Thank you very much, Brian, for your help!
I have just upgraded to curent trunk, but all remains the same - RTPs
from both endpoints arrives to the FreeSWITCH but no RTPs are send back
from FS (using ngrep I can't see even single RTP packet comming from FS).
I think it is due to my
After some testing I found following:
1. When I try to bridge call between endpoints, there are always no RTPs
from FreeSWITCH (like I wrote before).
2. When I just answer call on FS and play something localy (i.e. music)
everything works fine - FS sends RTPs like expected.
All tested with
Brian/Bakko,
Would you please tell me which softphone are you using?
As you know, my own one is not working and when I tried tcp with xlite
(providing transport=tls) I see in wireshark that it is still
transporting it over udp(!)
thanks allot in advance,
Tzury
On Sun, Aug 16, 2009 at 5:11 PM,
Tzury Bar Yochay tzury...@reguluslabs.com wrote:
Brian/Bakko,
Would you please tell me which softphone are you using?
As you know, my own one is not working and when I tried tcp with xlite
(providing transport=tls) I see in wireshark that it is still
transporting it over udp(!)
I've
I've successfully used TLS with FreeSWITCH at both ends (yes, that's with
FreeSWITCH itself as the softphone).
Well, as I said at the beginning of this thread, TLS works fine for me.
The problem is when using TCP (transport=tcp and not transport=tls)
Tzury Bar Yochay tzury...@reguluslabs.com wrote:
Well, as I said at the beginning of this thread, TLS works fine for me.
The problem is when using TCP (transport=tcp and not transport=tls)
I'm not sure whether that's supposed to use TLS. I suspect not.
I'm not sure whether that's supposed to use TLS. I suspect not.
Jason,
I think I confused you with this TLS/TCP thing.
For the sake of clarification, I am talking about TCP and _not_ about TLS.
That is simply transporting the signaling packets over TCP instead of UDP.
No TLS should be involved
Hi All,
Shall I log a JIRA for this issue?
Thanks
Bruce
Bruce McAlister wrote:
Hi All,
I have been having difficulty trying to build FreeSWITCH 1.0.4pre9 and
1.0.4.
I am running on Solaris 10 Update 5 on x86 hardware (32-bit).
The build fails with:
--- snip ---
make: Fatal
Tzury Bar Yochay tzury...@reguluslabs.com wrote:
I think I confused you with this TLS/TCP thing.
For the sake of clarification, I am talking about TCP and _not_ about TLS.
That is simply transporting the signaling packets over TCP instead of UDP.
No TLS should be involved at this stage. It
hi MikeJ,
i prefer creating MSI file that is easy to mintin$
unstid of using inno setup or advanced installer (not free), we can use
WIX (Windows installer XML) that is a open source one
we can create a customised MSI that fully install mor features,
including Sounds / MOH/...
and we can edit
Y make my tests with eyebeam.
I thing X-lite dont't support TCP transport.
BR
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Hi, could you please check the destination number in your dial string? If it
is the right format, one of the reasons could be that number is not in
service
when you get 403 response from the SIP gateway.
Thanks,
Chris
On Mon, Aug 17, 2009 at 8:10 AM, NOx-WHV enno.egb...@googlemail.com wrote:
You need to set from-domain on the gateway. And set the
effective_caller_id_number to just the number not the num...@host.
/b
On Aug 17, 2009, at 7:10 AM, NOx-WHV wrote:
Hi,
i have a problem using my freeswitch with a sipgate account. The
gateway
entry is o.k.. Freeswitch try a call,
Update, Math refactored the functions that caused some of the problems.
/b
On Aug 17, 2009, at 1:29 AM, kokoska rokoska wrote:
After some testing I found following:
1. When I try to bridge call between endpoints, there are always no
RTPs
from FreeSWITCH (like I wrote before).
2. When I
They make you pay for TCP which is weird since the RFC says TCP is
required.
/b
On Aug 17, 2009, at 7:18 AM, bakko wrote:
Y make my tests with eyebeam.
I thing X-lite dont't support TCP transport.
BR
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Hi,
i have just taken some pictures.
http://www.nabble.com/file/p25006858/pic1.JPG pic1.JPG
http://www.nabble.com/file/p25006858/pic2.JPG pic2.JPG
http://www.nabble.com/file/p25006858/pic3.JPG pic3.JPG
the third picture is taken by a softphone that works.
To Brian:
If i set the
FreeSWITCH works very well as a client :P
/b
On Aug 17, 2009, at 8:53 AM, Tzury Bar Yochay wrote:
my conclusion is then:
we are living in a sad world where there is no a production ready,
cross platform, fully compliant, decent and open source sip client
Well if you pay attention I told you in the last email... set the
param from-domain on the gateway to sipgate.de
/b
On Aug 17, 2009, at 8:36 AM, NOx-WHV wrote:
Hi,
i have just taken some pictures.
http://www.nabble.com/file/p25006858/pic1.JPG pic1.JPG
Brian West napsal(a):
Update, Math refactored the functions that caused some of the problems.
/b
Still no luck - no RTPs from FreeSWITCH.
For sure I'll make fresh svn checkout in a minute and let you know if it
helps...
Thank you very much for your help!
Best regards,
kokoska.rokoska
FreeSWITCH works very well as a client :P
I am currently porting it into iPhone and Symbian, I am almost done ;-)
anyway, seriously now, can one point to a wiki page about this?
How do I do that?
I would need 3 server instances to place a call, right?
if all else fails get me access to the machine please.
/b
On Aug 17, 2009, at 9:09 AM, kokoska rokoska wrote:
Still no luck - no RTPs from FreeSWITCH.
For sure I'll make fresh svn checkout in a minute and let you know
if it
helps...
