Thank you for you reply Mike!
'ignore_early_media=true' variable setting is the solution, but I figured it
out shortly after posting to the ML.
Best regards,
V. Panayotov
On Mon, Aug 24, 2009 at 10:46 PM, Michael Jerris m...@jerris.com wrote:
Do you have an answer in the dialplan for that
Hello Takeshi,
Thanks for your hint... it worked out... so to be precise:
VIA header of both INVITE and ACK messages MUST be identical (IP:PORT +
branch)... and you are right... it might not be according to SIP
specification. Anyhow, i get FS understand my ACK message.
Finally, here is what i
Hello,
I would like to dynamically add user to freeswitch. If I add a new file to
the directory dir, is there anyway to have freeswitch to read the new user
xml file without having to restart freeswitch?
Other than using flat file, is there anyway to add user to freeswitch user
api command?
You just need to reloadxml you don¹t have to restart the whole thing. You
can also use xml_curl to feed the users from a database see my contrib
directory (contrib/swk) for some example scripts and db code
From: Juan Backson juanback...@gmail.com
Reply-To: freeswitch-users@lists.freeswitch.org
Hello,
i'm trying to use freeswitch as a redirecting server meaning FS has to
receive an INVITE and according to some rules it will redirect calls to
other destinations.
CALLING_USERFREESWITCHSOMEWHERE
INVITE ---
Nowdays I 'm forced to put multiple | to find first free gateway, ie
sofia/gateway/panas111/1000|sofia/gateway/panas112/1000|sofia/gateway/panas113/1000
,
the whole sting is tooo long, is there any shorter way to write same thing? Like
sofia/gateway/panas*/1000 will try all gateways matching the
Maybe your load comes from disk access?
Try putting the sql and log directories on a ramdisk.
OTH,
-giovanni
On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjagatculj...@gmail.com wrote:
Hello,
i'm trying to use freeswitch as a redirecting server meaning FS has to
receive an INVITE and
Is this what you are after?
http://wiki.freeswitch.org/wiki/Mod_xml_odbc
Cheers,
Jim
On Tue, Aug 25, 2009 at 7:22 PM, Juan Backsonjuanback...@gmail.com wrote:
Hi Ken,
xml_curl is a great idea. Is there anyway to not having to setup another
HTTP server? For instance, can I have freeswitch
Hi Ken,
xml_curl is a great idea. Is there anyway to not having to setup another
HTTP server? For instance, can I have freeswitch to call an api or call a
lua or php or c script that will return the xml response? That way, I don't
need to maintain yet another service.
Thanks,
JB
On Tue, Aug
Hi,
I wrote that module, but been on vacation for a while :-) It's not
really finished yet, but it worked well for generating user directory
xml..
Some things that still need to be done:
- Fix it so that reloadxml works
- Don't write the generated xml always to disk before returning it to fs
Hey Giovanni,
thanks for the tip... indeed the db files were heavily used regardless if i
started freeswitch with nosql option (freeswitch -nosql)... FS was not
writing anything into that files ... instead it was just accessing it
This behaviour leads to a waste of 40% CPU time... waiting for
Everytime someone asks this , the resounding answer is use a 64bit os..
No question
Jay
On 25/08/2009, at 23:19, Tihomir Culjaga tculj...@gmail.com wrote:
Hey Giovanni,
thanks for the tip... indeed the db files were heavily used
regardless if i started freeswitch with nosql option
thanks for the feedback... this is something im going to do tomorrow...
what about other things?
On Tue, Aug 25, 2009 at 3:39 PM, Jay Binks jaybi...@gmail.com wrote:
Everytime someone asks this , the resounding answer is use a 64bit os..
No question
Jay
On 25/08/2009, at 23:19,
Definitely go for 64 bit OS.
If you want to be safe and sure, go for CentOS 5.2 64bit. Is the one
used both for development and for heavy duty production.
Also Ubuntu 8.04 is good.
Other versions/distros are less used by the community.
Adding RAM and CPUs helps to scale up.
