There should be several archive threads on this including:
http://n2.nabble.com/Using-Variables-in-Dialplans-tt2678222.html#a2678222
Mike
On Apr 29, 2009, at 10:30 AM, Alex Gusak wrote:
Hello.
Can I use custom variables in the condition field?
For example if i set my custom var
This was a bug that should be fixed now in trunk. Could you confirm
this please?
Mike
On Apr 27, 2009, at 9:12 AM, paul.degt wrote:
I use release 1.0.3. It's gone after I deleted default.xml with
pointers. Thanks for the hint.
Interesting that this file does not have any password
If you don't include the transport= in the message, the reply to that
register should still go back on tcp, but the calls to that registered
endpoint will go on upd.
Mike
On Apr 24, 2009, at 6:19 AM, Mikael Aleksander Bjerkeland wrote:
Hi,
I'm registering a Nokia N82 with FreeSWITCH,
There are extensive details and examples available at:
http://wiki.freeswitch.org/wiki/Mod_xml_curl
On Apr 23, 2009, at 3:07 AM, Meftah Tayeb wrote:
hello,
your FS-curl project is free ?
please cool you give it to me ?
if you dont know me, i'm DelphiWorld in the IRC Network
thanks
That all being said, I am sure we will need to make modifications to
make your caller id work as well, are there technical details available?
Mike
On Apr 23, 2009, at 6:58 AM, Fred-145 wrote:
xbipin wrote:
i was just thinking of throwing away my spa3102 and setting my
windows
machine
rfc 4733 attempts (and fails miserably) to clarify rfc2833. It
doesn't really change anything of any real substnace. Our handling
of these packets tries very hard to be strict in what we send and
loose in what we accept and we tend to interoperate pretty well with
most endpoints. That
I should clarify that we support the dtmf tones plus flash but no
other tones at this time.
Mike
On Apr 23, 2009, at 11:17 AM, Steve Underwood wrote:
Michael Jerris wrote:
rfc 4733 attempts (and fails miserably) to clarify rfc2833. It
doesn't really change anything of any real substnace
Hopefully the sangoma should work soon. Please contact sangoma to ask
when those cards will work on windows in FreeSWITCH.
Mike
On Apr 22, 2009, at 11:20 AM, Brian West wrote:
I don't know of any FXO cards for windows yet.
/b
On Apr 22, 2009, at 10:16 AM, xbipin wrote:
i was just
$${domain} (or any other preprocessor vars) will not be expanded on an
xml_curl return.
Mike
On Apr 22, 2009, at 12:31 PM, JuanMa wrote:
Peter Thanks for your reply:
I did what you said but FS still with the same problem, cant found the
user.
This is my reply to registration request
Those files in tree go with the rpm build with goes correctly to those
directories.
Mike
On Apr 21, 2009, at 4:06 AM, Fred-145 wrote:
Hello
I'm installing Freeswitch on a CentOS 5.3 test host. I noticed that
the
freeswitch.init.redhat contains paths that don't match a stock CentOS
Could you translate these into english?
On Apr 21, 2009, at 7:08 AM, Guido Kuth wrote:
I am playing around with FS (Windows) for one month now. First I
tried using FreeSwitch.NET which is a good class library for inbound
event socket. Unfortunatley it can't be used for outbound event
http://www.ietf.org/rfc/rfc3261.txt
8.1.1.7 Via
On Apr 21, 2009, at 10:48 AM, Fred-145 wrote:
Hello
I'm reading the examples, and found one on how to use a Sipura
SPA-2000:
http://wiki.freeswitch.org/wiki/Sipura_STUN
I was wondering what the VIA messages mean, and whether I should
FreeSWITCH has a generic ASR interface to plug into any asr engine.
The quality of the integrated free asr has little to do with VXML. If
you want more robust asr, you will likely need to pay for it.
Mike
On Apr 21, 2009, at 2:35 PM, mszla...@aol.com wrote:
Great Idea.
Try setting up
sound_prefix?
