I highly recommend you wipe the box/install and install from Scratch
using SVN trunk
/b
On Feb 10, 2009, at 7:43 PM, Jesse Peterson wrote:
> I'm not able to find any documentation on this setting. I think it may
> be newer than my version of FreeSwitch (1.0). What does it do?
>
> Thanks,
> -
Well if they are sending both they are broken. I would call and yell
at them and beat them with a cluebat.
/b
On Feb 11, 2009, at 10:42 AM, Dennis wrote:
> that is interesting. we are receiving the dtmf digits over 2833. might
> it be possible, that we receive 2833 AND inband (we asked our ca
turn on the start_dtmf app and dial digits from the outside.. if you
get duplicate digits then they are sending both.
/b
On Feb 11, 2009, at 11:14 AM, Dennis wrote:
> i can't tell, if they are sending both, but it seems so. we get 2833
> for sure. they were kind enough to give it to us, becaus
On Feb 11, 2009, at 12:23 PM, Dennis wrote:
> ok, i will try this, but how can it be possible, that inband tones are
> audible in conference, when we do not even have start_dtmf activated?
They aren't really sending 2833.
>
>
> i just don't understand, why it must be dtmf inband, if the tones a
Try answer or pre_answer before park.
/b
On Feb 11, 2009, at 12:37 PM, Dennis wrote:
> anthony, did you make the changes with "add {instant_ringback=true} to
> make ringback not wait for indication to generate ringback" for the
> described problem?
>
> we read something like this out of it, but
Can you show us what you're doing?
/b
On Feb 11, 2009, at 1:15 PM, Nik Middleton wrote:
I have a situation where FS aborts
I’m running an lua script with mysql statements
First time it runs, on hangup I get
[CONSOLE] switch_core_memory.c:374 switch_core_memory_reclaim()
Returning 4 recycl
How about getting a backtrace of the core dump and opening a jira?
http://wiki.freeswitch.org/wiki/Reporting_Bugs
/b
On Feb 11, 2009, at 1:35 PM, Nik Middleton wrote:
I was running in a screen session, so going back to the console it
shows it’s a seg fault
2009-02-11 19:27:53 [NOTICE] sofi
Please collect the backtrace and report it on Jira.
/b
On Feb 11, 2009, at 2:11 PM, Dennis wrote:
> this does not help. we are using socket outbound and everything worked
> before the changes yesterday.
>
> we have the same error with other dialplans.
__
Try starting it from your /usr/local/freeswitch/bin... ./freeswitch
it'll dump in the same folder.
/b
On Feb 11, 2009, at 2:20 PM, Nik Middleton wrote:
Where is the core dump written?
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show me "sofia status", Try changing the @ to a % but I really need
to see the sofia status output.
/b
On Feb 11, 2009, at 2:32 PM, Chris Elam wrote:
$cmd = "api originate sofia/mydomain.com/1...@192.168.15.50 &bridge(sofia/mydomain.com/1...@192.168.15.50
)";
The result I get is : -ERR D
try sofia/myinsideip/1000 and sofia/myinsideip/1001
I sure hope it doesn't say myinsideip on there and you only tried to
hide your IP.
/b
On Feb 11, 2009, at 2:54 PM, Chris Elam wrote:
> The % gives the same error. Here is the sofia status output:
>
> API CALL [sofia(status)] output:
>
remember its sofia/profilename/user%domain or sofia/domain/user the
latter requires an alias on the profile for the domain the user
registers with.
/b
On Feb 11, 2009, at 3:06 PM, Chris Elam wrote:
That's it, worked perfectly, thanks a bunch!
On 2/11/09 3:59 PM, "Brian We
Are you sure you rebuilt it clean? Are you doing anything special?
Changing any configs?
/b
On Feb 11, 2009, at 3:11 PM, Public Dump wrote:
After reading you suggestions I deployed the version from SVN today,
the problem persists.
