On Fri, Jul 11, 2008 at 10:50 AM, Faraz R. Khan [EMAIL PROTECTED]
wrote:
This seems to happen nightly. Let me explain the scenario, this is a
third world country and power outages happen regularly. The UPS on which
the phones are may die (and i'm sure it does). Seems like freeswitch
segfaults
On Fri, Jul 11, 2008 at 3:19 PM, Faraz R. Khan [EMAIL PROTECTED]
wrote:
No, I'm running the phoenix 1.0.0 tarball release
Should I go to SVN?
I would recommend trying it to see if the issue still exists with current
svn ...
--
wasim h. baig | principal consultant | convergence pk | +92 300
On Sat, Jul 12, 2008 at 5:24 PM, Faraz R. Khan [EMAIL PROTECTED]
wrote:
amazing. it crashed again
I am dumb founded. It can process thousands of calls through sipp
without problems. I have a sipp running right now and it has processed
thousands of calls. This time the crash happened on
On Mon, Jul 21, 2008 at 12:06 AM, Michael S Collins [EMAIL PROTECTED]
wrote:
If you remind me where in the configs to change the max sessions per
sec I will update the performance testing wiki page.
autoload_configs/switch.conf.xml:param name=sessions-per-second
value=30/
--
wasim h.
On Mon, Jul 21, 2008 at 9:46 PM, Michael S Collins [EMAIL PROTECTED]
wrote:
Please let me know what works for you and we will pick a time that
works for as many people as possible.
anything later than early morning westcoast, means not making it easy for
some other part of the world ...
so,
On Mon, Jul 21, 2008 at 10:00 PM, Wasim Baig [EMAIL PROTECTED] wrote:
On Mon, Jul 21, 2008 at 9:46 PM, Michael S Collins [EMAIL PROTECTED]
wrote:
Please let me know what works for you and we will pick a time that
works for as many people as possible.
anything later than early morning
On Fri, Jul 25, 2008 at 10:10 PM, Cesar Bermudez [EMAIL PROTECTED]
wrote:
On Jul 25, 2008, at 11:00 AM, Alex Kinch wrote:
Hi gang,
Are there any calling card billing packages out there that are
compatible with Freeswitch? Obviously I've heard of a2billing, but a
recent forum thread
On Wed, Jul 30, 2008 at 8:29 PM, Shehzad Pankhawala
[EMAIL PROTECTED] wrote:
Hi,
thanks mike,
I want to get CDR into MySql.
organizing CSV and XML files for each user is some how combursome..
You can load the cdr from csv into mysql, LOAD DATA INFILE etc,
rehup freeswith to create a new
On Sun, Aug 3, 2008 at 11:45 PM, Robert Dyck [EMAIL PROTECTED] wrote:
Links at http://wiki.freeswitch.org/wiki/Main_Page would be a nice touch.
Added under http://wiki.freeswitch.org/wiki/Documentation#Users
On Sunday 03 August 2008, Brian West wrote:
Channel Variables are located here
On Sun, Aug 3, 2008 at 10:02 PM, Henk Oegema [EMAIL PROTECTED]wrote:
In * I use the following to send a SMS message to my mobile phone, in case
of
no answer:
[macro-send_sms_message]
exten = s,1,NoOp(${ARG1})
exten =
s,n,System(/usr/bin/curl -s
On Mon, Aug 4, 2008 at 12:28 AM, Wasim Baig [EMAIL PROTECTED] wrote:
On Sun, Aug 3, 2008 at 10:02 PM, Henk Oegema [EMAIL PROTECTED]wrote:
How can I do something simular in FS?
You would probably use a language of your choice from
http://wiki.freeswitch.org/wiki/Languages_for_Call_Control
We're working on the billing solution for fs (astpp) fastpp?
And need some input on what is the best way to limit call length.
In asterisk we've been using L(x[:y][:z]) option to Dial.
Whats the recommended route for this in fs?
Does something like this exist? If not, can I add this to the if
On Thu, Aug 7, 2008 at 2:10 AM, Wasim Baig [EMAIL PROTECTED] wrote:
In asterisk we've been using L(x[:y][:z]) option to Dial.
to follow up, use sched_hangup for the x:
see
http://wiki.freeswitch.org/index.php?title=Misc._Dialplan_Tools_sched_hangup
and for :y and :z use sched_broadcast
see
I'm sure its documented in other places but
http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Shutting_Down_FreeSWITCH.E2.84.A2
has it (the E2.84.A2 i think is because of the superscript TM)
-wasim
On Fri, Aug 29, 2008 at 7:45 PM, Sunil Singh [EMAIL PROTECTED]wrote:
Thanks for the info.
On Tue, Sep 2, 2008 at 12:37 PM, Jon Bruel [EMAIL PROTECTED] wrote:
A number of applications and other functions are not documented in the
WIKI. For example, the application displace, is not documented. Where
can I search to find the documentation regarding this and other
applications?
