Hi,
Sometimes when I try to restart FS, I get the error while the mod_python is
loaded:
Python error: stdin is a directory, cannot continue
In some cases the error doesn't disappear until system reboot.
What causes such errors?
Thank you,
Vassil Panayotov
hello
i'm looking for a possibility to restart freeswitch like it is possible with
asterisk.
for me i tried to created a script that looks for open channels and if no
channel
is open it restarts freeswitch with the init script (not the most efficient
way).
i think i would be great if we would
Hi,
I'm newbie in FS. As far as I know for setting up custom variables in FS we
use this syntex in dialplan XML i.e.
action application=set data=ABC='value'/
But when I call this variable using eval application i.e.
action application=eval data=${ABC}/
the value I get from variable ABC is
Hi Nandy,
yes already tried this, but if I use proxy_media=true, FS makes no
control on the content of the RTP stream. But the pbm is that I need to
use this:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf_generate
This function enables transcoding of SIP_INFO or RFC2833 to
Hi,
I couple of my team members are working on translating a very long Asterisk
Dial Plan to FreeSWITCH XML Dial Plan. Now reference to wiki link below,
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#About_Dialplan_Variables
The dial plan variables are not getting initialized as
Why not make your gays have with all those programming skills try each one of
them. Say,
one guy programs in Perl, the other in PHP, still the other in Python, still
again
the other in Java and finally one in LUA. Take note, same dialplan project and
let them
not compare notes or translate the
Has anyone out there has the opportunity to get hands on a 5400zl series
Procurve?
FS on the Intel based module
That would be a sweet application!!
As far as I understand applications need to undergo some testing before they
can be run on the module.
anyone can comment??
rod,
it looks more complicated now when PEER C comes to the picture. i think
we'll have to wait for the availability of g729 on FS, as per Anthony's
post.
/nandy
On Fri, Sep 4, 2009 at 1:54 PM, rod kawa...@laposte.net wrote:
Hi Nandy,
yes already tried this, but if I use proxy_media=true,
On Thu, Sep 3, 2009 at 11:05 PM, Ahmed Munir ahmedmunir...@gmail.comwrote:
Hi,
I'm newbie in FS. As far as I know for setting up custom variables in FS we
use this syntex in dialplan XML i.e.
action application=set data=ABC='value'/
But when I call this variable using eval application
On Thu, Sep 3, 2009 at 10:59 PM, Muhammad Shahzad
shaherya...@googlemail.com wrote:
Hi,
I couple of my team members are working on translating a very long Asterisk
Dial Plan to FreeSWITCH XML Dial Plan. Now reference to wiki link below,
Before you go through all the trouble of translating
Look at the fsctl api on the wiki. It has what you need.
jmesquita
On 9/4/09, Christian Löschenkohl christian.loeschenk...@xpirio.com wrote:
hello
i'm looking for a possibility to restart freeswitch like it is possible with
asterisk.
for me i tried to created a script that looks for open
2009/9/3 Christian Löschenkohl christian.loeschenk...@xpirio.com
hello
i'm looking for a possibility to restart freeswitch like it is possible
with
asterisk.
for me i tried to created a script that looks for open channels and if no
channel
is open it restarts freeswitch with the init
Thank you so much. Of course we are not doing a blind translation, but at
the very basic we will need to get and set certain variable at different
stage of call processing.
Another question in same context, Can we do post-hangup call processing? I
mean like in Asterisk, we have extension h which
On Fri, Sep 4, 2009 at 12:25 AM, Muhammad Shahzad
shaherya...@googlemail.com wrote:
Thank you so much. Of course we are not doing a blind translation, but at
the very basic we will need to get and set certain variable at different
stage of call processing.
Another question in same context,
After some testing (fs_cli -x 'fsctl shutdown restart') I'm seeing
that all active calls are dropped and the freeswitch is restarted
On Fri, Sep 4, 2009 at 10:14 AM, Michael Collinsm...@freeswitch.org wrote:
2009/9/3 Christian Löschenkohl christian.loeschenk...@xpirio.com
hello
i'm
freeswi...@foosball fsctl
-USAGE: [send_sighup|hupall|pause|resume|shutdown [cancel|elegant|asap|
restart]|sps|sync_clock|reclaim_mem|max_sessions|max_dtmf_duration
[num]|loglevel [level]]
On Sep 4, 2009, at 3:35 PM, Anatoliy Kounitskiy wrote:
After some testing (fs_cli -x 'fsctl shutdown
Hi Folks,
I just tried to install FS with mod_skypiax on a new CentOS 5.2 machine.
