voyage linux is a stripped debian and i was using it on an alix board some
time ago... Asterisk was compiling on that without any issue. I beleive FS
will do the same.
T.
On Fri, Dec 11, 2009 at 2:57 AM, Brian May
br...@microcomaustralia.com.auwrote:
On Thu, Dec 10, 2009 at 03:53:32PM +1100,
Kristian,
from your experience, supposed we go for net5501 + a 4 - 8 FXS card, what is
the maximum simultaneous calls that this box can handle of course using g729
codec?
I used blackgin (IP08), alix2d3... and all of them were giving up on 6-7
simultaneous calls.
To be honest, i didnt run
ok, but how much smultaneous calls did you get on an alix board using
astlinux... for istnace, this is the question?
T.
On Thu, Dec 10, 2009 at 5:12 PM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
On Thu, Dec 10, 2009 at 9:26 AM, Frank Carmickle fr...@carmickle.com
wrote:
Hi Mike,
Lets suppose we have:
- 2 machines configured for high availability (LAN HA) in a master/slave
configuration with a floating public address on the master. (
http://www.ultramonkey.org/3/topologies/ha-overview.html)
- freeswitch installed on every machine configured to use
On Fri, Nov 27, 2009 at 11:00 AM, Steve Kurzeja steve.kurz...@gmail.comwrote:
Is that USD ? :)
i believe these are not Turkish liras
:P
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris m...@jerris.com wrote:
http://dev.mysql.com/doc/refman/5.1/en/connector-odbc-news-3-51-18.html
MySQL Connector/ODBC now supports batched statements. In order to enable
cached statement support you must switch enable the batched
this is how i do it from the dialplan:
extension name=ServiceLookup
condition field=destination_number
expression=^(300030)(.*)|^\+(300030)(.*)
action application=set data=bPfx=$1$3/
action application=set data=bNum=$2$4/
action inline=true application=set
after attended
transfer...
On Thu, Oct 15, 2009 at 11:54 AM, Tihomir Culjaga tculj...@gmail.com
wrote:
hi, any clue when can t38 be added?
Eventually. :) Of course, if we could get more to add to the bounty it
might grease the wheels of innovation.
http://wiki.freeswitch.org/wiki
Hi,
just a thing i noticed... the debug log and sip trace have different time
... one hour difference ... looks like UTC/GMT issue.
where do i set the time for siptrace correctly ?
2009-11-16 09:47:13.779070 [DEBUG] switch_core_state_machine.c:411
Brian is right,
pls, lets stop with exceptions and get stick to RFCs... otherwise it will be
a big mess ...
T.
On Wed, Nov 4, 2009 at 3:03 PM, Brian West br...@freeswitch.org wrote:
I'm going to say No!
/b
On Nov 4, 2009, at 2:23 AM, Dennis wrote:
is there a way to send something like
regards
Dave
- Original Message -
*From:* Tihomir Culjaga tculj...@gmail.com
*To:* freeswitch-users@lists.freeswitch.org
*Sent:* Tuesday, November 03, 2009 7:53 PM
*Subject:* Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with
FreeSwitch
you might read this before you bigin :P
regards
Dave
- Original Message -
*From:* Tihomir Culjaga tculj...@gmail.com
*To:* freeswitch-users@lists.freeswitch.org
*Sent:* Tuesday, November 03, 2009 7:53 PM
*Subject:* Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with
FreeSwitch
you might read this before you bigin :P
just an off-topic question but it concenns mass provissioning ... does
anyone know if there is an open TR069 platform we can work on?
T.
On Wed, Nov 4, 2009 at 1:16 AM, Michael Collins m...@freeswitch.org wrote:
On Tue, Nov 3, 2009 at 11:11 AM, Kristian Kielhofner
i tought so :PP
T.
On Sun, Nov 1, 2009 at 6:34 AM, Michael Jerris m...@jerris.com wrote:
This is a non working module, just a shell for development.
Mike
On Oct 30, 2009, at 5:52 PM, Tihomir Culjaga wrote:
does anybody know how does it work and how to use it in a dialplan
does anybody know how does it work and how to use it in a dialplan?
freeswi...@nemesis
freeswi...@nemesis
freeswi...@nemesis load mod_t38gateway
API CALL [load(mod_t38gateway)] output:
+OK
2009-10-30 22:44:38.204268 [NOTICE] mod_t38gateway.c:147 T.38 gateway
enabled
2009-10-30 22:44:38.204268
Handling of fastStart in CallProceeding is commented out in h323plus
library,
this is exploration from h323plus developers about this:
Yes that should be mera.
