http://wiki.freeswitch.org/wiki/Getting_Started_Guide
http://wiki.freeswitch.org/wiki/Interop_List
On Nov 25, 2009, at 1:36 AM, ovvenkat wrote:
Hi .
Could you please tell me, How to connect sip phone (which one is more
friendly with freeswitch) to freeswitch. How I can check whether
It sounds like the platform sdk is set up wrong. This used to be a problem
with older versions of express edition. Double check that your compiler works
at all with anything else.
Mike
On Nov 22, 2009, at 11:51 PM, 大泥人 wrote:
All,
I tried to compile FS source code under Windows while
I think this one is kept up to date, but we may re-do this at some point soon,
so it may get re-built.
http://svn.freeswitch.org/freeswitch.git/
Mike
On Nov 23, 2009, at 1:22 PM, William Suffill wrote:
Just wondering if anyone is keeping an update to date git repo of
FreeSwitch? I been
This looks like a nat issue to me, please re-test this against latest svn trunk
and if its still not working pastebin a full sip trace and report the link back
here.
Mike
On Nov 21, 2009, at 6:23 PM, RobertT wrote:
Yep, I use proxy media. First it started with 1.0.4 release, then I've
Try running the info app there and see if the info is anywhere in that output .
Mike
On Nov 23, 2009, at 5:36 AM, Albano Daniele Salvatore - Lavoro wrote:
Hi,
i'm writing some dialplan parts that get executed on execute_on_answer. In
this dialplan that get executed i need to make a
This is done automatically when you bridge 2 sessions together.
Mike
On Nov 23, 2009, at 6:45 AM, Oscav wrote:
How can we send the answer to the caller only when the callee answers, in
JavaScript??
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FreeSWITCH-users mailing list
Take a look at a pcap of the traffic, I suspect the other side still has media
flowing.
On Nov 23, 2009, at 7:00 AM, Maciej Aniserowicz wrote:
Hello,
I have 2 instances of FS: one controlled by my application (making calls
with TCP commands, recording sessions, listening to events etc) and
Yes please
On Nov 23, 2009, at 6:45 AM, Mike Tkachuk wrote:
Hello Anthony,
Is clear, thanks, I'll test and will let you know.
Should I add 'core-db-dsn' parameter description to Wiki? Maybe we need to
add this parameter also to sample conf files?
The Makefile rules that those are built with can all be found in
build/modmake.rules.in. I looked them over real quick and they look right,
maybe try throwing some debug echo statements in there or build with env var of
VERBOSE=1 to see more of what is going on and toss a patch to correct the
Sounds like they don't want your business that much. You can try using mrcp
with them , not sure if they have that released on their side or not. I think
the build integration for mrcp client just went into the windows build earlier
today. To be honest we used to have a pretty good
). My old-fashion
brute-force idea is to symlink the source src/mod/subdirs in the build
src/mod/subdirs right before line 12, changing line 12 to use
$(switch_builddir).
Does anybody have a better idea?
Thanks,
Robert
From: Michael Jerris [mailto:m...@jerris.com]
Sent
Default controls are hard coded. If you want to change them you must use a
name other than default.
Mike
On Nov 23, 2009, at 3:42 PM, Phillip Jones wrote:
Anthony - setting
control action=hangup digits=9/
or
control action=hangup digits=event/
does not make a difference, even
.
Most src/mod subdirs are not using automake and/or configure. They just have
a simple Makefile in with the source.
Robert
From: Michael Jerris [mailto:m...@jerris.com]
Sent: Monday, November 23, 2009 1:09 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users
That rev should have fixed that memory leak, could you test mod_local_stream.c
from rev 15430
(http://fisheye.freeswitch.org/browse/~raw,r=15430/FreeSWITCH/src/mod/formats/mod_local_stream/mod_local_stream.c)
with your current fs version to confirm this is the cause please?
