mszla...@aol.com pisze:
I have to change $${local_ip_v4} to 10.0.0.3 every time I update to a
new trunk and I have to go through vars.xml, etc changing
$${local_ip_v4} like you did.
Is there a way to change $${local_ip_v4} in one place. That way one
wouldn't have remember all the locations
I'm not quite sure what your asking.
Are you saying that I could run the latest FS svn but in a way that uses my
older configuration files? If so then I don't, and don't know how ... blush
blush.
If that's the easiest thing to do then please tell me how.
Thanks. Mark.
-Original
John,
Okay, a few things. First off, the wanpipe2.conf file has a booboo. This
line is WRONG:
TDMV_DCHAN = 0
For ISDN in North America you want:
TDMV_DCHAN = 24
Also, I recommend changing this line:
wbg1 = wanpipe2, , TDM_VOICE, Comment
To this:
wbg1 = wanpipe2, , TDM_VOICE_API,
mszla...@aol.com pisze:
I'm not quite sure what your asking.
Are you saying that I could run the latest FS svn but in a way that uses
my older configuration files? If so then I don't, and don't know how
... blush blush.
If that's the easiest thing to do then please tell me how.
Thanks.
Peter P GMX pisze:
I just wanted to know, how much memory overall is consumed by FS inkl.
all Libraries (when used on a Netbook with limited memory), so RES does
only show a portion of the overall RAM, FS uses incl. libraries.
So I did the following:
I restarted my laptop and noted the used
I just wanted to know, how much memory overall is consumed by FS inkl.
all Libraries (when used on a Netbook with limited memory), so RES does
only show a portion of the overall RAM, FS uses incl. libraries.
So I did the following:
I restarted my laptop and noted the used memory.
I deactivated
That was it.
I installed the hd sounds and it works now.
Thanks.
Brian West schrieb:
Chances are he just doesn't have the 16k sound files installed.
/b
On Apr 7, 2009, at 6:35 PM, Michael Collins wrote:
2009-04-08 00:43:24 [ERR] mod_sndfile.c:194 sndfile_file_open() Error
I track channels via mod_socket_event, I saw in source there is such flag
CF_VERBOSE_EVENTS to make all channel related events
contain extended data. Is it possible to set it via 'originate' or 'conference
xxx dial' commands.
This would ease my system. In scenario when I call user which is not
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello Anthony,
after one day running the actual trunk things looks much better than
before. FS started 24h ago with 129MB VRAM and grows to 136MB VRAM by
now. CPU is around 1.3%
Thanks for your work!
regards
helmut
On 06.04.2009 18:09, Helmut
Simply do a make current without a make samples. That way the conf
files in /usr/local/freeswitch/conf remain untouched.
I really very seldomly update the conf files.
Best regards
Peter
mszla...@aol.com schrieb:
I'm not quite sure what your asking.
Are you saying that I could run the latest
There is a dp_tools verbose_events can set that flag, you may try to
transfer into a dialplan or use the inline dialplan
try this, not tested.
originate sofia/gateway/my_gw/u...@domain.com
'verbose_events,playback:foo.wav,echo' inline
Here's the first draft: http://wiki.freeswitch.org/wiki/OpenWrt
Carlos
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Hi!
I wonder how to get out the member_id of a member in a conference room?
Currently, i found that it is increasing 1 each time, so the first one enter
the conference room, his/her member_id would be 1, and the second is 2. Sadly,
when i close a conference room (all members leave) and
Hi there,
I'm quite a newbie about freeswitch. I have an application (IVR) that
needs to have endpoints SIP to register,answer the calls and transfer them
to the right phones.(I( have my own SIP server).Moreover it needs also a
ASR/TTS API' set to communicate with my ASR/TTS engine ( just for
Thanks for the response Traun. The version of Python is 2.4.3, and I didn't
build it myself, I installed it with yum.
The version of Red Hat is 4.1.2-41.
import threading works fine, so I don't think it's a Python threading
issue.
The FreeSWITCH version I installed is the
to vietduc pisze:
Hi!
I wonder how to get out the member_id of a member in a conference room?
Currently, i found that it is increasing 1 each time, so the first one enter
the conference room, his/her member_id would be 1, and the second is 2.
Sadly, when i close a conference room (all
Hi,
I have mod_shout installed and I'm using session.recordFile to capture the
audio in a call. When I specify a local file mp3 or wav the audio is
captured fine. However, I'm using an icecast server to manage the audio for
me and when I specify a remote mp3
It's linux, yes.
The way I got around the problem that memory may not be freed is:
* to reboot the system.
* look for used memory
* start FS
* look for used memory
* calculate the difference
That way it showed 24-25M which I can understand.
