For what it's worth, using Asterisk recordings, I found FS to be better
than when played on an Asterisk system.
I came to the same conclusion early on that the included prompts with FS
were of a relatively poor nature. Not volunteering to record new ones,
but they do let the product down, as
jay binks jaybi...@gmail.com wrote:
Back in November, Brian ( BKW ) was raising money to get new sounds recorded
...
intending to have them for the 1.0.2 release..
I wonder if they made it in, or if they are still coming ...
Release 1.0.7 of the sound files was made available soon
There is also another side to make mimd to: the Asterisk sounds you
hear more often (the demo ones) are very long ones.
The ones of the FS demo are very very short (many times just one word)
and concatenated with the insertion of sleeps.
That is probably someway altering the equation between
Back in November, Brian ( BKW ) was raising money to get new sounds recorded
...
intending to have them for the 1.0.2 release..
I wonder if they made it in, or if they are still coming ...
Jay
On Tue, Feb 17, 2009 at 8:19 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote:
There is also
Paul D. wrote:
I re-tested calls to VM replacing some of FS prompts with * ones, and it
appears that * sounds were recorded with a better quality/higher volume,
so FS itself has nothing to do with that. That's solved. :-)
There's a long history of people in A/B listening tests reporting
Hi there,
Anyone knows how to configure FS in order to have log folder in
/var/log/freeswitch ?
I did ./configure --prefix=/opt/freeswitch and after install I got
logs, pid file and cdrs under /opt/freeswitch/log
Thanks in advance.
Pablo
___
David Knell napsal(a):
There's a long history of people in A/B listening tests reporting louder
as sounding
better on the same source material - even if the additional volume isn't
detectable
as such.
Yes, you are right :-) And therefor a lot of (nearly all of) European
TelCo operator
Hello FS Members,
Are there any example of FS running on a Thumb Flash USB?
Thanks in advance,
-E
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Maybe the sox script brian uses to downsample the files has a problem.
What if you download the 48k package (original) and listen to that?
On Tue, Feb 17, 2009 at 6:35 AM, David Knell d...@3c.co.uk wrote:
Paul D. wrote:
I re-tested calls to VM replacing some of FS prompts with * ones, and
Could be done ed... FS itself isnt that big
From: EdPimentl edpime...@gmail.com
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Tue, 17 Feb 2009 08:50:50 -0500
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB?
Hello FS
I would be glad to put a bounty for a FS(and Skypiax/softphone) running on
Flash Thumb Drive.
Project would be documented on Wiki for others to use and improve.
-E
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On 2/17/2009 1:18 AM, Tchavdar Paskov wrote:
I also have another question.
When i insert my custom query it looks like the profile is loaded
successfully but then when i place a call or use
lcr ## default /which is where i defined the sql query/ and
check the console output
Thanks Brian,
Actually we're using freeswitch ver 1.0.2.
Regards,
Message: 5
Date: Thu, 12 Feb 2009 15:48:00 -0600
From: Brian West br...@freeswitch.org
Subject: Re: [Freeswitch-users] Realm value
To: freeswitch-users@lists.freeswitch.org
Message-ID:
Its currently not possible.
/b
On Feb 16, 2009, at 11:33 PM, Cesar Cepeda wrote:
Hi,
I’m using FS with g279 on passthrough mode and I’m trying to play a
g729 file as ringback to the A-leg while bridging a call. As far as
I understand it should go something like this:
·
You can specify those as runtime argument.
$ ./freeswitch -h
these are the optional arguments you can pass to freeswitch
-nf-- no forking
-u [user] -- specify user to switch to
-g [group] -- specify group to switch to
-help
Thanks Mathieu.
So, it is not possible to set this at build time by passing a
parameter to the configure script...
IMHO, a symbolic link is not a good idea in production environments.
BR
Pablo
On Tue, Feb 17, 2009 at 1:05 PM, Mathieu Rene mrene_li...@avgs.ca wrote:
You can specify those as
Hello,
Our voicemail prompts playback much too quickly on FS v1.0.1. Any
suggestions to slow them down?
Thanks,
Maxim.