The problem is that Callproceeding does not always come from the remote it
may be generated by the gatekeeper.
this is a
P.S. people from russian community report what current version of module
work fine on fs
trunk version.
that's strange that they report it working as m_txAudioOpened is never
gonna be ready in pre_answer :P... i had to comment it to make it working.
anyhow, i moved everything to trunk
On Mon, Oct 26, 2009 at 4:41 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
The headers are used to pass the callee-id info back to the other side so
you have the id of who you called.
The standards have failed us in this case as everything does it differently
to the point that
On Mon, Oct 26, 2009 at 7:26 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
This is ridiculous but here it is
try r15230
add the profile param
param name=pass-callee-id value=false/
sorry for that but, this will save you a lot of e-mail explaining why calls
are not going
If one of the computers does a big download, it messes with FS in two ways.
If a connection is made, the voices are broken up, intermittent and
difficult to understand. If the download is long enough, the connection to
Flowroute is no longer usable due to registration failure.
In any
TCBTW: it really doesn't have sense to develop on 1.0.4 ... the proof of
TCconcept was done. I'm able to place calls in both directions so, lets
move
TCto trunk now.
i have my own voip infrastructure on my work, and it's better for me to use
it for tests,
and use my personal on work too, i
TC
TCit is gonna be easier to track.
TC
TCTomorrow i will test on 1.0.4 but please lets move to trunk.
i make it a bit later, to move tickets to jira and source to svn i
need some time to undertand how this system is works, especially jira.
audio issue is better now :)
however i have a
btw you are back with an old issue:
static const char modulename[] = h323;
static const char* h323_formats[] = {
G.711-ALaw-64k, PCMU,
G.711-uLaw-64k, PCMA,
GSM-06.10, GSM,
MS-GSM, msgsm,
SpeexNarrow, speex,
___
FreeSWITCH-users
i meant you switched PCMA and PCMU...
T.
2009/10/23 Georgiewskiy Yuriy bottle...@icf.org.ru
On 2009-10-23 10:16 +0200, Tihomir Culjaga wrote
freeswitch-us...@lists.fre...:
TC TC
TC TCit is gonna be easier to track.
TC TC
TC TCTomorrow i will test on 1.0.4 but please lets move to trunk
TC3. can we control mediaWaitForConnect flag within setup message via
config
TCfile setting?
what is mediaWaitForConnect flag, may be another trmin in h323?
If the calling endpoint sets the mediaWaitForConnect element to TRUE in the
Setup message, then
the called endpoint shall not send
TCcheck H225_Setup_UUIE H323SignalPDU::BuildSetup within
src/h323pdu.cxx
TC(H323plus)
i think it can be implemented later, but, why it may be needed? can you
explain some
situation where it need?
TC
TCyou should handle this and postpone pre_answer until you get an open LC.
TC
TCbool FSH323Connection::OnReceivedProgress(const H323SignalPDU pdu)
TC{
TCPTRACE(4,
mod_h323\t==FSH323Connection::OnReceivedProgress);
TC
TCPTRACE(4, mod_h323\t==FSH323Connection::OnReceivedProgress
-
TCdisabled pre_answer);
TC
TC
a solution to H323 endpoint = FS = SIP user no audio issue
is to disable a wait for tx Audio ... for case
SWITCH_MESSAGE_INDICATE_ANSWER:{
//m_txAudioOpened.Wait();
case SWITCH_MESSAGE_INDICATE_ANSWER:{
switch_log_printf(SWITCH_CHANNEL_LOG,
2009/10/23 Georgiewskiy Yuriy bottle...@icf.org.ru
On 2009-10-23 10:37 -0500, Anthony Minessale wrote
freeswitch-us...@lists.f...:
i have no way to install trunk at this time, i will go out of hospital
about one week later, after
this i will can try it on trunk.
