Hello Michael,
how much $$ are we talking about? I need this issue to be solved quickly
and it's worth to spend some money.
I've read the following post:
http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg05792.html
and have the same symptom with after hundreds of calls I
There NO previous version of FS installed before, and FS 1.0.2 is also
freshly installed.
Anthony Minessale-2 wrote:
please remove FS src and dest dir from your machine and recompile fresh
from
scratch.
On Tue, Jan 13, 2009 at 4:48 AM, shehzad p pmh...@gmail.com wrote:
Please
Is there a way to use the hardware timers e.g. of a PRI card in
fresswitch? Or other question: Is it recommended to use those if they
are available?
I have installed a dual PRI card, and show timer shows one soft timer.
Best regards
Peter
___
Thanks, will go and have a look at the developers list.
Scott
Jason White wrote:
Scott Ellis scott.el...@novatex.com.au wrote:
I have tracked down a set of au tones from the mailing list, which I am
going to verify. How do I go about getting these added into the default
build
Hi,
Im getting error on startup when executing freeSWITCH.exe , The procedure
entry point_apr_...@12 could not be located in the dynamic library
libaprutil.dll
--
/Lito
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
After poking around in the code, it looks like if I set param
name=enable-callerid value=false/ in openzap.conf.xml, it should
skip the GET_CALLERID state, and I should get the call answered straight
away.
mod_openzap.c
} else if (!strcasecmp(var, enable-callerid)) {
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Michael,
it must not be the case here, but I had the same error, when incomming
calles used a wrong numbering plan resp not the one, FS expected.
Just a hint.
regards
Helmut
Am 15.01.2009 09:20, schrieb Peter P GMX:
Hello Michael,
how much
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
today I moved the voicemail database from sqlite to mysql via odbc. FS
started up and connected successfully to the empty database.
Normally voicemail adds the neccessary database tables automaticly
during startup. In this case I forgot to
Helmut,
can you give me a hint, how you worked around this?
Best regards
Peter
Helmut Kuper schrieb:
Hi Michael,
it must not be the case here, but I had the same error, when incomming
calles used a wrong numbering plan resp not the one, FS expected.
Just a hint.
regards
Helmut
Am
Hi
I have a IP to GSM gateway which supports SIP. How I can send SMS to
the GSM phone using FreeSwitch + SIP GSM GW?
Thanks
Imthiyaz
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
So I decided to hack the code to see if I could just get it to do what I
wanted - assuming some kind of error in the options setting.
First I changed the state change code to just skip straight to IDLE
if (!event-channel-ring_count (event-channel-state ==
ZAP_CHANNEL_STATE_DOWN
Hello
I got problems with hanging spidermonkey sessions and need some advice on
how to debug them.
I've made a javascript queue application that uses mod_spidermonkey_socket.
It works fine for a while,
but after some calls I noticed that calls didnt get transferred to agents.
The reason was that
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Peter,
it was simply a change in our TDM Voice Switch. It used a different
numbering plan and we changed it to national to get it work with FS
and openzap in Q921/Q931 mode.
What I still search is a way to configure the numberplan in FS.
To make
Thank you very much for your help, I've realized I was specting to receive
my house phone number having a POTS line and that's not possible.
So, I've put my house number in openzap.conf:
[span zt]
name = OpenZAP
number = 91999
fxo-channel = 1
And I've added an extension on the default
FreeSWITCH + Centos 5.2 rolled into a VMWare appliance that's compatible
with VMWare for Windows, VMWare Player, and VMWare Fusion on the Mac.
Couple of things you need to know. 1) This is a Centos 5.2 X86_64 that was
updated before we started testing it. 2) FreeSWITCH SVN Trunk is installed
and
Hey guys,
I'm not trying to start 1 a day releases, Things just happened to fall that
way...
FreeSWITCH + Centos 5.2 rolled into a VMWare appliance that's compatible
with VMWare for Windows, VMWare Player, and VMWare Fusion on the Mac.
Couple of things you need to know. 1) This is a Centos 5.2
kokoska rokoska wrote:
Hi all,
I have just post two bounties
Where did you post these bounties? We've started moving bounties away
from the wiki and adding them to jira instead. ( so that progress can be
followed more closely )
-Ray
begin:vcard
fn:intralanman
n:Chandler;Raymond
adr:;;630
I added another arg to the list. I'll have to revisit this today to
make sure I did this right for your case.
/b
On Jan 15, 2009, at 1:43 AM, Juan Backson wrote:
Hi,
Is there a change in the playAndGetDigits api? In the old release,
11102, my lua script is working but is not working in
http://wiki.freeswitch.org/wiki/Report_Issue_Checklist
Please open a jira and include your script and a test case.