Regards
___
Quick note make sure you're 100% 64 bit.. if you need help with that I
can show you how on CentOS 5.2
/b
On Feb 11, 2009, at 4:34 PM, Nik Martin wrote:
> If any one needs a FreeSWITCH box with a public, static IP, I can
> provide them for you at a reasonable cost. I'm building a
> Virtualizat
Your VE must be 64bit also.
http://wiki.openvz.org/Install_OpenVZ_on_a_x86_64_system_Centos-Fedora
If you need the set util listed on that page let me know I have a
copy of it.
/b
On Feb 11, 2009, at 4:47 PM, Nik Martin wrote:
> I think the VE I've built is too, but uname is a bit cryptic:
It runs fine under OpenVZ pure 64bit...
/b
On Feb 11, 2009, at 4:55 PM, Nicolas Brenner wrote:
> Also be sure to test it right. I had a mediatemple VPS (they use
> Virtuozzo I think, the paid version of OpenVZ) and FS would not work
> right, I had multiple problems, then I switched to a real ser
n (light) production for about 2 weeks, with no issues
> so far. I'm going to build a pure 64 bit VE container though, and
> will run in that for a while too. Brian, you sid you have a readme on
> that?
>
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Lua has known issues with MySQL you must use latest SVN builds of the
luasql driver for that to avoid it.. and still its not stellar.. the
unixODBC one on the other hand works fine.
/b
On Feb 11, 2009, at 5:36 PM, Nik Middleton wrote:
I’ve abandoned LUA.
All sorts of problems (DTMF etc).
You can run a small SOHO operation on 256 megs
/b
On Feb 11, 2009, at 6:08 PM, EdPimentl wrote:
> 256 MB Ram . is this correct?... Does any VoIP provider to use
> this?
> -E
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first off don't use the session.originate
var new_session = new Session({var=val}sofia/blah/blah);
will do it all for you in one step. Also can you point me to where on
the wiki that keeps talking about session.originate? I need to clean
them off there.
/b
On Feb 11, 2009, at 6:09 PM,
Please refer to the extension in public.xml
and default.xml both will cause a deflect to be done so the 3 leg
call gets turned back into a 2 leg call. In some cases it might be
desired to do a 3 leg call so you can bill the party that caused the
302 and the original party also.
/b
On F
Nope its on auto pilot... we don't get passed the 302 from sofia. So
what you have there is all you can get at.
/b
On Feb 11, 2009, at 6:21 PM, Tchavdar Paskov wrote:
> Thank you Brian,
> is there any way to inspect what exactly is sent in 302 message and
> if possible to
Actually you can if you don't overload the machine like most VPS
providers do... The advantage with OpenVZ in this case is that you can
migrate the running FreeSWITCH instance between hardware nodes and not
drop calls at this size.
/b
On Feb 11, 2009, at 6:24 PM, EdPimentl wrote:
> Soho,,,
If you have outbound gateways registering make sure you set the
context and extension param on the gateway so it'll go to the right
spot. Recent changes made it work much smoother.
/b
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OK does it work now? We have tested this on various windows installs
among the team here and not seeing this issue... it was a known issue
back in Nov. or Dec. but thats long been fixed.
/b
On Feb 11, 2009, at 7:59 PM, Public Dump wrote:
http://files.freeswitch.org/freeswitch-snapshot.tar
ding ding ding .. yep!
"file /usr/local/freeswitch/bin/freeswitch" will also confirm
/b
On Feb 11, 2009, at 6:37 PM, Henry Huang wrote:
> Brian:
>
> I am also running my freeswitch on my own openVZ containers. Just
> how do you verify if the freeswitch is comp
This is normal. Are you using 5.0? can you include examples of how
you're doing this?
/b
On Feb 11, 2009, at 10:38 PM, pauld wrote:
> "TRANSCODING_NECESSARY"
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esf is for multi cast paging... it currently won't let you play
files... we would have to create a multicast playback application.
/b
On Feb 12, 2009, at 8:00 AM, Sluschny, Thomas wrote:
Hi,
i want to stream a file per IP multicast with mod_esf.