AMif you were on trunk
TC
TCI have enabled crash-protection and when i do SIP = H323 call it
doesn't
TCgenerate coredumps... it is just this thread that is crashing ... pls
check
TCthe log downbelow:
core dump in case enabled crash-protection may be unusable, i have a case
then
my module crash silently,
TCHi, here is the FS log without crash-protection:
TChttp://pastebin.freeswitch.org/10796 and here is the backtrace:
TChttp://pastebin.freeswitch.org/10797
i fix this crash already, please download latest version from same link
as previous, recompile and try again.
outgoing works, I can
TC
TCDo you need some logs ?
try disable medai-proxy, there is issue with rtp now then medai-proxy or
transcoding enabled.
Outbound calls:
disabled rtp proxy and it is still the same issue ... audio delay H323 =
SIP endpoint.
Inbound calls:
This is the extension i use to register
2009/10/22 Georgiewskiy Yuriy bottle...@icf.org.ru
On 2009-10-22 09:27 -0500, Anthony Minessale wrote
freeswitch-us...@lists.f...:
AMcrash protection has been completely removed from FreeSWITCH, I
certianly
AMhope you are working on this against SVN trunk?
i don't have trunk at this time,
On Thu, Oct 22, 2009 at 5:44 PM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
An update for Tony, Brian, Mike, and everyone on the list...
I was able to get some phone time with the team yesterday. Tony
worked on my machine, found the issue, and had it committed within 30
is strictly turned off on the machine.
Gabe
Kristian Kielhofner wrote:
On Thu, Oct 22, 2009 at 11:58 AM, Tihomir Culjaga tculj...@gmail.com
wrote:
One fax machine here in the office (still testing others) correctly
sends all fax pages. A minute or so after the fax is marked
2009/10/22 Georgiewskiy Yuriy bottle...@icf.org.ru
On 2009-10-22 16:53 +0200, Tihomir Culjaga wrote
freeswitch-us...@lists.fre...:
finally i fix this rtp bug, check new wersion please.
if course i can do that, but tomorrow morning ... i'm not in the office
anymore.
BTW: can we please move
simple:
action application=bridge data=h323/${number}/
if fs not registered on gk then data=h323/${numb...@xxx.xxx.xxx.xxx.
TC
TC2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3252 Found capability:
TCUserInput/PointDevice 14
TC2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3248
consider this:
context name=SIP_incoming
extension name=call-sip-extensions
condition field=destination_number expression=^(\d+)$
action application=set data=AUTHENTICATION_STATUS=0/
action application=transfer data=${AUTHENTICATION_STATUS} XML
Authen_Status/
2009/10/21 Georgiewskiy Yuriy bottle...@icf.org.ru
On 2009-10-21 09:34 +0200, Tihomir Culjaga wrote
freeswitch-us...@lists.fre...:
TC
TC
TC
TCI was using latest libpt.so.2.7-beta1.
TC
TCNow I went back to libpt.so.2.6-beta6 (can't find 2.6.5 you
mentioned...)
TCand FS is crashing
it depends of what you are trying to acheave one approach is with regex
check this: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_regex
you can set a different variable and have it true or false ... than you can
compare for false state...
well .. it is up to you ...
T.
On Wed,
FS = 10.4.62.7
SIP phone = 10.4.62.89
H323 endpoint = 10.1.14.153
TC2. hangup from sip side doesn't release the h323 leg (now the difference
is
TCthat FS is not complaining about thread mismatch ant it looks clean but
FS
TCdoesn't send any releasecomplete message... strange)
TC3.
TC
TCcall flow is SIP_user = FS = H323_endpoint is failing ..
coredumped
TChttp://pastebin.freeswitch.org/10703
i fix some bugs now,
ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/mod_h323.tar.bz2 this
is
updated version, try it, if you experience no audio try enable rtp proxy in
you are making FS to play wav file when sending a call in G711 or GSM or
some other codec.
you might use mod_native_filehttp://wiki.freeswitch.org/wiki/Mod_native_fileto
avoid transcoding.
T.
On Tue, Oct 20, 2009 at 9:56 PM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
Hello
Of course, I was listening to my A.M radio the other day and they said that
there was this new invention called the Internet that would let people send
documents to each other electronically. Maybe you should look into that.
Next thing you know they'll come up with telephones that people don't
hi, any clue when can t38 be added?
T.
On Thu, Oct 15, 2009 at 3:57 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
This is a known limitation until we add actual t38 support to the project.
On Wed, Oct 14, 2009 at 6:56 PM, Klaus Hochlehnert maili...@kh-dev.dewrote:
Hi,
?