-gm
Sincerely,
is a heavely multithreaded software, it benefits from number of CPUs
(or cores), RAM, and heavy duty kernel features (found in 64bit
kernels)
put all accesses on ramdisk, leave out the modules you don't use...
experiment, test, and find the best for your specific application/workload
test not
Max,
I would like to see something similar too. For example, it would be
wonderful if one could specify multiple gateways to try like this or
something similar:
action application=bridge
!-- set some kind of *bridge-specific* parameter/variable --
gateway order=1
Hi List,
i have some scripts to test our lab:
I have scripts to create sipp instances which act as individual
agents, which log on, take a random amount of calls, log off, wait a
bit, log on again, etc.
I have scripts to generate call traffic for our queues to saturate them
But what I have
Very Nice my friends i'm from Brazil i'm crazy for the tests ...
Eng Eder de Souza
On Mon, Aug 24, 2009 at 7:00 PM, Rodrigo P. Telles
telles-lis...@devel-it.com.br wrote:
It is very nice to hear that, great work!
Thanks to support FS.
Telles
Moises Silva wrote:
So, I finally took
Hello,
i was reading through the openzap wiki page and searched the wiki, the
only thing I found about dahdi is a note from January that freeswitch
does not work with dahdi right now.
Is this current information? Can't I use dahdi kernel drivers for
freeswitch? As the last update to the
Dear all,
In one of the applications I am writing I need to convert a recorded wav
to mp3.
After using
session.recordFile() and obtaining a foo.wav file, I am calling
session.execute(system,lmLameCmd);
to invoke lame for the conversion.
The system command looks like this:
lmLameCmd =
Try running it at the CLI and see if you see any errors. Also please
do not hijack threads. The original thread [Freeswitch-users] XML-
RPC on different ip than 0.0.0.0 which was hijacked by clicking
reply, changing the subject and clicking send. Please in the future
do not do that as it
I fired up Wireshark on each side and I can see the SIP register request coming
from the laptop, the Freeswitch server replies with a Destination Unreachable
(Port unreachable) message.
I rebooted the Server and now I get a Registration error; 405 Method not
allowed on the softphone and the
On Tue, Aug 25, 2009 at 9:35 AM, Raimund Sacherer r...@runsolutions.comwrote:
Hello,
i was reading through the openzap wiki page and searched the wiki, the only
thing I found about dahdi is a note from January that freeswitch does not
work with dahdi right now.
You did not search well
Take a look at http://jira.freeswitch.org/browse/FSCORE-422. This a
feature request I submitted. This problem it solves is different - but
the solution is the same. Perhaps you add your take to the comments
there.
On Tue, Aug 25, 2009 at 10:06 AM, Carlos S. Antunesc...@nowthor.com wrote:
Max,
On Aug 25, 2009, at 4:24 PM, Moises Silva wrote:
On Tue, Aug 25, 2009 at 9:35 AM, Raimund Sacherer
r...@runsolutions.com wrote:
Hello,
i was reading through the openzap wiki page and searched the wiki,
the only thing I found about dahdi is a note from January that
freeswitch does not
This suggestion violates the scope boundaries.
gateways are specific concept to mod_sofia so a gateway tag in action
(part of agnostic xml dialplan)
does not flow properly.
you can also use combinations of continue_on_fail and hangup_after bridge so
you can
just put each bridge statement in it's
I beleive this is following the right rfc rules for dialog matching.
If it is not, please open up a bug on jira.freeswitch.org with
references of what exactly is not right.
Mike
On Aug 25, 2009, at 2:51 AM, Tihomir Culjaga tculj...@gmail.com wrote:
Hello Takeshi,
Thanks for your hint...
You can also do xml config hooks in perl and some of the other
embedded languages.
Mike
On Aug 25, 2009, at 5:22 AM, Juan Backson wrote:
Hi Ken,
xml_curl is a great idea. Is there anyway to not having to setup
another HTTP server? For instance, can I have freeswitch to call an
api or
On Tue, Aug 25, 2009 at 7:20 AM, Mike Peace mpe...@edcogroupinc.com wrote:
I fired up Wireshark on each side and I can see the SIP register request
coming from the laptop, the Freeswitch server replies with a Destination
Unreachable (Port unreachable) message.
I rebooted the Server and
On Tue, Aug 25, 2009 at 2:06 AM, Raimund Sacherer r...@runsolutions.comwrote:
Hi List,
i have some scripts to test our lab:
I have scripts to create sipp instances which act as individual
agents, which log on, take a random amount of calls, log off, wait a
bit, log on again, etc.