Mike
On Nov 23,
I have done quite a few changes to the build system and correcting build
problems and other platform specific problems the last few days. Could
everyone on the list please take a little time out of their day and do a clean
fresh svn trunk checkout of FreeSWITCH and do a full build and report
Jira is the best, otherwise just mail me the patch and I'll take a
look. Also, I just synced lib up to current trunk. Can you take a
look at my last patch to the module to make it build please.
Mike
On Nov 22, 2009, at 1:02 PM, Arsen Chaloyan achalo...@yahoo.com wrote:
We discussed build
On Nov 22, 2009, at 11:51 PM, 大泥人 qinglan_z...@hotmail.com
wrote:
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FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
This was merged into trunk.
On Nov 20, 2009, at 12:34 PM, Brian West wrote:
Hope on IRC and talk to MikeJ in #freeswitch he can direct you better
on what to do vs not do since he maintains the builds system in
FreeSWITCH.
/b
On Nov 20, 2009, at 11:31 AM, Igor Neves wrote:
Ok,
no.
On Nov 19, 2009, at 1:36 AM, Eli Hayun wrote:
Hi
Is there is a way to intercept an event (for example : REGISTER) and
change one of its parameters (for example: the extension number) and
fire up the corrected event?
I need it to set the speedial of the phone value to be **x but to
I think a better approach here is to use spandsp. We already have some
groundwork done for this. If you are interested in contributing, please email
consult...@freeswitch.org and we can discuss further.
Mike
On Nov 19, 2009, at 6:54 PM, Klaus Hochlehnert wrote:
Hi,
one of my customers
check out sofia_contact function. If you use this in combination with binding
profiles together so they are one table I think this should work right.
Mike
On Nov 18, 2009, at 12:36 AM, Eli Hayun wrote:
Brian West wrote:
Why do you need to know the destination profile like that? You get
This issue is now fixed in trunk.
Mike
On Nov 17, 2009, at 9:05 AM, Christopher Z. wrote:
Hi,
I've got this error after make:
http://pastebin.freeswitch.org/11145
Any idea how to fix this error ?
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Kristian, catch up with me somewhere that I can get remote access to this build
environment so that we can sort this out.
Mike
On Nov 18, 2009, at 2:02 PM, Kristian Kielhofner wrote:
On Wed, Nov 18, 2009 at 12:45 PM, Steve Underwood ste...@coppice.org wrote:
Over time more and more of
Fixed in svn r15526 and other fixes in svn r15527.
mike
On Nov 18, 2009, at 11:40 AM, Robert Hadley wrote:
Hi All,
Anybody interested in helping fix the –srcdir option? I am trying to build
in a subdirectory off the Freeswitch source. I am working on it and finding
issues. However,
Okay, I'll ask the obvious question. Why are you passing record invalid file
paths and why should it not fail if you do?
Mike
On Nov 18, 2009, at 2:38 PM, William Kendi ... wrote:
While I was testing the mod dptools record application using invalid file
paths, i noted that the mod dptools
http://wiki.freeswitch.org/wiki/Mod_xml_curl
On Nov 15, 2009, at 11:39 AM, Samuel Mukoti wrote:
Hi,
I'm a newbie to FS, and I wanted to implement a setup where I
provision the sip endpoints though a SQL database like mysql and also
manage call routing too? Is this possible since I
It doesn't look like your call ever gets setup in this trace, if you enable the
sip trace you might see a bit more, but it looks like we are receiving a 480
response from the called phone.
Mike
On Nov 15, 2009, at 12:42 PM, vedama...@netscape.net wrote:
I am FS beginner and I have a basic
Patches to make this work would be gladly accepted.
Mike
On Nov 13, 2009, at 7:56 PM, Brian West wrote:
Don't use --srcdir we don't fully support that and the howto guides do not
mention it AT ALL. So doing things that are not in the howto aren't really
tested nor supported.
/b
On
Take a look at the freeswitch debug log, it should tell you exactly why it hung
up.
Mike
On Nov 12, 2009, at 10:01 AM, Lei Tang wrote:
Hi, I'm running a ivr script on FS, the call is from a softswitch to extenal
sip endpoint of FS.
