Hi Andrew,
Everything is running on an Ubuntu Hardy Xen domu with kernel
2.6.24-23-xen.
Erlang is version R12B5 and was compiled from source with options --
enable-hipe, --enable-smp-support en --enable-threads.
FS is trunk version 12197.
I did copy the configuration file to
freeswitch-users-boun...@lists.freeswitch.org wrote:
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Are you using cepstral 5.1?
There is a known issue with that release and it's closed
source so we
got this error when starting freeswitch -latest svn
2009-02-23 13:22:50 [CRIT] switch_loadable_module.c:840
switch_loadable_module_load_file() Error Loading module
/usr/local/freeswitch/mod/mod_sofia.so
**/usr/local/freeswitch/mod/mod_sofia.so: undefined
Hello,
today I found in FS logfile lines like this:
2009-02-23 15:27:12 [DEBUG] switch_ivr_originate.c:1605
switch_ivr_originate() Raw Codec Activation Success l...@8000hz 1 channel
20ms
It looks like L16 codec is used for incoming calls:
2009-02-23 15:27:03 [DEBUG] switch_core_session.c:523
I could compile and install FS 1.0.2 successsfully so, do I need to install
ODBC-devel package for 1.0.3 version?
Thanks,
Message: 5
Date: Thu, 19 Feb 2009 14:35:29 -0800
From: Michael Collins m...@freeswitch.org
Subject: Re: [Freeswitch-users] Realm Value
To:
You have something on your system thats causing the audio detect to
see you have odbc installed.. easiest way to get around this is to
just install the devel headers.
http://lists.freeswitch.org/pipermail/freeswitch-users/2009-February/011762.html
/b
On Feb 23, 2009, at 8:42 AM, Ali
On Feb 23, 2009, at 9:44 AM, Helmut Kuper wrote:
Hello,
today I found in FS logfile lines like this:
2009-02-23 15:27:12 [DEBUG] switch_ivr_originate.c:1605
switch_ivr_originate() Raw Codec Activation Success l...@8000hz 1
channel
20ms
It looks like L16 codec is used for incoming
I need someone with this issue to provide me ssh access to their box
so I can fix this problem for everyone. No one has done so yet.
Please find me on irc if you can provide access.
Mike
On Feb 23, 2009, at 10:16 AM, Brian West wrote:
You have something on your system thats causing the
Thanks Giovanni.
Were you planning to check in the sample skype.conf.xml into the default
FreeSWITCH conf folder? If so, just be aware the default config causes
freeswitch to hang right after a load mod_skypiax (if you do not have
skype running or specify a nonexistant skype user).
regards,
On Mon, Feb 23, 2009 at 11:09:28AM +0100, Leon de Rooij wrote:
Everything is running on an Ubuntu Hardy Xen domu with kernel
2.6.24-23-xen.
Oh, this might explain some things..
Erlang is version R12B5 and was compiled from source with options --
enable-hipe, --enable-smp-support en
Hi Mike,
thx. Today we had some failing test fax sessions (g711/PCMA) and my
first thought was that it could be caused by FS during transcoding. So I
looked into FS logfile and found those hints about transcoding. But
ringback shouldn't be the problem.
Fax path was from FAX device (source)
Hi Mike,
thx. Today we had some failing test fax sessions (g711/PCMA) and my
first thought was that it could be caused by FS during transcoding. So I
looked into FS logfile and found those hints about transcoding. But
ringback shouldn't be the problem.
Fax path was from FAX device (source)
Mesquita,
Relatively speaking, I feel like we are near the end of our project
roll out. Perhaps the case would be stronger once everything is
completed. At that time, I will be very glad to share the story on the
wiki -- and hopefully elsewhere!
Ben
On 2/21/2009 at 8:56 AM, João Mesquita
On Thu, Feb 19, 2009 at 5:04 AM, Rene Pankratz
r.pankr...@fh-wolfenbuettel.de wrote:
Hello,
when hanging up a call with portaudio automatically the next call that
is incoming or held is accepted.
Is it possible to configure PA that way, that after hanging up (doesn't
matter whether caller or
Hello,
if Cepstral 4.x is the way to go does anybody know where to get the demo
version?
BRs,
Claudio
I think you'll have to contact Cepstral on this one. I've tried to
find older revisions on their site and I can't find any way to get any
voices prior to 5.1.
-MC
Hello,
I run FreeSWITCH as a PBX solution for several companies, all sharing a
single server in a vritual pbx deployment. Dialplans and user directories
are all separate and handled per domains. Currently, there is about 250
phones set to use it, about 200 more will be migrated soon from Asterisk
Hi Guys
I'm having problems with seg faults about every 10 mins with call loads
200. I've processed the core dump
(http://pastebin.freeswitch.org/7436) but I'm unsure what I should be
looking for. I don't see the point where the crash occurred. Can
someone point me to where I should be
It looks like a file rewind operation.
does the lua script use the input callback to rewind a file?
It maybe be a race in some other thread can you paste a thread apply all
bt from the same core to look at the other threads.
On Mon, Feb 23, 2009 at 12:58 PM, Nik Middleton
Basically, you are trying to build what Empirix has with their Hammer tool.
You can create an application that is basically a mix of tshark and a
database feeder.
You sniff with tshark and going to basically pipe it to another application
that will read the pcap file, parse it, and load it into
Joseph Bajin napsal(a):
Basically, you are trying to build what Empirix has with their Hammer tool.
Thank you very much, Joseph, for your interest!
I have never heard about Empirix (I'll look at it), but what I'm trying
to build is something like SER/Kamailio/OpenSIPS sip_trace module.
You
Hello:
I have successfully loaded the Windows implementation (SVN 11602 -
02/02/09) from your site and it runs fine. I configured a Linksys SPA
2102 and have acquired dial tone and the '999X' tests work. I have not
been able to establish connection with either FreeWorldDialup or
Broadvoice
Leon,
I think I found the problem. I shouldn't have been defaulting to binding
to 127.0.0.1, instead the default should be 0.0.0.0. I've patched the
module to actually bind to 0.0.0.0 correctly and made it the default in
the config file. Erlang nodes by default bind to 0.0.0.0, so I decided
to
On Mon, Feb 23, 2009 at 4:47 PM, Stephen Walker swal...@sonasearch.comwrote:
Which files do I need to edit and what are the proper entries to enable
connection to FreeWorldDialup and Broadvoice? Example files and where they
reside in the file structure would be very much appreciated.
If you write it correctly it will work just fine. That is how most of
all the other correlation engines work. Your setup is not going to be
bigger than some of the large telecoms that use these systems today.
On 2/23/09, kokoska.rokoska kokoska.roko...@post.cz wrote:
Joseph Bajin napsal(a):
Joseph Bajin napsal(a):
If you write it correctly it will work just fine.
Yes, this is challenge I have talked about :-)
That is how most of
all the other correlation engines work.
I don't have enough informations but from what I heard from friendly
competitors they are usualy log (SIP|ISUP)
No, unfortunately the problem still persists. Portaudio still
automatically accepts/takes the next call.
René
On Thu, Feb 19, 2009 at 5:04 AM, Rene Pankratz
r.pankr...@fh-wolfenbuettel.de wrote:
Hello,
when hanging up a call with portaudio automatically the next call that
is incoming or
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