Hi.
We have a local network 192.168.1.0/24, where all the users are. Out
FreeSWITCH server is connected to this network, and has ip address
192.168.1.242. Through different network card it is connected to
external gateway, and has address 172.16.12.11 in this network.
I set up a test client with
Hi folks!
Suddenly I found this
http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-February/002015.htmltopic
and that explains a lot.
From there I see that sofia sends refresher messages for NATed client in
order to check if it still alive.
It means I have problems in my client. Sorry for
Anyone have this issue?
On Tue, Sep 29, 2009 at 9:15 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
I have just started to use dingaling again, and noticed I constantly
get a stun error.
2009-09-29 21:11:03.175002 [ERR] mod_dingaling.c:980 Stun Failed!
stun.fwdnet.net:3478
Yes, i had same problem, then i changed stun server to one of our own
servers. You may try some of public stun servers listed on below link,
http://www.voip-info.org/wiki/view/STUN
Thank you.
On Fri, Oct 2, 2009 at 2:58 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Anyone have
You can use the api and check that the channel is occupied with show
channels?
You can write a small javascript that checks if the channel is occupied by
means of session.execute api.
/aep
--
Stopping junk mailers is good for the environment
My SIP provider allows only one call (incoming or
Michael Collins m...@freeswitch.org said:
I believe the OpenZAP and 1 are coming from your conf file:
openzap.conf
[span zt PRI_1]
name = OpenZAP
number = 1
That is correct. If that information is removed, then X-Lite displays
FreeSWITCH
[Other: 00]
Are there any variables to set
hello,
i just got the last trunk and tried to compile it on one of my development
machines... Well configure fails on tiff-3.8.2 where it is unable to find
Makefile.in ... Can someone advice?
checking if g++ static flag -static works... yes
checking if g++ supports -c -o file.o... yes
checking
what if you are running some huge traffic e.g. 2000 calls with media?
a typical application for that is an IVR system handling several different
services. I'd like to dedicate some capacity for inbound on per service
basis.
e.g.
DID 10001 limit to 500 calls
DID 10002 limit to 400 calls
DID
anyhow, this is how it works for me!
include
context name=public
extension name=LNP
condition field=destination_number
expression=(^30)(.*)
action application=lnp_getprefix data=in $2, out
reroutingalias/
action
Hello Everybody,
I am new to freeswitch, so forgive me if I ask stupid questions. I
am planning a test setup consisting of:
1 - Pfsense router with the freeswitch package installed.
1 - Cisco WS-C3524-PWR-XL-EN 24 FE Switch, POE for the phones.
1 - LINKSYS SPA3000 to connect to my
Hey folks, the weekly conference call is starting. Please see the agenda for
instructions on dialing:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_02
Looking forward to speaking with you all!
-Michael
___
FreeSWITCH-users mailing list
On Fri, Oct 2, 2009 at 4:48 AM, russell.mosem...@cune.org wrote:
Michael Collins m...@freeswitch.org said:
I believe the OpenZAP and 1 are coming from your conf file:
openzap.conf
[span zt PRI_1]
name = OpenZAP
number = 1
That is correct. If that information is removed, then X-Lite
How would I configure FS to Call Forward All or Call Forward when Busy or
Call Forward when No-Answer? Can this be done at the server, rather than at
the phone?
Best Regards,
Jerry
___
FreeSWITCH-users mailing list
I only mentioned the OS I used as a reference for people. If they want to do
the same thing on another OS, then they might not have apt-get, etc.
Mike van Lammeren
On Thu, Oct 1, 2009 at 3:41 PM, Hadley Rich h...@nice.net.nz wrote:
On Thu, 2009-10-01 at 14:14 -0500, Anthony Minessale wrote:
The load balancer listens to the virtual IP address, and port-forwards to
one of the FreeSWITCH boxes. Each FreeSWITCH box listens for the same
virtual IP address for SIP registrations and connections, which is what
FreeSWITCH needs to bind to. All other traffic actually travels over their
real IP
Hi, I installed internal freeSWITCH in my LAN and want to see if I can dial
out / dial in. Could anyone suggest one Telephone Service provider which is
capable of connecting with FreeSWITCH and CHEAP/even FREE if possible? At
this moment, I want to prove it is working with the real outside world.
I am running the servers on the free version of VMware's ESX platform, but
only for development purposes. We will be setting up real machines sometime
in Spring 2010.
On Thu, Oct 1, 2009 at 2:12 PM, Even André Fiskvik grev...@me.com wrote:
That's very cool Mike!
I'm going to try to configure
Okay, I put a log up on the pastebin that shows the PUBLISH event coming
from a CounterPath Bria Professional phone. For some reason, FS is getting
an error and not relaying the presence status to the subscriber.
