Hmmm no MOH wont work... since I am planning on pulling more than just 2
members into the conference and I still need ringback for the later
members as well.
Is there a direct way for me to use conference number play audiofile
to play teletone directly? or should I just records some ringing if I
Hi all,
I completed the wiki page with the comments I made in the posts:
Re: [Freeswitch-users] FreeSwitch setup as a Dumb SBC
I detailed how to setup mysql/kamailio/carrierroute to use the
carrierroute module of kamailio for LCR.
I wrote this page using my memory and history of the linux
Thanks for that, coming from a C++ background it's a refreshing change
to be looking at something that seems logical and efficient.
I'd briefly looked at the event socket and wondered if that was the way
to go. I presume that there's some sort of event generation that can
trigger and external
Hi Nik,
Here's a snipped in Perl that launches an outbound call:
if (my $sock = IO::Socket::INET-new(Proto ='tcp', PeerAddr =
'127.0.0.1', PeerPort = 8021)) {
print $sock auth XXX\n\n;
print $sock api originate
Hi rod,
It's really amazing! Well described!
Could you please explain a bit why we used Kamailio?
Kind Regards
Saeed Ahmed Tariq
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of rod
Sent: Monday,
I believe that installing the libpcap and libpcap-dev packages may fix
your problem.
--
Raul
On Tue, 2009-02-03 at 12:55 +0100, Leon de Rooij wrote:
Hi all,
I've been trying to build new debs, but debuild seems to break..
I tried trunk rev 11608 and 1.0.3RC-1 and tried building the
In addition do David's suggestion, you probably want to have your
application to watch for some specific events after the call is
originated and take action based on them. For example, you could watch
for the CHANNEL_ANSWER event and play some audio file waiting for some
digit, which is generated
Hi Saaed,
thanks for encouraging.
I'm using Kamailio to get access to the carrierroute module.
Carrierroute is a module that is able to handle very large routing table
(excerpt from carrierroute page: This modules scales up to more than a
few million users, and is able to handle more than
Depending on what you want to do, I suggest having a look at
mod_event_socket: http://wiki.freeswitch.org/wiki/Mod_event_socket
That module is a socket based interface that provides a vast range of
options to control FreeSWITCH and its applications.
Just for the record, my application is entirely
There is a file format called tone_stream that I was trying to explain
yesterday.
tone_stream://teletone spec
or
tone_stream://path=/path/to/text_file.ttml
you can use this to play tones anywhere a filename is supposed to go.
I guess loopback really is your only option if you must generate
Hmm ok ... Ill try that In my head though the api call to originate
shouldent block? but I assume since it does my head is wrong.
Thanks you for the explanation. I think you can put this one to bed now
:-P
On Tue, Feb 03, 2009 at 07:54:29AM -0600, Anthony Minessale wrote:
There is a file
hmmm ok indeed.
small mods to js files to just play a lng tone_stream full of ringy
noises and then stop them in the on answer and I have what I wanted.
Thank you very very much for all your help.
On Tue, Feb 03, 2009 at 04:16:21PM +0200, Sias Mey wrote:
Hmm ok ... Ill try that In my head
Hello
I am trying to initiate a call from javascript, it works fine for local numbers:
session1.originate(session1,
{ignore_early_media=true}user/1...@192.168.1.122);
but when I am trying to connect through sofia gateway, the connection is not
being established:
Hi all,
I've been trying to build new debs, but debuild seems to break..
I tried trunk rev 11608 and 1.0.3RC-1 and tried building the packages
with:
debuild -i -us -uc -b
(which worked before)
And now it breaks at openzap with:
cc1: warnings being treated as errors
Newbie with FS, currently have Asterisk servers front ended by Openser
Question: I have around 400 sip remote clients, if I were to deploy FS,
do I need Openser? Is there any advantage in retaining Openser?
Regards
___
Freeswitch-users mailing list
you could use the intercept app to unpark the caller without using fifo out,
then it would only work if the caller existed.
On Tue, Feb 3, 2009 at 7:51 AM, Tamas Cseke cstomi.levl...@gmail.comwrote:
Hello,
We have a problem with mod_fifo.
we monitor fifo push event on event socket,
call
Lee,
You also might want to take a look at some of the examples in the
contrib folder in the source tree. There are several items there that
use the event socket. The event socket is extremely powerful and is
suitable for a wide range of applications. However, it isn't the only
way to do things.