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I'm having a problem that calls to the auto-attendant won't transfer. I know
this has been a problem in the past but thought that it was fixed. Whenever
I enter an extension (or press a key to transfer me to one), the call just
hangs up. I ran Freeswitch on the console, but all I see happening is
Our voicemail prompts playback much too quickly on FS v1.0.1. Any
suggestions to slow them down?
At this point the best thing for you to do is to update to the latest
trunk. We are 99.99% ready to tag 1.0.3RC2 which has significant
improvements over 1.0.1.
-MC
P.S. - you might find this page
You're missing some key information to help us answer your question.
First off we will need to know the SVN rev, Then you might want to
press F8 and check out the debug log. Chances are it'll tell you
exactly why. What concerns me is the fact that a .local domain is in
there. I wonder
On Tue, Feb 17, 2009 at 8:54 AM, Pablo Hernan Saro pablos...@gmail.com wrote:
Thanks Mathieu.
So, it is not possible to set this at build time by passing a
parameter to the configure script...
IMHO, a symbolic link is not a good idea in production environments.
BR
You can also specify these
I'm trying to get a clear picture of what you're trying to
accomplish. Why would you need/want to set a static realm? Anyway
can you collect sip traces?
/b
On Feb 17, 2009, at 8:51 AM, Ali Al-Rubaie wrote:
Thanks Brian,
Actually we're using freeswitch ver 1.0.2.
Regards,
We don't yet support localstatedir configure option. I expect we will
soon.
Mike
On Feb 17, 2009, at 12:59 PM, Pablo Hernan Saro wrote:
Hi Michael,
Thank you very much for your help. It meets my needs.
I was thinking in something like:
./configure --prefix=/opt/freeswitch
Having spent the last week developing a small js app, I ran some tests
today. With just 5 calls going on, I'm seeing huge delays from when the
call is answered to when the audio file is played. Sometimes it doesn't
even play at all!!
Example 3 calls and the matching playbacks
2009-02-17
if (first_session.ready()) {
console_log(notice,Session state=[ +
first_session.state + ] \n);
consoleLog(NOTICE, ready: Start DTMF\n);
first_session.execute(start_dtmf);
Is this the entire script?!
-MC
On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
if (first_session.ready()) {
console_log(notice,Session state=[ +
first_session.state + ] \n);
consoleLog(NOTICE,
Pretty much
I haven't included the on-event hooks as it never gets to the point
where they're called.
Only other thing is the dial it's self, attached below. However, I
notice in the default dial plan, if I call extension 1001 from 1000 it
takes about 2-3 seconds for the phone to ring. Is that
Hi all,
I have just upgraded to current trunk (before an hour or so),
configuration remain the same (served through mod_xml_curl), but
something has changed and I don'nt know where, what and why :-)
What's going on:
I have few sofia profiles and each of them has its own context. When
call
Make sure on outbound registrations/gateways you have the context and
extension params set.
/b
On Feb 17, 2009, at 2:17 PM, kokoska rokoska wrote:
Hi all,
I have just upgraded to current trunk (before an hour or so),
configuration remain the same (served through mod_xml_curl), but
.
Regards,
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Freeswitch
Hello, folks - I hope that I can reach someone who knows the answer to
this one:
I bought 2 Linksys SPA-922 phones from a guy on ebay. The phones are
locked by Webnet global Communications. From what I can tell, this
company went bankrupt, and the ebay seller bought the phones from a
okay, can you do the usual stuff and report a bug on jira? Not sure if
it's really bug but having you collect all of the data and submit a
bug report will assist us greatly.
-MC
On Tue, Feb 17, 2009 at 12:30 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
I'm starting to think it's a
Very sorry to hear you have to use Broken Software. But some good has
come of this if you update to rev 12113 or great you'll be 100% OK.
/b
On Feb 17, 2009, at 2:21 PM, Ali Al-Rubaie wrote:
I have to use a specific softphone, HelpCaster, but it can not pass
the authentication stage.
1) turn off crash protection.
2) you cant manipulate more that one call per script, design the script to
be run from the application interface so you originate the call with the api
interface and transfer the call to the script so each one has it's own copy
of the script.