/b
On Jan 15, 2009, at 5:20 AM, Jonas Gauffin wrote:
Hello
I got problems with hanging spidermonkey sessions and need some
advice on how to debug them.
I've made a
You can submit patches to http://jira.freeswitch.org
thanks,
/b
On Jan 15, 2009, at 1:16 AM, Scott Ellis wrote:
I have tracked down a set of au tones from the mailing list, which I
am
going to verify. How do I go about getting these added into the
default
build so that they are
Update and try now... I think we fixed this to not break API
compatibility.
/b
On Jan 15, 2009, at 1:43 AM, Juan Backson wrote:
Hi,
Is there a change in the playAndGetDigits api? In the old release,
11102, my lua script is working but is not working in the latest
release.
The error I am
Raymond Chandler napsal(a):
kokoska rokoska wrote:
Hi all,
I have just post two bounties
Where did you post these bounties?
I have posted them to the Boutny wiki page (at the bottom of the page):
http://wiki.freeswitch.org/wiki/Bounty
We've started moving bounties away
from the wiki
kokoska rokoska wrote:
Raymond Chandler napsal(a):
kokoska rokoska wrote:
Hi all,
I have just post two bounties
Where did you post these bounties?
I have posted them to the Boutny wiki page (at the bottom of the page):
http://wiki.freeswitch.org/wiki/Bounty
Could you repeat this test with debug loglevel turned on? (Press F8 or
type console loglevel 7). Please put the results in
pastebin.freeswitch.org.
-MC
On Thu, Jan 15, 2009 at 4:35 AM, Tomás tomasborre...@gmail.com wrote:
Thank you very much for your help, I've realized I was specting to
Hello Ken, hello all,
I just read about the FreeSWITCH VMware applicance. I'm curious about
your experiences with the audio quality on VMWare, so here's a new
thread.
I've installed freeswitch on VMware Server for Windows. The IVR audio
always plays choppy, while the server itself has no
On 1/15/09 11:01 AM, Remko Kloosterman r.klooster...@mtel.nl wrote:
Hello Ken, hello all,
I just read about the FreeSWITCH VMware applicance. I'm curious about
your experiences with the audio quality on VMWare, so here's a new
thread.
I've installed freeswitch on VMware Server for
If anyone figures this out please post it to this thread. I am working
on a wiki page for the VMWare appliance and I would like to be able to
inform people on how to handle this situation.
Also, IIUC, those running VMWare Fusion on Macs are not experiencing
this, correct? What about those using a
Thanks Helmut,
I cross-checked with our provider. They use national numbering plan for
our lines. So this didn't solve our problem.
I also ensured that the local language is DE and ZAP timing is dedicated
to span 1.
I changed the configs to debug mode for OpenZAP, so I hopefully will get
some
I have been running FreeSWITCH on a VM ever since I got involved in the
project. It's been almost a year now. I didn't do anything special - it
works fine. I get audio problems if I go over 10 or 15 simultaneous calls.
This is on the following setup:
VMWare Server 1.0.6 and VMWare Server 2.0 (2.0
Thanks Michael, that did get me a little further.
I renamed mod_managed_lib.dll to FreeSWITCH.Managed.dll and that definitely
had an effect. but now when I attempt to load mod_managed
FreeSwitch core dumps now.
I have tried mono 2.2 and mono 2.0.1
I am running . CentOS 5.2 x86 32bit
Raymond Chandler napsal(a):
kokoska rokoska wrote:
Raymond Chandler napsal(a):
kokoska rokoska wrote:
Hi all,
I have just post two bounties
Where did you post these bounties?
I have posted them to the Boutny wiki page (at the bottom of the page):
Hi,
Can somebody tell me how to achieve the same behavuior as session.waitForAnswer
via the socket interface?
That is, when i call a device, i want to block until the call is completely
answered (not just early media).
Thanks,
Klaus.
--
Sensationsangebot verlängert: GMX FreeDSL -
Then originate the call with {ignore_early_media=true}sofia/blah/blah,
It will not return till its actually answered.
/b
On Jan 15, 2009, at 12:55 PM, Klaus Teller wrote:
Hi,
Can somebody tell me how to achieve the same behavuior as
session.waitForAnswer via the socket interface?
That
Klaus,
What is your dialstring? If you ignore_early_media=true then I believe
it will have the same net effect, but it would be good to know exactly
what you're hoping to accomplish.