Best regards
Peter
Szymon Olko
Ciao Michele,
as a start is definitely better (and more gratifying) that you runs FreeSWITCH.
Then, if (and only if) there is a compelling reason that justify the
amount of time needed to develop a standalone application, go for it.
Sincerely,
Giovanni Maruzzelli
Peter P GMX pisze:
It's linux, yes.
The way I got around the problem that memory may not be freed is:
* to reboot the system.
* look for used memory
* start FS
* look for used memory
* calculate the difference
That way it showed 24-25M which I can understand.
I
mszla...@aol.com pisze:
OK , you're SVN updating on a Linux system but I'm using Windows. The
very few times I tried with Tortoise SVN I ran into problems were it
would fail because of some path not being present or some strange symbol
in a file or something else. Since I'm not experienced
You know if you keep doing a fresh checkout every single time then you
are wasting bandwidth... if its your only choice then do that but I
highly recommend you learn to use the tools properly. Our bandwidth
is kindly provided by Bandwidth.com and I would hate to just waste it
for no
Na jakiś czas, będę uczyć się, jak radzić sobie z Tortoise SVN błędów.
Mój polski nie jest zbyt dobre, ale dziękuję. Google pomaga.
-Original Message-
From: Szymon Olko so...@gcdf.pl
To: freeswitch-users@lists.freeswitch.org
Sent: Wed, 8 Apr 2009 10:36 am
Subject: Re:
We're seeing occasional one way audio issues for international calls
going out to one of several carriers. On roughly 2 out of 5 calls
outbound, there is no audio on the the calling party's side, however the
called party indicates they can hear the calling party perfectly well.
NAT is not involved
Brian West pisze:
You know if you keep doing a fresh checkout every single time then you
are wasting bandwidth... if its your only choice then do that but I
highly recommend you learn to use the tools properly. Our bandwidth is
kindly provided by Bandwidth.com and I would hate to just waste
Where are you?
/b
On Apr 8, 2009, at 1:14 PM, Szymon Olko wrote:
Regarding Cluecon, I would like to meet you all there, but in this
year it's to expensive and too far for me.
Szymon
Brian West
br...@freeswitch.org
-- Meet us a ClueCon! http://www.cluecon.com
Do you have any reason to be doing proxy media?
/b
On Apr 8, 2009, at 1:08 PM, Gabriel Kuri wrote:
I can turn off proxy_media and run a pcapsipdump if that will help?
Brian West
br...@freeswitch.org
-- Meet us a ClueCon! http://www.cluecon.com
Brian West wrote:
Do you have any reason to be doing proxy media?
no, not other than to fix the one way audio issue :)
I'd rather leave proxy_media off.
Gabe
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
Brian West pisze:
Where are you?
Poland, Wrocław.
http://maps.google.pl/maps?f=qsource=s_qhl=plgeocode=q=poland,+wroc%C5%82awsll=52.025459,19.204102sspn=6.979078,18.017578ie=UTF8ll=51.107833,17.038422spn=0.222023,0.563049z=11
/b
On Apr 8, 2009, at 1:14 PM, Szymon Olko wrote:
Regarding
Hello everyone,
I am trying to get a Java Sip client working with speex/16000.
FS sets the codec correctly and then starts sending packets to my client:
2009-04-08 21:46:34 [DEBUG] sofia_glue.c:2732 sofia_glue_negotiate_sdp() Audio
Codec Compare [speex:100:16000:0]/[SPEEX:99:16000:20]
Hi Adam,
I'm stumped .. I guess you could try the following:
* Try with the trunk version of freeswitch. I don't think it will matter,
but just in case
* Try to simulate the same test with a Lua script. Do you see the same
problem?
If those don't turn up anything, then the next logical step
Okay, a few things. First off, the wanpipe2.conf file has a booboo.
Don't think so.
This line is WRONG:
TDMV_DCHAN = 0
Not exactly. My understanding is you can use either:
wanpipeX.conf: TDMV_DCHAN = 0
zaptel.conf: dchan = 24 (or in our case 48 since it's the second span)
which
I want to run a script with a scheduler but I'm having a problem with how to
set up the originate in Javascript.
The originate would go something like:
originate
{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/12223334...@10.0.0.5:5061
GINO_ANS
I can get
Traun, thanks again for your help.
I followed your advice and I made some progress!
I tested with the latest trunk version and also with 1.0.2, and both
exhibited the same behavior.
I then tried writing a test script in Lua, and it worked fine.
So this meant the problem was in the Python module
This is because the Polycom doesn't support STUN, RPORT or any other
nat traversal technology. You have a couple of choices please review http://wiki.freeswitch.org/wiki/NAT_Traversal
Also review the NDLB-force-rport option for the sofia profile to
assume rport. CAUTION this breaks things
35 matches
Mail list logo