On Tue, Feb 17, 2009
Mark,
Sorry to say but I think you're pretty much SOL. I would check
voipsupply or the like for a replacement. On a side note you hijacked
the Big delays in playing audio files thread by clicking reply on
one of those messages then changing the subject and body... in the
future
Sadly, 73738# does not work.
Is there a jumper on the board or some other hardware fix for this?
Quoting Gabriel Kuri gk...@ieee.org:
Have you tried resetting the phone via the built-in IVR menu?
Pick up the handset and dial 73738#
This should reset the phone to factory defaults,
On Tue, Feb 17, 2009 at 12:48 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
1, OK,
2. Right now I have a php script calling bgapi via and event socket with the
call parameters. Is that what you mean? If not, can you give me a pointer?
I had assumed that every time I called
Brian West napsal(a):
Make sure on outbound registrations/gateways you have the context and
extension params set.
Thank you very much, Brian, for your suggestion!
I had context defined on all sofia profiles, but I didn't have extension
param set on gateways (but it works till I upgraded to
No problem. Just join us on IRC.. things move faster on there.
/b
On Feb 17, 2009, at 3:05 PM, kokoska.rokoska wrote:
If you have more hints, I be very happy :-)
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I did unplug the ethernet cable. I have never been able to make the
IVR work on any of the Linksys phones that I have. I must be doing
something wrong.
I will try to sniff the traffic on the phone when I start it up. I
will report back when I do.
Thanks so much -
Library Mark
Quoting
s it possible to run Skypiax on OS X? The wiki says Linux and Windows,
but says nothing about OS X.
I have been running FreeSWITCH on OS X for a couple of years now, and
love it. Adding Skype gateway would be really sweet.
Are there any plans for adding Skypiax to trunk, or do we have to
yes, bgapi counts as an API call. Ken Rice thinks this might be
related to a spidermonky concurrency issue...
Well that kinda fits, as I see the audio files stacking up, seems like
they're being queued.
Question is, what's my alternative, lua?
-Original Message-
From:
any hints?
what would be the best way to report CDRs for attended transfers??
We are using C with libxml to create a binary that can be called from a
script to rotate xml_cdrs and insert them on SQLite and would gladly submit
the code to your revision, advice and maybe even potential use.
I
Brian West wrote:
Don't they cryptographically sign the config also?
it's an option in the device... some providers do, some don't.
but it shouldn't matter too much if they're using https or not, as long
as the ata doesn't authenticate via certificate or something.
-Ray
yes, bgapi counts as an API call. Ken Rice thinks this might be
related to a spidermonky concurrency issue...
Well that kinda fits, as I see the audio files stacking up, seems like
they're being queued.
Question is, what's my alternative, lua?
Lua or C/C++, but Lua is the consensus for
I think Ctrix is working on mod_airpe in his branch for OS X.
/b
On Feb 17, 2009, at 3:19 PM, Ivan C Myrvold wrote:
s it possible to run Skypiax on OS X? The wiki says Linux and Windows,
but says nothing about OS X.
I have been running FreeSWITCH on OS X for a couple of years now, and
love
It depends on the whether you pass the option to the Linksys/Cisco
Profile Compiler to generate the config file. In any case, that
shouldn't be an issue.
Gabe
Brian West wrote:
Don't they cryptographically sign the config also?
/b
On Feb 17, 2009, at 2:58 PM, Gabriel Kuri wrote:
If you
On the slight chance they're not doing remote provisioning and the phone
is just simply locked with a username/password, you'll need to feed the
phone a TFTP server via DHCP Option 66 and setup a config file on that
tftp server with the name spa922.cfg.
Contact me off list about generating a
I'm about 99% positive that if https is enabled for remote provisioning,
the web server needs an SSL certificate signed by the Linksys
Enterprise CA, otherwise the phone will reject it.
