Hello,
hm kind of unclear Question. So I'm looking for a way to get the
affected number of rows after executing a delete statement via ODBC.
There is a function called SQLRowCount(), but I didn't found a
switch_odbc_* function in FS which allows me to call it.
On 18.11.2009 19:21, Helmut
Hello,
hm kind of unclear Question. So I'm looking for a way to get the
affected number of rows after executing a delete statement via ODBC.
There is a function called SQLRowCount(), but I didn't found a
switch_odbc_* function in FS which allows me to call it.
On 18.11.2009 19:21, Helmut
Samuel Mukoti samuelmuk...@gmail.com wrote:
I'm a new freeswitch user and am wondering what people do when setting
options in the freeswitch config files. Do people use special tools,
XML editors etc or is it just vi/emacs/Kate?
Emacs has an XML editing mode; Vim may have extensions for
I believe OBDC is the official way..
however id love look at doing this in a higher performance way, without the
single point of failure..
local memcache, in front of OBDC or something ??
not 100% sure of it, but just using a single central database is a little
bit of a concern in a carrier
If we could access mod_memcache for registration information that
would be ideal and highly robust, since you can share memcache with
external applications.
Lon
On Thu, Nov 19, 2009 at 2:07 AM, Leon de Rooij l...@scarlet-internet.nl wrote:
Well, you can of course easily have a loadbalancer with
Hi,
Can someone please help me understand a little more about Group configuration ?
I believe that Group Membership is configured in the
\conf\directory\default.xml file
I've done this and the caller groups seem to work fine.
However, each extension in the \conf\directory\default directory,
I'd have to double check all the sql used for registration, but I
doubt memcache is expressive enough to act as the registration store.
For instance, you can't get a list of registrations from it (sofia
status profile internal).
memcache is a keystore only.
That being said, one could use
Using lcr_auto_route + limit isn't really possible at this point. It
is on the list of things to do but is more complex than it seems on
it's surface.
mod_lcr just constructs dial strings, it doesn't do any call control.
It does provide enough information to do what you want via a scripting
Thx Jason for the reply,
I realise i was quite unclear in what i'm hoping to achieve. I wanted to
make a control panel for our office so that we can provision extensions at
the same time as we do users. We have a system much like the ubuntu ebox
that allows use to manage users for our
Hi Sam,
Take a look at mod_xml_curl. Pretty sure it'll do everything you're
looking for.
http://wiki.freeswitch.org/wiki/Mod_xml_curl
Also, I would browse the modules and look for other nifty
functionality that already exists before setting out to write
something new.
I have installed FreeSWITCH on our server, and need some help configuring our
FreeSWITCH instance. All of the numbers associated with our FreeSWITCH instance
are in the format: 1NPANXX (where NPA is the area code, and NXX are the
last 7 digits of the phone number).
I need
Hi Brian,
It just doesn't belong in user space or kernel space in the machine
for true performance you should do it in hardware... I'm pretty sure
the poor box would die if you tried it on 32 E1's at the same time.
Disagree somewhat. The challenge that echo cancellers further from the
If you are using Windows XP (or Vista for that matter), you may want to
look at tftpd32. Its more compact and uses less memory than Solarwinds
yet provides not only a tftp server but a dhcp and syslog server as well.
In the past, I've use it to upgrade, install, and troubleshoot a variety
of
Like I said,
The timer by default is designed to make sure that none of the audio is lost
for situations like FAX etc.
There are parameters you can configure to disable the timers that I
mentioned in the last email which will cause all of the audio to be jammed
into your ear like twiddlebugs if
On 11/19/2009 11:54 PM, David Knell wrote:
Hi Brian,
It just doesn't belong in user space or kernel space in the machine
for true performance you should do it in hardware... I'm pretty sure
the poor box would die if you tried it on 32 E1's at the same time.
Disagree somewhat.
Metik,
thanks a lot for the tip, I will certainly look at it, particularly if it
does DHCP too.
At the moment, I use my ADSL Router to provide DHCP to the network but I've
just discovered that you can't configure options in its DHCP server to point
to the TFTP server for the phone. At the
Some Cisco phones need DHCP option 150.
/b
On Nov 19, 2009, at 10:46 AM, Dave Stevenson wrote:
Metik,
thanks a lot for the tip, I will certainly look at it, particularly
if it
does DHCP too.
At the moment, I use my ADSL Router to provide DHCP to the network
but I've
just discovered
Yeah, roger that...
Here is an excerpt from the page I did on the Cisco 7960G HowTo:
http://wiki.freeswitch.org/wiki/Freeswitch_Cisco_7960G_Howto
It's for Linux, but you'll get some good pointers on the TFTP option you're
looking for.
I haven't provisioned any 7540's... Good luck!
Best
I don't think a 7540 exists.
/b
On Nov 19, 2009, at 12:11 PM, Karl J. Vesterling wrote:
I haven't provisioned any 7540's... Good luck!