Mike
On Apr 21, 2009, at 8:02 PM, Michael Collins wrote:
Kristian,
The symptom I'm experiencing is that no matter what language I
specify, it still plays the English sound files. Is that what you're
experiencing? I've run it with debug logging turned on and combed
through
You can use VXML without ASR, but ASR and TTS are both required parts
of the specs.
Mike
On Apr 21, 2009, at 10:41 PM, Brian West wrote:
Isn't ASR optional in VXML?
/b
On Apr 21, 2009, at 6:15 PM, David Knell wrote:
Out of interest, is that using some RAD tool or coding directly in
If someone has a way to make true mirrors that support read/write this
would be interesting.
Mike
On Apr 17, 2009, at 4:06 AM, Will Boyce wrote:
Special:Export will export a page to an XML format that can, in
turn, be imported.
There must be a way to automate that process (export extire
http://wiki.freeswitch.org/wiki/Special:Search?search=asteriskgo=Go
On Apr 14, 2009, at 1:09 AM, kunal rao wrote:
Hi
even I have downloaded FreeSWITCH and using MS Visual Studio 2008.
It is building properly. I now want to configure it properly. Can
you please give me directions and also
On Apr 10, 2009, at 1:35 AM, Traun Leyden wrote:
Hey you beat me to it. I was going to have a look this morning but
had no internet because some asswipe cut a bunch of fiber optic cables
and took out phone/internet for a big part of the bay area.
I haven't tried your patch yet, but I see
On Apr 10, 2009, at 1:35 AM, Traun Leyden wrote:
Hey you beat me to it. I was going to have a look this morning but
had no internet because some asswipe cut a bunch of fiber optic cables
and took out phone/internet for a big part of the bay area.
I haven't tried your patch yet, but I see
On Apr 10, 2009, at 4:59 AM, Peter Olsson wrote:
The spidermonkey modules core_db/odbc, curl, socket and teletone
fails to load in Windows. They just return error 1271 (Sym Error).
I'm not sure if this is a known issue, or if just doesn't work in
Windows :) I've been using the latest
On Apr 10, 2009, at 6:05 AM, Peter Olsson wrote:
When trying to load the mod_managed module it get an error that it
can't find FreeSWITCH.Managed.dll.
So my question is simply - where do I find this file, or how do I
build it? I'm using the VC++ Express edition when building, so I
The code that should work for this is on a box at Sangmoa under
testing right now on linux. It should be committed as soon as the new
driver is released (which the new module will require) at which point
it will just need build integration completed and proper testing on
windows.
Mike
I need some testers for systems using both libtool 2.2 and 1.5.x to
confirm the following patch:
http://jira.freeswitch.org/secure/attachment/11356/fs-r12922-libtool22.patch
In order to test you will need to do a complete fresh checkout, apply
this patch, then do a bootstrap, configure, etc.
On Apr 6, 2009, at 6:42 PM, mszlazak wrote:
NOTE ON UNRELATED ERROR: I don't use Lua but there was an error in
compiling
LUA on Windows with 2008 Express so I get a error loading
mod_lua.dll today.
This is fixed in svn a bit earlier today.
Mike
http://wiki.freeswitch.org/wiki/Mod_conference
On Apr 2, 2009, at 3:29 AM, bmsword wrote:
I want to use another softswitch conference that has been
deployed in freeswitch,How should I do?
___
Freeswitch-users mailing list
no, but they all auto-create. You can create a db and set up odbc,
start freeswitch, then dump your db schema. Also, please do not send
confidential emails to the mailing list.
Mike
On Apr 2, 2009, at 11:12 AM, Richard Lamkin wrote:
Are there documents or wiki page [I’ve missed during my
If you try to build just the sofia library, what are the first few
warnings and errors you get?
Mike
On Apr 1, 2009, at 6:41 AM, Lewis Liu wrote:
We download FreeSWITCH from SVN Trunk and want to build it on MS
Visual Studio 2008 with platform.
But we got one error message when we build
as replied earlier, if your doing nothing but consuming events, you
can just block instead of sleep:
con:pop(1)
there is also a msleep function that you can call the same way you do
console_log, it takes milli seconds as its arg. Note this should NOT
be used when you have a script
will not be consuming events.
can I get an example of how to use msleep in a lua script? This lua
script will be running in the background, and not part of a session
or event consumer. Thanks.
