Can you detail the exact callflow, config files and testing methods
you used to produce this issue? I suspect that this may be a real
issue but we ran sipp testing 2 days ago so your scenario must be
hitting an edge case we were not seeing. That being said we have made
no efforts to get
Can you run FS under valgrind and see if its really leaking?
As Brian mentioned Earlier, Memory pooling does not give up memory... I run
freeswitch at 1000s of concurrent channels...
You might not be able to do 5cps under valgrind (this is due to the
extensive memory checks valgrind does) but
Is there a simple explanation, in terms that a mere software engineer
can understand, what the MPL license means?
1) Can I sell FS under my own guarantee and support?
2) If I write code that connects FS to an external application that
I have written, does the external application have
can any1 plz compile the latest freeswitch for windows with the ability to
generate certificates for TLS protocol as i am just unable to do it with my
little knowledge for C and visual studio.
basically i looking for he latest windows freeswitch with TLS, as in plugins
etc included for
Is there a simple explanation, in terms that a mere software
engineer can understand, what the MPL license means?
You can use the code for whatever purpose you want, as long as you
give all changes you make to that code back to the project basically.
1) Can I sell FS under my own
Simon,
Welcome to FreeSWITCH.
On Thu, 31 Jul 2008, Simon Shaw wrote:
Is there a simple explanation, in terms that a mere software engineer
can understand, what the MPL license means?
1) Can I sell FS under my own guarantee and support?
So long as you give proper trademarks and
can you try trunk r9211 to make sure it's not fixed already and if not
provide a console trace at debug log level into a jira ticket.
On Wed, Jul 30, 2008 at 5:33 PM, Cesar Cepeda [EMAIL PROTECTED] wrote:
Hi,
I'm experimenting with mod_fifo, I'm having a problem when I insert a
consumer
Try ^5\d{4}$
/b
On Jul 31, 2008, at 8:55 AM, Simon Shaw wrote:
^5d{4}$
Brian West
sip:[EMAIL PROTECTED]
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Made the suggested change but the call is still failing. Here are the
traces:
2008-07-31 17:54:51 [NOTICE] switch_channel.c:534
switch_channel_set_name() New Channel sofia/internal/[EMAIL PROTECTED]
[554b908a-7b88
-7e42-b4ec-6f9c5a57eac1]
2008-07-31 17:54:51 [INFO] mod_dialplan_xml.c:222
This is why.
On Jul 31, 2008, at 9:57 AM, Simon Shaw wrote:
sofia_glue_ext_address_lookup() Stun Failed! stun.freeswitch.org:
3478 [Timeout]
Brian West
sip:[EMAIL PROTECTED]
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Both FS and the asterisk box I would like to trunk are in the same
private network, I have no reason or ability, (IT guy will take weeks to
open up a port on the firewall) to connect to a Stun server in my test
environment.
1) Is there a way to disable this? I tried to disable it by
The stun is failing.. either hard code the ext-sip-ip and ext-rtp-ip
or fix the stun.
On Jul 31, 2008, at 10:43 AM, Simon Shaw wrote:
Both FS and the asterisk box I would like to trunk are in the same
private network, I have no reason or ability, (IT guy will take
weeks to open up a port
if you register a chat interface with FS you can route im to it from any
protocol by using the + delim
e.g. mod_conference uses [EMAIL PROTECTED] [EMAIL PROTECTED]
On Thu, Jul 31, 2008 at 9:48 AM, Alois Komenda
[EMAIL PROTECTED] wrote:
Is there any way to inform a module of incoming SIP
OK, commented out the lines of code in sofia_glue.c that run the stun
test and now I see an INVITE message being sent to the trunk, however
the To field is [EMAIL PROTECTED] instead of 58661 as I would have expected.
2008-07-31 19:26:19 [WARNING] sofia.c:75 sofia_handle_sip_r_notify()
delete
Why are you touching the code? Just edit the sip profile and set the
ext-sip-ip and ext-rtp-ip or remove them that is what triggers the
stun lookup. In the default config its set to
stun:stun.freeswitch.org which in your case that is failing.