-MC
On Thu, Jan 15, 2009 at 10:55 AM, Klaus Teller klaus.tel...@gmx.net wrote:
Hi,
Can somebody tell me how
Hello,
I'm running fs 1.0.2 on CentOS 5.2
I've been trying to setup my fs to talk with googletalk following the
instructions in
http://wiki.freeswitch.org/wiki/Mod_dingaling#Sample_Configuration
I got the error of TLS not supported so i:
INSTALLED:
yum install gnutls-devel gnutls
REMOVED:
rm
On 01/13/2009 04:00 PM, Anthony Minessale wrote:
So I can supply you with 250 thousand lines of C code that make your
application possible.
but you are not willing to show me the silly js code that may be the
cause of your crash?
What security purposes are you kidding?
I just need to
install gnutls and dev packages and reconfigure/recompile
/b
On Jan 15, 2009, at 1:21 PM, Milena wrote:
Hello,
I'm running fs 1.0.2 on CentOS 5.2
I've been trying to setup my fs to talk with googletalk following
the instructions in
Thanks folks! ignore_early_media=true solves my problem. The dialstring was
just sofia/gateway/blah/blah.
Klaus.
Original-Nachricht
Datum: Thu, 15 Jan 2009 11:06:57 -0800
Von: Michael Collins m...@freeswitch.org
An: freeswitch-users@lists.freeswitch.org
Betreff: Re:
Daniel-Constantin Mierla wrote:
On 01/13/2009 04:00 PM, Anthony Minessale wrote:
So I can supply you with 250 thousand lines of C code that make your
application possible.
but you are not willing to show me the silly js code that may be the
cause of your crash?
What security purposes
Hello,
Isn't that what I did?
if not, what is the right way to install gnutls and dev packages and
reconfigure/recompile
thank you
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
Well the right way is depends on your distro. Once you have it
installed I would ./bootstrap.sh and ./configure again to be safe.
/b
On Jan 15, 2009, at 1:39 PM, Milena wrote:
Hello,
Isn't that what I did?
if not, what is the right way to install gnutls and dev packages
and
Hello,
I have recently found out about FS and how great it is.
We are trying to use FS as a voip client for radio shows.
We have been using Trixbox and Skype but Skype isn't getting it done.
I have heard about how great the celt codec is but I don't have
enough 'skill' to compile both FS and celt
You'll never fix this. Voice is a latency specific application unless you
figure out how to manipulate time. Any virtualization platform is going to
provide less timing granularity than raw hardware.
Hello Ken, hello all,
I just read about the FreeSWITCH VMware applicance. I'm curious about
Matthew,
I am not berating him, I am trying to convince him to give me the script
that causes his crash.
It seems ridiculous to me that he should be worried about what I will do
with his js code when I am clearly
only interested in finding out what causes his issue. And there is irony
for him to
That won't eliminate the problem. Just reduce the possibility of it
happening.
Trust me... I've got a large ESX infrastructure, and there is no way that a
software based Voice platform is going to provide skip free audio in a
virtualized environment.
-Original Message-
From:
I did some more tests. When I sequentially setup calls (only one
simultaneous call at a time), it works for hundreds of calls.
As soon as I setup 2 calls in parallel ist fails aber a number of calls.
Please find another debug ouput (now with Q.921 debug also).
The log starts with the latest
Michael Collins wrote:
If anyone figures this out please post it to this thread. I am working
on a wiki page for the VMWare appliance and I would like to be able to
inform people on how to handle this situation.
Also, IIUC, those running VMWare Fusion on Macs are not experiencing
this,
Oops, it took me a little while to realize what you meant and why make alone
wouldn't work, thank you very much sir, it all works fine now.
2009/1/15 Milena testeado...@gmail.com
Hello,
Isn't that what I did?
if not, what is the right way to install gnutls and dev packages and
To the contrary, we have had quite good results in virtualized
environments and you don't really need timing that is that accurate to
make it work. We work quite well on amazon EC2 for example. There
are 2 issues I know about with vmware, 1 is you need to set a setting
on the host to
To the contrary, we have had quite good results in virtualized
environments and you don't really need timing that is that accurate to
make it work.
If you don't handle RTP, I'm sure it is amazing. However, if you have to do
voicemail, stream audio from the server or do any kind of actual
Hello all,
let me also give some experience from the VirtualBox side (Community
Version).
Host machine
==
AMD X2 64 3800 with 8GB of RAM
OS is a generic Debian 4.0R5 with Kernel 2.6.18-6-amd64
No special parameters in the Kernel.
Started with VirtualBox 1.5 and now on 2.0.x
Client
We have people running FreeSWITCH in vmware and xen with media and
considerable load and it doesn't have a problem. We also work very
well inside OpenVZ.
