* Hi,*
*
*
*I want to check value given to the javascript with conditions whether it is
voicefile, extension or mobile Number when i press the dtmf value.*
*
*
*Steps i need to check in javascript:*
*
*
*When i Press the DTMF value 1 it should check the 3 condition*
*
*
If the Value for
*Hi Rupa,
I get core dump segmentation fault in freeswitch machine frequently. can
any one assist me what is error in the freeswitch. i have pasted the logs in
freeswitch pastebin.
This is the link
http://pastebin.freeswitch.org/9854http://pastebin.freeswitch.org/9851
Can some one assist me
*Hi,
Problem has been resolved.
**Thanks for reply from Meftah Tayeb,** Michael Jerris, Rupa Schomaker and
freeswitch-users.
--
Thanks with Regards,
N.Baskar
*
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FreeSWITCH-users@lists.freeswitch.org
.
Crash Protection [Disabled]
Max Sessions[1000]
Session Rate[30]
SQL [Enabled]
freeswi...@baskar 2009-07-20 10:55:05.812500 [ERR] sofia.c:801 Error
Creating SIP UA for profile: internal-ipv6
One more question in windows whether it is possible to connect the ODBC
connection through JavaScript
*Hi Meftah Tayeb**,*
*One more question in windows whether it is possible to connect the ODBC
connection through JavaScript in freeswitch.
I have configured inbound in Linux it is working fine but same script i
tried in windows but i get this error. I have installed and configured MYSQL
*Hi Michael Jerris,
In linux if ODBC modules to be load in freeswitch i can load by this command
make mod_spidermonkey_odbc-install
make install
But in windows how can i enable the modules for mod_spidermonkey?
I checked whether mod spidermonkey is loaded by this command load
mod_spidermonkey
*Hi,
i have changed the openzap.conf file but still i get the same error
**[span wanpipe 1]
number = 1
trunk_type = e1
b-channel = 1:1-15
d-channel = 1:16
b-channel = 1:17-31*
*
freeswi...@localhost.localdomain load mod_libpri
API CALL [load(mod_libpri)] output:
-ERR [module load file routine
*Hi,
I have configured outbound call through JavaScript it is working fine but i
want the conversation to be recorded .
Javascript:
sessionA = new Session({ignore_early_media=true,
origination_uuid=+argv[0]+}sofia/default/sip:+argv[0]+@
192.168.1.135:5066);
sessionB = new
*Hi Michael Jerris,
Is there any other possible way to queue the inbound call in JavaScript.
I am working on this process:
step 1: I want the inbound call to be in queue through JavaScript
step2: Then JavaScript will check most waiting agent and bridge the call to
the most waiting agent ( this
*Hi,
**I have installed the latest trunk with A102D Sangoma card when i load the
openzap i get this error
**
freeswi...@localhost.localdomain load mod_openzap
2009-06-30 14:47:33 [NOTICE] zap_io.c:2626 zap_global_init() Modules
configured: 1
2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config()
*Hi,
I have configured inbound through JavaScript it is working well.
Through dialplan i have configured FIFO it is also working fine but i want
to configure FIFO (First In First Out) through JavaScript.
Is there any link or examples to configure FIFO through JavaScript which
will assist me to
*Hi,
I have some questions can any one assist me and answer this query
Question 1: Can we able to execute all the api command through JavaScript
using session.execute.
Question 2: How to kill the session using uuid_kill whether it is possible.
**If yes means how we will use uuid_kill in
Hi,
I have configured inbound in FS SVN Trunk. i have written small program
for inbound call to bridge. i have used session fifo
session.execute( fifo, sales_fifo_1 out wait undef
'/usr/local/freeswitch/sounds/en/us/callie/time/8000/tomorrow.wav' );
session.execute(bridge,
*Hi,
I have an issue in transfer the call through JavaScript session.
In JavaScript Session i have dialed 2 numbers
One is mobile number and another one is extension number
I want both the call to transfer in the conference room using JavaScript
session
Whether transfer is possible in
*Hi,
Michael Collins
Step1: I get Mobile Number and Extension Number from Database and pass
those value to JavaScript.
Step2: JavaScript will dial the both Mobile and Extension Number. After some
time agent want to transfer the call to conference room.
Step3: Then agent will dial another
Hi,
I have designed a CRM in that caller will login and he will wait for the
call.
when the campaign start it get no from database and dial the number and
bridge to the caller.
*Step1: I get Mobile Number and Extension Number from Database and pass
those value to JavaScript.
*
Who or
Hi,
In JavaScript session i have one question:
Step1: I have written one small JavaScript program first dial the one mobile
number and one extension example: 98417988741001
Step2: In that same JavaScript itself i want to transfer both the mobile
number and extension into conference room
Hi,
Karl Vesterling Thanks for the reply But i need the above process to be done
through javascript session.
