun 18, 2009 at 5:19 PM, Michael Collins m...@freeswitch.orgwrote:
Hello FreeSWITCHers out there! I have it on good authority
Hi All,
I've tested this new variable and everything works grand. I've tested
recording to wav,mp3 and shoutcast and in all cases the sample rate is set
correctly. I was about to post an entry on the wiki but I discovered a very
similar variable already there called record_rate. I've tested this
On Jun 18, 2009, at 11:54 AM, Andy wrote:
1) I notice that when I change the sample rate it automatically
changes the bit rate too. I understand why this is the case but
wondered if it was just as easy to be able to control the bitrate as
well as the sample rate.
If you're talking about
Thanks Brian,
So, just to calrify will the base call always be 8kHz?
On a related note, do you happen to know the bitrate of each open
channel/live call? Is it 16 kilobits per second like the recorded audio? I
need to do some calculations on the badwidth required to handle a certain
number
Most calls are at 8kHz. The formula for bandwidth is sampling rate *
bit rate * channels, so 8000 * 8 * 1 = 64000 = 64kbps (per way).
Math
On 18-Jun-09, at 1:01 PM, Andy wrote:
Thanks Brian,
So, just to calrify will the base call always be 8kHz?
On a related note, do you happen to know
look in mod_shout you'll see my calculations.. I think it has to be
multiples of 16 if I recall.
/b
On Jun 18, 2009, at 1:01 PM, Andy wrote:
On a related note, do you happen to know the bitrate of each open
channel/live call? Is it 16 kilobits per second like the recorded
audio? I need
The call rates we support are 8, 16,32 and 48k
/b
On Jun 18, 2009, at 1:01 PM, Andy wrote:
Thanks Brian,
So, just to calrify will the base call always be 8kHz?
On a related note, do you happen to know the bitrate of each open
channel/live call? Is it 16 kilobits per second like the
- plus UDP/RTP overhead. Budget 10 calls/megabit for G.711 and you'll
have a bit of headroom available.
--Dave
Most calls are at 8kHz. The formula for bandwidth is sampling rate *
bit rate * channels, so 8000 * 8 * 1 = 64000 = 64kbps (per way).
Math
On 18-Jun-09, at 1:01 PM, Andy
or go over the limit and you'll have Max Headroom =D
On Thu, Jun 18, 2009 at 1:16 PM, David Knell d...@3c.co.uk wrote:
- plus UDP/RTP overhead. Budget 10 calls/megabit for G.711 and you'll
have a bit of headroom available.
--Dave
Most calls are at 8kHz. The formula for bandwidth is
Thank you for all the patience and effort. You've done a great work! Have a
great meal!
On Thu, Jun 18, 2009 at 12:48 PM, Michael Collins m...@freeswitch.orgwrote:
Thank you so much! The devs are really loving this.
-MC
On Thu, Jun 18, 2009 at 11:40 AM, Saeed Ahmad
Hi, I am new user of Freeswitch, I am having trouble doing basic
configurations. Somebody could help me how to configure a simple extension?
Thanks
sorry for my bad english
--
Mario Uzae
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Sure, but you need to provide more details, what do you want to do exactly?
On Thu, Jun 18, 2009 at 12:58 PM, Mario Guerra Uzae da Silva -
mariou...@gmail.com wrote:
Hi, I am new user of Freeswitch, I am having trouble doing basic
configurations. Somebody could help me how to configure a
Hi Guys,
This one has me a little baffled. If have a recent build (in the last
week) of FS installed on two near identical HP servers. One happily
runs 400 concurrent calls at around 50% CPU. The other can only run
around 50 calls without the CPU going to 98%. Identical configs and lua
I have defined the following template in autoload_config/cdr_csv.conf.xml:
template
name=sql${caller_id_name},${caller_id_number},${destination_number}
,${context},${start_stamp},${answer_stamp},
${end_stamp},${duration},${billsec},${hangup_cause},${uuid},${ble
Is this possibly an issue to do with a newer tickless kernel?
see
http://www.nabble.com/FreeSWITCH-under-the-Linux-2.6.29-kernel-td23248559.html
Tony
On Thu, Jun 18, 2009 at 3:54 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Hi Guys,
This one has me a little baffled. If
Hello everyone,
I've setup one of my build servers to do a fresh check out of SVN
trunk and build AstLinux with it every day at 2AM EST. The ISO and
build log (for the curious) are available here:
http://mirror.astlinux.org/freeswitch/daily/
I just ran a test build but daily builds will
So I rebooted, installed some OS updates, synched up, and am running again.
I've also been doing closer comparisons between the conference I'm running
and the same phones through VOIP to other locations (like between the phones
without the conference).
The lag isn't as bad as it was, a
OK, so I did some more experimenting today. I found a problem with the code
I'm using (again, this is off the current trunk, but with some small
modifications):
conference_member_say in mod_conference.c is simply not working. There are
several messages in there that can theoretically tell the
Hello, I am planning to build a plataform to sell content, pictures, tones,
MMS, etc.
Do you know wich GSM 3G boards should work? Anyone has done this?
Greetings!