After having followed the big help doc from the wiki page (
http://wiki.freeswitch.org/wiki/Skypiax#An_example_of_Skypiax_and_FreeSWITCH_installation_on_CentOS.2C_from_scratch),
I hit an error when running multi.sh
Jingwei,
those are normal warnings made by the Skype client (not by
mod_skypiax), you just have to edit /etc/alsa/alsa.conf and comment
out hdmi lines. Is a problem with a lazy implementation of that file,
that supposes you got an hdmi.
The other warning is because there are some files missing
Hi Giovanni,
That's a big relief. Thanks a lot for the reply :)
Regards,
-Jingwei
On Fri, Sep 4, 2009 at 4:25 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote:
Jingwei,
those are normal warnings made by the Skype client (not by
mod_skypiax), you just have to edit /etc/alsa/alsa.conf and
On Fri, Sep 4, 2009 at 1:19 AM, Brian Westbr...@freeswitch.org wrote:
There will not be an authorization header on the first register attempt...
it only happens once we are 401/407'ed and the phone comes back and
registers again.
/b
Alas, I cannot change the way the provider's gateway works.
:-) My fault, I would have to document this.
I'll do pretty soon.
Sorry about that, and thanks for reporting!!!
-gm
Sincerely,
Giovanni Maruzzelli
Cell : +39-347-2665618
On Fri, Sep 4, 2009 at 10:38 AM, Jingwei Yangjingwei.y...@gmail.com wrote:
Hi Giovanni,
That's a big relief. Thanks
What you're looking for is :
fs_cli -x 'fsctl shutdown elegant restart'
:)
It will restart the freeswitch after all calls are hanged up.
On Fri, Sep 4, 2009 at 10:46 AM, Seven Dudujinf...@gmail.com wrote:
freeswi...@foosball fsctl
-USAGE: [send_sighup|hupall|pause|resume|shutdown
Thanks Anthony,
that did the trick.
Best regards
Peter
Anthony Minessale schrieb:
you can edit mod_xml_curl.c line 64
and increase XML_CURL_MAX_BYTES
On Thu, Sep 3, 2009 at 12:31 PM, Peter P GMX prometheus...@gmx.net
mailto:prometheus...@gmx.net wrote:
Hello,
in a B2BUA
On Fri, Sep 4, 2009 at 10:56 AM, Giovanni Maruzzelligmar...@celliax.org wrote:
:-) My fault, I would have to document this.
http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#Error_and_warnings_at_the_starting_of_Skype_clients_on_Linux
-giovanni
Updated the wiki page with references to other errors/warnings as well :-)
-giovanni
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi mayamatake...@gmail.comwrote:
Hello,
I'm testing FS support for the header Path (FS is behind opensips).
It pretty much works: I tested calling from one user to the other and calls
work perfectly.
However, I've noticed that when I register my
That's efficient :) By the way, do you have any idea about this warning?
ALSA lib pcm_dmix.c:1008:(snd_pcm_dmix_open) unable to open slave
On Fri, Sep 4, 2009 at 5:47 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote:
On Fri, Sep 4, 2009 at 10:56 AM, Giovanni Maruzzelligmar...@celliax.org
I had a similar problem when I needed to talk to a gateway using g729
while g711 was used by default. The following works for me:
vars.xml
(...)
X-PRE-PROCESS cmd=set
data=global_codec_prefs=PCMU,PCMA,g7...@32000h,g7...@16000h,G722,GSM,G729,G723/
X-PRE-PROCESS cmd=set
Hi,
i am have FS SVN revision 14760, i am trying to use mod_xml_curl against
mod_dingaling. When i call xml_curl url in browser i get mod_dingaling
configuration correctly, also when i do reload mod_dingaling it fetches its
configuration from xml_curl correctly. BUT when i try to use dl_login
Hello Anthony,
2009/9/2 Anthony Minessale anthony.miness...@gmail.com:
yes if you have a version that only has log-file you can use that.
if you find me on irc and send me the credentials privately I will examine
your box for you.
thanks for that offer, but the box is pretty deep inside our
I cannot change the way SIP Authentication works. The first register
is always sent without an authorization header then is challenged. If
you're getting an instant 403 then you have something wrong in your
config and the remote system doesn't like it. Please contact your
provider and
Worst offenders (leakers over 100K). The last one is the worst (672M)
-- looks like a lua script. What are you doing in lua again?
==28624== 105,725 bytes in 1,804 blocks are still reachable in loss
record 497 of 529
==28624==at 0x4022AB8: malloc (vg_replace_malloc.c:207)
==28624==by
Hi
On Thu, Sep 3, 2009 at 6:52 AM, Mathieu Renemrene_li...@avgs.ca wrote:
See
http://wiki.freeswitch.org/wiki/Authoring_Freeswitch_Modules#Subscribing_to_events
Thanks.