I added two dialplan in public dialplan xml file. as flow:
I have asked you before to please not cross post to both mailing
lists. Please refrain from this in the future.
Mike
On Nov 9, 2009, at 6:36 PM, srinivasula reddy wrote:
Hi,
From Freeswitch there is continuously Request: Notify (Messages-
waiting) requests are comming, i didnt
If you can figure out a clean way for us to do this with proper ifdefs
in tree in a way that will not break others that would be the most
preferred.
Mike
On Nov 8, 2009, at 1:03 PM, Bruce Fletcher wrote:
OK, I'll ignore that MacPorts patch for now and try to find a better
approach.
I'll
You don't have ext-rtp-ip set in your config.
Mike
On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote:
Hi!
I have FS natted and am connecting with an 'external' extension that
is registered to FS. ie the extension 2000 is registered on the
internet with a public IP through my router to
0
NOMEDIA false
LATE-NEGfalse
PROXY-MEDIA false
AGGRESSIVENAT true
STUN-ENABLEDtrue
STUN-AUTO-DISABLE false
On Mon, Nov 9, 2009 at 7:28 AM, Michael Jerris m...@jerris.com
wrote:
You don't have ext-rtp-ip set
Those vars were not even available in 1.0.3. I can't recall if they
were in 1.0.4 or if you will need to use the latest 1.0.5 pre-release.
Mike
On Nov 7, 2009, at 9:26 AM, Steven Brown wrote:
Hi
I've been trying to experiment with leg_delay_start when bridging to
two mobiles via a
You can use EventConsumer class for this, I am afraid its not very
documented, but I do recall either a sample or discussion on the
mailing list that you should be able to find.
Mike
On Nov 7, 2009, at 12:38 AM, lakshmanan ganapathy wrote:
Ya. I have done that event processing with ESL.
looks like ogg devel packages are installed but ogg lib is not?
On Nov 7, 2009, at 3:59 AM, Sean Ferguson wrote:
FreeSWITCH seems to be unable to read MP3 files, citing that it's an
unknown format. Looking through the log, I found this during startup:
2009-11-07 02:43:45.749328 [CRIT]
It cleans up after itself fine, but it is an indication of some issue
in the code we need to address. if you can reproduce this in svn
trunk, please file a bug on jira.freeswitch.org with details how to
reproduce.
mike
On Nov 5, 2009, at 12:44 PM, Artem Shiyanov wrote:
Hello!
I have
This would be specific to the zaptel driver for that card, not openzap.
mike
On Nov 5, 2009, at 1:43 PM, Fred-145 wrote:
Hello
As an alternative to more expensive alternatives like OpenVox or
Sangoma,
I'd like to order an X100P clone from www.x100p.com for use in France.
According to a
Call loop?
On Nov 4, 2009, at 10:25 AM, Diego Viola wrote:
Hello,
I tried to help Roy with this issue yesterday, I saw that calls
couldn't go through and then I made a sofia profile internal
siptrace on.
Then I found a message like SIP/2.0 503 Maximum Calls In Progress
and saw he
It means you need to go change the setting from the broken defaults,
thats all.
Mike
On Nov 4, 2009, at 1:48 PM, Dave Stevenson wrote:
Michael et al - and specifically, the FS Developers,
this is all the more annoying given the fact that the SPA-3102 was
bought specifically to run with
That is correct.
Mike
On Nov 2, 2009, at 4:24 AM, Lei Tang wrote:
Hi all,
The problem is solved. I ask the softswitch to send only sdp in
INVITE message, then It works.
I think sofia doesn't support multipart content currently. is it
right?
Please re-try with latest svn trunk.
Mike
On Nov 2, 2009, at 9:36 AM, Humberto Quintana wrote:
Thanks for you answers guys,
I test the parameters you suggested
but still no audio due to the lack of reINVITE. By the way I'm using
1.0.4 but I also tried 1.0.5pre3.
One particular
This may be possible with tcp, how could this work on udp? Can you
provide an rfc reference on this?