Can some one assist me to solve this problem.
Thanks in advance.
--
Warm Regards,
N.Baskar
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Freeswitch-users mailing list
Hi,
Raul Fragoso:*I can to get that through Mysql DB but i want to know whether
there is any **api command to check the status of the extension(1000).*
*If the agent register 1000 in xlite.
The registered users can be viewed by api command api sofia status profile
default.
After that if i want
*Hi,
I can able to play the voice file. But I need to play the voice file thrice
**or else in a particular period of time using session.streamfile in
JavaScript .
Please assist me for the above process.
--
Warm Regards,
N.Baskar
*
___
Hi,
There is any api command to check the status of the extension whether the
agent is in ideal or in calling .
Can any one asset me to solve the problem. Thanks in advance
--
Warm Regards,
N.Baskar
___
Freeswitch-users mailing list
*Hi,
Michael Collins: I will explain in Detail
If the agent register 1000 in xlite.
The registered users can be viewed by api command api sofia status profile
default.
After that if i want to view the status of that extension 1000 whether he is
in calling are in ideal.
*
*There is any api
*Hi,
Now i can able to load the mod_java in the freeswitch console.
After that i have followed these method to run the PhoneTest.java
*
*1) verified my classpath in the java.conf.xml: option
value=-Djava.class.path=/usr/local/freeswitch/scripts/freeswitch.jar/*
*2)my PhoneTest.class is
*Hi Brian West,*
* I have installed the latest SVN Freeswitch trunk but still i get the same
error. How can i over come this problem.
2009-04-14 12:44:26 [NOTICE] modjava.c:244 mod_java_load() Java Framework
Loading...
2009-04-14 12:44:26 [ERR] modjava.c:133 load_config() Error loading
*Hi,
I have installed latest java version jdk1.6.0 in this path
/usr/java/jdk1.6.0_04/bin
I have reconfigured FS ./configure --with-java=/usr/java/jdk1.6.0_04/bin,
make, make install
But when i run freeswitch in console i get this error.
2009-04-14 15:00:22 [ERR] modjava.c:133
Hi,
My Java.conf.xml
configuration name=java.conf description=Java Plug-Ins
!-- Path to the Java 1.6 virtual machine to use --
javavm path=/usr/java/jdk1.6.0/jre/lib/i386/client/libjvm.so/
!-- Options to pass to Java --
options
!-- Your class path (make sure freeswitch.jar is on
*Hi,
I have not edited the java.conf.xml
*
*my libjvm.so file is loacted in this paths*
* [localhost ~]# locate libjvm.so
/usr/java/jdk1.6.0_04/jre/lib/i386/client/libjvm.so
/usr/java/jdk1.6.0_04/jre/lib/i386/server/libjvm.so
/usr/lib/gcj-4.1.1/libjvm.so
Hi,
I have loaded the java module in freeswitch. But when i run freeswitch
in console i get this error.
2009-04-13 17:38:33 [NOTICE] modjava.c:244 mod_java_load() Java Framework
Loading...
2009-04-13 17:38:33 [ERR] modjava.c:133 load_config() Error loading
*Hi,
I have seen the above mail. In that all of you tried to created dynamic
conference through diaplan itself using the database to insert the uuid,
caller_id_number, destination_number, etc .Can one guide me set the dynamic
conference and Schema for the dynamic conference.
I have tried the
*Hi,
I am using TDMAPI with EuroISDN.
*
*My openzap.conf*
*[span wanpipe]
name = OpenZAP
number = 1
trunk_type = E1
b-channel = 1:1-15
d-channel = 1:16
b-channel = 1:17-31
[span wanpipe]
name = OpenZAP
number = 2
trunk_type = E1
b-channel = 2:1-15
d-channel = 2:16
b-channel = 2:17-31
*
*My
Hi,
I am using sangoma A102 card with freeswitch. I have updated all the
changes in the freeswitch and have loaded openzap also. But still i cant
able to make an outbound call.
*openzap.conf*
[span wanpipe]
name = OpenZAP
number = 1
trunk_type = E1
b-channel = 1:1-15
d-channel = 1:16
*Hi Michael Collins,
When i load mod_openzap i can able to get this output **Successfully
Loaded [mod_openzap**] with one error message
2009-03-13 10:21:42 [ERR] mod_openzap.c:1898 load_config() Error starting
OpenZAP span 1 mode: -1264601207 dialect: -1264601162 error:
I have pasted full
Hi,
I am using javascript to store uuid, phone_no, endpoint_disposition and
hangup cause in my MYSQL. I can get the session UUID , Phone_no, endpoint
disposition but i cant get the originate disposition.