Gabe
but it shouldn't matter too much if they're using https or not, as long
as the ata doesn't
Hi guys,
I noticed that the debian build is missing lines for shout.conf.xml and does
not install mod_flite (if its built) . This can be fixed by adding the
following lines to debian/freeswitch.install
opt/freeswitch/mod/mod_flite*
opt/freeswitch/conf/autoload_configs/shout.conf.xml
and
Please submit all patches and changes via jira if possible
http://jira.freeswitch.org
Thanks,
Brian
On Feb 17, 2009, at 4:40 PM, Dan wrote:
Hi guys,
I noticed that the debian build is missing lines for shout.conf.xml
and does not install mod_flite (if its built) . This can be fixed
by
Oh, and thanks for the info!
-MC
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local e = freeswitch.Event(custom,
dialer::dialer-result);
e.addBody(custom_msg);
e:fire(e);
The wiki page (http://wiki.freeswitch.org/wiki/Lua#event:fire) shows
that you fire thusly:
in lua you call methods with a colon :
e:addBody(blah);
calling with a . implies you are going to supply the obj too
e.addBody(e, blah);
On Tue, Feb 17, 2009 at 5:11 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Hi Guys,
I'm having real problems doing something trivial,
On Tue, Feb 17, 2009 at 3:25 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
in lua you call methods with a colon :
e:addBody(blah);
calling with a . implies you are going to supply the obj too
e.addBody(e, blah);
Also, there is an explicit example here:
I've got it working now thanks
I've also added a working example to the Wiki (lua/addBody) which was
empty
Regards,
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 17
Err, that's what I just posted :)
Regards,
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 17 February 2009 23:30
To: freeswitch-users@lists.freeswitch.org
Subject: Re:
Good... keep up the good work adding more docs. ;)
/b
On Feb 17, 2009, at 5:33 PM, Nik Middleton wrote:
Err, that's what I just posted :)
Regards,
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FreeSWITCH now compiles in AsLinux:
http://www.astlinux.org
AstLinux with the new bootloader Runnix (or you could just use
syslinux) boots from flash. It also boots from PXE, ISO, disk, etc.
Pretty much anything :).
FreeSWITCH (compiled against uClibc) is about 3.3MB and as Tony
pointed out,
Great news!!! Good Job!
/b
On Feb 17, 2009, at 5:43 PM, Kristian Kielhofner wrote:
FreeSWITCH now compiles in AsLinux:
http://www.astlinux.org
AstLinux with the new bootloader Runnix (or you could just use
syslinux) boots from flash. It also boots from PXE, ISO, disk, etc.
Pretty much
wow... this is awesome !
good job mate.
On Wed, Feb 18, 2009 at 9:43 AM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
FreeSWITCH now compiles in AsLinux:
http://www.astlinux.org
AstLinux with the new bootloader Runnix (or you could just use
syslinux) boots from flash. It
I'll shortly post some docs on the php fs_sock. There's also a couple
of bugs in it that I've fixed.
I ran 10,000 events, which completed in around 20 seconds, all received
and processed flawlessly. A new one on me was arrayshift. To think that
I messed around in C for ages with circular
On Tue, Feb 17, 2009 at 3:43 PM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
FreeSWITCH now compiles in AsLinux:
Nice work! I'll go tell our friends over in the Yahoo financial forums
- I'm sure they're dying to hear about it! ;)
-MC
___
I ran 10,000 events, which completed in around 20 seconds, all received
and processed flawlessly. A new one on me was arrayshift. To think that
I messed around in C for ages with circular buffers, this is so simple.
Excellent! You're officially deputized to add any Lua examples you
create. We
And you ran this in lua?
/b
On Feb 17, 2009, at 6:07 PM, Nik Middleton wrote:
I ran 10,000 events, which completed in around 20 seconds, all
received
and processed flawlessly. A new one on me was arrayshift. To think
that
I messed around in C for ages with circular buffers, this is
awesome work! on a slightly related [embedded] note, do you know if any
work has been done to port FS to any of the Analog Blackfin MCUs? I'd be
interested in hearing if anyone has had any such luck.
Gabe
Kristian Kielhofner wrote:
FreeSWITCH now compiles in AsLinux:
http://www.astlinux.org
Ah yes, the pinnacle of online discussion! ;)
On Tue, Feb 17, 2009 at 7:10 PM, Michael Collins m...@freeswitch.org wrote:
On Tue, Feb 17, 2009 at 3:43 PM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
FreeSWITCH now compiles in AsLinux:
Nice work! I'll go tell our friends over in
I don't think so but something tells me that FreeSWITCH won't do too
well without an MMU and the external libs and modules could cause
quite a problem. Not that it is impossible but the uh, performance,
would be interesting...
