30 retries that are queued up in the sip
stack.
at 100cps you will be generating this problem 100 times per second and
queue up countless unfinished dialogs thus
eating up the cpu.
On Mon, Aug 24, 2009 at 12:19 PM, Tihomir Culjaga tculj...@gmail.comwrote:
Hello,
I've been
Date: Mon, 24 Aug 2009 14:15:40 -0500
Subject: Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand
ACK message
In your scenario you need to add [peer_tag_param] at the end of the to on
the Ack.
/b
On Aug 24, 2009, at 2:03 PM, Tihomir Culjaga wrote
hard coded IP's in the fields that
shouldn't be there.
/b
On Aug 24, 2009, at 3:37 PM, Tihomir Culjaga wrote:
Hello Brian,
it doesn't work .. tried this today as well:
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http
sipp_cmd: sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1 -l
4000
scenario file: uac_redirect.xml
FS dialplan: public.xml
SIP trace: trace.log
Here it is... sorry for not including at first...
... documentation hate that :)) ... but thats my life actually... thats what
i do for living :)
of course i can go forward and document a lot of things... a lot of
docummentation is pending to completing the project i currently have using
FS.
well ... my soul? .. is it really necessary?
T.
On
:
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
shouldn't that be
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
On Mon, Aug 24, 2009 at 5:48 PM, Tihomir Culjaga tculj...@gmail.comwrote:
... documentation hate that :)) ... but thats my life actually
Is there any way to make FS complain about what header is wrong ?
T.
On Tue, Aug 25, 2009 at 1:14 AM, Tihomir Culjaga tculj...@gmail.com wrote:
this is original: Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7085-1-0
http://sipp.sourceforge.net/doc3.0/reference.html
*[branch]* - Provide
Hello Takeshi,
Thanks for your hint... it worked out... so to be precise:
VIA header of both INVITE and ACK messages MUST be identical (IP:PORT +
branch)... and you are right... it might not be according to SIP
specification. Anyhow, i get FS understand my ACK message.
Finally, here is what i
Hello,
i'm trying to use freeswitch as a redirecting server meaning FS has to
receive an INVITE and according to some rules it will redirect calls to
other destinations.
CALLING_USERFREESWITCHSOMEWHERE
INVITE ---
Hey Giovanni,
thanks for the tip... indeed the db files were heavily used regardless if i
started freeswitch with nosql option (freeswitch -nosql)... FS was not
writing anything into that files ... instead it was just accessing it
This behaviour leads to a waste of 40% CPU time... waiting for
, Tihomir Culjaga tculj...@gmail.com wrote:
Hey Giovanni,
thanks for the tip... indeed the db files were heavily used regardless if i
started freeswitch with nosql option (freeswitch -nosql)... FS was not
writing anything into that files ... instead it was just accessing it
This behaviour leads
Exactly... the scenario i use seems operating on a single thread... why is
that ? can it be changed?
T.
On Tue, Aug 25, 2009 at 5:31 PM, Michael Jerris m...@jerris.com wrote:
Actually in this case, we are bound to one thread in sofia.
Mike
On Aug 25, 2009, at 9:47 AM, Giovanni Maruzzelli
well :) ... this is something we are going to change tomorrow of course
will let you posted.
T.
On Tue, Aug 25, 2009 at 6:11 PM, Michael Collins m...@freeswitch.org wrote:
Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have
sense to move my OS to 64 bit? ... will FS
are probably hitting all the default stuff to save
that last dialed number etc
that exists in the default config.
sigh, we go through this every new guy doing load testing.
On Tue, Aug 25, 2009 at 11:34 AM, Tihomir Culjaga tculj...@gmail.comwrote:
Exactly... the scenario i use seems
this , the resounding answer is use a 64bit os..
No question
Jay
On 25/08/2009, at 23:19, Tihomir Culjaga tculj...@gmail.com wrote:
Hey Giovanni,
thanks for the tip... indeed the db files were heavily used regardless
if
i started freeswitch with nosql option (freeswitch -nosql
clear... thanks!
On Tue, Aug 25, 2009 at 7:55 PM, Michael Jerris m...@jerris.com wrote:
On Aug 25, 2009, at 1:40 PM, Michael Collins wrote:
On Tue, Aug 25, 2009 at 10:31 AM, Tihomir Culjaga tculj...@gmail.comwrote:
Of course i removed everytihng from teh dialplan except my extension
Hi Giovanny,
regarding ubuntu, did you mean 8.04 server or desktop ?
On Tue, Aug 25, 2009 at 3:41 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote:
Definitely go for 64 bit OS.
If you want to be safe and sure, go for CentOS 5.2 64bit. Is the one
used both for development and for heavy
intanto e il centos che si sta installando :)
grazie.
T.
On Wed, Aug 26, 2009 at 10:25 AM, Giovanni Maruzzelli
gmar...@celliax.orgwrote:
netbook remix
joking! Server 64bit :-)
-gm
On Wed, Aug 26, 2009 at 10:08 AM, Tihomir Culjagatculj...@gmail.com
wrote:
Hi Giovanny,
regarding
Hi Steve,
what exactly do you need?
i was experimenting with V.17/V.29 - V.34 fax interworking
well... I have V.34 machines ready over here... I just need to reconfigure
some CPE to point towards FS and thats it what setup do you need ?
T.
On Mon, Aug 31, 2009 at 5:09 PM, Steve
hello,
i'm trying to build mod_opal and getting this error:
making all mod_logfile
making all mod_loopback
making all mod_native_file
making all mod_opal
Compiling mod_opal.cpp...
quiet_libtool: compile: g++ -g -ggdb -I.
