Hi folks,
i try javascripts example for conference application and get this error,
2008-08-28 20:59:55 [ERR] mod_spidermonkey.c:3303 js_api_use() Error
loading ODBC
2008-08-28 20:59:55 [ERR] conference.js:10 mod_spidermonkey()
ReferenceError: ODBC is not defined
i have follow the instruction
First you have to add
use(ODBC);
in your javascript.
On Thu, Aug 28, 2008 at 9:06 AM, mashudi [EMAIL PROTECTED] wrote:
Hi folks,
i try javascripts example for conference application and get this error,
2008-08-28 20:59:55 [ERR] mod_spidermonkey.c:3303 js_api_use() Error
loading ODBC
Hi everyone,
I would like to join the choir of Tom Warren: HA with state
synchronisation would not just be a killer, it will pave the road for
the FreeSWITH all the way to the entrance of the Telco's.
As a former co-owner of a hosted PBX Telco, I would like to share my
experience about
Hello :
http://wiki.freeswitch.org/wiki/Enterprise_deployment_UltraMonkey
I don't really test it .
I do have some experience like yours , using openser plus some asterisks,
but the openser dispatcher module even I use hash over Call-Id ,
I still get some fail experience , e.g. un-hold after
Hello :
Is it possible to directly mix 3 way voice ?
Not putting eny leg into holding music.
I found the script originate a session is quite different than an
agent call. Some channel attributes missing .
# calltest.js
n_sess = new Session() ;
res = n_sess.originate(n_sess,
hi,
actually to try out freeswitch to the fullest, i need to get it running
using TLS and that also on a windows server as my isp blocks plain SIP and
my dedicated server is windows 2003. the problem is i cant get TLS running
on windows using freeswitch and what i was told and what i read was
Hi all,
My concern is to terminate a call using a specific timeout value. I have
referred to the docs, what I found is, *originate_timeout*, *
progress_timeout*, and *call_timeout*. Do we have something like *
talk_timeout*? or anything for similar purpose?
Thanks.
--
Best,
Adeel Ansari
Hi All,
I have created a IVR menu where in the digits is ended with #,
could anyone point me why a menu with ending digits (e.g. 12345#)
doesn't work?
entry action=menu-exec-app digits=/^(\d{5})#$/
param=transfer $1 XML default/
Does Freeswitch IVR doesn't support # in digits?
1.With which application and API is possible to keep full content of RTP
audiostream? Our FreeSWITCH is configured in such manner that RTP-traffic
directing throw FreeSWITCH, but not between UAs.
2. Which audioformat is better to use, if we wish to make format's
transformation using, for
On Aug 28, 2008, at 6:18 AM, Adeel Ansari wrote:
Hi all,
My concern is to terminate a call using a specific timeout value. I
have referred to the docs, what I found is, originate_timeout,
progress_timeout, and call_timeout. Do we have something like
talk_timeout? or anything for similar
On Aug 28, 2008, at 5:15 AM, ??? ??? wrote:
1.With which application and API is possible to keep full content of
RTP audiostream? Our FreeSWITCH is configured in such manner that
RTP-traffic directing throw FreeSWITCH, but not between UAs.
The default behavior is for the audio to
Show me the whole XML for this. I suspect maybe you don't have the
digit-len set correctly.
/b
On Aug 28, 2008, at 7:21 AM, Sheeju Alex wrote:
Hi All,
I have created a IVR menu where in the digits is ended with #,
could anyone point me why a menu with ending digits (e.g. 12345#)
Here is the one I have setup:
-- /etc/odbc.ini
[freeswitch]
Description = MySQL ODBC Driver Testing
Driver = MySQL
Socket =
Server = localhost
User = myuser
Password = mypass
Database = test
Option =
Port = 3306
Stmt =
and the param in my FS config.
param name=odbc-dsn
Bipin,
I'm working on an openssl build under VS2008 but it's not easy.
If you wish to try compiling sofia with TLS, I suggest you download the
latest openssl build (0.9.8i) and compile it (you'll need Perl to do that).
If done correctly, this should give you the necessary lib files to link with
so do i need to compile sofia or openssl as the openssl u told me, i have the
compiled version directly from the site for windows platform. does it need
to be re compiled or how is it all as i am a newbie in programming and
compiling
UV-5 wrote:
Bipin,
I'm working on an openssl build
if ur using the TLS funstion successfully then can u plz give me those files?
UV-5 wrote:
The compiled version from openssl site doesn't include the lib files (I
have
no idea why). You'll have to compile those yourself. Unfortunately, you
can
only do that using Perl, so it's quite a
You escaped the * in this one.. but in your # one you didn't... You'll
need to do the same for #
/b
On Aug 28, 2008, at 10:17 AM, Sheeju Alex wrote:
entry action=menu-exec-app digits=/^(\d{5})\*$/
param=transfer $1 XML default/
Brian West
sip:[EMAIL PROTECTED]
Just a thought, but # arrives encoded as %23 in events. Is it appearing
as such in the IVR
menu string - it'd explain why the regex match fails.