I have
On Tue, Aug 25, 2009 at 7:35 AM, Raimund Sacherer r...@runsolutions.comwrote:
On Aug 25, 2009, at 4:24 PM, Moises Silva wrote:
On Tue, Aug 25, 2009 at 9:35 AM, Raimund Sacherer
r...@runsolutions.comwrote:
Hello,
i was reading through the openzap wiki page and searched the wiki, the
only
Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have
sense to move my OS to 64 bit? ... will FS gain more preformance ?... I mean
will FS perofomr drastically better 20%+ ?
If you really want to get on the same page as the developers then get the
64bit CentOS 5.3 loaded on
On Tue, Aug 25, 2009 at 8:29 AM, Michael Jerris m...@jerris.com wrote:
You can also do xml config hooks in perl and some of the other embedded
languages.
FYI, here's a page to go you started:
http://wiki.freeswitch.org/wiki/Mod_perl_and_Generating_XML
-MC
Exactly... the scenario i use seems operating on a single thread... why is
that ? can it be changed?
T.
On Tue, Aug 25, 2009 at 5:31 PM, Michael Jerris m...@jerris.com wrote:
Actually in this case, we are bound to one thread in sofia.
Mike
On Aug 25, 2009, at 9:47 AM, Giovanni Maruzzelli
well :) ... this is something we are going to change tomorrow of course
will let you posted.
T.
On Tue, Aug 25, 2009 at 6:11 PM, Michael Collins m...@freeswitch.org wrote:
Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have
sense to move my OS to 64 bit? ... will FS
I get a bash: sofia: command not found. Is there something I need to add to my
config to use these commands?
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, August 25, 2009 10:55 AM
To:
I wish I had a nickel for every guy struggling with sipp load testing vs
real world traffic.
On Tue, Aug 25, 2009 at 1:51 AM, Tihomir Culjaga tculj...@gmail.com wrote:
Hello Takeshi,
Thanks for your hint... it worked out... so to be precise:
VIA header of both INVITE and ACK messages MUST
You also should put your extension first in your dialplan ahead of the
default extensions which match every call and do a lot of db access for
record keeping etc.
The single thread in sofia is part of their concurrency model.
The single thread acts as a scheduler and indicates to us that an
This requires invasive changes in the sofia-sip stack to get thread-
pooling working again. I am sure they would accept patches if you can
provide some that fully address any issues that may come up from
adding this such as race conditions.
Mike
On Aug 25, 2009, at 12:34 PM, Tihomir
Of course i removed everytihng from teh dialplan except my extension :)
when exactly do you react and bring up a new thread ? ... is it on INVITE or
on 1st 1xx response ?
i beleive i can have several lets call it SIP interfaces ... on different
ports 5060, 5070, 5080 ... every interface will
if you want to run that at your prompt instead of at the fs_cli you
can do this:
function sofia() { fs_cli -x $(echo sofia $@); }
(thanks ray for the bash foo)
Mike
On Aug 25, 2009, at 1:04 PM, Mike Peace wrote:
I get a bash: sofia: command not found. Is there something I need to
add to
On Tue, Aug 25, 2009 at 10:04 AM, Mike Peace mpe...@edcogroupinc.comwrote:
I get a bash: sofia: command not found. Is there something I need to add
to my config to use these commands?
Hehe, sorry. The sofia command is for use at the FreeSWITCH command line.
If you have FS running as a daemon
Hi Giovanny,
thanks for your help,
everything that heavyly accesses the disk is on ramdisk now...
hopefully will get some real traffic pretty soon...
On Tue, Aug 25, 2009 at 3:47 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote:
is a heavely multithreaded software, it benefits from number of
Ok, figured out from your post to load the FS-CLI in order to run the Sophia
status Michael mentioned.
Now I get: [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting [Socket
Connection Error] when running ./fs_cli from a terminal window in
/usr/local/freeswitch/bin where fs_cli is located.
On Aug 25, 2009, at 1:40 PM, Michael Collins wrote:
On Tue, Aug 25, 2009 at 10:31 AM, Tihomir Culjaga
tculj...@gmail.com wrote:
Of course i removed everytihng from teh dialplan except my
extension :)
when exactly do you react and bring up a new thread ? ... is it on
INVITE or on 1st
It means freeswitch is not running. You need to start it with /usr/
local/freeswitch/bin/freeswitch -nc
@MikeJ: Nice bash scripting :D
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 25-Aug-09, at 1:53 PM, Mike Peace
mod_sofia will take care of spawning the session thread once it
authenticated the call and loaded all the variables related to the
call such as the caller profile (callerid, destination number, etc).