-I/home/tculjaga/freeswitch-trunk/src/include
hhmmm :))
is there any doc following up mod_opal ?
I really don't want to waste your time :)
T.
On Tue, Sep 1, 2009 at 1:00 AM, Michael Collins m...@freeswitch.org wrote:
On Mon, Aug 31, 2009 at 3:48 PM, Tihomir Culjaga tculj...@gmail.comwrote:
hello,
i'm trying to build mod_opal
hi, It went well
obviously FS needs v3_6 opal :)
thx.
On Tue, Sep 1, 2009 at 8:09 AM, Tihomir Culjaga tculj...@gmail.com wrote:
hhmmm :))
is there any doc following up mod_opal ?
I really don't want to waste your time :)
T.
On Tue, Sep 1, 2009 at 1:00 AM, Michael Collins m
oh good, on remote router/dsl modem (whatever doing NAT) never use upnp,
never use ALG, just do a simple NAT and it is alway gonna work!
T.
On Tue, Sep 1, 2009 at 2:06 AM, e schmidbauer e.schmidba...@gmail.comwrote:
i cannot reach the remote endpoint. the remote endpoint can reach a
locally
ok, please can you provide a tcpdump/wireshark sniff on before and after
that linksys.
this is something trivial.
T.
On Tue, Sep 1, 2009 at 6:22 PM, e schmidbauer e.schmidba...@gmail.comwrote:
I put tomato on the router and still no success. upnp is enabled,
should i disable it? what do you
-users-boun...@lists.freeswitch.org] *För *Tihomir Culjaga
*Skickat:* den 1 september 2009 08:09
*Till:* freeswitch-users@lists.freeswitch.org
*Ämne:* Re: [Freeswitch-users] mod_opal
hhmmm :))
is there any doc following up mod_opal ?
I really don't want to waste your time :)
T.
On Tue
Hi guys,
just a quick question... is it possible to do a reliable on the fly T30
T38 transcoding at all ... what is the status of T.38 on FS ?
T,
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it that was fully
implemented.
Tim
*From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Tihomir
Culjaga
*Sent:* Wednesday, September 02, 2009 3:33 PM
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* [Freeswitch
anyone knows anything about this?
T.
On Wed, Sep 2, 2009 at 10:53 PM, Tihomir Culjaga tculj...@gmail.com wrote:
I will put several nickels saying it is impossible :)
seriously, can it be done?
T.
On Wed, Sep 2, 2009 at 10:35 PM, Tim Meade tim.me...@millicorp.comwrote:
I am very
, Tihomir Culjaga tculj...@gmail.comwrote:
anyone knows anything about this?
T.
On Wed, Sep 2, 2009 at 10:53 PM, Tihomir Culjaga tculj...@gmail.comwrote:
I will put several nickels saying it is impossible :)
seriously, can it be done?
T.
On Wed, Sep 2, 2009 at 10:35 PM, Tim Meade tim.me
it may be that you have 2 configured nics and FS hooks up to your 1st one
only ignoring the 2nd one. This happend to me. You need to configure the 2nd
IP address explicitly.
T.
On Sat, Sep 5, 2009 at 12:15 AM, Ujjval Karihaloo
ujj...@simplesignal.comwrote:
Hi,
I just installed
check your sip profiles
/usr/local/freeswitch/conf/sip_profiles/external.xml
param name=context value=public/
/usr/local/freeswitch/conf/sip_profiles/internal.xml
param name=context value=default/
/usr/local/freeswitch/conf/vars.xml
!-- Internal SIP Profile --
X-PRE-PROCESS cmd=set
Hi Woody,
well, it is quite hard to answer you back with this logs...
you didn't tell us:
1. what machine are you running (CPU/RAM)
2. what distro are you running - 32 or 64 bit (i had some lets say
experience with a wrong selection :P)
3. what is your configuration
Hi,
i just have a maybe dummy question but it is still a question :P
*action application=record
data=${recordpath}/${service_instance}/${record_filename} 20 200/*
in my case ${service_instance} is something dynamic and has to be created on
the fly.
Is there any way FS can create a
...@scarlet-internet.nlwrote:
Hi,
You could use a system call for that:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_system
regards,
Leon
On Sep 14, 2009, at 3:58 PM, Tihomir Culjaga wrote:
Hi,
i just have a maybe dummy question but it is still a question :P
*action
nice ... thx.
T.
On Mon, Sep 14, 2009 at 4:41 PM, Evgeniy Zolotov zolo...@altron.ua wrote:
This works for me:
action application=mkdir data=${filebase_dir}/
You must set ' filebase_dir ' before.
- Original Message -
*From:* Tihomir Culjaga tculj...@gmail.com
*To:* freeswitch
Hi,
is there any way to route fax calls according to the call capability?
I mean .. if the fax call supports T.38 i'd like to route it to a T.38
capable gateway. All other fax calls (meaning inband) should be handled by
FS/SpanDSP.
Of course, I know that every fax call starts as a voice call and
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
hi,
I've build a custom module for FS and everytihng work well except reloadxml
command :P... m'I missing something in my module? ... i used mod_skeleton as
a template when i started.
When i start the FS without my module reloadxml works fine ... as soon as i
include my module within
, Tihomir Culjaga wrote:
perfect,
thanks.
T.
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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
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
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
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
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
, 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.
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
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 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
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
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
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 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'.
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
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
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
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
?
-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
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,
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
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
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
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