Javascript :
session.setVariable(session.uuid, ses_uuid: + session.uuid);
Hi,
One thing i forget to tell i can able to get this Disconnection
causeand Disconnection
code in Freeswitch console but when i set in the variable i did not get the
cause or cause code in javascript.
I use these line in Javascript i get the output in the Freeswitch console.
Hi Anthony Minessale,
I have added these lines in my javascript with your *guidance. *But still i
did not get any status like busy , no answer, etc .
session.setVariable(cause_code, session.causecode);
session.setVariable(cause_name, session.cause);
I Get this output only for all the call:
*Hi,
I want to get the events for the tone detect but i cant able to get any
events,
Procedures i follow to done detect:*
Step 1:I have added the line in default.xml
extension name=Local_Extension
condition field=destination_number
expression=^(1[0-9][0-9][0-9])$
action
*Hi Michael Jerris,*
*1. When we call a busy number wont it detect busy signal?*
*2. If we get the SIP response code for busy as 486 can we detect the tone?*
*3. And more over in the client side only we are using softphone, the other
end is connected with E1 in the audiocode, so since it seems
Hi,
Conference is working well in recent days, but suddenly it in not working
when i transfer the call ,the call get disconnected with the voice file
playing Bye...
What is the error?
When i transfer the call why the call get hangup
api originate sofia/internal/1003%172.20.191.227
*Hi **Michael,*
Step i follow for the Tone Detect process
Thanks for the Reply It is useful for me
On Sat, Dec 27, 2008 at 11:14 PM, Michael Jerris m...@jerris.com wrote:
On Dec 27, 2008, at 5:56 AM, Baskar wrote:
*Hi **Michael,*
*
I try to detect the tone
*Hi **Michael,*
*Steps I follow for the Tone Detect process*
*
**Step1: **From X-lite i called my no (eg: 1007==9841799874 )
**Step2: Then i run the JavaScript in that also i have given same no
(9841799874)
* *Step3: While i run the JavaScript i should get the busy tone detect but i
cant ???*
Hi Michael Jerris,
I will explain what i am currently doing :I don't understand
Step 1: From the xlite phone I have dialed a number and we were on the
conversation with one extension (1007 is my extension and my mobile No
9841799874)
Step 2: From the freeswitch console I am executing a
*Hi **Michael,*
*
I try to detect the tone before answering the call.
Is there any module for tone detect to be enabled*
* I have set ignore_early_media=False **(False is case sensitive?)*
* But still no Tone is Detected.*
*--
Warm Regards,
N.Baskar
*
*Hi Michael,
*
* I have updated all the changes what u said, But still i did not get
any tone detect in the script *
*session1 = new Session();
session1.originate(session1, {ignore_early_media=True}sofia/default/
39841799...@172.20.191.228);
session1.execute(tone_detect, test 400 r +30 hangup
*Hi,
This is my JavaScript for tone detect
session1 = new Session();
session1.originate(session1, {ignore_early_media=false}sofia/default/
39841799...@172.20.191.228);
session1.execute(tone_detect, test 400,25 r +1 hangup 'normal_clearing'
1);
session1.execute(bridge,
*Hi,
I am using JavaScript file to detect busy tone signals but I cant able to
detect the busy tone signals
*
*My JavaScript*
*
session1 = new Session();
session1.originate(session1, {ignore_early_media=true}sofia/default/
39841799...@172.20.191.228);
session1.execute(tone_detect, busy 400 r);
Hi,
*I have newly installed freeswitch in another machine.
**After starting the freeswitch I try to dial a extension from console but
when i call extension 1002 from freeswitch console, call is connected to
extension 1002, but FS is aborted but call is established in1002.*
*When i dial from
Hi,
*
**It is possible to dial outbound through console dialing. Yes means me
How ?**
Without using the softphone how can i dial outbound from freeswitch
console itself. *
*
I want to Know without using any softphone for calling.
It is possible in asterisk. we can dial from console
*Hi Giovanni Maruzzelli*,
To list the available devices i have given this command *pa devlist*
*output:*
[EMAIL PROTECTED] pa devlist
2008-12-02 11:27:34 [CONSOLE] switch_console.c:255 switch_console_process()
Unknown Command: pa
But when i check in my system *hwconf *there is auido drives
*Hi,
I have updated all the above events you told .It's working fine but when i
call extension 1002 from freeswitch console, call is connected to extension
1002, but FS is aborted but call is established in1002. what shall i do.
what was the error.
Full freeswitch get cut.*
*output:*
[EMAIL
*Hi,*
*It is possible to dial outbound through console dialing. Yes means me
How?*
*Another question whether there is any api command for console dialing.*
*
Thank you with regards,
N.Baskar
*
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Hi,
Thanks for the support from *Brian West, Michael S Collins,Birgit Arkesteijn,
Cesar Cepeda, Michael Jerris, Gopala krishnan*.