I can stream IP multicast with:
pa call
You still didn't answer my question. How are you trying to do this
from the dialplan.
/b
On Feb 12, 2009, at 8:08 AM, pauld wrote:
> Yes I am using 5.1, I haven't done anything special other than
> followed
> wiki and then the advice given here to create symlinks in FS lib dir
> to all
>
You could store the data in globals and then restart the profiles via
XML PRC. ie global_setvar, reloadxml, sofia profile blah restart.
/b
On Feb 12, 2009, at 5:05 AM, Rene Pankratz wrote:
> Hello,
> we want to use mod_pa as a softphone, that registers to a
> SIPregistrar.
> But the usern
Are you calling via a gateway?
/b
On Feb 12, 2009, at 2:34 AM, Helmut Kuper wrote:
> Hi,
>
> any ideas how to get FS's BYEs authenticated ?
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You could but I think you want to stream RTP to a multicast it would
be better off building an rtp format mod so you can record rtp://
x.x.x.x:5000 and play from rtp://y.y.y.y:5000
/b
On Feb 12, 2009, at 10:12 AM, Sluschny, Thomas wrote:
Hi Brian,
i thought if i can stream from portaudio
Dialplan or language method...btw if you're on IRC its better to ask
there.. faster response... ;)
/b
On Feb 12, 2009, at 11:51 AM, Nik Middleton wrote:
HI,
Is there an equivalent function in FS to waitforexten ? Closest
I’ve seen is collectInput?
Right now I’m using stream file, whic
Dialplan isn't for writing IVR's... doing so is against the design of
FreeSWITCH.. you can do simple things in dialplan but more complex
stuff needs to be in a language.
/b
On Feb 12, 2009, at 12:01 PM, Shelby Ramsey wrote:
> Nik,
>
> I'm not sure if this is the right way ... but I use
> a
Well when I do this:
r...@taz [Thu Feb 12 02:20 PM] /usr/src/freeswitch.trunk
<13>:file /usr/local/freeswitch/bin/freeswitch
/usr/local/freeswitch/bin/freeswitch: ELF 64-bit LSB executable, AMD
x86-64, version 1 (SYSV), for GNU/Linux 2.6.9, dynamically linked
(uses shared libs), for GNU/Linu
deflect takes one arg. and that isn't one. Try a SIP uri... not a
sofia/ string. ie sip:b...@host:5080
/b
On Feb 12, 2009, at 2:44 PM, jonathan augenstine wrote:
I am trying to use the deflect command to transfer an inbound call.
The call is established and the command seems to complete
What SVN rev?
/b
On Feb 12, 2009, at 3:41 PM, Ali Al-Rubaie wrote:
> Hi,
>
> How can the default value of "realm" be changed? I had changed the
> command:
>
>
>
> in the file internal.xml but FS still uses the server IP address as
> the challenge realm.
>
> Thanks in advance!
>
__
This is prob. why we don't see this crazy stuff on CentOS since the
compiler is 4.1.2
/b
On Feb 12, 2009, at 4:34 PM, Andy Spitzer wrote:
> Possibly. A recent (last year?) GCC change caused some order of
> operations to change, and so code that inadvertently relied on the
> previous behav
YOU DO! ;) Its a user edited content portal.
/b
On Feb 12, 2009, at 4:58 PM, Nik Middleton wrote:
>
> Not sure who updates the WIKI, but it's wrong on collectinput for the
> example. In the call, dtmf needs quotes, ie "dtmf"
___
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Would need a sip trace to know. TPORT_LOG=1 ./freeswitch
/b
On Feb 13, 2009, at 8:17 PM, Maxim Karp wrote:
Hello,
I am using a SNOM 320 and Windows Mobile 6 VoIP softphone on two
separate extensions. When dialing from the SNOM to the WM6 device I
get ringback on the SNOM but when callin
You have let the names of the profiles confuse you. Chances are
you're trying to hair pin the calls out and back into the same nat.
That usually doesn't work. You will need to give me more details
about your setup.