Are you suggesting that I should process the call externally instead of
using the dialplan? That would be neat as the audio file select could
be driven from the db select for the number. I presume that I could
also bridge the call to another number as well dependant on DTMF
selection?
Regards
Nik,
There are a lot of ways to make FS dial out and deliver messaging etc. We
are going through the process of replacing * for this purpose. For us
(getting started with the help of our friends here on the list) it has been
pretty easy.
With * we were using AMI to originate calls ... to
Jacek,
I had a similar problem once. It actually depends on your sip gateway,
but I was able to solve the problem by setting the caller id, ie:
session1 = new Session();
session1.setCallerData(caller_id_name, 8280052500);
session1.setCallerData(caller_id_number, 8280052500);
You forgot to tell us what revision of the code you're on?
/b
On Feb 3, 2009, at 11:16 AM, e schmidbauer wrote:
hey everyone. just wondering if anyone has tested recording
conferences at 48000h celt to a shoutcast stream or wav file.
we are able to have cd quality conferences with 3 members
YOU should NEVER use this method or call setCallerData at all you
should use the correct methods to override the callerid.
If its a B-Leg born from an A-Leg you use these on the on the A-Leg:
http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name
On Tue, Feb 3, 2009 at 8:53 AM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Are you suggesting that I should process the call externally instead of
using the dialplan? That would be neat as the audio file select could
I'm not saying you should, merely that you could. What I did was
Oops! Well, fortunately I don't use that voip provider anymore (nor the script).
Thanks Brian.
Nicolas
On Tue, Feb 3, 2009 at 2:25 PM, Brian West br...@freeswitch.org wrote:
YOU should NEVER use this method or call setCallerData at all you
should use the correct methods to override the
There is also an event socket library written in C called esl that is in the
fs tree in the libs directory.
This has the ability to establish connections both inbound and outbound from
FS.
There is also a perl module FreeSWITCH::Client that mr collins may be
interested in in the tree as well.
Well Openser has better NAT handling than Asterisk for a start. In
addition it takes the load off of Asterisk with regards to
registrations. Further, I'm able to have multiple asterisk servers
fronted by Openser
Finally, I've numerous posts that * chokes with sip clients 200. I
couldn't afford
FreeSwitch is very capable of handling high call setup loads... The question
is what do you consider high setup loads?
Where it is true, OpenSER/SIP/whatever its called this week can handle a
much higher packet per second load then freeswitch, freeswitch on the other
hand is capable of handling
If you're telling me that FS can handle the figures quoted, that's
plenty enough for me. I have 5,000 lines PSTN /channels, possibly
double that shortly. I need to fill all of them as quickly as possible
and maintain that level for a given period of time. So I guess I'm in
the upper medium end
Hi Rod,
Great info, Thanks!
Glad to see others are interested in the same concept.
My reasons for SER as routing core and implementation is slightly different
yet similar.
I like your Redirect model, with that you are truly using your Kamailio as
route server only. I would imagine very scalable.
In every one of my SIP sessions FreeSwitch appears to be inserting ..
Contact: sip:mod_so...@xxx.xxx.xxx.xxx:5060
Is this normal?
I only ask as it is causing some of my end points to RE-INVITE back to this
after the initial ( INVITE 100 Trying--- 200 OK )
call setup.
Actually I currently deploy FreeSWITCH for high volume usage using
FreeSWITCH + mod_easyroute (I'm the author) and an advanced LCR module that
does things like load balancing across multiple media gateways, auto route
advance, and a few other nifty things... (this LCR module uses a proprietary
im using the latest svn of freeswitch
On Tue, Feb 3, 2009 at 12:23 PM, Brian West br...@freeswitch.org wrote:
You forgot to tell us what revision of the code you're on?
/b
On Feb 3, 2009, at 11:16 AM, e schmidbauer wrote:
hey everyone. just wondering if anyone has tested recording
FreeSWITCH Version 1.0.trunk (11567)
check out these sample recordings
http://bwrl.org/recordings/2009-01-31-12-07-49.mp3
http://bwrl.org/recordings/2009-01-31-12-07-49.wav
http://bwrl.org/recordings/test2.mp3
http://bwrl.org/recordings/test2.wav
the conferences were recorded as wav files, i then
We are attempting distributed radio. We plan on having the hosts of the
shows join the conference using CELT. But callers to the show would be
joining using regular phones therefore using lower end codecs. I will be in
the IRC shortly.