DTMF is working fine in barging and Conference.
--
Warm Regards,
N.Baskar
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Hi cesar,
If i have added these line in mod_commands.c
stream-write_function(stream,+OK\n); just after inserting the DTMF
before the goto done;
When i compile by command *make* it get these error
*Compiling mod_commands.c...
mod_commands.c: In function âunload_functionâ:
mod_commands.c:869:
Hi,
We need the above process to be done on the event socket through API
commands.
Thanks
Baskar.N
On Wed, Nov 19, 2008 at 4:18 PM, Rupa Schomaker (lists)
[EMAIL PROTECTED] wrote:
On 11/19/2008 4:31 AM, Baskar wrote:
Hi,
I want to have the following scenario to be done
Hi,
I want to have the following scenario to be done on the conference room.
1. I am calling the 1st person and talking with him.
2. When I require, I need to hold the 1st person and dial out 2nd person.
(till such time 1st person to be on hold by hearing hold music).
3. I need to unhold the
Hi,
I want to pass the DTMF digits through api command
i find the api command *api uuid_send_dtmf* *uuid* dtmf_data
I just want to know what is dtmf_data what is the value to pass in that
parameter
Thanks in advance
--
Warm Regards,
N.Baskar
___
Hi,
i have tried it before itself first i pass one digit
api uuid_send_dtmf c08f77be-fbed-44c3-a2a7-8650d88b0e33 *2 *
*
output:*
Content-Type: api/response
Content-Length: 14
-ERR no reply
Then i passed all the values in the barging
api uuid_send_dtmf baf82956-111d-4cd8-9568-47010ac8bd20
Hi Brain,
I am working on DTMF signals during *eavesdrop* and in* CONFERENCE *
DTMF signal is *not working* through* even socket api command *
I tried in conference also when we manually done in softphone it work . when
i press the # button it hangup and * for mute etc. it works fine but when
Hi,
In barging if we want to pass the DTMF signals. For example in barging
- 2 to speak with the uuid
- 1 to speak with the other half
- 3 to engage a three way
- 0 to restore eavesdrop.
- * to next channel.
I want pass these DTMF signals through event socket api uuid_send_dtmf
Hi,
*Just i want to call the 1002==9841799874 and i want to call another no
9840544078 and add in to the conference room *
*First method i tried :*
api originate sofia/internal/1002%172.20.191.227 bridge(sofia/default/
[EMAIL PROTECTED]) 1002 call 9841799874
api uuid_transfer
, at 9:59 PM, Baskar [EMAIL PROTECTED] wrote:
Hi,
*
Recoding is done through default.xml.
For past 1 month recoding is working fine.
Suddenly for past 3day recording is not working.
I did not modify any thing in deafult .xml .
*
* My deafult.xml file*
?xml
Hi,
When i load the mod_spidermonkey command i get all mod_spidermonkey loaded
successfully
[EMAIL PROTECTED]* load mod_spidermonkey*
2008-10-20 12:29:46 [CONSOLE] mod_spidermonkey.c:944 sm_load_file()
Successfully Loaded [/usr/local/freeswitch/mod/mod_spidermonkey_teletone.so]
2008-10-20
Hi,
I have done all the changes but i cant able to run jsrun odbc.js
[EMAIL PROTECTED] 2008-10-20 19:34:29 [ERR]
switch_odbc.c:160 switch_odbc_handle_connect() STATE: 01000 CODE 0 ERROR:
[unixODBC][Driver Manager]Can't open lib '/usr/local/lib/libmyodbc3.so' :
/usr/local/lib/libmyodbc3.so: file
Hi,
I have read the above mail list but i cant able to clear this ODBC error.
while installing the unixodbc i deed not get any error but when i load the
mod_spidermonkey_obdc i get these errors
[EMAIL PROTECTED] load mod_spidermonkey_odbc
2008-10-17 14:18:00 [CRIT] switch_loadable_module.c:767
, at 1:47 AM, Baskar wrote:
Hi,
I have read the above mail list but i cant able to clear this ODBC error.
while installing the unixodbc i deed not get any error but when i load the
mod_spidermonkey_obdc i get these errors
[EMAIL PROTECTED] load mod_spidermonkey_odbc
2008-10-17 14:18:00
Hi,
I don't understand.could you please tell me in brief? . I want to check my
channel status whether it is ringing or answer or hangup.
On Mon, Sep 29, 2008 at 11:38 PM, B Karthik [EMAIL PROTECTED] wrote:
Set hangup_after_bridge variable to false and run the info command after
bridge, you
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