Mike
On Oct 24, 2009, at 8:13 AM, Dennis wrote:
ok, as written, i come back after some tests with fs and a thomson
cirpack.
it did not work - at least in our tests.
we are using
You actually can use these in conditions. Just need to be careful
that the var you are conditioning on is already set.
Mike
On Oct 22, 2009, at 1:54 PM, Michael Collins wrote:
On Thu, Oct 22, 2009 at 5:51 AM, Rupa Schomaker r...@rupa.com wrote:
cond would be helpful here? I updated the
We still do plan on branching 1.0 into bugfix only. This has not yet
happened but may happen at some point after 1.0.5. In the mean time,
the vast majority of the work lately has been fixes with small feature
improvements, most all of this would stay in a 1.0 branch even if we
were
Have you answered the call?
On Oct 30, 2009, at 11:34 AM, Rob Forman wrote:
Hm, strange. I haven't seen that before. Can you pastebin your logs
at debug level?
On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote:
It's strange... a tcpdump tells me that there is no DTMF from my
provider
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?
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see rupa's explanation below.
On Nov 1, 2009, at 1:24 AM, Michael Collins wrote:
How would you do an expression like: if $x 24 in a condition tag?
Just curious. I would like to make sure that is properly documented.
-MC
On Thu, Oct 22, 2009 at 5:51 AM, Rupa Schomaker r...@rupa.com
The libcurl is broken on your distro. You can fix this by configuring
with --without-libcurl which will use our working in tree copy instead
of the broken one from your distro.
Mike
On Oct 28, 2009, at 1:39 PM, Erwin Davis wrote:
Hi, I got an error in loading mod_spidermonkey. my fs in
You probably should not be calling that function, what are you trying
to do?
Mike
On Oct 27, 2009, at 8:48 AM, srinivasula reddy wrote:
Hi,
when i am calling switch_xml_open_root(1,err) . i am getting this
warning message.
HEAP[FreeSwitch.exe]: Invalid Address specified to RtlFreeHeap
just api execute reloadxml
Mike
On Oct 28, 2009, at 12:37 AM, srinivasula reddy wrote:
Hi mike,
thank for your reply.
i am trying to call that function from swig.cs. its working fine
first time with the warning information, then onwards it is not
working.
is there other way can i
New sofia profile param as follows:
!-- set this param to false if your gateway for some reason
hates X- headers that is is supposed to ignore--
!--param name=pass-callee-id value=false/--
On Oct 26, 2009, at 12:16 PM, Anthony Minessale wrote:
Thus perpetuating the wild-west of sip
I did test this on trunk and it seems to work right:
freeswi...@default sofia_gateway_data
-ERR Parameter missing
Mike
On Oct 22, 2009, at 3:58 PM, Michael Collins wrote:
What SVN rev of FS? What operating system? If you're not on the
latest then do a make current and get to the latest SVN
This appears to be some sort of ice implementation? We don't support
sip ice at this time.
Mike
On Oct 21, 2009, at 7:58 AM, ineya ineya wrote:
Codecs are fine. I spent much time experimenting with codecs and
completely missed, that freeswitch is modifiyng the SDP record.
When phone A is
The syntax is different, but the api is the same as lua:
So you need an API object in order to use it. I don't know the
syntax for creating an api obj in Java but in Lua it goes like this:
api = freeswitch.API();
res = api:execute(sched_api,+300 none my_api my_api_args)
create the API
This should now be fixed in latest svn trunk.
Mike
On Oct 21, 2009, at 12:45 PM, Keith Laaks wrote:
Hi,
Hope someone knows how I am able to get around this one. Here goes...
Did an upgrade to trunk (from a July vintage build) last week and
noticed calls out to a provider were now
REPORTING is the state that it writes to CDR. If you have calls stuck
in this state, take one and try to use uuid_kill on it and see if it
goes away, then get a core off of it and pastebin the thread apply all
bt (with no other calls up). What modules are you using for cdr and
with what
If you really want to access this information outside I would strongly
recommend using odbc instead of the internal sqlite db, it does not
handle locking contention well. If you need access to things in the
core db (like show calls and show channels information) you will need
to write a
we added some params for new better automatic nat handling, grep the
new defailt configs for localnet and you will find what you are missing.
