Hi,
I am having problem with around 1 % of the channels always get zombilized.
What I want to do is to have a background thread that regularly check all
the channels that have been in existance for like 1 hr, and then check to
see if there is any RTP coming in and going out. If there is no
do
that for you.
Unless something else is going on.
/b
On Dec 16, 2009, at 6:33 AM, Juan Backson wrote:
Hi,
I am having problem with around 1 % of the channels always get
zombilized.
What I want to do is to have a background thread that regularly
check all the channels that have
Hi,
I found that with bypass_media=true, freeswitch would change c= to FS's own
IP.
I think this is a misconfiguration. Does anyone know what config could have
caused that?
thanks,
jb
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In the following trace,102 is FS IP, 104 is calling party and 13 is
called party.
with bypass_media, FS still changesc=IN IP4 192.168.1.102
Any idea why?
freeswi...@localhost.localdomain recv 951 bytes from
udp/[192.168.1.104]:5060 at 22:56:33.782715:
Hi,
Instead of using show calls count to obtain the current call count stat, I
am writing some C code to increment a counter during on_answer_hook and
decrement the counter during on_hangup_hook.
It looks like my counter result is very closed to show calls count when
the traffic is low, like 50
Hi,
If I am using proxy_media=true, bypass_media=false, is there anyway of
modifying o= and c= so that it won't show the IP of the far-end B leg?
I am using fs as b2b2a and I want to hide the far-end ip as much as
possible.
I got to hide the IP for invite by modifying the sdp within C code, but
Hi,
I tried to use the variable remote_media_ip from within dialplan, but it is
not returning anything.
Does anyone know when this variable gets set and how to have this variable
to be set as soon as an INVITE hit freeswitch?
Thanks,
jb
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Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 24-Nov-09, at 10:22 AM, Juan Backson wrote:
Hi,
I tried to use the variable remote_media_ip from within dialplan,
but it is not returning anything.
Does anyone know when this variable
Hi,
I am using 1.0.4 version of freeswitch and I am doing proxy_media for all
calls. Basically, I just proxy all media from one gateway to another with
freeswitch serving as a middleman.
In the outgoing invite, I found that the owner line ( o= ) in SDP is showing
the originator's IP which I
Hi,
Is there anyway to configure freeswitch so that it won't retry the SIP
message when 200 OK is not received?
thanks,
jb
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the set_user app
On Thu, Oct 8, 2009 at 5:20 AM, Juan Backson juanback...@gmail.comwrote:
Hi,
My application fails to set the appropriate variables using directory xml
after using the latest trunk as of yesterday.
My curl looks like:
document type=freeswitch/xml
section name=directory
Hi,
I am still stuck in trying to get curl directory variables to show up in
channel even after the user has registered.
Is this something that gets changed in the latest trunk or is it just my
mis-configuration?
please help.
jb
On Sat, Oct 10, 2009 at 6:05 PM, Juan Backson juanback
Hi,
My application fails to set the appropriate variables using directory xml
after using the latest trunk as of yesterday.
My curl looks like:
document type=freeswitch/xml
section name=directory
domain name=192.168.1.102
user id=22
params
param name=password
Hi,
Does anyone have any luck on porting freeswitch to blackfin + uclinux?
Is this a feasible option?
jb
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Hi,
I need to have local_media_ip, local_media_port, remote_media_ip, and
remote_port to be available in the b-leg. Is there anyway to do that?
Since these variables are only created after the call has been answered, I
can't do application name=export
this MESSAGE.
Is there anyway to solve this problem.
thx,
jb
On Sun, Sep 6, 2009 at 2:36 AM, Brian West br...@freeswitch.org wrote:
Not automatically. But you could use the event socket to get the
message and forward it via ESL.
/b
On Sep 5, 2009, at 1:26 PM, Juan Backson wrote:
If so
Hi,
I am getting no dial-string available error when using xml_odbc module to
bridge a call. How can I resolve this problem?
EXECUTE sofia/internal/180...@192.168.1.130 bridge(user/180001)
2009-09-05 16:31:29.853456 [INFO] mod_xml_odbc.c:401 DEBUG in
xml_odbc_search, header
Hi
Is there anyway to use freeswitch to redirect chat message the same that it
redirects SIP message?