The $1 results from not doing a capture in
Speaking of the Rosetta Stone page, it could use some love and
attention. If you have any Asterisk and FS knowledge we'd love to have
you add something to the Rosetta Stone page. Even if you add just one
tip or trick it would be welcomed heartily. It could save a lot of time
and energy on the
Hi Mike,
Any chance of you posting that information?
If you did so already, sorry, I might have missed out on your posting.
Thanks, Birgit
:(
I am a slacker. Stand by and I will get an XML-CDR wiki page started
and I'll get these vars documented first.
Please give me an hour or so and
On Thursday 31 July 2008 19:28:57 unknown wrote:
It's not a problem.
do this
Perl script: src/scripts/socket/fs.pl
Thanks Chris.
I'm not familiar with Perl, so a more detailed explanaition
is highly appreciated.
Where is: src/scripts/socket/fs.pl ?
I have a file fs.pl in
freeswitch# perl fs.pl
FreeSWITCH
That's it.
Chris
On Thu, Jul 31, 2008 at 2:45 PM, Henk Oegema [EMAIL PROTECTED]wrote:
On Thursday 31 July 2008 19:28:57 unknown wrote:
It's not a problem.
do this
Perl script: src/scripts/socket/fs.pl
Thanks Chris.
I'm not familiar with Perl, so a
Also make sure the FreeSWITCH/ dir is in your site_perl dir
/b
On Jul 31, 2008, at 1:54 PM, unknown wrote:
freeswitch# perl fs.pl
FreeSWITCH
That's it.
Chris
Brian West
sip:[EMAIL PROTECTED]
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FYI,
I've added stubs and info to the XML CDR wiki page. I'm still fleshing it
out so keep checking back:
http://wiki.freeswitch.org/wiki/Mod_xml_cdr#Reference_Information
If you have anything to add to this page please email me or email the list
because I'm actively editing right now. Let's
Just as a suggestion, it would be nice a -r option in the freeswitch
binary, could this be done in the future? =D
Diego
On Thu, Jul 31, 2008 at 2:58 PM, Brian West [EMAIL PROTECTED] wrote:
Also make sure the FreeSWITCH/ dir is in your site_perl dir
/b
On Jul 31, 2008, at 1:54 PM, unknown
I know there is fs.pl, and it's indeed useful, but an -r option in the
binary itself would be nicer ;-)
Just as a suggestion, don't take it bad =D.
FS rocks!
Diego
On Thu, Jul 31, 2008 at 3:57 PM, Henk Oegema [EMAIL PROTECTED] wrote:
On Thursday 31 July 2008 20:54:53 unknown wrote:
Jonahtan,
I'm using an AudioCodes Gateway. I changed the parameter you mention and
indeed the problem went away!
Thanks a lot! :)
What I don't understand is why the GW doesn't detects a broken connection
the first time the consumer enters the fifo?
Cesar Cepeda.
-Mensaje original-
In the sip_via_host that is going to be the first (next hop) host in
the via. We don't currently go through the entire list of via headers
and turn them into variables. It could be added, but I would want a
compelling reason to add the overhead.
Mike
On Jul 31, 2008, at 4:01 AM, Alois
I wondered if attaching FS to a a screen and in bg mode would be good
workaround, so that one can reattach to that screen from any other shell
later ?
Regards,
ashutosh
On Fri, Aug 1, 2008 at 1:32 AM, Diego Viola [EMAIL PROTECTED] wrote:
I know there is fs.pl, and it's indeed useful, but an -r
Hi
I have a situation is which I need to do jitter buffering.