Mike
On Oct 18, 2009, at 11:14 PM, Chris Fowler wrote:
I've tried all sorts of debug and parameter changes over the weekend,
but still can't figure
There is an event you can send as well to switch them, it your trying
to switch it via event socket, that should be better, its not on the
wiki, but
a session message with
eavesdrop-command header with data as the same as dtmf
should do the trick
Mike
On Oct 16, 2009, at 11:54 AM, Nikita
Try starting out reading this.
http://wiki.freeswitch.org/wiki/Mod_managed
Mike
On Oct 19, 2009, at 9:14 AM, srinivasula reddy wrote:
Hi,
How can i use freeswitch.managed project. what are the parameters
for calling Execute method? and how can i call?
any help
You need a sofia profile for each identity, if your using multiple
external ip addresses, you will need a profile for each. If you are
using bgp or something of the sort and only using one external ip, you
can use a single profile and route using standard routing.
Mike
On Oct 13, 2009, at
I can't recall if we ever exposed an option for this, take a look at
sofia-sip and see if they have a tag to enable this, if so it would
probably be a fairly simple patch to add.
Mike
On Oct 15, 2009, at 3:20 PM, Alexandre Savard wrote:
Hi,
Does Freeswitch support TLS
inline is new, it won't work unless your using recent trunk. That
being said, read is not being run inline, so the set is actually being
run before digits_dialed is set. You will most likely need to use
transfer in this situation.
Mike
On Oct 19, 2009, at 12:53 AM, Mark Campbell-Smith
This is what force-rport is supposed to do. That being said, I can't
tell from your trace where it is actually going to, just what it says
in the packet, which can be different.
Mike
On Oct 19, 2009, at 3:23 AM, Tzury Bar Yochay wrote:
Hi,
I am struggling with a cellular operator which
Try out trunk and see if this issue is resolved please.
mike
On Oct 19, 2009, at 3:11 AM, Durk de Beer wrote:
Hello,
This is something I came across on Freeswitch 1.0.4
First let me explain what I'm trying to do.
I want Free-Switch to behave as a proxy so in the settings section
of
is the syntax for this session
message?
I tried this:
sendmsg e8e4f0ed-a0cc-4dff-b7e1-09eeade5df05
eavesdrop-command: 1
but it doesn't work.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-
users-boun...@lists.freeswitch.org] On Behalf Of Michael
we do have a license for this, people didn't seem to like it last time
we looked at it, I can't recall why.
On Oct 19, 2009, at 4:24 PM, Roberto Martins wrote:
what about http://www.atlassian.com/software/confluence/ they give
free licenses to open source project, and FS is using JIRA.
-please_enter_pin_followed_by_pound.wav res 1 9/ in my
case) or is it not 'set' until after the transfer?
action inline=true application=set data=code=$
{digits_dialed}
Thanks!
On Tue, Oct 20, 2009 at 12:32 AM, Michael Jerris m...@jerris.com
wrote:
inline is new, it won't
.
--matt
On Sat, Aug 8, 2009 at 12:42 PM, Michael Jerris m...@jerris.com
wrote:
http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP
turn the logging all the way up and see what it says.
Mike
On Aug 4, 2009, at 2:51 PM, Matthew Fong wrote:
Hi Mathieu, thanks for the reply. I enabled sip
I think we strongly lean towards using a sub-domain (if necessary) and
maintaining other language content in the same wiki in alternate
pages. If the wiki software we are using is not effective to create
multiple languages we should find a way to do it all in one. We can
set up an
If you don't have working stun, jingle is not going to work very
well. It is a required part of the protocol. You need to be able to
determine your external ports for media on each call, using a host
name will not do this for you.
Mike
On Oct 16, 2009, at 10:48 AM, Brian West wrote:
sched_api is a fsapi command not a dialplan application, I believe
sched_hangup is both.