--matt
2009/3/31 Michael Jerris m...@jerris.com
as replied earlier, if your doing nothing
http://www.freeswitch.org/docs/class_event_consumer.html#a0
It takes 2 args, not one to specify the subclass
Mike
On Mar 31, 2009, at 12:11 PM, Matthew Fong wrote:
Thanks, the freeswitch.msleep(5000) works!
Any comment about the first Q...
con = freeswitch.EventConsumer(CUSTOM my::event);
On Mar 30, 2009, at 10:33 AM, xbipin wrote:
can any1 tell me where can i find a live cd image with the basic
stuff to run
FS and FS with all it tools installed and WITH A GUI, something like
a pbx
in a flash iso image so windows users like me find it easier to get
testing
with FS
please see my previous response in this thread.
MIke
On Mar 28, 2009, at 5:09 AM, Moiz Chinoy wrote:
I am trying it on windows.
Where to get gnutls-devel for windows?
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
Look closer at the logs or sip trace, this sounds like a failed
session timer to me.
Mike
On Mar 27, 2009, at 8:34 AM, Andy Ayers wrote:
Thanks for your help folks, the ping parameter seems to have
resolved the gateway connection issue but I now seem to be having a
related issue with
fixed revision 12805.
Mike
On Mar 27, 2009, at 9:49 AM, Brian West wrote:
http://jira.freeswitch.org/browse/ESL-7
I think this might apply to you.
/b
On Mar 27, 2009, at 6:48 AM, Matthew Fong wrote:
/usr/bin/ld: cannot find -lruby
collect2: ld returned 1 exit status
Brian West
You seem to be confusing your standards, those 2 specs are about tel:
uri's not sip: uris. Sending a tel uri I am not sure we can do, where
would we send it to?
Mike
On Mar 27, 2009, at 6:11 AM, James H Thompson wrote:
I need to generate calls with Invite URIs in this format:
INVITE
Another example of a fatal issue was the optimizer in gcc was breaking
openzap code even with -O2.
Mike
On Mar 27, 2009, at 12:12 PM, dujinfang wrote:
On Mar 26, 2009, at 10:59 PM, Anthony Minessale wrote:
We do not support ubuntu interpid, it has at least 3 known fatal
issues not
There is no known working build on windows for the tls with
freeswitch. We would be happy if someone would submit a full working
build.
Mike
On Mar 26, 2009, at 2:59 AM, Moiz Chinoy wrote:
Hi,
I am having trouble compiling iksemel for google talk. There errors
are in gnutls.h...
I
Actions are all run AFTER all conditions are parsed so the nated var
is not set yet. you can do a single condition in this case, and set
nated for use elsewhere if you need it in the actions.
Mike
On Mar 26, 2009, at 10:27 AM, Rodrigo P. Telles wrote:
Hi Guys,
I'm trying to do some
http://n2.nabble.com/freeswitch-users-f2379917.html
Mike
On Mar 26, 2009, at 12:01 PM, Tim Ringenbach wrote:
Is there nothing out there that integrates a forum with a mailing
list?
It seems like one could display the mailing list archives exactly
like a
forum, and allow users to
it was never really fixed as no one let me into their machine to
troubleshoot.
Mike
On Mar 25, 2009, at 4:01 PM, chevio wrote:
How was this fixed ?, I am experiencing the same problem.
Chevio
Shelby Ramsey-2 wrote:
Thanks for the help. That did the trick.
SDR
Thanks for access to your machine. The issue was that the odbc
detection was trying to use odbc if either the libs or headers were
found, not only if both were found. I fixed the detection to not try
to use odbc if the headers were not installed. Installing unixODBC-
devel package of
=as4863e49a
To: sip:70...@b-pbx-lab-1.mynet.net;tag=FgDae7QaetHgm
Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
CSeq: 103 CANCEL
Content-Length: 0
2009/3/24 Michael Jerris m...@jerris.com
This means we
I still need access to a machine in this state so I can debug and fix
this issue.