Best Regards,
Jerry
_
From: João Mesquita
Michael Collins m...@freeswitch.org said:
do something like:
name = XYZ Corp
number = 8005551212
I was expecting that information to be filled with the caller name and
number. It doesn't really help if someone calls from outside the
business, and it looks like my business is calling me.
Hey Erwin,
Can't give any personal recommendations, but on the FS site, there's several
examples. Some have free or cheap in the name. Might be a good place to
start, plus the means to connect is demonstrated to-boot.
http://wiki.freeswitch.org/wiki/SIP_Provider_Examples
Regards,
Mike G.
On
Hello!
For dialing in, there are a number of sites that provide free DIDs,
such as http://freephonelines.ca/ .
For dialing out, you can get 1.5 cents per minute calling to N.
America from http://les.net/ .
Mike
On 2009-10-02, at 11:50 AM, Erwin Davis davis.er...@gmail.com wrote:
Hi, I
connect to sqlite directly with sqlite3 app and try that sql stmt and see
why it doesn't match anything.
sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db
select
Hi!
Callcentric
http://wiki.freeswitch.org/wiki/Provider_Configuration:_Callcentric
offers a package called IP Freedom
http://www.callcentric.com/rate_plans01.php. It costs nothing and will
allow you to test FS.
Carlos
Erwin Davis wrote:
Hi, I installed internal freeSWITCH in my LAN and
On Fri, Oct 2, 2009 at 10:24 AM, russell.mosem...@cune.org wrote:
Michael Collins m...@freeswitch.org said:
do something like:
name = XYZ Corp
number = 8005551212
I was expecting that information to be filled with the caller name and
number. It doesn't really help if someone calls from
I put the sqlite3 select query in the paste bin, and prior to that, I
entered the .dump command. The select command came back with a ...
prompt which I don't understand. I don't know enough about sqlite3 to know
what that means?
Best Regards,
Jerry
_
From: Anthony Minessale
You are missing the trailing ;
On Fri, Oct 2, 2009 at 1:28 PM, Jerry Richards
jerry.richa...@teotech.com wrote:
I put the sqlite3 select query in the paste bin, and prior to that, I
entered the .dump command. The select command came back with a ...
prompt which I don't understand. I don't
Can you pastebin a dialplan snippet (or put it here) so I can see what
you're doing?
-MC
It is the stock FS configuration with a small change. We're still testing
things, getting them to work. This is from public.xml. It detects calls to
internal 71xx extensions and transfers them. The
Is there benchmark test results on how many simultaneous calls Freeswtich can
do (with RTP anchored through it) vs the Asterisk.
For any hardware/CPU/Mem that anyone may have performed this performance
testing.
Any numbers on average how much Freeswitch scores over the Asterisk in terms of
On Fri, Oct 2, 2009 at 11:53 AM, Russell Mosemann russell.mosem...@cune.org
wrote:
Can you pastebin a dialplan snippet (or put it here) so I can see what
you're doing?
-MC
It is the stock FS configuration with a small change. We're still testing
things, getting them to work. This is
Hi,
for example here: http://blogs.zdnet.com/Greenfield/?p=214
We *replaced* a cluster of *10 Asterisk* servers with a *single
FreeSwitch*server, said Chris Parker, director of systems for a large
publicly traded
CLEC. Parker says hes getting several hundred concurrent calls on a single,
cool. can you pastebin a debug log on an incoming call?
-MC
Here you go.
http://pastebin.freeswitch.org/10570
One thing I notice is that in the second line, the caller number is missing.
2009-10-02 16:42:39.539736 [NOTICE] ozmod_libpri.c:772 -- Ring on channel 1:1
(from to 7100)
If libpri
On Fri, Oct 2, 2009 at 2:53 PM, Russell Mosemann
russell.mosem...@cune.orgwrote:
cool. can you pastebin a debug log on an incoming call?
-MC
Here you go.
http://pastebin.freeswitch.org/10570
One thing I notice is that in the second line, the caller number is
missing.
2009-10-02
Exactly. Turn on q931 debugging and try again:
oz libpri debug 1 all
PB the results again and we'll check it out.
-MC
Here's the next one. I'm not sure what to look for, but nothing pops out right
away.
http://pastebin.freeswitch.org/10571
--
Russell Mosemann
Dear all,
I am in the process of implementing IVR server in Perl using event
outbound socket.
Let take the following scenario.
There are three menus in the IVR. First menu will authenticate you,
second menu get the option value from you,. third menu will give the you the
You can use play_and_get_digits command or the read command.
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_read
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_read
Thanks,
Vinuth.
On Sat, Oct 3, 2009 at 9:54
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