Mike
On Oct 13, 2009, at 6:14 AM, Henry Huang wrote:
Hi:
I am using mod_java. And in my script I was able to achieve using:
execute(sched_hangup, +300 alloted_timeout);
However, when I try to run
I would love to see this work in tree, but i am pretty sure it has
never worked. I would gladly accept patches that implement this.
Mike
On Oct 14, 2009, at 2:33 AM, Simon J Mudd wrote:
br...@freeswitch.org (Brian West) writes:
You shouldn't have to make clean usually ... doing so might
There was just a bunch of work on UPDATE, can you confirm this is the
same behavior with trunk?
On Oct 14, 2009, at 6:55 AM, Peter P GMX wrote:
Hello,
we have the following problem.
2 Fax machines are communicating via Freeswitch. One is externally
attached via a Telco who is able to
Try turning up all the sofia debug to 9.
Mike
On Oct 14, 2009, at 2:16 AM, Szasz Szabolcs wrote:
Hi,
Did anybody set up TLS between Freeswitch and Audiodes MP11X ? I got
to work TLS between freeswitch and a softphone (phonerlite), but I
have problem with Audiocodes during the TLS
, 2009 at 5:04 AM, Michael Jerris m...@jerris.com
wrote:
We don't have session messages directly exposed, except for things
like display, respond, and deflect. What specifically are you trying
to send ?
Mike
On Oct 9, 2009, at 3:34 PM, Matthew Fong wrote:
I'm used to using the onInput
Group information is not stored in sqlite, it is pulled from the xml
registry (switch_xml_locate_group function can find them) . Also,
please do not cross post between lists.
http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups
http://wiki.freeswitch.org/wiki/Mod_commands#in_group
On Oct 7, 2009, at 10:48 AM, srinivasula reddy wrote:
Hi
can any please tell me where registered calls are stored, so when
incoming call came to mod_sofia.c how it will check it is registered
or not?\\
Calls are not registered and calls have nothing to do with
registration. Users
On Oct 9, 2009, at 2:41 AM, Gabriel Gunderson wrote:
On Fri, Oct 9, 2009 at 12:08 AM, Michael Collins
m...@freeswitch.org wrote:
Thanks for reporting back. Please let all the Asterisk users know
that they
are welcome to join us in #freeswitch on irc.freenode.net and that
they will
On Oct 9, 2009, at 10:40 AM, Maciej Aniserowicz wrote:
Hello,
The issue is resolved. I feel stupid, because Michael Jerris was
right the first time. Setting external_rtp_ip and external_sip_ip to
$${local_ip_v4} made it work.
But the strange thing is: it SOMETIMES worked before without any
There is this endless push and pull on this topic, those who want them
assume it should be default, those who don't assume that should be
default. This probably needs a configuration option defaulting to
pass them (those who don't want to pass them are usually a bit more
educated and
We don't have session messages directly exposed, except for things
like display, respond, and deflect. What specifically are you trying
to send ?
Mike
On Oct 9, 2009, at 3:34 PM, Matthew Fong wrote:
I'm used to using the onInput callbacks inside lua and javascript to
listen for dtmf
anyway).
Codecs are the same on both legs:
read codec/read rate: PCMU 8000
write codec/write rate: PCMU8000
MA
Michael Jerris wrote:
What codecs are all the call legs using, also, please try current
svn
trunk.
Mike
On Oct 7, 2009, at 3:39 AM, Maciej Aniserowicz wrote
On Oct 11, 2009, at 5:44 PM, Diego Viola wrote:
Maybe you want enable_heartbeat or sched_heartbeat from mod_dptools?
You can pass your parameters in second to these two.
Example:
action application=enable_heartbeat data=1/
action application=sched_heartbeat data=1/
Where 1 in this case
I am still working on the new build system for esl, stay tuned for
more info soon, it should be in 1.0.5.