-metik
- Original Message -
*From:* Tihomir Culjaga tculj...@gmail.com
*To:* freeswitch-users@lists.freeswitch.org
*Sent:* Tuesday, October 13, 2009 3:24 PM
*Subject:* Re: [Freeswitch-users] SIP Overlap support?
i never found it working properly... i always had some interoperability
2009/10/14 Georgiewskiy Yuriy bottle...@icf.org.ru
On 2009-10-14 08:59 +0200, Tihomir Culjaga wrote
freeswitch-us...@lists.fre...:
try sow start on h323 channel, there is a bug in faststart, i will fix it
later.
there are few things,
1. capability PCMU/PCMA needs to be inverted
2. when
2009/10/14 Georgiewskiy Yuriy bottle...@icf.org.ru
On 2009-10-14 09:58 +0200, Tihomir Culjaga wrote
freeswitch-us...@lists.fre...:
i need trace level 4 from mod_h323 and debug log of entire call, tcpdump
may be needed later, i have no way
to test it on this time, i do it later.
Ok
On Wed, Oct 14, 2009 at 10:16 AM, Tihomir Culjaga tculj...@gmail.comwrote:
2009/10/14 Georgiewskiy Yuriy bottle...@icf.org.ru
On 2009-10-14 09:58 +0200, Tihomir Culjaga wrote
freeswitch-us...@lists.fre...:
i need trace level 4 from mod_h323 and debug log of entire call, tcpdump
may
tracking
resources and not fragment the community if possible.
/b
On Oct 12, 2009, at 2:43 PM, Tihomir Culjaga wrote:
hi,
finally i compiled it right ... had a stupid issue with ekiga and
wrong ptlib in place...
anyhow, i loaded the module and will continue the tests
tomorrow ...first
On Tue, Oct 13, 2009 at 8:31 AM, Brian West br...@freeswitch.org wrote:
I wouldn't call it donating per se... Its just giving it a place to
live with easy access for end users without having to do anything
extra go get it! ;)
/b
I agree with you Brian.
what about some console logs sip traces ?
T.
On Tue, Oct 13, 2009 at 10:56 AM, srinivasula reddy
srinivas.ksvre...@gmail.com wrote:
Hi,
two users are registered in freeswitch, when i making call to another user
i am getting 606 error,
any help
--
Srinivasula Reddy K
Temporarily unavailable with reason header cause=606;
text=user-not-registered. This also happened with other consoles.
Thanks
SRINIVAS
On Tue, Oct 13, 2009 at 2:35 PM, Tihomir Culjaga tculj...@gmail.comwrote:
what about some console logs sip traces ?
T.
On Tue, Oct 13, 2009 at 10:56 AM
to reuse our issue tracking
SD resources and not fragment the community if possible.
SD
SD /b
SD
SD On Oct 12, 2009, at 2:43 PM, Tihomir Culjaga wrote:
SD
SD hi,
SD
SD finally i compiled it right ... had a stupid issue with ekiga and
SD wrong ptlib in place...
SD
SD anyhow, i loaded
the wireshark file, any help?
thanks
srinivas
On Tue, Oct 13, 2009 at 4:02 PM, Tihomir Culjaga tculj...@gmail.comwrote:
and you are sure both users are registered to the same context and your
dialplan is correct ?
T.
On Tue, Oct 13, 2009 at 11:13 AM, srinivasula reddy
srinivas.ksvre
2009/10/13 Georgiewskiy Yuriy bottle...@icf.org.ru
On 2009-10-13 13:35 +0200, Tihomir Culjaga wrote
freeswitch-us...@lists.fre...:
this morning me bring in hospital, and now i cannot make much work,
i think return to the ranks in 1-2 week.
damn, hope you will recover soon... take it easy
you need a softswitch i'm afraid a SIP phone is not designed for
overlap...
T.
On Tue, Oct 13, 2009 at 5:26 PM, Dennis oderm...@googlemail.com wrote:
how could we try? we played arround with a snom phone (snom seems to
support something in this direction, but are not shure, how we can
?
i do think some softphone can do it but i forgot which one it was either
snom or grandstream
On Tue, Oct 13, 2009 at 12:12 PM, Tihomir Culjaga tculj...@gmail.comwrote:
you need a softswitch i'm afraid a SIP phone is not designed for
overlap...
T.