Mike
On Mar 23, 2009, at 3:31 PM, Raul Fragoso wrote:
Running the following command as root should install the ODBC
development package:
yast install unixODBC-devel
After that, run configure again and make
On Mar 19, 2009, at 7:54 AM, Gilles wrote:
Michael Jerris There is currently no openzap (sangoma, etc) support
on windows, we hope this will be coming soon.
I found an alternative: The Linksys 3102 VoIP gateway. It's cheaper
too.
Would you say the Windows port of Freeswitch is ready
On Mar 17, 2009, at 10:31 PM, Jason White wrote:
Brian West br...@freeswitch.org wrote:
if you installed the ssl devel stuff AFTER you configured you'll need
to reconfigure.
I'm reasonably sure it was installed already, unless it was pulled
in recently
by a package upgrade.
The
There is currently no openzap (sangoma, etc) support on windows, we
hope this will be coming soon.
Mike
On Mar 17, 2009, at 5:20 AM, Gilles wrote:
Hello
For single-host settings, getting customers to buy a separate server
just to run Freeswitch is overkill, so I'm thinking about selling
Which data?
On Mar 16, 2009, at 11:58 AM, Ali Al-Rubaie wrote:
Hi,
Is it possible to access FS DB to retrieve data? Where can i find
details about that?
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
What is a pbm?
On Mar 13, 2009, at 4:51 AM, rod kawa...@laposte.net wrote:
Hello,
I'm now running r12590, the pbm was still there, but this was
because of
a broken dialplan.
I'm using this for exceeded limit:
extension name=TOO_MANY_CALLS
condition field=destination_number
The continue works fine, it just hangs up befoer that due to:
action application=set data=hangup_after_bridge=true/
Mike
On Mar 11, 2009, at 2:46 AM, mszla...@aol.com wrote:
I have the following in my dialplan.
Individually, each extension does what it's suppose to do when
If your actually playing back a file (that blocks) it should work
fine. Ringback goes in the background so it doesn't wait for it to
finish before it does the ringback
Mike
On Mar 11, 2009, at 4:07 AM, Helmut Kuper wrote:
Hello,
I'm looking for a way to play a file exactly once in early
Much more than an archive, nabble makes a full embeddable forum that is
linked to the mailing list. We will be embedding this in our webpage soon
for the best of both worlds, a forum and a mailing list without the
additional overhead of having to monitor 2 things.
Mike
Ken Rice-3 wrote:
On Mar 5, 2009, at 12:03 PM, Ben Holtsclaw wrote:
I agree with Harry. I do not like the mailing list. Those that do
like the mailing list always advocate Nabble. For those that
advocate that solution, do you even realize that you can't post on
Nabble unless you are subscribed to the
On Mar 5, 2009, at 3:39 PM, Kristian Kielhofner wrote:
A bunch of telephony geeks and a 1900 number - what could go wrong?
Anyways, I too don't understand why people prefer forums.
I follow dozens of mailling lists and a half a dozen e-mail addresses
without ever leaving my mail client.
Due to licensing reasons, you can not port a gpl piece of code to
FreeSWITCH due to restrictions imposed by the gpl so it is not
possible to do this unless all copy-write holders approve a license
change.
Mike
On Mar 4, 2009, at 10:31 PM, Gerry Hull wrote:
I hear rumors that someone is
The debug logs should give you more information about what is
happening here.
Mike
On Mar 3, 2009, at 10:53 AM, Ali Al-Rubaie wrote:
Hi,
During proxy authentication, I got mandatory IE missing error in
the response as shown below. How this error can be resolved?
Could you please post this to jira along with a thread apply all bt of
a core file taken from the process with the stuck sessions.
Mike
On Mar 2, 2009, at 2:06 AM, rod wrote:
Hi All,
I ran some longer tests with FS 1.0.3 acting as an SBC.
The test machine has the following specs:
-
I think any issues we have are related to pri, the analog doesn't seem
to generate any major bug reports.
Mike
On Mar 2, 2009, at 6:47 AM, Fred wrote:
Thanks guys for the feedback. So, the OpenZap driver isn't ready for
production yet?