/b
On Feb 13, 2009, at 9:41 PM, Woody Dickson wrote:
> Hi,
>
> I tried t
FreeSWITCH default config already has this feature. Register two
phones... 1000 and 1001 both with password of 1234 then you can call
between them. That will work exactly as you want out of the box.
Expect more simplified configs to show up after 1.0.3.
/b
On Feb 14, 2009, at 6:02 PM, xs
And are you planning on contributing anything useful back to the
community? Or just take take take?
/b
On Feb 14, 2009, at 5:04 PM, JCATS wrote:
>
> Have you planned any predictive dialer features ( like VICIDIAL )?
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Well the config itself is easy to follow. The problem is coming from
Asterisk you have the wrong mindset to approach most things in
FreeSWITCH.
http://svn.freeswitch.org/svn/configs/ (softphone is the best small
config to look at)
The default config is easy to strip down once you take a fe
I haven't ever experienced this issue can you maybe elaborate on the
issue a little more? We usually hear that the audio quality is much
better... have you tried latest SVN trunk? If resampling was involved
it might cause some audio issues but those were usually gain issue and
that has si
What is odd some people have reported the same issue with Asterisk. I
would like to get to the bottom of it but nobody can provide any more
detail on what might be going on and I haven't experienced this issue
with the 30 or so phones I have on my desk I highly recommend you
try SVN t
This was a problem with the resampler which was replaced... we use the
resampler in Speex now which will not exhibit the problem.
/b
On Feb 14, 2009, at 9:18 PM, Jason White wrote:
> I sometimes get audio distortion in the above situation if anyone
> speaks too
> loudly. I suspect clipping s
Yes this issue has already been fixed in SVN Trunk. I recommend you
update.
/b
On Feb 15, 2009, at 10:29 AM, Dale Trub wrote:
The bug I describe sure looks a lot like:
http://jira.freeswitch.org/browse/FSCORE-266
We have a direct Metaswitch-> FS connection, and both machines in
the same L
Since nobody can reproduce it... not sure how we can proceed... have
you done a fresh checkout from SVN trunk and tried again?
/b
On Feb 15, 2009, at 1:16 PM, Public Dump wrote:
So, no ideas left how to fix this problem ?
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I'm not able to reproduce this issue.. can you verify the codecs are
what you think they are on both Asterisk and FreeSWITCH.
/b
On Feb 15, 2009, at 8:04 PM, Paul D. wrote:
> Well, I tried several call scenarios:
> 1. Call from X-Lite or Linksys to VM.
> 2. Call from X-Lite or Linksys to a con
Also you didn't try SVN Trunk?
/b
On Feb 15, 2009, at 8:04 PM, Paul D. wrote:
> I have now * 1.6.5 and FS 1.0.3RC1
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U
I think this is in the process of getting corrected to beh the
"debian" way. Please join on IRC and interact with everyone related
to this.
/b
On Feb 15, 2009, at 8:09 PM, Jason White wrote:
> I've found the cause of my problem:
> As of the 12018 build, FreeSWITCH is searching for its confi
Please open a jira http://jira.freeswitch.org
/b
On Feb 15, 2009, at 9:02 PM, Jason White wrote:
Jason White wrote:
I decided to try rev. 12027, which, on the same machine (Debian
Sid) fails to
build with the following error:
x86_64-linux-gnu-gcc -c -ggdb -I. ./dftables.c
In file included
Try "pa switch x", x being the call number.
/b
On Feb 16, 2009, at 1:59 AM, Jan Fricke wrote:
> Hello,
> I'm using Freeswitch (1.0.trunk) as a softphone with mod_pa. My GUI
> communicates with freeswitch via xml-rpc and fetches calls with "pa
> list".
> If somebody is calling, the state of the
Reloadacl would have to be called in either case.
/b
On Feb 16, 2009, at 9:18 AM, Adam Long wrote:
Hi Anthony, could he use mod_xml_curl for this to serve up a dynamic
acl.conf.xml?
Or would reloadacl have to be called somehow?