I have tried this method without success and finally replaced the
voicemail section in dialplan by a spidermonkey script
2009/9/4 Rupa Schomaker r...@rupa.com:
Worst offenders (leakers over 100K). The last one is the worst (672M)
-- looks like a lua script. What are you doing in lua again?
i feel kinda dumb to double post, but here it is again :)
the setup is the same as in
Doesn't that look like a pool that isn't being destroyed?
On Fri, Sep 4, 2009 at 9:10 AM, Rupa Schomakerr...@rupa.com wrote:
Worst offenders (leakers over 100K). The last one is the worst (672M)
-- looks like a lua script. What are you doing in lua again?
==28624== 672,268,288 bytes in
personally i would blame xmlrpc (which is no xml :) for it.
Just my 2cent
Beni.
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
There are other smaller leakers. xmlrpc is leaking, but the leaks are
very small compared to the lua leak. Same with spidermonkey_curl - it
is leaking but not too terribly much. I'll hop on #freeswitch in a
bit and see if anyone has an idea.
On Fri, Sep 4, 2009 at 9:35 AM, Benedikt
Hi Michael,
Why is it not recommended to do the brdge app right in the script? The
reason I ask this, I did have lot of trouble using Park/Fifo app in the
script and the whole thing started working after I did the UUID transfer and
have the things I wanted executed as part of the Dial plan.
that looks to me like luarun being called on a script that never terminates.
could your script be ending up caught in an endless loop or blocking on
something?
On Fri, Sep 4, 2009 at 9:42 AM, Rupa Schomaker r...@rupa.com wrote:
There are other smaller leakers. xmlrpc is leaking, but the leaks
Hi all
anyone know where I can find UK English recordings for the FS prompts
(assuming there are any)? (I've googled to no avail). Alternatively is
there a list of the text used so we can record our own?
Regards
Brian
___
FreeSWITCH-users mailing
We don't currently have a full set of UK English prompts, the prompts
list (soon to be updated with some new prompts) is available at:
http://svn.freeswitch.org/svn/freeswitch/trunk/docs/phrase/phrase_en.xml
If you are going to get a set professionally recorded, we would be
happy to host
Hello all,
We are now on line and welcoming callers. Here's the agenda so far:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_04
Come join the conference
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
1-213-799-1400
-MC
___
I'm in, very cool =D
Diego
On Fri, Sep 4, 2009 at 4:18 PM, Michael Collins m...@freeswitch.org wrote:
Hello all,
We are now on line and welcoming callers. Here's the agenda so far:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_04
Come join the conference
For the ones SIP challenged:
call Skype the skypeuser skypiax5 and then press 1
-gm
On Fri, Sep 4, 2009 at 6:43 PM, Diego Violadiego.vi...@gmail.com wrote:
I'm in, very cool =D
Diego
On Fri, Sep 4, 2009 at 4:18 PM, Michael Collins m...@freeswitch.org wrote:
Hello all,
We are now on
Does anyone know of a SIP provider or network directory? A list of all
the public service provider or networks? Gizmo, Google Voice, etc? Or
Vitelity, iCall?
Lon Baker
Kickass Pixels
-
+1-415-894-0184
-
http://kickasspixels.com
http://twitter.com/kickasspixels
http://www.linkedin.com/in/lonbaker
Under the Minimum/Recommended System Requirements, what is meant by We
recommend you plan for 50% duty cycle? What is this duty cycle?
Also, I see that the system requirements indicate Freeswitch recommends 1GB
RAM and 50MB disk space. I guess I'm wondering how the number of extensions
and
On Fri, Sep 4, 2009 at 4:45 PM, Brian Westbr...@freeswitch.org wrote:
I cannot change the way SIP Authentication works. The first register is
always sent without an authorization header then is challenged. If you're
getting an instant 403 then you have something wrong in your config and the
Try filling out contact-host too. But if the far end gets pissed
about your contact they are broken.
/b
On Sep 4, 2009, at 2:22 PM, Dmitry Bely wrote:
Well, you are right. Looks like the problem is not with authorization
but in the line
Contact: sip:gw+1.2@5.6.7.8:5080;transport=udp
I'm started to suspect another thing.. Successful register (SIP phone) contains
REGISTER sip:Domain SIP/2.0
while unsuccessful one is
REGISTER sip:1.2.3.4 SIP/2.0
What parameter is responsible for Request-URI? Note that I need both
IP address for proxy and symbolic name for SIP domain (which
show me your XML for the gateway please.