/b
On Jan 15, 2009, at 2:37 PM, Gregory Boehnlein wrote:
If you don't handle RTP, I'm sure it is amazing. However, if you
have to do
We have people running FreeSWITCH in vmware and xen with media and
considerable load and it doesn't have a problem. We also work very
well inside OpenVZ.
I'd be very interested in seeing that, and knowing how it was done.
___
Freeswitch-users
Ok if can summarize a little of the intention of releasing this VMWare
image. Its really there so you guys can get it and check it out. I
personally don't believe in running such services on a virtual machine (too
many nightmare stories from the 'day job' from such things)
However, for testing
On that note the OpenVZ instances could live migrate from box to box
without dropping calls and usually had a small acceptable blip in audio.
/b
On Jan 15, 2009, at 2:59 PM, Gregory Boehnlein wrote:
We have people running FreeSWITCH in vmware and xen with media and
considerable load and it
On Jan 15, 2009, at 1:02 PM, Brian West br...@freeswitch.org wrote:
On that note the OpenVZ instances could live migrate from box to box
without dropping calls and usually had a small acceptable blip in
audio.
I'd say a small blip is quite acceptable compared to the alternative!
-MC
/b
On that note the OpenVZ instances could live migrate from box to box
without dropping calls and usually had a small acceptable blip in
audio.
OpenVZ is not a hypervisor. It essentially runs all of it's applications
natively on the CPU. I would expect that it would work under OpenVZ or other
Hi,
Need your help on this. I have the following Javascript statement:
session.execute(bridge,sofia/gateway/sip.gafachi.com/someNumber) in a file
called gafachiDialout.js
Then, i have the following extension in default.xml:
extension name=6337
condition field=destination_number
Hi,
I have successfully installed and configured the FS thanks to the community
help. Greatly appreciate all.
Now I have some basic error:
I can dial out from extension 1000 (all default ext) to any number not in the
same network. I got the other number rung, and answered, but cannot hear
did you check our firewall? and various nat settings?
/b
On Jan 15, 2009, at 3:42 PM, Will Smith wrote:
Hi,
I have successfully installed and configured the FS thanks to the
community help. Greatly appreciate all.
Now I have some basic error:
I can dial out from extension 1000 (all
Thank you so much for responding.
Yes, I checked those, everything looks fine, and infact, if the audio stream is
blocked by firewall or nat setting, how can I inject the audio file and hear it
played on both ends. But as you suggest, I will doublecheck those values.
Thanks again
--- On Thu,
Hello again FreeSwitchers,
I have built the 1.02 on 10.4.11(OSX) and had no problems with that.
I have never been able to build from the SVN, but that is another story.
Now that I have migrated to 1.02 I was wondering if I can get some
help on a long standing issue I have with starting FS at
Lot's of experience and suggestions here. Thanks.
I believe it should be theoretically possible to have blip-free RTP
streaming through the appliance. Most windows ethernet drivers allow for
QoS packet scheduling. If the VMware network bridge driver honors this
and syncs the buffers at 20ms
I found this:
When I call the outside number, first, cannot hear or be heard, then when I put
the line on hold, the other party can hear the MOH, and when I switch it back,
now we can talk. Something goes wrong here
--- On Thu, 1/15/09, Brian West br...@freeswitch.org wrote:
From: Brian
open a jira and attach a svn diff and we'll have a look
thanks
On Thu, Jan 15, 2009 at 5:14 AM, Scott Ellis scott.el...@novatex.com.auwrote:
So I decided to hack the code to see if I could just get it to do what I
wanted - assuming some kind of error in the options setting.
First I changed
Have you looked at creating a system level startup item in
/Library/StartupItems ?
Also, to build from source you need the latest DevTools Kit from apple
installed. (I don't know if the latest will work w/ 10.4)
Ken
On 1/15/09 3:54 PM, Martin Joseph ast...@stillnewt.org wrote:
Hello again
Your build issue is with your autotools install, I have seen issues if
you have ever installed any of the autotools from macports or fink.
If you want to build from svn you can run bootstrap on another box (a
linux box perhaps) and then tar up that dir and move it to your mac.
We
Could you imagine a large software company saying anything other than
you have not supplied enough information for us to reproduce this bug?
Between the time wasted writing a longer response, and the image it
creates for clueless users/customers of the developers and the support
process, it just
Hi Tim,
I'm having exact same problem, try renaming mod_managed_lib.dll to
FreeSWITCH.Managed.dll and then load.
Michael confirmed this is supposed to be the case and is building a patch
for the Makefile.
