Are those in the Tarball?
On Sep 15, 2009, at 11:47 PM, Nandy Dagondon nandy1...@gmail.com
wrote:
it's working now. the problem? it's the configure script itself.
some ^M characters somehow crept into the line containing
ac_config_files. tks for the tip Andrew!
/nandy
On Wed, Sep 16,
Hi,
I'm newbie in FS. I want to know how to process invalid extension in FS?
Because I want to prompt the IVR if invalid extension is dialled.
Kindly advice me.
--
Regards,
Ahmed Munir
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Ahmed Munir ahmedmunir...@gmail.com wrote:
I'm newbie in FS. I want to know how to process invalid extension in FS?
Because I want to prompt the IVR if invalid extension is dialled.
You could write an entry at the end of the dial-plan that matches any
extension and invokes the IVR.
hi mike, i download the tarball file to check the configure script. it's
clean. so, there must be an error during my first download or build. - nandy
On Wed, Sep 16, 2009 at 3:54 PM, Michael Jerris m...@jerris.com wrote:
Are those in the Tarball?
On Sep 15, 2009, at 11:47 PM, Nandy Dagondon
Hi,
Didn't see this one come through before when I posted it, so sending it
again - apologies if it did come through.
I've moved this discussion to users as it seems my query is moving in
that direction :)
So, upon looking at limit_hash, it appears to do what I need to do.
My question then
Hi,
Currently, the invite message looks as follows
INVITE sip:1...@client_ip:5060 SIP/2.0
Via: SIP/2.0/UDP SERVER_IP;rport;branch=z9hG4bKgvD702De7e0Se
Max-Forwards: 69
From: Extension 1001 sip:1...@server_ip;tag=2rH67Q3aa1rpe
To: sip:1...@client_ip:5060
Is there a way to configure FS so the
Hello
I am trying to set up a shared mailbox per group of extensions where the
mailboxes are [2-9][2-9][0,2,4,6,8]0 and the phones are [2-9][2-9]\d[1-9]. I
mostly have it working except I can't seem to figure out how to get mwi for the
phones in the group. I've tried a number of things. I
On Wed, Sep 16, 2009 at 5:40 AM, Matt Riddell li...@venturevoip.com wrote:
My question then becomes, how do I set a hash for an originated call?
It seems that limit_hash is an application rather than a channel
variable, and so far I've been doing most things without touching the
dialplan.
hi , i m very new to the FreeSwitch..
can any one tell me how to add a new user.
i have already tried creating a new user by creating a
$INSTALL_DIR/conf/directory/default/pankaj.xml :
include
user id=pankaj
params
param name=password value=pankaj/
param name=vm-password
Yes you're missing a switch_xml_free(xml); some place.
/b
On Sep 16, 2009, at 8:49 AM, Tihomir Culjaga wrote:
hi,
I've build a custom module for FS and everytihng work well except
reloadxml command :P... m'I missing something in my module? ... i
used mod_skeleton as a template when i
FS loads all users from $INSTALL_DIR/conf/directory/ and you did it correct.
freeswitch.xml:
section name=directory description=User Directory
X-PRE-PROCESS cmd=include data=directory/*.xml/
Than, you need to check sip profiles. By default FS will accept
registrations on internal
perfect,
thanks.
T.
On Wed, Sep 16, 2009 at 4:05 PM, Brian West br...@freeswitch.org wrote:
Yes you're missing a switch_xml_free(xml); some place.
/b
On Sep 16, 2009, at 8:49 AM, Tihomir Culjaga wrote:
hi,
I've build a custom module for FS and everytihng work well except
hi,
I've build a custom module for FS and everytihng work well except reloadxml
command :P... m'I missing something in my module? ... i used mod_skeleton as
a template when i started.
When i start the FS without my module reloadxml works fine ... as soon as i
include my module within
I will like to update the wiki to spell out clearly the differences
between this three commands
I have a IVR running in 4600 and the FS box has IP address 192.168.46.15
originate sofia/192.168.46.15/1001 4600
originate sofia/internal/1...@192.168.46.15 4600
originate
hello
version : 1.0.4 std. tarball
- the wiki example for php outbound socket connection leaks memory without the
async option
- the memory used is never given back
- async isn't that usefull for us - we want to query databases, set variables
and so on
no wait statements are possible
Might I ask what you are working on?Its interesting to hear what
people are doing with FreeSWITCH.
/b
On Sep 16, 2009, at 9:17 AM, Tihomir Culjaga wrote:
perfect,
thanks.
T.
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On Wed, Sep 16, 2009 at 1:53 AM, Tristan Mahé t.m...@telemaque.fr wrote:
Hi,
Count on me for answering questions on IRC when I'm in, and for subprojects
I'm in too as you know ;)
Merci!