The setup is as follows:
(local sip user)freeswitch---(remote gateway)
The leg that needs de-jittering is the RTP travelling from the remote
gateway to the local sip user. I only want Freeswitch to enable
jitterbuffer on
On Thu, Jul 31, 2008 at 03:13:45PM -0500, Cesar Cepeda scribbled:
# Jonahtan,
#
# I'm using an AudioCodes Gateway. I changed the parameter you mention and
# indeed the problem went away!
#
# Thanks a lot! :)
#
# What I don't understand is why the GW doesn't detects a broken connection
# the
It would be cool also if you could re-attach FS from internet and from
the local host, with -r (resume) -h (host) -p (port). So you could
re-attach not just locally but from any host.
Just a though ;-)
Diego
On Thu, Jul 31, 2008 at 6:41 PM, Ashutosh [EMAIL PROTECTED] wrote:
I wondered if
while I totally agree...
id love to see it... but Id also love the core devs to keep working on more
important stuff..
after FS is setup and running, your not going to need to attach to the
console to do stuff.
and when you do, fs.pl is there for the occasional usage.
so yea... I guess its one
OK, I'll post this problem, reproduction method and memory usage on JIRA.
Thank you very much for your helps.
Sangwoo Jin.
-Original Message-
From: [EMAIL PROTECTED] [mailto:freeswitch-
[EMAIL PROTECTED] On Behalf Of John Skopis (Lists)
Sent: Thursday, July 31, 2008 9:43 PM
To:
Anyone developing a GUI for freeswitch
Ilan Perez
Webmaster
0432 326 017
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www.wikipbx.com
2008/8/1 Ilan Perez [EMAIL PROTECTED]
Anyone developing a GUI for freeswitch
*Ilan Perez*
*Webmaster*
0432 326 017
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So... anyway to see it working? or is this commercial?
thxs
d
On Thu, Jul 31, 2008 at 10:32 PM, Lito Manansala [EMAIL PROTECTED]
wrote:
www.wikipbx.com
2008/8/1 Ilan Perez [EMAIL PROTECTED]
Anyone developing a GUI for freeswitch
*Ilan Perez*
*Webmaster*
0432 326 017
Hello ,
I would like to share with you the email below that I exchanged with the
representative of Mera Systems. I was about to evaluate their product when
I stumbled with FS. He, of course, is trying to say that FS is not able to
support more than 300 simultaneous calls and that I should buy
opensource offcourse, its working I tested already couple of months ago
On Fri, Aug 1, 2008 at 11:06 AM, David Villasmil
[EMAIL PROTECTED] wrote:
So... anyway to see it working? or is this commercial?
thxs
d
On Thu, Jul 31, 2008 at 10:32 PM, Lito Manansala
[EMAIL PROTECTED] wrote:
d to help setup my own environment
Ilan Perez
Diagnostic Devices
Webmaster
0432 326 017
8347 2244
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Villasmil
Sent: 01 August 2008 13:07
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] GUI
Could someone please confirm this behavior and comment on whether it
is expected or not?
Using the api or webapi from mod_xml_rpc, one can fetch CLI commands.
Some commands seem to work well using either method and others do not.
Try these two commands and tell me what you see:
Hi Erol,
A dialplan extension like this:
extension name=callthrough
condition field=destination_number expression=^0
action application=socket data=127.0.0.1:6100 async full /
/condition
/extension
results in all calls to numbers starting with 0 being handled by a
server on
Hi Ron,
Have you set the various ulimits as described here?
http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations
--Dave
Hello,
Can someone please tell me all the places in the config files where
the number of concurrent calls can be limited?
I am doing SIP-SIP calls with
I think you meant this command:
http://fs.ip:8080/webapi/sofia?status%20profile%20internal
Anyway, I've been digging into the XML-RPC internals and couldn't make much
sense of it either - especially with mod voicemail.
It would be great if we had a reference to all xml-rpc links.
I'd be happy
I have installed FS from svn trunk.
It works fine but some modules are not loaded even if they compiled, and
loaded in modules.conf.xml
While start freeswitch and load that module the following error displayed:
2008-08-01 10:58:43 [CRIT] switch_loadable_module.c:756
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