If you want to check the source, this is done in
sofia_handle_sip_i_invite()
On Tue, Aug 25, 2009 at 10:31 AM, Tihomir Culjaga tculj...@gmail.comwrote:
Of course i removed everytihng from teh dialplan except my extension :)
when exactly do you react and bring up a new thread ? ... is it on INVITE
or on 1st 1xx response ?
i beleive i can have several lets call it SIP
Giovanni,
you mean like this message?
Unable to determine location for device. Voicemail password set via FreePBX
will not be valid.
This is a known FreePBX issue. http://www.freepbx.org/v3/ticket/36
Let's keep in ming FreePBX v3 is a developer release and as such many
features are in flux and
Hello All,
Does anyone know what the capacity of a stand-alone Freeswitch, in terms of
how many users?
Also, when that number is exceeded, how can Freeswitch server be distributed
to accommodate a larger installation?
Best Regards,
Jerry
___
clear... thanks!
On Tue, Aug 25, 2009 at 7:55 PM, Michael Jerris m...@jerris.com wrote:
On Aug 25, 2009, at 1:40 PM, Michael Collins wrote:
On Tue, Aug 25, 2009 at 10:31 AM, Tihomir Culjaga tculj...@gmail.comwrote:
Of course i removed everytihng from teh dialplan except my extension :)
Is FreePBX V3 based on Freeswitch?
Michael DiMartino | Director of IT | Open Access, Inc.
115 Bi County Blvd | Farmingdale, NY 11735
631.227.1034| 631.694.6730 FAX |631.988.6060 MOBILE
www.openaccessinc.com
From: freeswitch-users-boun...@lists.freeswitch.org
Carlos,
you're very kind, as always.
I'm aware that this is a dev preview, and I'm interested just in that,
to begin getting acquainted with the framework (and adding support to
the endpoints/trunk I take care of).
I probably have not got the logic right :-) (I tried both Windows
Installer and
Hello All,
I noticed Freeswitch becomes the middle-man, handling RTP traffic for an
active call. How do I configure it so it allows the two SIP endpoints to
send RTP packet to each other directly?
Best Regards,
Jerry
___
FreeSWITCH-users mailing
It supports FreeSWITCH and soon other engines as well. More
information is at:
http://www.freepbx.org/freepbx-v3
Mike
On Aug 25, 2009, at 2:20 PM, Michael Di Martino wrote:
Is FreePBX V3 based on Freeswitch?
Michael DiMartino | Director of IT | Open Access, Inc.
115 Bi County Blvd |
On Tue, Aug 25, 2009 at 11:35 AM, Jerry Richards jerry.richa...@teotech.com
wrote:
Hello All,
I noticed Freeswitch becomes the middle-man, handling RTP traffic for an
active call. How do I configure it so it allows the two SIP endpoints to
send RTP packet to each other directly?
Check
Deployable scalability varies based on a number of things... Number of users
registering, how often they register, concurrent call volume, call rate
(calls/second) etc... Defining that a little better may illicit a better
response... But generally FS can scale into the 1000s of concurrent calls
Just remember that bypass_media is a special mode and not everything will
work when a call is in that mode... Don¹t expect to do anything that
remotely relies on media being in the FS box...
From: Michael Collins m...@freeswitch.org
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Tue, 25
You can start it without the -nc switch and it'll stay active on the
current terminal so you can look for any errors.
Just do:
/usr/local/freeswitch/bin/freeswitch
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On
It was started from the services configuration window first and I still get the
error. I set the service not to load rebooted then ran it from the prompt as
you suggested. It stated 3585 Backgrounding I then ran ps -eaf from the
prompt and that PID didn't show up. Then when I attempted to run
OK,
Received a different error: Cannot Initialize [[error near line 294]: unclosed
!--]
?
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Mathieu Rene
Sent: Tuesday, August 25, 2009 2:02 PM
To:
It means you have an error in your XML configuration. You can open /
usr/local/freeswitch/log/freeswitch.xml.fsxml to figure out where it
is (but do not modify this file, as it is generated from all the other
xml files).