The
On Tue, Sep 2, 2008 at 6:02 PM, Steve Underwood [EMAIL PROTECTED] wrote:
spend his entire life afloat outside territorial waters. :-)
anybody up for a new business opportunity? FloatSwitch global codec
conversion farm ... located 22km offshore all major IXs and MAEs
--
wasim h. baig |
On Thu, Sep 18, 2008 at 4:24 PM, xbipin [EMAIL PROTECTED] wrote:
has any1 been able to use ASTPP billing with freeswitch and if so then my
yes, darren and a few others have, i have it under testing but not
production
next question, is it possible to use it on a win2k3 platform and not
On Fri, Sep 19, 2008 at 6:41 PM, Gopal krishnan [EMAIL PROTECTED] wrote:
Hi,
Is there any way that Asterisk dialplan can be used for freeswitch, since
there is a flle extensions.conf in PREFIX/conf directory, is th possible
with this file I can write a normal asterisk dialplan so that it
On Sat, Sep 20, 2008 at 2:07 AM, xbipin [EMAIL PROTECTED] wrote:
whats the amount ur looking for this bounty, mayb guys like me who wanna
use
dwiebe can let you know more about that
it as a proper softswitch can pitch in a little and can also help in testing
and coding.
what is sorely
On Wed, Sep 24, 2008 at 1:50 AM, Michael Collins [EMAIL PROTECTED]wrote:
Jon,
Are you saying that you cannot pass a comma-separated list of variables
like this?
action application=bridge
On Thu, Sep 25, 2008 at 3:12 AM, John Nicholson [EMAIL PROTECTED]wrote:
Anyone had any luck getting the Sangoma A 102 card into a 1 U box?
I'm looking to build 2 freeswitch servers for redundancy, and was
wondering if anyone has had any luck getting these cards to work with
1U riser cards and
On Sat, Sep 27, 2008 at 10:43 AM, preetha Ayyappan
[EMAIL PROTECTED] wrote:
Thanks wasim.I have changed the line
my $dbh =
DBI-connect(DBI:mysql:database=freeswitchdb;host=localhost,freeswitch,1234)
in /usr/src/freeswitch/scripts/contrib/wasim/cdrload.pl .then i called an
extension.Now the
Sat, Sep 27, 2008 at 2:26 PM, preetha Ayyappan
[EMAIL PROTECTED]wrote:
Actually now I can able to save the call detail records in the database
through php.But the csv file does not contains the disposition details.How
can i get the disposition in csv file?
the variable is known as
On Sun, Sep 28, 2008 at 9:50 PM, Jon Bruel [EMAIL PROTECTED] wrote:
I have made some load test, where an Asterisk server was controlled to
make a high number of calls to a FreeSWITCH, which was registered on
Asterisk. Each had its own server. The calls made to the FreeSWITCH were
answered
hemant:
umm.. yes, we want
-wasim
On Sun, Oct 12, 2008 at 2:53 AM, hemant kumar [EMAIL PROTECTED] wrote:
I have developement agreement and also distribution for g729. If you
guys want I can prepare a build of freeswitch with g729 provided there
is enough demand.
On 10/12/08, Michael
On Thu, Oct 23, 2008 at 12:26 AM, Anthony Minessale
[EMAIL PROTECTED] wrote:
the variables would be set on the b leg of the call not the a leg so you
would get them
from the cdr.
Since this is a pain for some people I added code so this information will
be set on the A leg too
so update
On Wed, Oct 29, 2008 at 7:20 AM, Peter P GMX [EMAIL PROTECTED] wrote:
I did a svn log:
/usr/src/freeswitch/libs/apr# svn log
r9605 | mikej | 2008-09-20 02:05:00 +0200 (Sa, 20 Sep 2008) | 1 line
hack for now until we
I spoke to Jim about this back in January, to see if he would be able to add
OSP / FS integration to transnexus.
Last we left it at was that he'd try to get in touch with some of the
developers at VoN and see what could be done.
Perhaps, its a good time to re-initiate that discussion.
-wasim
On
They should be FXS to SIP from what I can see on the net. I guess the
simplest answer to Wellie question is to power a couple up and try them out.
-wasim
On Sat, Nov 1, 2008 at 9:51 PM, Anthony Minessale
[EMAIL PROTECTED] wrote:
Are these analog to digital multiplexers? Like turn 24/30
Please check if your net interface is full duplex (ethtool should give you
this info).
We had this problem once, and spent weeks trying to identify and turned out
to be a half-duplex.
Also, see if you can offload checksum to the NIC if it'll do that.
-wasim
On Tue, Nov 4, 2008 at 8:20 AM,
On Wed, Nov 19, 2008 at 11:07 AM, Faisal Maqsoodi
[EMAIL PROTECTED]wrote:
Faisal:
Welcome to the wonderful word of open source telephony.