On Tue, Feb 3, 2009 at 4:21 PM, Brian West
Ken Rice napsal(a):
...
On Registrations we have experienced Registration/second rates exceeding 150
registrations per second using mod_xml_curl to feed the users directory. I
suspect, this number can be greatly increased if we were to feed directory
with something that cut out the apache and
What does it look like if you serve the directory from the static xml file
out of curiosity.
On Tue, Feb 3, 2009 at 4:11 PM, kokoska.rokoska kokoska.roko...@post.czwrote:
Ken Rice napsal(a):
...
On Registrations we have experienced Registration/second rates exceeding
150
registrations
You're doing distributed radio right? So callers are calling in with
CELT from all over the place? Can you contact us on IRC because we
are very interested in debugging this issue.
You can get us on IRC #freeswitch on irc.freenode.net
Thanks,
/b
On Feb 3, 2009, at 2:59 PM, e
Anthony Minessale napsal(a):
What does it look like if you serve the directory from the static xml
file out of curiosity.
Good question :-)
I have never thing about it, becasue I need dynamic users.
But it should show up very impressive number :-) I'll try it tommorow
(here is midnight) and
Yes this is normal. Your contact is mod_sofia ... why would it
change?Remember its a B2Bua. Now you can put param name=NDLB-to-
in-200-contact value=true/ in your sofia profile but be warned it
will break some devices.
/b
On Feb 3, 2009, at 2:24 PM, Adam Long wrote:
In every one
no,
there is no way to do that.
On Tue, Feb 3, 2009 at 11:09 AM, Helmut Kuper helmut.ku...@ewetel.dewrote:
Hi,
has anybody an idea?
regards
Helmut
Am 02.02.2009 19:00, schrieb Helmut Kuper:
Hello,
today I searched for a way to limit the number of menu repeatings in
mod_voicemail
Can you get me a sample of the recording to listen to?
/b
On Feb 3, 2009, at 2:26 PM, e schmidbauer wrote:
im using the latest svn of freeswitch
On Tue, Feb 3, 2009 at 12:23 PM, Brian West br...@freeswitch.org
wrote:
You forgot to tell us what revision of the code you're on?
/b
latest isn't a number... Can you provide the exact SVN rev you're on?
/b
On Feb 3, 2009, at 2:26 PM, e schmidbauer wrote:
im using the latest svn of freeswitch
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
Never tried hah...
From: Anthony Minessale anthony.miness...@gmail.com
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Tue, 3 Feb 2009 16:34:38 -0600
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] FreeSwitch setup as a Dumb SBC
What does it look like if you
Hi,
I was trying to record a background music with a user's voice at the
same time. I did a playback and started recording. But the recorded
user's voice and the background music is about 0.5 second out of sync. I
also tried to use uuid_displace instead of playback, but I got the same
result.
I
I did the test when I start looking at FS.
With 10 000 files in conf/directory/default mounted as a ramdisk (if not
in Ramdisk, the IO are too high) and an intel quad core q9550 (2.83Ghz)
with 4GB RAM and the db also in Ramdisk, I was stuck at approx 150cps
with a very high CPU usage. The
Hi anthony,
I Modified the whole architecture of call routing system,
Now after getting required routes, script exit and,
control comes back to Dialplan, and call is bridged there,
And call hangup, CDR is posted to cdr.php file (using xml_cdr).
So now there is no blocking statement (bridge or
Hi Adam,
I detailed a bit more my previous mail on this page:
http://wiki.freeswitch.org/wiki/SBC_Setup
Round robin is managed by the carrierroute module. Carrierroute will
reply based on the probability you defined for a route, so if you define
0.3 and 0.7 for the same prefix, your traffic
One more thing,
I worked on a setup like yours:
- Kamailio as a registrar that do the routing decision
- FS as a SBC
What you have to do is just append an header with Kamailio and send the
invite to your FS server using something like that (use of pseudo
variables in Kamailio):
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