However, when I do this on my Cent OS 5.2 it now loads successfully but
equivalent but that exists and still no a no go. Any ideas would be
very welcome? Thank you!Regards, -Adam -- next
part -- An HTML attachment was scrubbed... URL:
http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/73ac27e4/attachment-0001.html
but that
exists and still no a no go.
Any ideas would be very welcome? Thank you!
Regards,
-Adam
-- next part --
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--
Message: 3
Date: Thu, 15 Jan 2009 17:50:30 +1100
From: Scott Ellis scott.el...@novatex.com.au
Subject: [Freeswitch-users] zapata.conf
try http://files.freeswitch.org/freeswitch.msi
On Thu, Jan 15, 2009 at 2:03 PM, Terrance Harris tharris...@gmail.comwrote:
Hello,
I have recently found out about FS and how great it is.
We are trying to use FS as a voip client for radio shows.
We have been using Trixbox and Skype but Skype
://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/73ac27e4/attachment-0001.html
--
Message: 3
Date: Thu, 15 Jan 2009 17:50:30 +1100
From: Scott Ellis scott.el...@novatex.com.au
Subject: [Freeswitch-users] zapata.conf immediate=yes in Asterisk -
Freeswitch
is it only a problem in js
what if you call the bridge app in the dialplan?
On Thu, Jan 15, 2009 at 3:20 PM, Klaus Teller klaus.tel...@gmx.net wrote:
Hi,
Need your help on this. I have the following Javascript statement:
session.execute(bridge,sofia/gateway/sip.gafachi.com/someNumber) in a
Hi Anthony,
The problem exists also when i call
session.execute(bridge,sofia/gateway/sip.gafachi.com/number);
I tend to believe that this is a firewall issue. Would you confirm?
Klaus.
Original-Nachricht
Datum: Thu, 15 Jan 2009 17:42:46 -0600
Von: Anthony Minessale
Hello,
From what I heard celt isn't included in the most recent windows builds.
I would have to build FS and celt from the source to get it enabled.
On Thu, Jan 15, 2009 at 5:31 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
try http://files.freeswitch.org/freeswitch.msi
On Thu,
Yeah I compile mono ... i tried both 2.0.1 and 2.2 both error on loading the
mod_managed.
Tim
_
Windows Live™: Keep your life in sync.
I would like to be able to place a call on hold on one extension, walk
to another phone and then dial a sequence (like the barge sequence) say
55+extension number and have the call taken off hold and transferred to
the extension I am on.
Has anyone done this? (Before I try and work it out for
You would use a combination of storing the UUID... in the internal
db... see insert in the default dialplan... then a code to get that
out of the db... then run intercept on it using the value returned
from the db. See default config's
Store it something like this:
action application=db
Wouldnt that be call parking??
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_park
I have been told that would be better o use mod_fifo instead... It
would be nice if someone would post something on mod_fifo wiki page
about how to do fancy call parking with mod_fifo (even tho it might
Well, sorry. That would be better, wouldnt it?
http://wiki.freeswitch.org/wiki/Mod_fifo#Park_Time_Out_Example
Mesquita
On Jan 15, 2009, at 11:36 PM, Scott Ellis wrote:
I would like to be able to place a call on hold on one extension, walk
to another phone and then dial a sequence (like the
It is kind of - but slightly different, and simpler for the users.
Scott
Joo Mesquita wrote:
Wouldnt that be call parking??
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_park
I have been told that would be better o use mod_fifo instead... It
would be nice if someone would post
Thanks Brian, I had started looking at this, and I think I was heading
in the direction you describe - now I can pursue that with a bit more
confidence!
So even if we do not originate the call, the last dialled extension
would still be valid as it would be set up during the bridging process?
The key is the uuid.. In FreeSWITCH the uuid is the only bit you
really need to know to do anything with the session.
/b
On Jan 15, 2009, at 9:12 PM, Scott Ellis wrote:
Thanks Brian, I had started looking at this, and I think I was
heading in the direction you describe - now I can pursue
So for this scenario, I think I need to store the UUID of both sides
before every bridge that I do, that way it will always reflect the most
recently connected call to an extension - either as source or
destination.
I found the log action, so now I can spit out debug information as I
work
On Jan 15, 2009, at 3:10 PM, Michael Jerris wrote:
Your build issue is with your autotools install, I have seen issues if
you have ever installed any of the autotools from macports or fink.
I have never used Fink or Macports so that isn't it. In fact the
supposed statements made to the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
I found, that FS doesn't maintain D-Channel's state correctly.
I have a PRI with disabled layer 2 and 3 on TDM side. When FS starts up
I get this on console:
2009-01-16 08:16:10 [DEBUG] ozmod_isdn.c:1441 zap_isdn_run() ISDN thread
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