Mike
On Oct 11, 2009, at 5:36 PM, Herman Griffin wrote:
Although probably not the best solution, I figured out a way to make
it compile and install:
I removed all of the -Werror
On Oct 9, 2009, at 7:58 AM, srinivasula reddy wrote:
Hi all,
does any know about How apr_queue is maintaing and retriving all
registered and all stuff
parse error
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What codecs are all the call legs using, also, please try current svn
trunk.
Mike
On Oct 7, 2009, at 3:39 AM, Maciej Aniserowicz wrote:
Sorry about posting several questions at once, I wasn't aware it's
rude.
Let's concentrate on this issue then.
I use FS rev 14994. Phones on
switch_ivr_async.c:480
On Oct 5, 2009, at 3:16 AM, Maciej Aniserowicz wrote:
Hi,
When I record a call in FS, it only creates a 388-byte-long wav
file. The conversation is no written there, and FS deletes the file
when the session finishes.
What can cause this strange behavior?
Incorrect NAT configuration so one of the boxes is not actually
getting a BYE.
On Oct 5, 2009, at 3:13 AM, Maciej Aniserowicz wrote:
Hi,
When I use two FreeSWITCH instances ('internal' and 'external'), all
users register to the 'external' instance which acts as a gateway by
'internal'
On Oct 6, 2009, at 4:14 AM, Yehavi Bourvine wrote:
Hello,
We have Polycom and SNOM phones running with FreeSwitch. The
Polycoms have shared lines defined and the SNOMs have both shared
lines and BLFs (defined as extensions in the phone config). I've
tried supporting both, but have
As I said in the duplicate thread, the voip codecs issue has been
resolved in trunk, I had a change 1/2 done waiting for testing and it
is now complete.
Mike
On Oct 6, 2009, at 12:30 AM, David Clark wrote:
No I found the one header. I added it to the include list for the
project. It
I am not sure what you mean, do you think that fixes from today should
somehow go somewhere else before we do a release?
On Oct 6, 2009, at 3:21 AM, Vladimir Elizarov wrote:
Brian West пишет:
Because TRUNK is stable... its only fixes going in usually and if
things do break they don't stay
Could you open a bug on jira.freeswitch.org as a feature request to
make this a configurable param. (patches that do it even better)
Mike
On Oct 6, 2009, at 12:55 PM, Christian Damianidis wrote:
I’ve tested this and making the change from ANY to BASIC worked.
Thanks for the help.
It no
'tones' are standardised and the ones on the wiki are correct?
Also, I guess this doesn't work with media bypass (which I don't
use).
Thanks!
On Mon, Oct 5, 2009 at 9:56 AM, Michael Jerris m...@jerris.com
wrote:
check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
http://wiki.freeswitch.org/wiki/Mod_voicemail#voicemail_inject
On Oct 5, 2009, at 9:46 AM, Nagalenoj wrote:
Is it possible to treat a recorded voice as voice mail?
Assume that, I've recorded a conversation and I want this recorded
file to
be treated like voicemail. So, I could check it
On Oct 5, 2009, at 10:49 AM, Woody Dickson wrote:
Is is possible to override any of the setting specified in the
conference profile?
Just the flags you can pass per user such as pin and mute
What I want to do is to have a default profile, and be able to
modify certain fields if
voip codecs is fixed, ptlib I can't recall if we ever did full build
integration or if you needed to manually download the libraries, can
someone who has done mod_opal build on windows comment?
Mike
On Oct 5, 2009, at 5:14 PM, David Clark wrote:
Ok I found spandsp.h. It is a case of the
there is a profile param to enable 3pcc. It should be documented in
the default configs.
Mike
On Sep 29, 2009, at 5:22 PM, Jerry Richards wrote:
Hello All,
I have an internal extension that needs to send an INVITE without
SDP body
(Content Length 0). Freeswitch is replying with 480
This sounds like a bug in the snom to me, we keep changing the expire
on to the future so it should never expire in the first place. You
will have to look at a longer running sip trace to see what exactly is
going on.
Mike
On Oct 1, 2009, at 4:52 AM, Helmut Kuper wrote:
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