Edwin
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Thanks! I added a link from the wiki...
http://wiki.freeswitch.org/wiki/Download_FreeSWITCH#Linux
On Thu, Jun 18, 2009 at 1:36 PM, Kristian
Kielhofnerkristian.kielhof...@gmail.com wrote:
Hello everyone,
I've setup one of my build servers to do a fresh check out of SVN
trunk and build
Thanks for all of your help!
On Thu, Jun 18, 2009 at 6:26 PM, Darin Weeks d...@unwire.it wrote:
Thanks! I added a link from the wiki...
http://wiki.freeswitch.org/wiki/Download_FreeSWITCH#Linux
On Thu, Jun 18, 2009 at 1:36 PM, Kristian
Kielhofnerkristian.kielhof...@gmail.com wrote:
Hello
Do you have any way to ensure that those variables are populated? Can you
manually set those in the dialplan? Also, are you doing a leg only or b leg
only or both?
-MC
On Thu, Jun 18, 2009 at 2:54 PM, Lars Zeb larc...@yahoo.com wrote:
I have defined the following template in
Hi,
I need to have the hability to negotiate the codec in a session (using
proxy media or bypass media), unfortunally I've been unable to achive
this due the documentation that I've found about it's vague.
I've already tried using absolute_codec_string and everything that
says in
I also saw the option for the "announce-count" conference parameter
(which i assume is what you're trying to use) and it didn't seem to
work for me either. I couldn't figure out whether I was doing
something wrong or if it was not working - that's why I implemented it
in JS. Looking at the
I was indeed looking at announce-count, but from the code, it looks like
that was designed to announce to the caller how many people were on the
conference only when they were joining and the number was over a threshold
specified in the profile. Not exactly what I was looking for, but it did
help
FYI, the devs report that they are at the restaurant! Last chance to
pitch in and feed the troops. :) hit the paypal button on the main
FreeSWITCH page:
http://www.freeswitch.org
Keep those devs happy and fed and version 1.0.4 will be here before
you know it!
-MC
I got one... But its a secret...
mercutioviz wrote:
I am not aware of anyone who will give you free access to any kind of PSTN
network. If you do find someone please let us in on the secret. :)
-MC
On Thu, Jun 18, 2009 at 2:58 AM, Edmar Cruz darklio...@yahoo.com wrote:
Hi is there
try to run verbose_event before answer or bridge might help.
On Jun 19, 2009, at 3:54 AM, Lars Zeb wrote:
I have defined the following template in autoload_config/
cdr_csv.conf.xml:
templatename=sql${caller_id_name},${caller_id_number},$
Trying to do a local test for faxing. Keep getting an error. Can someone tell
me how to correct this?
Tim
default dialplan:
extension name=test_rxfax_stream
condition field=destination_number expression=^8000$
action application=answer /
action application=playback
http://versafon.com/versafonweb/Software.jsp
Essentially it's a wrapper around inbound socket interface, not all
events supported yet, and not all event parameters/variables. It's multi
threaded and scaled well in testing.
We offer commercial support and development for FreeSwitch as well.
Right now, I am working on a board that will soon support all those features
but it isn't compatible to FreeSWITCH just yet.
Other then that, there was thread here before discussing PorTech GSM
gateways. They might be able to help.
If you are interested in using other platform with the Khomp
As far as using multiple digits in the conference controls, that doesn't seem
possible. I was hoping I could make all the commands require a preceding *,
like *1 for mute, *2 for lock, etc but that didn't work. I'm sure that
could be added, but then you have other silly issues to worry about...
extensions from 1000 - 1019 are available with password 1234 by
default conf
On Jun 19, 2009, at 12:58 AM, Mario Guerra Uzae da Silva - wrote:
Hi, I am new user of Freeswitch, I am having trouble doing basic
configurations. Somebody could help me how to configure a simple
extension?
I would like to thank everyone for Dinner... we had a great time...
now MORE CODE!!!
/b
On Jun 18, 2009, at 7:51 PM, Michael S Collins wrote:
FYI, the devs report that they are at the restaurant! Last chance to
pitch in and feed the troops. :) hit the paypal button on the main
FreeSWITCH
I've been using multiple digits successfully right from the start, about 2
or 3 weeks ago. They do the separation of *1 and *10 the same way as several
other systems -- by time. If you dial *, then 1, then wait past a timeout,
then 0, you'll get *1, and *10 if you did it faster. I've tested by
Tim,
We need some information, specifically we need you to turn on debugging at
the console and give us the log from start of call to the very end. Go to
the CLI and press F8 (or type console loglevel debug) and then initiate
the call. Capture everything from the CLI from start to finish, then
Hi Edwin,
Rather than using a GSM/3G card, you might do better to find a mobile
services aggregator which covers the locations you're interested in -
MBlox or Sybase365 would be two places to start - and use them. You'll
get scalability, better reliability, etc.; be warned that MMS is *still*
a
Well crap, I must have had something else screwed up then with the
multiple digits... I will try it out again soon, thanks for putting me
back on track. I made some more changes to the wiki that hopefully
clean up some confusion on a few things like the caller-controls.
Actually, that's another good reason to do those wiki and/or code
comments changes... most likely, the reason you thought it couldn't be done
is that you tried it and it didn't work... but you tried it on the default
profile before you realized that it was hard coded. I know that's what I did
and
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