Regards,
-Adam
install gdbm-devel and db4-devel.
/b
On Feb 16, 2009, at 9:24 AM, Andrea wrote:
>
> making all mod_perl
> Creating mod_perl.so...
> /usr/bin/ld: cannot find -ldb
> collect2: ld returned 1 exit status
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You can send them directly to me br...@freeswitch.org
Thanks,
/b
On Feb 16, 2009, at 6:08 PM, Paul D. wrote:
> I was trying to send tcp dumps today, but the message was rejected
> because of its size (zipped). How do I send them?
>
>
> Anthony Minessale wrote:
>>
>> The typing it takes to start
Its currently not possible.
/b
On Feb 16, 2009, at 11:33 PM, Cesar Cepeda wrote:
Hi,
I’m using FS with g279 on passthrough mode and I’m trying to play a
g729 file as ringback to the A-leg while bridging a call. As far as
I understand it should go something like this:
· originat
You're missing some key information to help us answer your question.
First off we will need to know the SVN rev, Then you might want to
press F8 and check out the debug log. Chances are it'll tell you
exactly why. What concerns me is the fact that a .local domain is in
there. I wonder
I'm trying to get a clear picture of what you're trying to
accomplish. Why would you need/want to set a static realm? Anyway
can you collect sip traces?
/b
On Feb 17, 2009, at 8:51 AM, Ali Al-Rubaie wrote:
Thanks Brian,
Actually we're using freeswitch ver 1
Make sure on outbound registrations/gateways you have the context and
extension params set.
/b
On Feb 17, 2009, at 2:17 PM, kokoska rokoska wrote:
>
> Hi all,
>
> I have just "upgraded" to current trunk (before an hour or so),
> configuration remain the same (served through mod_xml_curl), but
Very sorry to hear you have to use Broken Software. But some good has
come of this if you update to rev 12113 or great you'll be 100% OK.
/b
On Feb 17, 2009, at 2:21 PM, Ali Al-Rubaie wrote:
>
> I have to use a specific softphone, HelpCaster, but it can not pass
> the authentication stage.
Mark,
Sorry to say but I think you're pretty much SOL. I would check
voipsupply or the like for a replacement. On a side note you hijacked
the "Big delays in playing audio files" thread by clicking reply on
one of those messages then changing the subject and body... in the
future
Don't they cryptographically sign the config also?
/b
On Feb 17, 2009, at 2:58 PM, Gabriel Kuri wrote:
> If you need help generating a config for the phone, with Linksys'
> special config tool, contact me offlist.
>
> Gabe
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No problem. Just join us on IRC.. things move faster on there.
/b
On Feb 17, 2009, at 3:05 PM, kokoska.rokoska wrote:
> If you have more hints, I be very happy :-)
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I think Ctrix is working on mod_airpe in his branch for OS X.
/b
On Feb 17, 2009, at 3:19 PM, Ivan C Myrvold wrote:
> s it possible to run Skypiax on OS X? The wiki says Linux and Windows,
> but says nothing about OS X.
> I have been running FreeSWITCH on OS X for a couple of years now, and
> lo
Please submit all patches and changes via jira if possible
http://jira.freeswitch.org
Thanks,
Brian
On Feb 17, 2009, at 4:40 PM, Dan wrote:
Hi guys,
I noticed that the debian build is missing lines for shout.conf.xml
and does not install mod_flite (if its built) . This can be fixed
by
Good... keep up the good work adding more docs. ;)
/b
On Feb 17, 2009, at 5:33 PM, Nik Middleton wrote:
> Err, that's what I just posted :)
>
> Regards,
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Great news!!! Good Job!
/b
On Feb 17, 2009, at 5:43 PM, Kristian Kielhofner wrote:
FreeSWITCH now compiles in AsLinux:
http://www.astlinux.org
AstLinux with the new bootloader Runnix (or you could just use
syslinux) boots from flash. It also boots from PXE, ISO, disk, etc.
Pretty much anyth
And you ran this in lua?