Thank you very much for your help!
Best regards,
Well if you append ;transport=tcp on the bridge lines it will use TCP .
/b
On Aug 17, 2009, at 9:06 AM, Tzury Bar Yochay wrote:
FreeSWITCH works very well as a client :P
I am currently porting it into iPhone and Symbian, I am almost
done ;-)
anyway, seriously now, can one point to a wiki
On Aug 17, 2009, at 5:29 AM, Meftah Tayeb wrote:
hi MikeJ,
i prefer creating MSI file that is easy to mintin$
How is this useful vs. something that is already maintained?
unstid of using inno setup or advanced installer (not free), we can
use WIX (Windows installer XML) that is a open
On Sun, Aug 16, 2009 at 4:24 AM, Woody Dickson woodydick...@gmail.comwrote:
Hello,
I find hangup_hook, but I would like to define different actions for
different hangup codes. Is there anyway to do that?
I can think of at least two ways you could do this: one that uses only the
dialplan
Brian West napsal(a):
if all else fails get me access to the machine please.
/b
I'm sorry, but even after fresh svn checkout all goes wrong :-)
If you be so kind to look at that machine, I'll be very glad.
I still think it will be some trivial, stupid, misconfiguration, but
can't find
shinzon.pub and get on IRC and msg bkw_ and i'll take a look.
/b
On Aug 17, 2009, at 11:46 AM, kokoska rokoska wrote:
I'm sorry, but even after fresh svn checkout all goes wrong :-)
If you be so kind to look at that machine, I'll be very glad.
I still think it will be some trivial, stupid,
Brian West napsal(a):
shinzon.pub and get on IRC and msg bkw_ and i'll take a look.
/b
OK, I do it.
But wait few minutes, please:
1. I should asleep my chidren :-)
2. Before a while I discovered that if I disable IPv6 networking in my
CentOS, the RTPs from FreeSWITCH works great (FS
Let me apologize to waste your time, Brian (and others too).
I'm still not 100% sure where problem is, but I'm nearly sure it is not
related to FreeSWITCH.
I found similar problems with SMB - with IPv6 active some clients (even
with only IPv4 stack) can't connect to server. Disabling IPv6
I found this same issue on my machine.
If you could compile a esl_wrap.o then you have to generate a libesl.so with
a cmd like this:
g++ -shared esl_wrap.o -o libesl.so
Then in your code, do something like this:
/* Test.java */
import org.freeswitch.esl.*;
class Test
{
public static void
can someone post a patch to that makefile to jira.freeswitch.org please.
Mike
On Aug 17, 2009, at 4:04 PM, Fernando Testa wrote:
I found this same issue on my machine.
If you could compile a esl_wrap.o then you have to generate a
libesl.so with a cmd like this:
g++ -shared esl_wrap.o -o
I used to be able to dial 88+extension to eavesdrop, but now it is killed
right after the call is answered by the extension. Can anyone tell me what I
have done wrong?
I am running version 14534 on Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7
10:39:21 EDT 2009 i686 i686 i386 GNU/Linux.
To overcome the G729 transcoding issue with voicemail, I'm using an Audiocodes
Mediant 1000 for transcoding. Our SIP trunk provider and all of our phones use
G729 exclusively.
When a call needs to go to voicemail, the call is bridged to the M1000, which
transcodes to G711, and returns the call
Matt,
Okay the good news is vmd should be able to handle these cases. The
bad news is for whatever reason they are not getting detected at the
moment.
vmd-not-panasonic-home-ans.wav is a sine at ~1400Hz you can change
MAX_FREQ to 1450 and play with MIN_AMPL if that still doesn't help.
The
Matt,
You must first capture the audio beeps and verify that they are sine
waves. If not, simply tweaking the algorithm will not give you better
results.
It might be possible to use FFT and I would be happy to help you
implement such a solution but keep in mind FFT is very very demanding
on the
On Mon, Aug 17, 2009 at 1:18 PM, Lars Zeb larc...@yahoo.com wrote:
I used to be able to dial 88+extension to eavesdrop, but now it is killed
right after the call is answered by the extension. Can anyone tell me what I
have done wrong?
I am running version 14534 on Linux fs
Done!
Check out patch at http://jira.freeswitch.org/browse/FSBUILD-185
Testa
On Mon, Aug 17, 2009 at 5:18 PM, Michael Jerris m...@jerris.com wrote:
can someone post a patch to that makefile to jira.freeswitch.org please.
Mike
On Aug 17, 2009, at 4:04 PM, Fernando Testa wrote:
I found this
Hi FS Team,
Thanks for your great work at Cluecon, we came back to Brazil with some
good ideas (+ Snom 360 + Sangoma B600) ;-)
See you again next year!
Rodrigo Telles
Devel-IT
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Thanks, I'll get that merged in and clear from this we need to start
using proper autoconf checks for java.
mike
On Aug 17, 2009, at 5:35 PM, Fernando Testa wrote:
Done!
Check out patch at http://jira.freeswitch.org/browse/FSBUILD-185
Testa
On Mon, Aug 17, 2009 at 5:18 PM, Michael Jerris
On Aug 17, 2009, at 4:30 PM, Justin Miller wrote:
Is there a way to end the transcoded call legs, and bridge to the
phone from the original call leg? This would free up the M1000, and
just seems like a better way to do things.
You might consider using a REFER if your endpoints support it.
Hi,
I'm trying to have an audio as ringback (WAV, 8khz, mono) when originating a
call. Unfortunately, it doesn't seem to work; the RINGING teletone is being
used instead of my audio. I have debug level enabled on Freeswitch but i
don't find anything suspicious.
is there anything specific i
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