On Tue, Oct 13, 2009 at 5:26 PM
2009/10/12 Georgiewskiy Yuriy bottle...@icf.org.ru
On 2009-10-08 20:32 +0200, Tihomir Culjaga wrote
freeswitch-us...@lists.fre...:
TCHi Yuriy,
TC
TCcan you share what you have so far, I'm sure we can help with RTP
part...
ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code
hi,
can't make it...
subZero:~/freeswitch-trunk$ make mod_h323
making all mod_h323
Compiling mod_h323.cpp...
quiet_libtool: compile: g++ -g -ggdb -I/usr/local/include/ptlib
-I/usr/local/include/openh323 -I. -DPTRACING=1 -D_REENTRANT -fno-exceptions
-I/home/tculjaga/freeswitch-trunk/src/include
this is up to your phone # means address complete and you phone sends
the number you dialed into an INVITE message.
if you want to support FAC with # you should modify the phone's dialplan and
make it expect more digits... for certain prefixes.
T.
On Sun, Oct 11, 2009 at 12:10 PM, Henry
Hi Yuriy, did you manage to do something with H323plus and FS ?
btw: have you checked Objective OpenH323
http://www.obj-sys.com/telephony-objective.shtml ?
This looks better to me as it is lighter and can be easily customized.
T.
2009/10/8 Georgiewskiy Yuriy bottle...@icf.org.ru
On
yep, you made the point :P
T.
2009/10/8 Georgiewskiy Yuriy bottle...@icf.org.ru
On 2009-10-08 16:20 +0200, Tihomir Culjaga wrote
freeswitch-us...@lists.fre...:
TCHi Yuriy, did you manage to do something with H323plus and FS ?
i already doing it, but now it not in usable state.
TCbtw
Hi Yuriy,
can you share what you have so far, I'm sure we can help with RTP part...
T.
2009/10/8 Georgiewskiy Yuriy bottle...@icf.org.ru
On 2009-10-08 13:25 -0400, Tuyan ?zipek wrote
freeswitch-us...@lists.freesw...:
TzHi,
Tz
Tz2009/10/8 Georgiewskiy Yuriy bottle...@icf.org.ru:
Tz On
k
2009/10/8 Georgiewskiy Yuriy bottle...@icf.org.ru
On 2009-10-08 20:32 +0200, Tihomir Culjaga wrote
freeswitch-us...@lists.fre...:
TCHi Yuriy,
TC
TCcan you share what you have so far, I'm sure we can help with RTP
part...
I think there is a few days and i make it work, after this i
it happen.
We need to work on it ourselves or pay to the people that knows how to do
it to do it for us.
There is no other way I think.
Diego
On Tue, Oct 6, 2009 at 11:41 PM, Tihomir Culjaga tculj...@gmail.comwrote:
Diego,
what i'm pointing here is the situation where you have a great
On Wed, Oct 7, 2009 at 2:40 PM, Claudiu Filip clau...@globtel.ro wrote:
Hi Tihomir,
I've done some tests to see how suitable is freeswitch as a
SIP/H323 translator and you are right about the fact that H323
'alert+open logical channel' will generate a SIP '200 OK'.
hello guys,
i was playing with mod_opal to see if i can make it working ... well it
seems SIP-H323 interworking is not tuned at all.
I have a call from a registered sip user (1001) to PSTN via mod_opal
include
extension name=EMERGENCY
condition field=destination_number
issue will not be addressed but there is no promise
how fast it will be.
On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga tculj...@gmail.comwrote:
hello guys,
i was playing with mod_opal to see if i can make it working ... well it
seems SIP-H323 interworking is not tuned at all.
I have
diego.vi...@gmail.com wrote:
Instead of complaining and demanding things for free, people should start
to put their money where their mouth is.
Diego
On Tue, Oct 6, 2009 at 8:47 PM, Tihomir Culjaga tculj...@gmail.comwrote:
hi Anthony,
it is somewhere here:
switch_status_t
, Oct 7, 2009 at 12:58 AM, Jason White ja...@jasonjgw.net wrote:
Tihomir Culjaga tculj...@gmail.com wrote:
I understand your financial point of view, but anyhow while the entire
world
is wants sip and trying to move to sip, the reality is quite different.
The
majority of voice traffic
it works,
thx!
T.