If so, how can it be done?
thx,
jb
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at 11:51 PM, Juan Backson juanback...@gmail.comwrote:
Hi,
In a new environment, I am getting the following error when building the
latest freeswitch from svn. Does anyone know how to resolve it?
gcc -I/usr/src/freeswitch-snapshot/src/include
-I/usr/src/freeswitch-snapshot/libs/libteletone
Hi,
Things are working find before I tried using public IP ( behind NAT ) to
register IP phones. I am getting:
2009-09-02 20:46:50.575837 [WARNING] sofia_reg.c:1713 Can't find user
[180...@public-ip]
You must define a domain called 'public-ip' in your directory and add a user
with the
Hi,
In a new environment, I am getting the following error when building the
latest freeswitch from svn. Does anyone know how to resolve it?
gcc -I/usr/src/freeswitch-snapshot/src/include
-I/usr/src/freeswitch-snapshot/libs/libteletone/src -fPIC -Werror
-fvisibility=hidden
Hi,
I tried to make install mod_xml_odbc and load it in freeswitch, but I am
getting:
2009-08-28 00:46:55.848087 [CRIT] switch_loadable_module.c:871 Error Loading
module /usr/local/freeswitch/mod/mod_xml_odbc.so
**/usr/local/freeswitch/mod/mod_xml_odbc.so: invalid ELF header**
I had to manually
Hi,
ok, the module can be loaded, but it now complains about odbc. I can't find
anything missing in my odbc.ini. Could someone please point me to the right
direction?
2009-08-28 02:22:35.670284 [ERR] switch_odbc.c:188 STATE: IM002 CODE 201
ERROR: [unixODBC]Missing server name, port, or
Hi,
Finally, I got xml_odbc running, but it does not really work well for me. I
am getting:
2009-08-28 03:33:47.459383 [ERR] mod_xml_odbc.c:325 Stopped rendering
template, called xml_odbc_render_template more than [32] times, probably
looping.
2009-08-28 03:33:47.459383 [ERR] mod_xml_odbc.c:408
Hi Leon,
Thanks for your help.
I have changed it according to your comment but I am still getting the
looping error.
Would you please take a look see what else I did wrong?
Also, sip_user is an integer field, so I can't really use ''. Is there
anyway to get around that?
configuration
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?
--
*From: *Juan Backson juanback...@gmail.com
*Reply-To: *freeswitch-users@lists.freeswitch.org
*Date: *Tue, 25 Aug 2009 16:21:58 +0800
*To: *freeswitch-users@lists.freeswitch.org
*Subject: *[Freeswitch-users] reload user data
Hello,
I would like to dynamically add user to freeswitch
Hi
I tried to download non-US sound files, but I am getting this error:
--21:50:09-- http://files.freeswitch.org/freeswitch-sounds-fr-1.0.10.tar.gz
Resolving files.freeswitch.org... 69.174.57.101
Connecting to files.freeswitch.org|69.174.57.101|:80... connected.
HTTP request sent, awaiting
Hi,
Does anyone know the purpose of fifo_orbit_announce?
When does fifo_orbit_announce get played?
Thanks,
JB
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Hi,
I am getting some strange vg malloc error message in switch_core_hash_insert.
Does anyone know what is wrong with these few lines? Am I missing
something?
switch_core_hash_init(hash,pool);
param_name =switch_core_sprintf(pool,%s, key);
param_value =switch_core_sprintf(pool,%s, value);
Does anyone know how to take the epoch time in switch_event_t and convert it
into a format such as Sat Jul 5 02:44:33 2009?
Is there any existing facility that I can use for this purpose?
br,
JB
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Hi,
I would like to implement a random route selection based on some arbitrary
percentage.
Does anyone know if there is any good way of doing that within freeswitch?
If there isn't any api that I can use, does freeswitch has any random
generator that I can be used for this purpose?
br,
JB
Hi,
Is there a sample module that I can take a look at on how to do that?
I don't understand how to get the registration request and how to pass back
auth result to freeswitch.
JB
On Mon, Aug 3, 2009 at 8:42 PM, Brian West br...@freeswitch.org wrote:
You could build your own module to do it
Hi,
Other than curl, is there anyway to do dynamic registration?
It there anyway to embed a script in freeswitch to do the authorization?
Thanks,
Anne
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Hi,
I am trying to use the fifo app, but I am hitting on the following error.
Does anyone know how to resolve the No code is found error?
Dialplan: sofia/internal/1...@192.168.1.102 Regex (PASS) [internal-call]
destination_number(5501) =~ /^(.*)$/ break=on-false
Dialplan:
Hi,
I read the fifo section of the wiki and what is not clear are:
What is the meaning of fifo_orbit_announce?
What is the meaning of fifo_override_announce?
Is it possible to create a scenario where the caller can hear Agent #123 is
going to attend to your call?
Any help will be greatly
Hi,
I am getting problem when one UA is xlite and another UA is another
sip application.