Okay, what's your IRC nick and when are you generally on line? Also, I'm
pretty sure that you're fluent
Either:
1) Provide a simple self-contained example that demonstrates the leak
or
2) Run your application with FreeSWITCH under valgrind and provide the
final output. To run freeswitch under valgrind:
On Tue, Sep 15, 2009 at 1:46 PM, roberto miles.c...@gmail.com wrote:
Hello,
Someone could tell me what happens to the project, it seems that is no
longer available in github ?
http://github.com/diego/freeswitch-card/
thanks,
I haven't seen Diego Viola on line for a day or two. He can
Or you setup a gateway and set the from-domain
/b
On Sep 16, 2009, at 10:00 AM, João Mesquita wrote:
Is this what you are looking for?
http://wiki.freeswitch.org/wiki/Channel_Variables#sip_invite_domain
jmesquita
On Wed, Sep 16, 2009 at 10:58 AM, Tzury Bar Yochay tzury...@reguluslabs.com
as a good fs user - of course i am :-) - i made a jira on this
MODAPP-336 to be precise
i hope this helps to solve my problem
br
On 2009-09-16 17:05, Rupa Schomaker wrote:
Either:
1) Provide a simple self-contained example that demonstrates the leak
or
2) Run your application with
If you are a Debian person and have experience creating .debs then read
on...
Frank Carmickle has graciously volunteered to assist with creating a FS deb
config. I've heard that others have been doing something similar or are
interested. If you are such a person then please email Frank and me off
I think you're referring to the SIP SIMPLE implementation as the default FS
presence mechanism. This is fine and I can use that protocol. The question
I still have regards the plain text content in the body of the SIP MESSAGE
method. What is the format of this plain text for presence that is
On Wed, Sep 16, 2009 at 1:23 AM, Ahmed Munir ahmedmunir...@gmail.comwrote:
Hi,
I'm newbie in FS. I want to know how to process invalid extension in FS?
Because I want to prompt the IVR if invalid extension is dialled.
Kindly advice me.
Is this an invalid extension that was dialed by a
In my attempts to receive a fax from a PSTN fax machine, the transaction
fails with error code 13 Unexpected message received. Verbose logging
is on for mod_fax. Here is an exerpt:
#
2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:
On 09/17/2009 12:08 AM, Travis Stutsman wrote:
In my attempts to receive a fax from a PSTN fax machine, the transaction
fails with error code 13 Unexpected message received. Verbose logging
is on for mod_fax. Here is an exerpt:
#
2009-09-15
And make sure verbose is set to true in ./conf/autoload_configs/
fax.conf.xml.
On Sep 16, 2009, at 11:50 AM, Steve Underwood wrote:
On 09/17/2009 12:08 AM, Travis Stutsman wrote:
In my attempts to receive a fax from a PSTN fax machine, the
transaction
fails with error code 13 Unexpected
Alrighty. Here is mod_fax from beginning to end.
#
2009-09-15 10:41:26.433382 [DEBUG] mod_fax.c:591 Raw read codec
activation Success L16 2
2009-09-15 10:41:26.433382 [DEBUG] mod_fax.c:607 Raw write codec
activation Success L16
2009-09-15
I think you need to upgrade your version before we even take a look at
that... You are so far behind trunk right now and lots of things have been
changed since then.
Not sure if this would solve your problem but not a lot of ppl will look at
your problem when you run this version.
jmesquita
On
Hi, count on me for testing and answering questions on Windows and spanish
support.
Diego
http://lacarretade.blogspot.com/
--- On Wed, 9/16/09, Michael Collins m...@freeswitch.org wrote:
From: Michael Collins m...@freeswitch.org
Subject: Re: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting
well, it is a specific module for delivering some sort of services...
Actually, we are trying to build a SIP Application Server based on
freeswitch in core... the server will/should be in charge of delivering
various services e.g.
international call routing (route calls to international
On Wed, Sep 16, 2009 at 7:26 AM, Alberto Escudero aep.li...@it46.se wrote:
I will like to update the wiki to spell out clearly the differences
between this three commands
I have a IVR running in 4600 and the FS box has IP address 192.168.46.15
originate sofia/192.168.46.15/1001 4600
Hello Mindaugas,
It was not really a matter of Freeswitch + mod_radius_cdr not being
good for me, or for what we needed it to do, but rather more of a
resource and time constraint based decision. If we had 'C'
knowledgeable resources readily available, the Freeswitch and Radius
customizations
The problem i am facing is the following:
Extension 4600 is a Javascript IVR that starts by session.aswer()
I want to originate a call to leg 1 and then connected to the IVR when the
leg 1 has answered.
If I run
originate sofia/192.168.46.15/1001 4600
call is transfer to extension 4600 *IVR*
On Wed, Sep 16, 2009 at 11:43 AM, Alberto Escudero aep.li...@it46.sewrote:
The problem i am facing is the following:
Extension 4600 is a Javascript IVR that starts by session.aswer()
I want to originate a call to leg 1 and then connected to the IVR when the
leg 1 has answered.
If I run
I am also available for FS configuration on various Linux distributions and
Wiki / documentation.
Thank you.
On Wed, Sep 16, 2009 at 11:44 PM, Diego Toro dft...@yahoo.com wrote:
Hi, count on me for testing and answering questions on Windows and spanish
support.
Diego
Thank you for your answer.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of email
lists
Sent: 2009 m. rugsėjo 16 d. 21:39
To:
Hi guys!