On Mon, Oct 5, 2009 at 12:31 AM, Michael Jerris m...@jerris.com wrote:
I updated the tiff lib to build better inline, try make tiff-reconf
Mike
On Oct 2, 2009, at 8:05 AM, Tihomir Culjaga wrote:
hello,
i just got the last trunk and tried to compile it on one of my
hi Mark,
This is an inbound call leg and media channel (so far) is open in reverse
direction only (application ringback). I'm afraid you have to answer the
call to be able to hear the fax tone.
T.
On Mon, Oct 5, 2009 at 2:32 PM, Michael Jerris m...@jerris.com wrote:
Fax tones are not
also, you can store files in PCMA/PCMU format and avoid transcoding at
all... and as said disk space is cheap.. go get some...
On Sat, Oct 3, 2009 at 7:07 PM, Diego Viola diego.vi...@gmail.com wrote:
Why is not recommended?
On Sat, Oct 3, 2009 at 2:52 PM, Brian West br...@freeswitch.org
hello,
i just got the last trunk and tried to compile it on one of my development
machines... Well configure fails on tiff-3.8.2 where it is unable to find
Makefile.in ... Can someone advice?
checking if g++ static flag -static works... yes
checking if g++ supports -c -o file.o... yes
checking
what if you are running some huge traffic e.g. 2000 calls with media?
a typical application for that is an IVR system handling several different
services. I'd like to dedicate some capacity for inbound on per service
basis.
e.g.
DID 10001 limit to 500 calls
DID 10002 limit to 400 calls
DID
anyhow, this is how it works for me!
include
context name=public
extension name=LNP
condition field=destination_number
expression=(^30)(.*)
action application=lnp_getprefix data=in $2, out
reroutingalias/
action
as a more familiar interface for those coming over from asterisk.
Mike
On Sep 24, 2009, at 6:10 AM, Tihomir Culjaga wrote:
hello,
i'm on latest trunk and for some reason i cannot get timestamps dumped in
my cdrs. I use mod_cdr_csv with default settings plus i enabled to get both
a and b legs
should i move this to the DEV mailing list ?
T.
On Fri, Sep 25, 2009 at 4:12 PM, Michael Jerris m...@jerris.com wrote:
nothing I can think of, set up a test box that is not in production and
lets figure out what is wrong.
Mike
On Sep 25, 2009, at 7:22 AM, Tihomir Culjaga wrote:
Hi
does it mean, if i encode my voice files in g729 i can use mod_nativefile to
playback to a call using 729 codec?
T.
On Fri, Sep 25, 2009 at 8:30 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
fixed in latest trunk,
please test
thank you
On Fri, Sep 25, 2009 at 6:17 AM, Hound Dog
hello,
i'm on latest trunk and for some reason i cannot get timestamps dumped in my
cdrs. I use mod_cdr_csv with default settings plus i enabled to get both a
and b legs dumped.
cdr_csv.conf.xml:
configuration name=cdr_csv.conf description=CDR CSV Format
settings
!-- 'cdr-csv' will
, this is the desired outcome. I was planning of using FreeSWITCH +
MySQL to do this. How do I do this inline?
On Wed, Sep 23, 2009 at 12:49 AM, Tihomir Culjaga tculj...@gmail.comwrote:
so, you say ...
CallingParty = AS5300
A: aNum
B: didNum
AS5300 = PSTN
A: 1 + didNum
B: prefix (actually
endpoints that you are sending/receiving calls to/from It is useful to
have a separate configuration (other than dialplan) when you need to specify
credentials for GW to register somewhere, to specify domain, etc, etc ...
T.
On Wed, Sep 23, 2009 at 9:30 AM, Anil Kumar S. R.
and this is not enough for you?
!--- The *%* behind the username tells FS to lookup the user in it's
local sip_registration database --
action application=bridge data=user/${dialed_extension}@
${domain_name}/
!--- x.x.x.x in the line above is the IP address to the FreeSWITCH
=password value=test/
param name=register value=true/
param name=caller-id-in-from value=true/
param name=sip-port value=5060/param
/gateway
/include
What can be still wrong here?
Regards,
Filip
Tihomir Culjaga schrieb:
hi Filip,
for calling a user... please read
well .. it is AS .. it can be SIP or H323 ... well if it is hooked to a PGW
it is MGCP but i doubt... so it is either SIP or H323.
i will put a nickel for H323 :P
T.