/b
On Feb 17, 2009, at 6:07 PM, Nik Middleton wrote:
>
> I ran 10,000 events, which completed in around 20 seconds, all
> received
> and processed flawlessly. A new one on me was arrayshift. To think
> that
> I messed around in C for ages with circular buffers, this
So you sticking with the Astlinux name? or switching it to something
more general?
/b
On Feb 17, 2009, at 6:20 PM, Kristian Kielhofner wrote:
> I don't think so but something tells me that FreeSWITCH won't do too
> well without an MMU and the external libs and modules could cause
> quite a pro
Please update to SVN Trunk and try again... what are the specs on your
machine?
I have been testing PocketSphinx the past couple of days on linux
again and its fine.
/b
On Feb 18, 2009, at 9:09 AM, Moiz Chinoy wrote:
Hi,
I have downloaded and build the Freeswitch from
http://files.fre
Please go get an SVN client for windows... svn update vs downloading
the tarball every day will save bandwidth. ;)
/b
On Feb 18, 2009, at 9:49 AM, Moiz Chinoy wrote:
System specs:
- Intel Core 2 Duo
- 2.00 GHZ CPU
- 1 Gb Ram
I will download the latest from here
http://files.freeswitch.or
Learn C and write it all in C.
/b
On Feb 18, 2009, at 3:56 PM, Nik Middleton wrote:
> Astererisk happily does around 200 calls, I'm hoping FS will do better
> or I've just been wasting my time. Is there a more efficient way of
> doing this?
___
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Looks like someone jumped the gun... just get SVN trunk... we are in
the process of release right now.
/b
On Feb 18, 2009, at 8:00 PM, Philip Patterson wrote:
Hi All.
Have a fresh server and going to install FS on it. Went to the
download page (http://wiki.freeswitch.org/wiki/Installati
Thats one I think Anthm will need to chime in on... maybe skypiax
isn't sending the right indications to cause the core to trigger the
ringback.
/b
On Feb 18, 2009, at 8:53 PM, Carlos Talbot wrote:
> Giovannia,
>
> great work on mod_skypiax. I've been testing it under Windows and it
> soun
go try now! ;)
/b
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It has to be in trunk to be in the MSI... I don't want to cause
confusion ... Now that 1.0.3 is tagged we can put it in trunk?
/b
On Feb 19, 2009, at 12:07 AM, Giovanni Maruzzelli wrote:
> Would be *very* nice to have skypiax in MSI, thank you!
___
No clue, I haven't ever seen that behavior on linux. Maybe you can
try to narrow it down and report it on jira.. chances are its a bug in
the pocketsphinx libs.
/b
On Feb 19, 2009, at 1:29 AM, Moiz Chinoy wrote:
>
> Often in the log I see cryptic characters in the XML part returned by
> ASR
ocket) Not sure those names are official but we have been calling
them that ;)
Thanks,
Brian West
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Plus its not an exact science in the first place.
/b
On Feb 19, 2009, at 3:47 PM, Michael Collins wrote:
This is really advanced stuff. You're going to need to pay someone who
really understands DSP and programming. You might want to start with
consult...@freeswitch.org.
-MC
___
Pika hardware already works with OpenZAP.
/b
On Feb 19, 2009, at 4:13 PM, jonathan augenstine wrote:
> I have heard a rumor that Pika support was being developed for
> Freeswitch. Is that still going on? Can someone tell me if the
> rumor is true or not, and if so, what is the status of th
Just make sure you send the call out an ipv6 profile and it'll work.
/b
On Feb 19, 2009, at 8:52 PM, Jason White wrote:
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it all depends on what you're doing.. can you elaborate?
/b
On Feb 20, 2009, at 4:18 AM, shehzad p wrote:
> Recently I faced some performance bottleneck by using Javascript.