Can anyone call me out on this assumption?
On Tue, Feb 17, 2009 at
No, js, I was trying to break the fs_sock.php, though I found the time
was dependant on how much I echoed to the screen.
I expect lua to be even faster
Regards,
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org]
Hello FreeSWITCHERS,
My company is currently creating a suite of applications which uses
FreeSWITCH as the back-end for an IP-PBX solution. We currently have a
prospect to have our first customer installation - a governmental
department. That is a tender to have an IP-PBX installation to connect
Kristian,
You're my hero, if I hadn't come across astlinux 3 years ago, I wouldn't
be doing this stuff right now. Not too sure if that's a good thing
though ;)
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org]
I have Freeswitch running successfully with a fairly basic config. Nat
traversal is working well on both the client and server side. I want to start
running all RTP streams through a media gateway, and use Freeswitch for SIP
registrations and signalling only.
I believe that I need to have
you could set the variable bypass_media to true before you call bridge
action application=set data=bypass_media=true/
action application=bridge data=sofia/internal/someu...@somehost/
that will negotiate a point to point media connection between the caller and
callee
On Tue, Feb 17, 2009 at
I looked at that, but I think that will cause issues with the NAT traversal.
Our phones will all be in external networks. I forgot to mention that.
-Original Message-
From: Anthony Minessale [mailto:anthony.miness...@gmail.com]
Sent: Tuesday, February 17, 2009 05:34 PM
To:
That and the lack of an FPU, I'm curious how that would affect FS,
especially with transcoding. At one point I was interested in building a
little embedded PBX running FS on the Blackfin MCU. Since Analog Devices
seems fairly open with the Blackfin, I thought it might be a good
choice, but I'm not
I really need to work on that name but in the meantime it seems like
people are interested. Check it out:
http://www.astlinux.org/node/41
It's just a little ISO, download it and give it a shot! ;)
--
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
Ok Tomorrow is BKW's b-day...
Lets as a community see if we cant pull together and get him something nice
off his Amazon wish list...
Visit his wish list here...
http://www.amazon.com/gp/registry/wishlist/1BWDJUX5LYQE0
If you want to chip in contact me on IRC (I'm SwK) or send something via the
We could ban him from IRC for the day... that would be a gift :)
On Tue, Feb 17, 2009 at 19:48, Ken Rice kr...@freeswitch.org wrote:
Ok Tomorrow is BKW's b-day...
Lets as a community see if we cant pull together and get him something nice
off his Amazon wish list...
Visit his wish list
a man
geez im interested in this ..
I hope it ends up kicking ass ! :)
Congrats, you are awesome.
Jay
On Wed, Feb 18, 2009 at 11:41 AM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
I really need to work on that name but in the meantime it seems like
people are
Brian,
We'll figure something out...
On Tue, Feb 17, 2009 at 8:36 PM, Brian West br...@freeswitch.org wrote:
So you sticking with the Astlinux name? or switching it to something
more general?
/b
--
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
Nik Middleton wrote:
I'll shortly post some docs on the php fs_sock.
don't waste your time... There's a php .so for ESL now, and i'll
probably be removing the fs_sock from tree sometime very soon... maybe
replacing it with some specific api classes... i'm not sure on that part
yet.
-Ray
Pretty much all the codecs (mod_voipcodecs, mod_speex, mod_ilbc,
mod_g722_1, mod_celt) and the resampler all have fixed point
implementations (in tree) as well.
Mike
On Feb 17, 2009, at 7:44 PM, Gabriel Kuri wrote:
That and the lack of an FPU, I'm curious how that would affect FS,
Nik,
Thanks but I'm not sure I want to take the credit (blame?) for that! ;)
On Tue, Feb 17, 2009 at 7:25 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Kristian,
You're my hero, if I hadn't come across astlinux 3 years ago, I wouldn't
be doing this stuff right now. Not too
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