On Tue, Sep 22, 2009 at 6:49 PM, Tihomir Culjaga tculj...@gmail.com wrote:
so, you say ...
CallingParty = AS5300
A: aNum
so, you say ...
CallingParty = AS5300
A: aNum
B: didNum
AS5300 = PSTN
A: 1 + didNum
B: prefix (actually the PSTN subscriber's number)
well, without a doubt... you can manipulate whatever number you want ... you
just need to find the best way to do it. This depends of the number of DIDs
you
of a pretty
serious issue.
Are you using 2 separate fresh checkouts for both suncc and gcc builds
because it's not possible to switch the same source tree once it's already
configured for one of them.
On Tue, Sep 22, 2009 at 11:29 AM, Tihomir Culjaga tculj...@gmail.comwrote:
Hi Anthony
well ... shame on me :P
thx anyway...
T.
On Tue, Sep 22, 2009 at 10:12 PM, Diego Viola diego.vi...@gmail.com wrote:
He's doing an extra effort... just compile it as you would normally and you
will have the debug symbols.
On Tue, Sep 22, 2009 at 8:11 PM, Diego Viola
I didn't say i have a working FS on blackfin... i just said i've ported a
lot of software to blackfin and it was always floating point, fork vs
vfork ... main issues... but why do you think it cannot be done?
T.
On Mon, Sep 21, 2009 at 6:08 AM, Hadley Rich h...@nice.net.nz wrote:
On Mon,
its a waste of time ... i doubt it can be done.
T.
On Mon, Sep 21, 2009 at 10:56 AM, Fred-145 codecompl...@free.fr wrote:
Or as a more affordable solution... is it possible to connect an
entry-level
GSM phone to a PC running Freeswitch and use this as a poor man's gateway?
--
View this
Hi Guys,
I have an issue running FS... it crashes apparently without leaving any log
... not even a core dump is left.
The machine is dual AMD opteron quad core with 8 GB RAM and i'm running 75
simultaneous calls (with media) with a rate of 5 calls per second.
As i was not able to reproduce
switch.conf.xml (btw: in console you can enable/disable logging on the fly -
F8/F7)
param name=loglevel value=debug/
your relevant sip profile:
param name=sip-trace value=yes/
T.
On Sun, Sep 20, 2009 at 4:14 AM, Klaus Teller klaus.tel...@gmx.net wrote:
Hi,
Say i have an inbound VoIP/SIP
hi,
well, yes, it should be possible to crosscompile freeswitch on that
platofrm... this is a totally different topic and to be honest i really
don't see the point doing this. When i did it last time (porting stuff to
Blackfin), it took several days of hard work.
This is an external
might have with RTP so check the wiki for NAT config as
well.
T.
On Sat, Sep 19, 2009 at 7:50 AM, pankaj anand pankajanan...@gmail.comwrote:
@Tihomir Culjaga
HI folks,
thanx for such a quick reply.
Q. what I want to achieve with FreeSwitch ?
A: I want to enable the outside
btw, you can check this GW:
http://www.edgepbx.cn/shop/index.php?controller=productproduct_id=12
i have it on my desk and it works as a charm...
T.
On Sat, Sep 19, 2009 at 1:47 PM, Alberto Escudero aep.li...@it46.se wrote:
If you can wait a few weeks, it will be one :) available and
in other works,
what are you trying to achieve?
where do you want send calls?
what is 192.168.1.50?
where/how are you originating calls from?
basically can you please tell us what is your call flow scenario otherwise
we can't help?
T.
On Fri, Sep 18, 2009 at 4:15 PM, Brian West
hi Filip,
for calling a user... please read this first:
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered_User
for making a GW register into e.g. asterisk please use this:
include
gateway name=gw01
param name=username value=USERNAME_ON_ASTERISK/
param
FS loads all users from $INSTALL_DIR/conf/directory/ and you did it correct.
freeswitch.xml:
section name=directory description=User Directory
X-PRE-PROCESS cmd=include data=directory/*.xml/
Than, you need to check sip profiles. By default FS will accept
registrations on internal
perfect,
thanks.
T.
On Wed, Sep 16, 2009 at 4:05 PM, Brian West br...@freeswitch.org wrote:
Yes you're missing a switch_xml_free(xml); some place.
/b
On Sep 16, 2009, at 8:49 AM, Tihomir Culjaga wrote:
hi,
I've build a custom module for FS and everytihng work well except
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