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http
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 Al-Rub
The problem comes up that the default is 30... the chances are that
your phone doesn't set the mode= line so we default to 30 when this
takes place. Not setting the mode= line in the FMTP usually means
30ms... which is the default. So to force this always to 30 you can
allow i...@30i, be
http://www.bkw.org/pizza_gram.tar.gz
/b
On Feb 24, 2009, at 1:36 PM, mszla...@aol.com wrote:
I'm getting this error message trying out the pizza demo in FS 1.0.3:
"Can't open dictionary C:\Program Files\FreeSWITCH\grammar
\default.dic"
I didn't have this before where there was no default.
You're not in 1.0.3 you're in SVN trunk... The reason I know this is
that wasn't changed till AFTER 1.0.3 was tagged and a new file set was
used... please go into your libs dir and wipe out pocketsphinx and
sphinx base.. then let it redownload them. I'll make you a new
tarball of the new g
You can do something like this "sofia/blah/somenum...@someip:
5060;this=rocks"
/b
On Feb 25, 2009, at 9:22 AM, Josh Forman wrote:
I'm trying to edit the sip headers to make the from field look like
this:
From: ;tag=gK0a00d6ea.
I know that to read that data on an incoming sip message it is i
Alex,
If you want to update to svn trunk the tts-engine and tts-voice are
now valid options on the menu. They were not before (But the wiki
said they were). So to cut confusion I made them work... if you do
not wish to upgrade you'll need to set the tts_engine and tts_voice
variab
It will actually add it to both places
INVITE sip:1...@conference.freeswitch.org;this=rocks SIP/2.0
Via: SIP/2.0/UDP 99.185.85.3;rport;branch=z9hG4bK0Kaa1322U42eK
Max-Forwards: 69
From: "1004" ;tag=1SparjgraS69m
To:
I verified it does indeed add it in both places.
/b
On Feb 25, 2009, at 10:3
Can you report these issues in jira?
http://jira.freeswitch.org
/b
On Feb 25, 2009, at 12:27 PM, Alexander de Greiff wrote:
> (with the current trunk i have all sorts of other compile problems
> (mod_fax, python) but i will work this out following the build
> instructions again).
If he's on 1.0.3 I don't think it has php in it..
/b
On Feb 25, 2009, at 1:31 PM, Mathieu Rene wrote:
> FreeSWITCH will listen on a socket allowing clients to send commands /
> receive events.
> ESL is a library to ease the creation of applications connecting to
> that socket.
>
> To install the
SP,
That won't go into the from. You can't add params to the from unless
you have svn rev 12287 or higher. I added the ability to set
"sip_invite_params, sip_invite_to_params, sip_invite_from_params" to
sofia_glue.c, I added two lines and changed two lines to make this
possible.
I also realized I broke backwards compatibility for anyone using
sip_invite_params so I corrected that in rev 12288
/b
On Feb 25, 2009, at 2:13 PM, SP wrote:
> data="sip_invite_domain=some.domain;this=rocks"/>
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Fre
I seen that yesterday... looks interesting.
/b
On Feb 25, 2009, at 3:00 PM, Kristian Kielhofner wrote:
> Hello everyone,
>
> I just ordered one of these:
>
> http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp
>
> Just over $110 with shipping but they are expe
Read this http://wiki.freeswitch.org/wiki/Dialplan_XML#About_Dialplan_Variables
You can't condition on a variable you set in the same extension
because the set happens later. Thus its not possible to do what
you're doing... Once you understand the dialplan is just a list of
instructions tha
can you include the backtrace? We might have already fixed this one.
http://wiki.freeswitch.org/wiki/Reporting_Bugs
/b
On Feb 26, 2009, at 9:11 AM, Gopalakrishnan A.N wrote:
Hi,
I have installed Freeswitch 1.0.3. I am using event socket with
Javascript. When I try to dial the script w
What I would need is a debug log... ie press F8, attach that info to a
jira http://jira.freeswitch.org and please assign it to me "brian" is
the user.
/b
PS: do not paste logs in the comment box.. Attach them instead.
On Feb 26, 2009, at 9:19 AM, Alexander de Greiff wrote:
> ano
Alex,
Mine changes voices every time.. can you post your ivr.conf.xml along
with the report?
/b
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