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Well, you'd have another nickel from over here, then.
If I can get this working before I'm tasked with something else I'll write
up something more on the wiki about Freeswitch and SIPp, but I'm not sure
I'll get that chance.
BB
On Tue, Aug 25, 2009 at 11:05 AM, Anthony Minessale
here's the one i use for making a call waiting x seconds and hanging up
http://www.freeswitch.org/eg/load_test/dft_cap.xml
This requires that the sipp terminate all the calls.
careful with sipp, it's like a roach motel, you can get stuck trying to make
it work and never get it to produce
Hi Brian,
From the CLI
freeswi...@open46 system /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav
/tmp/foo.mp3 -S
2009-08-25 22:41:51.556484 [NOTICE] mod_commands.c:3386 Executing command:
/usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S
API CALL
maybe it's writing some err to stderr that is being suppressed somehow
On Tue, Aug 25, 2009 at 3:46 PM, Alberto Escudero-Pascual (lists)
aep.li...@it46.se wrote:
Hi Brian,
From the CLI
freeswi...@open46 system /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav
/tmp/foo.mp3 -S
2009-08-25
That was it! I had been try to setup the conference java script example and I
made an error in the conference-conf.xml. Thanks a lot to everyone!
Can somebody point me in the right direction to get my new T-1 taking to the
Sangoma A101?
I have it installed and setup the Wanpipe driver off
Hi there...
So a few things on this.
1) We have a module that's still being worked on called Sip Interface that
allows you to configure Sip Profiles in FreeSWITCH. Unfortunately we don't have
the ability to easily import your existing SIP profiles, and by NOT displaying
them in the UI your
Darren,
thanks! it helps a lot.
If we svn up in /var/www/freepbx we got the trunk?
On Tue, Aug 25, 2009 at 10:59 PM, Darren Schreiberd...@d-man.org wrote:
Hi there...
So a few things on this.
1) We have a module that's still being worked on called Sip Interface that
allows you to
Can somebody point me in the right direction to get my new T-1 taking to
the Sangoma A101?
Are you using PRI or just a 24 channel T1, aka CAS or RBS? Check out this
page for some handy information:
http://wiki.freeswitch.org/wiki/OpenZAP
-MC
P.S. - If you join #openzap and/or #freeswitch
I have looked at that but I am confused on which files need to be edited. Since
I have already installed in Wanpipe mode with the Sangoma card I skipped
straight to the Wanpipe section. It mentions setting the [span wanpie PRI_1]
etc in the openzap.conf then further down it mentions editing the
Darren,
I shall look forward to check version 3 of FreePBX for FS.
I hope it not trixswitch anyone aware of status on trixswitch (not
that I m a big fan of Trixbox, but quite a while back I had seen
trixswitch iso so wondering if anything is progresing on that end.
Thanks Regards,
Mitul
Anthony,
Yes, you are right, I was thinking strictly in terms of SIP gateways. I
guess that instead on the tag gateway, one could use channel? For
example:
action application=bridge
!-- set some kind of *bridge-specific* variable --
channel order=1 data=sofia/gateway/gw-1/1$1/
channel
I am setting the caller id like this in my ESL script:
@con.sendRecv(api originate
{origination_caller_id_number=15103245678}sofia/gateway/junctionnetworks/1...@number}
park())
And the caller id comes out as all zeroes. The sip trace shows the from
as shown in the sofia status command. This is
That still wouldn¹t work... An action has 2 parameters application and
data... And deeper then that and you have to start re-arranging all sorts of
things...
Continue_on_fail and hangup_after_bridge like tony pointed out are what you
want if you don¹t want to use the | delimiting ... I use these
You can try playing with this in your gateway profile
param name=caller-id-in-from value=true/
Not sure what it'll do to your registration, give it a try
On Tue, Aug 25, 2009 at 21:51, Shameem Shiek gshfre...@gmail.com wrote:
I am setting the caller id like this in my ESL script:
Continue_on_fail and hangup_after_bridge like tony pointed out are what you
want if you don’t want to use the | delimiting ... I use these all the time
with gateway counts 10 just stacking additional actions for each bridge
line
Let's imagine that I need to call 1000,1001,1002 via
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