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Thanks Guys,
I had not realised until the last couple of days that DHCP did more than
just providing the IP address to the client. I have been happily just doing
that for a few years now without anything other than my Router providing the
DHCP function. It's only now that I have taken the
I had almost the same problem- it was needed to record everything, even if
the record path doesn't exist - it was requested to create the needed path.
For this purpose I've used event_socket command api system ..., precisely,
api system mkdir -p path
And after this command I've started recording.
Hi,
i'm trying to configure freeswitch with a VOIP provider, exsorsa, that
uses OpenSER. Exsorsa use as own gateway, another provider, Eutelia,
that it uses Cisco (or, at least, this appears in headers).
Short story:
If i try to setup my Eutelia account all works perfectly while
I'm going to guess gw+exsorsa is what they don't like. try extensions-
in-contact=true on the gateway config.
/b
On Nov 19, 2009, at 1:29 PM, Albano Daniele Salvatore - Lavoro wrote:
Hi,
i'm trying to configure freeswitch with a VOIP provider, exsorsa,
that uses OpenSER. Exsorsa use as
Hi there!
I've got annoying FS behavior:
There are 2 channels executing the same Java application (application itself
is an IVR). If I try to bridge them with uuid_bridged then both channels are
killed. Here is a log from FS console:
uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2
Hello,
I just pasted a log in the Pastebin with Freeswitch logging enabled. Does
anyone know a way to prevent FS from connecting the call prior to the callee
answering?
Best Regards,
Jerry
-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Thursday,
set enable-3pcc to proxy instead of true
On Thu, Nov 19, 2009 at 2:00 PM, Jerry Richards
jerry.richa...@teotech.comwrote:
Hello,
I just pasted a log in the Pastebin with Freeswitch logging enabled. Does
anyone know a way to prevent FS from connecting the call prior to the
callee
On Thu, Nov 19, 2009 at 3:33 AM, Dave Stevenson
steve...@primrosebank.netwrote:
Hi,
Can someone please help me understand a little more about Group
configuration ?
I believe that Group Membership is configured in the
\conf\directory\default.xml file
I've done this and the caller groups
On Thu, Nov 19, 2009 at 12:18 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
set enable-3pcc to proxy instead of true
FYI, the wiki entry is here:
http://wiki.freeswitch.org/wiki/Sofia.conf.xml#enable-3pcc
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On Thu, Nov 19, 2009 at 11:46 AM, Artem Shiyanov shiya...@gmail.com wrote:
Hi there!
I've got annoying FS behavior:
There are 2 channels executing the same Java application (application
itself is an IVR). If I try to bridge them with uuid_bridged then both
channels are killed. Here is a log
I don't see any hangups here, are you talking about the BREAK signals?
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 19-Nov-09, at 11:46 AM, Artem Shiyanov wrote:
Hi there!
I've got annoying FS behavior:
There are 2
Thanks Michael,
I think I've got it !
regards
Dave
- Original Message -
From: Michael Collins
To: freeswitch-users@lists.freeswitch.org
Sent: Thursday, November 19, 2009 8:25 PM
Subject: Re: [Freeswitch-users] Extension Configuration - XML File Entriesfor
Group
Hi again !
I have FreeSwitch configured such that if someone dials in from the PSTN line,
a group of phones ring.
If nobody answers, the group extension number (100) picks up the call and voice
mail kicks in.
So far, so good, each of the individual phones logs a missed call and anyone in
the
We have removed the two modules using the reference code from
BroadVoice and added a lib with a new interface from Steve Underwood
and mod_bv.c using this lib... We know their is ONE last bug to be
fixed in the lib before its working so please do not open any jira's
if you try to run it
On Wed, Nov 18, 2009 at 6:54 PM, John Platts john_pla...@hotmail.comwrote:
I have installed FreeSWITCH on our server, and need some help configuring
our FreeSWITCH instance. All of the numbers associated with our FreeSWITCH
instance are in the format: 1NPANXX (where NPA is the area code,
On Thu, Nov 19, 2009 at 12:49 PM, Brian West br...@freeswitch.org wrote:
We have removed the two modules using the reference code from
BroadVoice and added a lib with a new interface from Steve Underwood
and mod_bv.c using this lib... We know their is ONE last bug to be
fixed in the lib
Anthony Minessale wrote:
Like I said,
The timer by default is designed to make sure that none of the audio is
lost for situations like FAX etc.
Right, that makes sense. I've updated the wiki entries I made to warn
about this.
We do not use sleep for the timers we have timer objects into
On Fri, 2009-11-20 at 00:15 +0800, Steve Underwood wrote:
The audio path between kernel and user space is not stable with any
current PC based telephony system. At some point in the day the odd
chunk of data is lost here and there, whether you use asterisk,
callweaver, yate or FS, with
Hi everyone,
I want to verify what the wiki says, you can use a radius server as
the data source for your registrations?
Lon
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Is there any way that each extension in the group can be notified that a
group Voice Mail is waiting to be picked up so that each phone shows the
message waiting indication ?
Wouldn't this be simply accomplished by setting the vicemail as box 100 for
each of the users (such as ext 1011xx)?
Thanks Joseph,
that would be one way, but it would mean that everyone had a common mailbox
for all calls, I just wanted to do it for calls coming in on the PSTN line.