I've tested FreeSWITCH conference module performance trying to figure out
maximum number of simultaneous calls my FS box can serve. It took all 100%
of CPU with only 50 calls (in average depending on conference rate) and
leaking stream handle messages started appearing.
The environment
Yes, it did work! No we do not need to pay for several GSM calls to test
a IVR script!
/aep and gmaruzz
--
Stopping junk mailers is good for the environment
On Wed, Sep 16, 2009 at 11:43 AM, Alberto Escudero
aep.li...@it46.sewrote:
The problem i am facing is the following:
Extension 4600
Has anyone integrated Vestec Speech Recognition with FreeSwitch? It's
$99/port...http://www.vestec.ca/
They have a C/C++ api, looks pretty simple. Alas, no MRCP until 2010.
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FreeSWITCH-users@lists.freeswitch.org
I would be really interested to replay your test on Linux. Would you be
willing to provide me all the details and relevant files so I can reproduce
the test with a Linux box here?
If yes, contact me offlist and we can work together on this.
Regards,
jmesquita
On Wed, Sep 16, 2009 at 2:56 PM,
I would be very interested in this also.
/b
On Sep 16, 2009, at 4:52 PM, João Mesquita wrote:
I would be really interested to replay your test on Linux. Would you
be willing to provide me all the details and relevant files so I can
reproduce the test with a Linux box here?
If yes,
A few thing stuck out to me ...
Mainly 50 calls and transcoding speex.
Try it again with g711 and see how you go.
Also not sure windows 7 is going to perform as good as other options,
could be wrong though .
Jay
On 17/09/2009, at 3:56, Роберт Тверитнер
siniy...@gmail.com wrote:
Hi
Yah speex is a cpu hog (while you can tune it to use less)! Granted
it uses less bandwidth but on the server side it doesn't scale very
well.
/b
On Sep 16, 2009, at 5:11 PM, Jay Binks wrote:
A few thing stuck out to me ...
Mainly 50 calls and transcoding speex.
Try it again with g711
hello
since installing the latest trunk 14894 my local streams / moh don't work
anymore
no config file has changed, the files are in place
show_local_stream outputs
default,/opt/freeswitch/sounds/music/8000
moh/16000,/opt/freeswitch/sounds/music/16000
I think the file was there but deleted by FreeSWITCH if it thinks it
was too short (like 3 seconds?). If I'm not wrong, someone requested
this feature becuase FreeSWITCH left too many small recordings.
On Sep 17, 2009, at 1:27 AM, João Mesquita wrote:
I think you need to upgrade your
Hi Travis,
That's a pretty weird call. It looks like you have a long delayed echo.
See below.
On 09/17/2009 01:21 AM, Travis Stutsman wrote:
Alrighty. Here is mod_fax from beginning to end.
#
2009-09-15 10:41:26.433382 [DEBUG] mod_fax.c:591
this makes sense. a workaround would be to provide an optional variable to
delete recording file if it's less than N seconds. otherwise, it defaults to
a preset duration.
/nandy
On Thu, Sep 17, 2009 at 7:46 AM, Seven Du dujinf...@gmail.com wrote:
I think the file was there but deleted by
Folks;
I give credit where credit is due, and I thank Brian K. West
What For:
This was found to be a compounded problem. (Cisco was part of it...
But the real problem was the linux kernel...)
Suffice it to say, without the kernel bug, the cisco bug wouldn't have
been easily found.
What
we support both application/dialog-info+xml (snom maybe a few others, I
can't keep track) and pidf used by polycom and eyebeam
On Wed, Sep 16, 2009 at 10:50 AM, Jerry Richards jerry.richa...@teotech.com
wrote:
I think you're referring to the SIP SIMPLE implementation as the default
FS
Karl Vesterling k...@ken-ton.com wrote:
What Kernel Bug:
It's a kernel bug that corrupted the sqlite database.
This caused Freeswitch to refuse the phones registration request.
Please take this up with your Linux distribution as a bug report related to
the kernel, and persist with it until
Hi:
Since svn version 13523 to current I get this error:
make[5]: swig: Command not found
make[5]: *** [mod_lua_wrap.cpp] Error 127
make[4]: *** [all] Error 1
make[3]: *** [mod_lua-all] Error 1
make[2]: *** [all-recursive] Error 1
Making all in build
+ FreeSWITCH Build Complete
Which distro is this?
/tyn
On Thu, Sep 17, 2009 at 12:07 AM, Jason White ja...@jasonjgw.net wrote:
Karl Vesterling k...@ken-ton.com wrote:
What Kernel Bug:
It's a kernel bug that corrupted the sqlite database.
This caused Freeswitch to refuse the phones registration request.
Please take
On Wed, Sep 16, 2009 at 9:10 PM, Karl Vesterling k...@ken-ton.com wrote:
It's a kernel bug that corrupted the sqlite database.
This caused Freeswitch to refuse the phones registration request.
This in turn caused the phones to re-register.
Problem was, with 10 phones, 6 lines each, perpetually
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