___
There are examples on the wiki for this.
Mike
On Mar 1, 2009, at 3:10 PM, Rex_Alex rex.alex...@yahoo.com wrote:
Hi Shelby Ramsey,
I would like to do the same in php script.
Please post me a sample.
Thanks,
Rex.
Shelby Ramsey wrote:
Rex:
The basis for xml_rpc or mod_event is
You should be able to do loglevel of console
Mike
On Feb 26, 2009, at 7:41 PM, Michael Collins m...@freeswitch.org wrote:
Is there a way of displaying a console message not related to a log
level?
I’ve got the console only reporting errors now, but it would be ni
ce to be
able to
Please report this bug to jira.freeswitch.org.
On Feb 24, 2009, at 2:27 AM, Rene Pankratz wrote:
No, unfortunately the problem still persists. Portaudio still
automatically accepts/takes the next call.
René
On Thu, Feb 19, 2009 at 5:04 AM, Rene Pankratz
r.pankr...@fh-wolfenbuettel.de
The web version of this list is available at:
http://www.nabble.com/Freeswitch-users-f32209.html
Mike
On Feb 24, 2009, at 2:08 PM, Fred wrote:
Hello
Maybe this question has been raised before, but if not: There's so
much traffic in this mailing list that I was wondering if adding a
On Feb 23, 2009, at 9:44 AM, Helmut Kuper wrote:
Hello,
today I found in FS logfile lines like this:
2009-02-23 15:27:12 [DEBUG] switch_ivr_originate.c:1605
switch_ivr_originate() Raw Codec Activation Success l...@8000hz 1
channel
20ms
It looks like L16 codec is used for incoming
I need someone with this issue to provide me ssh access to their box
so I can fix this problem for everyone. No one has done so yet.
Please find me on irc if you can provide access.
Mike
On Feb 23, 2009, at 10:16 AM, Brian West wrote:
You have something on your system thats causing the
Look up furthur, there will be an error around where it builds or
links the core. Try typing make core.
Mike
On Feb 22, 2009, at 1:30 PM, Shelby Ramsey sicfsl...@gmail.com wrote:
Hello,
I'm getting this all over the place today:
make[5]: *** No rule to make target `/usr/src/freeswitch/
There should be more errors above that? Are you cutting off some of
them in your paste?
On Feb 22, 2009, at 2:26 PM, Shelby Ramsey sicfsl...@gmail.com wrote:
Mike,
This is what I get when I run make core (after I did a checkout on
latest svn trunk, ./bootstrap.sh, ./configure):
can you please contact me off list and get me information to access
your box so I can try to correct this for good in tree.
Thanks
Mike
On Feb 22, 2009, at 7:46 PM, Shelby Ramsey wrote:
Thanks for the help. That did the trick.
SDR
___
Can you re-test this with current svn trunk. I believe this was fixed
yesterday.
Mike
On Feb 19, 2009, at 9:19 AM, Alexandru Nedelcu wrote:
OK, so effective_caller_id_number is the same as
origination_caller_id_number set on the B-leg (cool).
Unfortunately origination_caller_id_number on
We don't yet support localstatedir configure option. I expect we will
soon.
Mike
On Feb 17, 2009, at 12:59 PM, Pablo Hernan Saro wrote:
Hi Michael,
Thank you very much for your help. It meets my needs.
I was thinking in something like:
./configure --prefix=/opt/freeswitch
Pretty much all the codecs (mod_voipcodecs, mod_speex, mod_ilbc,
mod_g722_1, mod_celt) and the resampler all have fixed point
implementations (in tree) as well.
Mike
On Feb 17, 2009, at 7:44 PM, Gabriel Kuri wrote:
That and the lack of an FPU, I'm curious how that would affect FS,
I have reverted this patch, it should be in /opt/freeswitch/conf in
trunk now properly.
Mike
On Feb 16, 2009, at 1:06 AM, Michael Jerris wrote:
This patch was incorrect and was supposed to be reverted. I will
correct this error.