I ve just started reading on freeswitch. From where should i
start?
http://wiki.freeswitch.org/wiki/Main_Page
On Mon, Nov 24, 2008 at 11:42 AM, Faisal Maqsoodi
[EMAIL PROTECTED]wrote:
Root user using sudo su - . But with limited permissions.
then you need to unlimit them ... or compile as a different user
and then sudo as root to install
--
wasim h. baig | principal consultant | convergence pk |
On Mon, Nov 24, 2008 at 11:44 AM, Chav Paskov [EMAIL PROTECTED] wrote:
On ubuntu, you type:
sudo apt-get install gcc-c++
If it doesn't work, you type:
sudo apt-get install g++
if using another distro try yum
and try configure again you might need to install the devel packages as
well.
On Mon, Nov 24, 2008 at 11:54 AM, Wasim Baig [EMAIL PROTECTED] wrote:
On Mon, Nov 24, 2008 at 11:44 AM, Chav Paskov [EMAIL PROTECTED] wrote:
On ubuntu, you type:
sudo apt-get install gcc-c++
If it doesn't work, you type:
sudo apt-get install g++
if using another distro try yum
and try
On Mon, Dec 15, 2008 at 8:25 PM, Saeed Ahmed saeedahmad1...@gmail.comwrote:
Hi,
Salaam Saeed.
I am very new Freeswitch.
Welcome.
Till now I've some experience with Asterisk.
Be prepared to be amazed.
Can someone explain me the following things:
1. Can I connect my TDM switch to
On Wed, Jan 7, 2009 at 2:48 PM, Saeed Ahmed saeedahmad1...@gmail.com wrote
Do we have a call center graphical solution which works with FS? Same as
VICIDIAL which work with asterisk.
No, but I've got a pot going to bribe outtolunc in porting gnudialer to fs
... anyone else interested in
On Mon, Jan 12, 2009 at 4:49 PM, David Knell d...@3c.co.uk wrote:
In case anyone's interested, I've documented how we interfaced FS with
Lumenvox via MRCP using FS' event socket and unicast interfaces and a
bit of Perl here:
http://www.softivr.com/wiki/index.php/FreeSWITCH_MRCP_in_Perl
Three
On Wed, Feb 4, 2009 at 8:20 PM, Jonas Gauffin jonas.gauf...@gmail.comwrote:
I'm behind NAT. Is it FS that picks the random port, or the FW?
the FW
I've mapped port 5060 to the freeswitch ip in my FW.
thats inbound, now you need to tell your firewall to nail port 5060 on the
outbound side
On Sat, Feb 7, 2009 at 1:30 AM, Michael Giagnocavo m...@giagnocavo.netwrote:
$22K would buy quite a few machines with many core Xeons. I just don't see
how it'd be effective at that price. Not to mention a yearly figure.
G729 is roughly 25 MIPS (encode+decode), coppice, please correct as
On Mon, Feb 9, 2009 at 10:16 AM, Andrew Thompson and...@hijacked.us wrote:
On Sun, Feb 08, 2009 at 10:50:31PM -0600, Ken Rice wrote:
Also depending on what your Timeframe is like there is a distributed
queue
mechanism with skills based routing on the way...
It even managed to route 2
2009/3/17 Anthony Knight tntkni...@gmail.com
I'm thinking about doing a project that would use FreeSWITCH as an IVR, with
callers being routed in by both ISDN PRI, and also SIP trunks, with
occasional bridge calls between callers.
I'm wondering in what use cases hardware echo cancellation on
On Tue, Mar 17, 2009 at 2:02 PM, Gilles codecompl...@free.fr wrote:
I don't know of FXO PCI cards to connect an XP/Vista host to a phone
line. Does someone know of such a thing?
Sangoma makes a low cost 4FXO, 1FXS.
http://sangoma.com/products_and_solutions/hardware/analog_telephony/b600.html
On Sun, May 24, 2009 at 8:17 PM, bakko asannu...@gmail.com wrote:
I just installing FS in my new server. When i can make tests y will write
you.
bakko:
its really quite simple ... and should work straight out of the box
first install your server -
On Mon, May 25, 2009 at 8:58 AM, mashudi mashudifl...@telkom.co.id wrote:
Anybody could share the profile configuration for softphone register
behind NAT?
http://wiki.freeswitch.org/wiki/Natted_Softphone_ATA
http://wiki.freeswitch.org/wiki/NAT
http://wiki.freeswitch.org/wiki/NAT_Traversal
On Mon, May 25, 2009 at 4:13 PM, Simon Tennant si...@imaginator.com wrote:
...actually it's not quite simple. Nokia handsets have a myriad of
other options that help fine tune things like STUN server resolution and
are only editable after installing some nokia supplied software.
concur, but
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