--Dave
Hi Brian,
Here is the XML I am using in ivr.conf.xml
menu name=conf_ivr
greet-long=phrase:conf_ivr_main_menu
I don't have those files yet
Long story
Will let you know when I do. ;-)
Or even better, when I'll have a working SLN for openssl...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of xbipin
Sent: Friday, August 29, 2008 1:44 AM
To:
That depends on the phone. Yesterday the Snom 7.3.7 firmware was
sending the # at %23 and it shouldn't but not when you're live on the
phone dialing digits on a call thats up.. its RTP out of band.. it
should show up at #
/b
On Aug 28, 2008, at 10:52 AM, David Knell wrote:
Just a
Hi,
I am a newbie in this list and I am finding openser a lot along with
freeswitch and asterisk.
First what is the use of openser with free switch/asterisk, is it required
with free switch if yes in what kind of environment.
It helps in scaling the solution or if natting than its not clear.
I tried escaping #, but it didn't work. According to normal regex
matching escaping # is not required I guess.
So now I have tried both and is not working.
I am using X-Lite, could you please let me know how to check the
whether it is arriving as %23 or something else.
Here is some debug
The biggest gotcha here is you need to have versions compiled with
the same compiler version as the rest of freeswitch or strange things
can result. Maybe we will have to make some pre-compiled versions and
allow them for download so we can cleanly integrate this into the build.
Mike
On
All,
I am attempting to allow a single line (1005) to be called which
simultaneously allows multiple threads (lines 1006 and 1007) to apply
simultaneous IVR.
I would thing such grouping is common.
Using the below it appears that line 1005 will not bridge to 1006 and 1007
since they are in the
Brian, I will be using # as a delimiter to authenticate a PIN. Say
for e.g I will be using to authenticate a card or bank number.
Thanks
SA
On Thu, Aug 28, 2008 at 9:52 PM, Brian West [EMAIL PROTECTED] wrote:
Why are you even bothering with a #? If you lower your digit count by
1 it'll just
They aren't calling via FreeSWITCH I suspect. They'll have to be
registered to FreeSWITCH for any call to originate from FreeSWITCH to
the phone.
/b
On Aug 28, 2008, at 11:52 AM, Robert Clayton wrote:
Yet 1006 and 1007 can be called directly without being registered.
Brian West
Hi,
I can call extensions like 1000 with
extension name=extensions
condition field=destination_number expression=^(\d+)$
action application=bridge data=sofia/doublenat/$1%voipclic.com
/
/condition
/extension
and I can call PSTN with
extension name=PSTN
For extensions you usually have a 4 digit number so you would do
this: ^(\d{4})$ then for the pstn you would not want to match just
all digits like that. ^(\d{7,15})$
/b
On Aug 28, 2008, at 1:01 PM, Cliconnect wrote:
Hi,
I can call extensions like 1000 with
extension
Thank you Brian,
it worked.
regards
Duan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
West
Sent: Thursday, August 28, 2008 11:07 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] dialpaln
For extensions you
I think that UltraMonkey doc was more hypothetical. I would test it
first.
/b
On Aug 28, 2008, at 2:33 AM, Lee JJ wrote:
Hello :
http://wiki.freeswitch.org/wiki/Enterprise_deployment_UltraMonkey
I don't really test it .
I do have some experience like yours , using openser plus some
This is all architecture decisions that really anyone here can only
give you input.. but in the end you'll have to make that decision
yourself.
Do you have to use OpenSER with FreeSWITCH? No!
OpenSER does help with scaling large deployments.
I know a few people that are just deploying
Didn't you say you're doing SIP? To not have it put into hold music
set the variable hold_music=silence and nobody will get music while
setting up the threeway call.
/b
On Aug 28, 2008, at 2:46 AM, Lee JJ wrote:
Hello :
Is it possible to directly mix 3 way voice ?
Not putting eny leg
it is already
use(ODBC);
var DSN = DB;
var DB_USER = conce;
var DB_PASS = password;
var line = \n;
var db = new ODBC(DSN, DB_USER, DB_PASS);
db.connect();
var sql;
var dtmf = new Object();
var replay = 1;
function mycb (session, type, data, arg) {
Hi,
Can the default voicemail file path be configured to another location?
Any hint on where to look would be appreciated.
Thanks,
JB
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
If you are running the latest code as of today you can do storage-
dir param on the voicemail profile to specify the directory.
ie param name=storage-dir value=/tmp/
/b
On Aug 28, 2008, at 9:35 PM, Juan Backson wrote:
Hi,
Can the default voicemail file path be configured to another
The commands (2 separate ones) are more like:
originate sofia/gateway/gizmo1/6098989898
bridge(sofia/gateway/gizmo9/0116054545454)
sched_hangup +120 uuid alotted_timeout
Where uuid is the channel UUID being assigned to that originate request
_
From: [EMAIL PROTECTED]
I have already tried it. This way its working, actually. :)
Anyways, can you please tell how meet the dial plan using original command.
I mean,
*originate sofia/gateway/gizmo1/6098989898
bridge(sofia/gateway/gizmo9/0116054545454) *
I am trying quite a few but not meeting the criteria, I suppose.
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