Maybe that's not possible though ?
regards
Dave
- Original Message -
From: Joseph L. Casale jcas...@activenetwerx.com
He should be able to just use Additional Option to add option 150 (and
the associated IP address to which the TFTP server is bound).
Brian West wrote:
Some Cisco phones need DHCP option 150.
/b
On Nov 19, 2009, at 10:46 AM, Dave Stevenson wrote:
Metik,
thanks a lot for the tip, I
Hi,
one of my customers is willing to contribute for t38 integration.
The basic idea is to connect HylaFAX to FS:
t38modem - FreeSWITCH - Media Gateway with t38 support
All this without media proxy.
Another idea might be to implement t38 origination/termination with a class 1
modem
I have upgraded FreeSWITCH several times recently for testing purposes. Also,
my router's configuration has changed slightly as I have moved from tunneled
IPv6 to a new native IPv6-over-ADSL trial.
However, the problem now is related to my ISP's IPv4-only SIP service, and the
symptoms are as
I think the fix for this is coming to an SVN repo near you... so give
it a few and update.
/b
On Nov 19, 2009, at 7:15 PM, Jason White wrote:
I have upgraded FreeSWITCH several times recently for testing
purposes. Also,
my router's configuration has changed slightly as I have moved from
On 11/20/2009 05:15 AM, David Knell wrote:
On Fri, 2009-11-20 at 00:15 +0800, Steve Underwood wrote:
The audio path between kernel and user space is not stable with any
current PC based telephony system. At some point in the day the odd
chunk of data is lost here and there, whether you
It now works.. update and have fun!
/b
On Nov 19, 2009, at 3:01 PM, Michael Collins wrote:
On Thu, Nov 19, 2009 at 12:49 PM, Brian West br...@freeswitch.org
wrote:
We have removed the two modules using the reference code from
BroadVoice and added a lib with a new interface from Steve
Thanks for your answers Rob and Shelby. I found more info on apt-get and
ran it against all the missing dependences noted. I also ran through the
sequence of commands Shelby suggested. In the end, running dpkg
-checkbuilddeps I got the following in return
dpkg-checkbuilddeps: Unnet
Cool, I will explore that option when I have some time.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rob Forman
Sent: Wednesday, November 18, 2009 11:02 PM
To:
I'm using ekiga with mod_fsv, trying to record and play back video. When
I dial the record extension, it seems to record something, as the video
file gets bigger. Trying then to dial the extension for play back, just
hangs up, with freeswitch saying:
od_fsv.c:247 File version does not match!
Your annoying behaviour is the exact behavior you should be getting
considering what you told FS to do.
As soon as you call uuid_bridge you are transferring both legs of the call
to bridge to each other.
This means your java app must exit so the channels can connect to each
other.
remember that
Ujjval Karihaloo ujj...@simplesignal.com wrote:
Getting error below..not sure whats wrong..which line number in what file
does this refer to?
freeswitch/log/freeswitch.xml.fsxml
This will be due to a syntax error somewhere in your configuration.
Ubuntu has pretty good package management with apt-get and should work
well for a beginner. The recommended OS (though FreeSWITCH runs on a
wide variety of platforms) is 64-bit CentOS. You can get it here: http://www.centos.org/
if you'd like, but at this point I think it's fine to just
I really didn't change anything.
I was running 1.0.4 and now built from SVN...I see the oddly placed entry in
ivr.conf.xml
sscode_invalid.wav/configuration...removed it now its error on somewhere here:
configuration name=xml_rpc.conf description=XML RPC
settings
!-- The port where you
Does svn update try to merge the config files..Need some help, I think it has
added some entries in my config files that is causing tag mismatches..
Please advise how to get back my orig config?
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
Nothing will replace your configs automatically in the whole build
system.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 19-Nov-09, at 9:22 PM, Ujjval Karihaloo wrote:
Does svn update try to merge the config
check out sofia_contact function. If you use this in combination with binding
profiles together so they are one table I think this should work right.
Mike
On Nov 18, 2009, at 12:36 AM, Eli Hayun wrote:
Brian West wrote:
Why do you need to know the destination profile like that? You get
Ujjval Karihaloo ujj...@simplesignal.com wrote:
Does svn update try to merge the config files..Need some help, I think it
has added some entries in my config files that is causing tag mismatches..
Building and installing FreeSWITCH won't interfere with your configuration.
I would suggest using
On Thu, Nov 19, Joseph L. Casale wrote:
Is there any way that each extension in the group can be notified that a
group Voice Mail is waiting to be picked up so that each phone shows the
message waiting indication ?
Wouldn't this be simply accomplished by setting the vicemail as box 100 for
Brian West br...@freeswitch.org wrote:
I think the fix for this is coming to an SVN repo near you... so give
it a few and update.
Thanks Brian!
I'll watch the svn logs and update when the fix lands.
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Update and see if the problem is gone.
/b
On Nov 20, 2009, at 1:01 AM, Jason White wrote:
Thanks Brian!
I'll watch the svn logs and update when the fix lands.
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