Mike
On Feb 15, 2009, at 9:13 PM, Brian West wrote:
I
This patch was incorrect and was supposed to be reverted. I will
correct this error.
Mike
On Feb 15, 2009, at 9:13 PM, Brian West wrote:
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
its just event plain CUSTOM conference::maintenance no bgapi see:
http://wiki.freeswitch.org/wiki/Event_Socket
for more info.
MIke
On Feb 14, 2009, at 12:45 AM, Adam Wilt wrote:
I'm trying to use custom events for a conference call in a Python
script. I set-up the events in the
I updated the version of the speex library we use in tree last night
and it may cause some build issues for those with current working
copies. To fix this issue you can type make speex-reconf
MIke
___
Freeswitch-users mailing list
Is this running on 64 bit os or 32?
On Feb 12, 2009, at 4:23 AM, Public Dump p...@suspiria.net wrote:
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.
If using gayeway it should already do this.
On Feb 12, 2009, at 3:34 AM, Helmut Kuper helmut.ku...@ewetel.de
wrote:
Hi,
any ideas how to get FS's BYEs authenticated ?
On 11.02.2009 13:41, Helmut Kuper wrote:
Hello,
my FS is connected to my SIP-DDI softswitch, which requires all SIP
You can change the config files on disk and then issue reloadxml or
use mod_XML_curl
Mike
On Feb 12, 2009, at 6:05 AM, Rene Pankratz r.pankr...@fh-wolfenbuettel.de
wrote:
Hello,
we want to use mod_pa as a softphone, that registers to a
SIPregistrar.
But the username and password need
I don't have a 64 bit windows box/os to get this working. Someone
with access to such a box would have to set this up and submit a patch.
Mike
On Feb 11, 2009, at 2:49 PM, Public Dump wrote:
… did anybody succeed with this ? The solution for VS2008 does not
seem to have a valid 64bit
If your in a conference and your hearing other people hitting dtmf
digits that IS inband, it means that the place upstream that is doing
inband to 2833 conversion is not properly clipping the dtmf, this
probably needs to be fixed on that device.
Mike
On Feb 10, 2009, at 9:58 AM, Dennis
It sounds like your automake got screwed up with some new changes. I
tried and was unable to reproduce this issue, can you test a fresh
checkout and see if you still see this issue?
Mike
On Feb 9, 2009, at 3:27 AM, Helmut Kuper wrote:
Hello,
update, when I remove all ozmod_ from
We can not add apr dependency in openzap, we should use the native
openzap calls instead. If there is anything you NEED that you don't
have, please let me know and we will try to add replacement functions.
Mike
On Feb 9, 2009, at 1:17 PM, Helmut Kuper wrote:
Hello Anthony,
:D yes
Not sure what you mean by playback speed. All the prompts for
voicemail are defined in the phrase macros in the configuration.
Mike
On Feb 6, 2009, at 7:32 PM, Maxim Karp mk...@securesilence.com
wrote:
Hello all,
Can anyone please let me know how I might be able to configure the
This should now be fixed in trunk in revision 11632. Can you please test
and confirm.
Mike
On 2/4/09 9:27 AM, Cavalera Claudio Luigi claudio.caval...@italtel.it
wrote:
Hello,
I'm trying to compile a brand new fs on a clean system.
Revision: 11630
After the usual ./bootstrap.sh
We need to add more than this including detection in openzap
configure.in if libpcap is available (headers and lib) and if not,
disabling the functionality.
MIke
On Feb 2, 2009, at 1:04 PM, Helmut Kuper wrote:
Hello,
today I uploaded a little patch for openzap concerning missed
This is fixed in svn trunk now, I will have to do a little bit of
autoconf work to get the build w/o libpcap working again but expect
that this weekend.