When I call from xlite to a sip application, I am getting noise:
I have tried these:
extension name=redial
condition field=destination_number expression=^3000
action application=bridge
,PCMU,8000,PCMU,8000
a6f1e90c-f6a9-4ac1-9f26-fe08c5c0dd74,outbound,2009-05-23
11:25:30,1243092330,sofia/internal/454044009539026,CS_EXCHANGE_MEDIA,1000,1000,192.168.1.193,454044009539026,,,XML,public,PCMU,8000,PCMU,8000
Thanks for any suggestion.
Thanks,
JB
On Sat, May 23, 2009 at 3:11 PM, Juan
Hi,
I notice that there is a newly-developed mod_easyroute model available. Has
anyone used it with large amount of routes ( ex 1M ) on a high traffic
scenario? For that kind of scenario, would it be better to consider using
out-going event socket to serve that purpose? I would greatly
Hi,
I am running some continuous testing hitting FS. There are a couple errors
that gets popped up occasionally and I am trying to find out why. In one of
the trace, I am seeing FS not sending 183. The weird thing is that this
problem is not happening everything, but on a very rarely basis.
Hi,
I am running a test for a set of 290 calls to be fired to freeswitch once
every second. The console logs show the following for one of those 290
calls. Could someone please help me out to fix this problem? What could be
causing this? I recalled being able to run thousands of calls without
Hi,
I don't think so because I am testing everything within LAN. Also, amount
290 calls, only 1 gets that error.
JB
On Mon, Jan 19, 2009 at 10:38 PM, Brian West br...@freeswitch.org wrote:
You have NAT issues... ACK Timeout.
/b
On Jan 19, 2009, at 5:06 AM, Juan Backson wrote:
BYE sip
Hi,
Is there a change in the playAndGetDigits api? In the old release,
11102, my lua script is working but is not working in the latest
release.
The error I am getting is Error in playAndGetDigits expected 10..10
args, got 9 .
Thanks,
JB
___
the profile with stun enabled for your testing.
On Fri, Dec 26, 2008 at 5:33 AM, Juan Backson juanback...@gmail.com wrote:
Hi,
Is there any hard limit set on the number of RTP sessions for
Freeswitch? I am seeing freeswitch start sending out BYE after the
number of RTP session reaches 3000
Hi
I am getting the following strange error while running stress test on
freeswith. When the number of sessions reaches 3000, I get the
following error:
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR]
Hi,
I am running some stress testings on freeswitch. When the number of
RTP ports reached around 1248 - 1250, freeswitch starts to pop out No
RTP ports available! error:
2008-12-23 13:14:02 [CRIT] sofia_glue.c:562
sofia_glue_tech_choose_port() No RTP ports available!
OS is Centos 5.2 64 bits
Hi,
I have openser set up as sip registrar, location, and redirect server. When
a INVITE is received in openser, it would send back a 301 to redirect the
user agent to one of the freeswitch servers. The question I have is that
what mechanism can be used within freewitch to tell whether the
Hi,
I tried using curl to configure switch.conf to set the min rtp port and max
rtp port. The request did arrive in webserver and the response is correctly
returned.
However, Freeswitch still did not use the port in that specified range. So,
I manually modified the switch.conf under
Hi,
I tried to use the Phrase API for playAndGetDigits, but it only works if the
phrase does not take any input. When an input is specified, it gives error:
2008-11-21 22:42:55 [ERR] switch_ivr_play_say.c:201
switch_ivr_phrase_macro() Can't find macro ivr_prompt,welcome.
2008-11-21 22:42:55
Hi,
Here is the exact lua script:
digits = session:playAndGetDigits( 1, 1, 3, 3000, #,
phrase:enter_userid,phrase:invalid_input,[1|2|3|4|5|6])
If I repeatedly enter 9, here is what would happen:
z
2008-11-12 22:48:22 [DEBUG] switch_ivr_play_say.c:269
switch_ivr_phrase_macro() Handle
with
line-by-line lua script to reproduce this feature.
JB
On Fri, Nov 7, 2008 at 7:57 PM, Gonzalo Servat [EMAIL PROTECTED] wrote:
On Fri, Nov 7, 2008 at 9:44 AM, Juan Backson [EMAIL PROTECTED] wrote:
Hi Gonzalo,
Here is the lua code I am using
digits = session:playAndGetDigits(1, 1, 3
Hi,
I am encountering an issue when using playAndGetDigit. I have the regex set
as 1|2|3|4|5|6, and then pressed 9, it gives error as expected. Then, I
pressed 2. This time it passes, but the digit being returned by
playAndGetDigit is 92, instead of just 2.
Also, the voice prompt seems a bit
Hi Gonzalo,
Here is the lua code I am using
1.