/b
On Sep 4, 2009, at 3:43 PM, Dmitry Bely wrote:
I'm started to suspect another thing.. Successful register (SIP
phone) contains
REGISTER sip:Domain SIP/2.0
while unsuccessful one is
REGISTER sip:1.2.3.4 SIP/2.0
What parameter is responsible for
Jerry,
As far as I understand freeswitch, it using kernel to thread and this
operation eats good amount of RAM, but since the internal strructure
of fs is to store all these sip details in runtime sqlite db, which is
compressed text data earlier written in XML but while fs loads this
On Sat, Sep 5, 2009 at 1:08 AM, Brian Westbr...@freeswitch.org wrote:
show me your XML for the gateway please.
/b
It's fairly standard:
!--
Shell provider account should work with most providers.
--
include
user id=$${default_provider}
gateways
gateway
Can you send it to me with the data filled out off list please.
/b
On Sep 4, 2009, at 4:33 PM, Dmitry Bely wrote:
It's fairly standard:
!--
Shell provider account should work with most providers.
--
___
FreeSWITCH-users mailing list
Hello,
It may sound a bit stupid but still wanna ask out here, if there is
any way to replicate FreeSWITCH wiki mirror for local reference or
mainataining local copy instead of Reading online which costs a lot in
developing countries like that of ours and Asia/Africa in general.
We tried
I will look into the MediaWiki docs to see what's available. In the meantime
you will probable need to use the recent changes link on the navigation
bar.
-MC
On Fri, Sep 4, 2009 at 2:39 PM, Mitul Limbani mi...@enterux.com wrote:
Hello,
It may sound a bit stupid but still wanna ask out here,
Hello,
I have a question, but Iââ¬â¢m not certain whether this is a FreeSWITCH
issue, or
something specific to Perl.
I have setup a Perl application (ââ¬Ålistenerââ¬Â) that monitors the
events of my
FreeSWITCH server via a Telnet socket. So far, the application seems to work
very
you should use ESL lib and the supplied perl mod
from FS build root
cd libs/esl
make
make perlmod
cd perl
copy ESL.pm and ESL.so into your INC path
see the examples in that same folder.
On Fri, Sep 4, 2009 at 5:40 PM, Tina Martinez t...@a2unlimited.com wrote:
Hello,
I have a question, but
Mike,
I m not sure if we can program httrack to pickup changes automatically
(I.e. Looking at recent changes) so this brings us back to square one,
instead we can setup media wiki here and do rsync with fs wiki box.
Thanks Regards,
Mitul Limbani,
Founder CEO,
Enterux Solutions Pvt. Ltd.,
From the mod_nibblebill documentation:
At the end of a call, the module sets a variable named
nibble_total_billed. You can use mod_cdr to record this variable to
your CDR log.
Is it possible to do the same with mod_xml_cdr?
I'm looking for a simple way of billing my CDRs and this one looks
All the variables are there in XML_CDR too.
/b
On Sep 4, 2009, at 6:28 PM, Rogelio Perez wrote:
From the mod_nibblebill documentation:
At the end of a call, the module sets a variable named
nibble_total_billed. You can use mod_cdr to record this variable to
your CDR log.
Is it possible
Hello,
I'm a new Freeswitch user. After some reading I put Freeswitch (Version 1.0.4)
to work as Session Border Controller. I have only one problem that I dont know
how to solve it ( or which parameter to set) and I'd appreciate if someone
could give me a clue about this.
Kamailio is
Hi,
I just installed freeswitch as a replacement for our Asterisk Server. I want
to untimately do Conferencing with it as I have heard is it pretty good at it.
I have it compiled and up and running. However, when I provision a
Sofphone/Xlite to register with it to run basic tests, it does
make sure your firewall is not up
/b
On Sep 4, 2009, at 5:15 PM, Ujjval Karihaloo wrote:
Hi,
I just installed freeswitch as a replacement for our Asterisk
Server. I want to untimately do Conferencing with it as I have heard
is it pretty good at it.
I have it compiled and up and
I'm going to gess the call-id is the same for the second
transaction... can you provide a more detailed trace?
/b
On Sep 4, 2009, at 11:06 AM, Humberto Quintana wrote:
Hello,
I'm a new Freeswitch user. After some reading I put Freeswitch
(Version 1.0.4) to work as Session Border
If you do event plain all from the FS CLI you should see the variable
exported on the CHANNEL_HANGUP_COMPLETE event, with the other CDR variables
as well. These information should be available on mod_xml_cdr and
mod_cdr_csv as well.
Diego
On Fri, Sep 4, 2009 at 11:28 PM, Rogelio Perez
I have a call transfer problem with Freeswitch
Here is the call flow:
I call from the PSTN (A party) into my Polycom phone (B-party) which is
registered to FreeSwtich. The Freeswtich is setup not to route media as I have
an SBC acting as a mirror proxy that will do all the NAT and media
Would that be firewall on the CentOS machine that FS is installed on?
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West
Sent: Friday, September 04, 2009 5:35 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re:
67 matches
Mail list logo
