I'm working on a tool to block SPIT calls. One important piece of
information in that context is the outbound proxy of the caller. With
certain assertions implied (no spoofing) this is the same as the domain part
of the From uri. So at the moment I can use that as well. But the complete
route liste
You are using a recent fedora that ships with a libcurl with broken
dependencies. You either need to install a non broken libcurl (I
don't think they have any non broken packages) or configure freeswitch
with the --without-libcurl option to force freeswitch to not use the
system libcurl.
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
switch_loadable_module_load_fil
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 as
Gang,
On our weekly conf call Ray (intralanman) had a great idea: to help
get the default configs documented we are going to have a "mod of the
week" and invite all community members to assist with the
documentation process.
To help out you can do any or all of the following:
Write the wiki
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 me
Hi Erol,
A dialplan extension like this:
results in all calls to numbers starting with 0 being handled by a
server on localhost
listening on port 6100. That server has full control over the call.
Change the IP
address to run the server on a different host.
Cheers --
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:
http://fs.ip:
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
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]>
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 th
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*
www.wikipbx.com
2008/8/1 Ilan Perez <[EMAIL PROTECTED]>
> Anyone developing a GUI for freeswitch
>
>
>
>
>
> *Ilan Perez*
>
> *Webmaster*
>
> 0432 326 017
>
>
>
> ___
> Freeswitch-users mailing list
> Freeswitch-users@lists.freeswitch.org
> http://list
Anyone developing a GUI for freeswitch
Ilan Perez
Webmaster
0432 326 017
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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: f
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 of
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 att
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 fir
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 th
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 media and everything is working fine, except that
at about 260 call legs, the above SIP response is returned to the originating
GW.
Thanks in a
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 a
Still searching event_socket module.
If we set up dialplan, all received calls will routed based on dialplan rules.
But at the same time, if there is remote process connected to FS through
socket_module, dialplan or remotes process will have priority to control FS
behavior. Can remote process
If it is not on that page, we probably don't have a formal concise
list anywhere. If you ask about specific features here, we should be
able to answer and explain what is there already.
Mike
On Jul 31, 2008, at 4:26 AM, Simon Shaw wrote:
In the list of features on the spec-sheet in the FS
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 Ko
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-
>
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:
>> frees
On Thursday 31 July 2008 20:54:53 unknown wrote:
> freeswitch# perl fs.pl
> FreeSWITCH>
>
> That's it.
:-)
Henk
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UNSUB
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, unkn
On Wed, Jul 30, 2008 at 05:33:15PM -0500, Cesar Cepeda scribbled:
# Hi,
[snip]
# . After exactly 10 seconds the consumer channel is hung up by FS
# with reason UNKNOWN.
[snip]
# It would seem that the problem is that sofia detects that I'm not sending
# any audio and hangs up.
Cesar,
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 no
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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Freeswitc
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 P
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-1.
On Jul 30, 2008, at 12:37 PM, Martin Joseph wrote:
> Hello FreeSWITCHers!
>
> I am new to FreeSWITCH, having played with/used asterisk for a couple
> of years and being based on the mac platform, I was looking for a
> superior choice and FreeSWITCH seems to fit the bill.
>
> I have built and am
> 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
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 li
It's not a problem.
do this
Perl script: src/scripts/socket/fs.pl
as in http://wiki.freeswitch.org/wiki/Rosetta_stone
Chris
On Thu, Jul 31, 2008 at 1:15 PM, Henk Oegema <[EMAIL PROTECTED]>wrote:
> How do I (re)connect to a running FS? (like -rv parameter in * ) to get
> the
> FS CLI prompt bac
How do I (re)connect to a running FS? (like -rv parameter in * ) to get the
FS CLI prompt back?
[EMAIL PROTECTED]:/usr/local/freeswitch/bin# ./freeswitch
Cannot lock pid file /usr/local/freeswitch/log/freeswitch.pid.
I get this message because FS is already running.
Even better: where do I f
http://wiki.freeswitch.org/wiki/Bypass_Media
/b
On Jul 31, 2008, at 11:50 AM, Simon Shaw wrote:
Thanks for that, got it working now.
I notice that the SDP in the outgoing INVITE message is changed so
that the media will flow via the trunk, is there any way to override
this?
Simon
As far as I know, there is no ability to generate certificates for TLS on
any windows release.
Since I also needed that, I shared my experience in this temporary wiki page
until an automated script is written for windows (brutally mimicking the
gentls_cert script):
http://wiki.freeswitch.org/wiki/G
Thanks for that, got it working now.
I notice that the SDP in the outgoing INVITE message is changed so that
the media will flow via the trunk, is there any way to override this?
Simon
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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 your
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 su
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
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 o
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
changing
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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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 dia
Is there any way to inform a module of incoming SIP Instant Messages (e. g.
sent to a special user)?
--
Alois Komenda
Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK
smime.p7s
Description: S/MIME cryptographic signature
___
Fre
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
On 28/07/08 15:42, Michael S Collins wrote:
> Birgit,
> I have deciphered many of these. I will post my info when I get in
> front of my computer. I proba
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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UN
I am running FS of a windows box IP 10.10.4.204 and have successfully
managed to demo most of the default features.
I attempted to extend the configuration by adding a gateway as follows
in $(prefix)\conf\dialplan\default.xml, where 10.10.7.95 is the address
of a working asterisk test box I have
The core functionality to do it is there... and the ability to push
XML to a phone via an event.
On Jul 31, 2008, at 2:00 AM, Matt Darnell wrote:
> Are there any docs on the enhanced Snom support? I read that it
> allows for key system functionality.
>
> -Matt
Brian West
sip:[EMAIL PROTECTED]
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
> co
Sangwoo Jin wrote:
> I don't have gotten the same result in testing MOH calls with 15 CPS.
> The memory usage of freeswitch on testing doesn't grow at some point.
> But, The memory usage of freeswitch on testing bridged calls with 5CPS and 1
> second duration was growing endlessly. I have watched T
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
> 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 g
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 generating
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 t
In the list of features on the spec-sheet in the FS Wiki, one of the
features is "Basic IP/PBX features". Do you have a list of specific
features that you support?
Thanks,
Simon
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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 you
Hello Freeswitchers,
is it possible to get a list of all hosts, which are conained in the Via
header, over the Event Socket?
Most important for me is to know the first proxy, which forwarded the call
(i. e. outbound proxy of the caller). But a complete list of involved
forwarding hosts would be n
Dear All,
I am not sure who is an expert in PBX/ FS and who isnt but I dont know
that I certainly am not J
in light of that I really need some help getting
things moving with this system. Just before I start, I want to say that the
wiki isnt the be-all and end all in terms of answers. I have
I have fs running in a 64MB system, a 450MHz arm with no cache (or hard
drive).
I have very few calls running at the same time ~5 channels max, and I see a
memory usage of about 20-23MB. I have not done any load testing like you
describe, but during a bridged call the fs CPU load is not measurable
On Thu, Jul 24, 2008 at 7:06 AM, Anthony Minessale
<[EMAIL PROTECTED]> wrote:
> The first round of improvements are in from our 1.0 line!
> Enhanced SNOM support.
Are there any docs on the enhanced Snom support? I read that it
allows for key system functionality.
-Matt
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