2. digits = session:playAndGetDigits(1, 1, 3, 3000, #*,
phrase:admin_menu, phrase:invalid_input, 1|2|3|4|5|6)
Thanks,
JB
Show us the relevant code you're using.
- Gonzalo
___
Hi,
I am working on developing a very simple custom mod that needs to push out
data to an external HTTP service, similar to the way curl works. Does
anyone have experience in writing a Freeswitch mod that pushes out data to
an external HTTP server and can give me a hand? I am hoping to learn
Hi,
I found out that if fifo_music is set before fifo out wait, the consumer
would get hung up after 30 s.
action application=set
data=fifo_music=/usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav/
action application=fifo data=myfifo out wait /
If I remove
error.
Any help in getting my FS to work again would be grealy appreciated.
Thanks,
JB
On Wed, Sep 24, 2008 at 10:46 PM, Michael Jerris [EMAIL PROTECTED] wrote:
On Sep 24, 2008, at 12:08 AM, Juan Backson wrote:
Hi,
My lua scripts were working fine until I updated with SVN. I am
installation?
Thanks,
JB
On Wed, Sep 24, 2008 at 11:06 PM, Michael Jerris [EMAIL PROTECTED] wrote:
That looked like an error loading luasql. Nothing changed recently in lua
or the freeswitch objects exposed to it.
Mike
On Sep 24, 2008, at 10:57 AM, Juan Backson wrote:
Hi Michael,
Thank you
Hi,
I just refreshed another working FS enviornment with the latest SVN, that
environment was working fine until I installed the latest build. The luasql
error starts to pop up. The only working environment I have now is running
on last week's build.
If I run run require(luasql.mysql) on
Hi,
Can the default voicemail file path be configured to another location?
Any hint on where to look would be appreciated.
Thanks,
JB
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Hi,
This is due to missing doPost() in my servlet.
It is fixed now. Please ignore.
Thanks,
JB
On Wed, Aug 27, 2008 at 5:03 PM, Juan Backson [EMAIL PROTECTED] wrote:
Hi,
I am trying to get Freeswitch to post CDR to HTTP using xml_cdr_mod.
Here is the error in console:
2008-08-28 01:13:11
you seem to want.
On Mon, Aug 25, 2008 at 1:57 AM, Juan Backson [EMAIL PROTECTED]wrote:
Hi Brian,
If I try
e = freeswitch.Event(message);
e:addBody(thisisatestevent,abcd)
session:sendEvent(e)
2008-08-25 23:09:30 [ERR] mod_lua.cpp:176 lua_parse_and_execute() Error
Hi,
I tried to fire custom event with the following example:
e = freeswitch.Event(message);
e:add_body(mymsg,abcd)
session:sendEvent(e)
But I am getting this error from freeswitch console:
2008-08-25 05:49:33 [ERR] mod_lua.cpp:176 lua_parse_and_execute()
Hi,
Is there anyway that I can tell if one of either leg A or leg B is a client
behind NAT? I would like to find a way to do that in order to dynamically
determine whether bypass_media or proxy_media should be used.
Thanks,
JB
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Hi,
We are running some stress tests again freeswitch using sipp.
At 100 concurrent call, Freeswitch starts to throw out some strange error
messages:
[EMAIL PROTECTED] 2008-07-21 06:27:51 [CRIT] switch_core_session.c:826
switch_core_session$
2008-07-21 06:27:51 [CRIT] sofia.c:3177
in
conjunction with OpenZAP. Do you have any reason to not use your phones
three way feature? We do support that without any modification to your
dialplan. I'll dig up the commit message on that app and wiki the usage
example.
/b
On Jul 4, 2008, at 10:45 AM, Juan Backson wrote:
Hello Brian
Hello,
The wiki page on threewaycall -
http://wiki.freeswitch.org/index.php?title=Misc._Dialplan_Tools_threewayaction=edit
is empty. Does anyone know how to use this function call? Any input will
be great.
Regars,
Juan
On Fri, Jul 4, 2008 at 2:03 AM, Juan Backson [EMAIL PROTECTED] wrote
Hello Brian,
I am using SIP.
JB
On Fri, Jul 4, 2008 at 11:32 PM, Brian West [EMAIL PROTECTED] wrote:
Are you going to be using SIP or TDM?
/b
On Jul 4, 2008, at 6:24 AM, Juan Backson wrote:
Hello,
The wiki page on threewaycall -
http://wiki.freeswitch.org/index.php?title=Misc
Hello
I am new to Freeswitch. Can someone show me an example on implementing a
3-way calling feature within the dialplan?
Regards,
Juan
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