Mike
On Jan 30, 2009, at 4:35 PM, Dan wrote:
Hi guys I pulled FS from trunk today to build some debian packages
and ran into this
On Jan 28, 2009, at 5:24 AM, Cavalera Claudio Luigi wrote:
freeswitch-users-boun...@lists.freeswitch.org wrote:
Out of curiosity which SIP messages have you been watching for on the
event socket? Also, how are you connected to the event socket? Are
you
subscribing to all events and
On Jan 23, 2009, at 11:04 AM, Dennis wrote:
is it possible to define a profile and its params for a conference
dynamically over socket outbound?
in the moment, if we want to have multiple profiles for different
clients, we (have to) setup a profile in the conference.conf -
otherwise we get
If you can get it to work in openzap, it will be easy enough for me to
port when we do the windows driver integration for openzap.
Mike
On Jan 21, 2009, at 10:01 AM, Helmut Kuper wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
It's a standalone program to profe concept. So I
One issue with the service is we have no console to dump errors too,
it sounds like it is failing one of the startup requirements like
config files being there. Are you able to start it in non service
mode? If so, check permissions on the freeswitch dir that the user
running the service
On Jan 20, 2009, at 9:20 AM, Paul D. wrote:
Hi,
I am fairly new to FS. Is there a way to switch off/on caller ID
presentation dynamically, similar to callingpres in *?
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_privacy
___
You can disable the core db using -nosql or if you want to remotely
have access to an odbc db of calls you can make an event socket
listener to make your own db.
Mike
On Jan 19, 2009, at 1:04 PM, Anthony Minessale anthony.miness...@gmail.com
wrote:
it's not possible, the core is only
All long running non js code should be wrapped in the suspend/resume
gc stuff. For example:
cb_state.ret = BOOLEAN_TO_JSVAL(JS_FALSE);
cb_state.saveDepth = JS_SuspendRequest(cx);
args.input_callback = dtmf_func;
args.buf = bp;
args.buflen = len;
To the contrary, we have had quite good results in virtualized
environments and you don't really need timing that is that accurate to
make it work. We work quite well on amazon EC2 for example. There
are 2 issues I know about with vmware, 1 is you need to set a setting
on the host to
Your build issue is with your autotools install, I have seen issues if
you have ever installed any of the autotools from macports or fink.
If you want to build from svn you can run bootstrap on another box (a
linux box perhaps) and then tar up that dir and move it to your mac.
We
I noticed tonegroup=es. What country are you in and do you know what
method they use to do dtmf. Most likely we need a small tweak to set
the dtmf method for your country.
Mike
On Jan 14, 2009, at 9:05 AM, Anthony Minessale wrote:
number = 1
This value should be set to the DID of the
Sip cause code to Q.850 cause code translations can be found in
RFC4497 section 8.4.4.
FreeSWITCH uses Q.850 codes internally so you will typically see those
in the logs. We do pass the sip cause codes across a sip to sip bridge.
Mike
On Jan 14, 2009, at 9:18 AM, Alexandru Nedelcu wrote:
Please follow up on this issue on jira.
On Jan 12, 2009, at 8:08 AM, ahgindia wrote:
Hello,
I have tried the new freeswitch release 1.0.2.
But while starting phase only, I encountered freeswitch crashes
frequently.
Here it is :
That is correct, if you want to use the input callback you need to use
the session methods (streamFile ?) to play the file.
Mike
On Jan 8, 2009, at 9:33 PM, Erik Wickstrom wrote:
It seems that it just doesn't work while doing an api call such as
session.execute(playback,
Update on this... the latest of anything to speak of in the standards
process is that they are killing dtmf-relay. The last draft on it was:
http://tools.ietf.org/html/draft-kaplan-sipping-dtmf-package-00
Which says snom is sending it right. We will continue to accept both
ways but need
Can you get a recording of calling your voicemail and post it online
somewhere, I am sure there are some on the list who could tweak to fix
this.
Mike
On Jan 9, 2009, at 8:59 AM, Adam Wilt wrote:
Thanks for the replies. I wrote a script in SpiderMonkey to place a
call, and upon connct
This is broken from a change that just went in this afternoon.. I will fix
it shortly.
Mike
On Fri, Jan 9, 2009 at 11:14 PM, Tim B timb0...@hotmail.com wrote:
1. downloaded from svn
2. built freeswitch .. runs fine
3. tried to build mod_managed ... following error:
[r...@phone2
401 - 500 of 827 matches
Mail list logo
