This is not currently possible. It's something that could be added
but would require a rework of mod_local_stream
Mike
On Sep 21, 2008, at 3:15 PM, Cesar Cepeda wrote:
Hi,
I need to create and destroy local_streams dynamically, that is, I
need to be changing the MOH of several fifo’s in r
On Sep 21, 2008, at 4:11 PM, xbipin wrote:
>
> ok i marked bypass media as well as proxy media to default which was
> like
> commented, means marked as comment. With proxy media to enabled i
> used to
> get the error of cant find codec but after making it all to default
> config,
> im now g
On Sep 21, 2008, at 5:00 PM, xbipin wrote:
>
> file doesnt exist but doesnt it need to create it by itself as u
> never know
> whose gonna call when and where.
> basically im looking for it to create the file and record in it so i
> can get
> a different file for each user, whats the point cr
On Sep 22, 2008, at 12:49 AM, preetha Ayyappan wrote:
I have put the calltest.js in /usr/local/freeswitch/scripts and
changed sofia to openzap/default/[EMAIL PROTECTED] in the coding and
i got the error:
Error:
2008-09-22 10:13:26 [ERR] switch_core_session.c:249
switch_core_session_outg
tcapi is currently a developer only project. You can catch up to the
developers and discuss any contribution you can offer in the #tcapi
channel on irc.freenode.net.
Mike
On Sep 22, 2008, at 12:44 PM, xbipin wrote:
>
> hi,
>
> i did a snv update for tcapi which is the web frontend for
>
I think that macro IS defined, this sounds like a messed up
bootstrap. Could you bootstrap and configure again and see if it helps?
Mike
On Sep 23, 2008, at 6:52 AM, Jon Bruel wrote:
I do have c++ installed, and the release version 1.0.1 did install
OK. On the other hand 1.0.1 came with th
On Sep 24, 2008, at 12:08 AM, Juan Backson wrote:
> Hi,
>
> My lua scripts were working fine until I updated with SVN. I am
> starting to get errors in my lua scripts that use luasql lib.
>
> Does anyone know what may be causing the problem and how can I fix it?
We have not made any major c
We never added this to mod_spidermonkey, we did add it to mod_lua and
some others. You could either add the capability to mod_spidermonkey
or have a lua script launch at startup that starts a js using jsrun.
Mike
On Sep 24, 2008, at 1:40 AM, preetha Ayyappan wrote:
> Hi,
> I am trying to st
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_privacy
What your doing just sets the privacy flags, it still sends the caller
id information, just with the hide flags turned on. You may want to
set caller id number and name explicitly blank if you do not trust the
downstream to handle
On Sep 24, 2008, at 9:33 AM, Ryan, Jay wrote:
Hi,
I am new to freeswitch. I have a few questions:
1. Is there a way to get a VoiceXML browser (e.g. i6net) glued into
freeswitch?
It would require some coding, but the major pieces and interfaces are
there to do so.
2. How about a more
08 at 10:46 PM, Michael Jerris <[EMAIL PROTECTED]>
wrote:
On Sep 24, 2008, at 12:08 AM, Juan Backson wrote:
> Hi,
>
> My lua scripts were working fine until I updated with SVN. I am
> starting to get errors in my lua scripts that use luasql lib.
>
> Does anyone know wh
On Sep 24, 2008, at 6:17 PM, Jair Santos wrote:
Hi,
If I call ext 1000 the voicemail system answer on timeout . If I
call a DID that is linked to that same extension it returns a busy
signal when it is trying to call the VM.
In my public.xml I have
expression="^(3105
We have confirmed that the change in svn revision 9605 broke this. I
am going to look at ways to fix this in the morning.
Mike
On Sep 24, 2008, at 8:50 PM, Juan Backson wrote:
Hi,
The one that is working has version: FreeSWITCH Version 1.0.trunk
(9588)
The one that is not working has ver
On Sep 25, 2008, at 2:31 AM, sambasivarao Vemula wrote:
HI,
I want establish trunk between asterisk and free switch .Is there
any special procedure for establishing trunk..?
Please forward configure details.
Regards
Samba
DISCLAIMER == This e-mail may contain privileged and
con
On Sep 25, 2008, at 4:47 AM, preetha wrote:
>
> Hi,
> when i try to run a sample odbc code from freeswitch console like
> [EMAIL PROTECTED]> jsrun odbc.js
>
> I found the following error:
> API CALL [jsrun(odbc.js)] output:
> OK
>
> [EMAIL PROTECTED]> 2008-09-25 19:44:35 [ERR] mod_spidermonkey.c:
On Sep 25, 2008, at 7:19 AM, Peter P GMX wrote:
> I figured out (via ngrep) that freeswitch didn't even try to contact
> the
> registered phone. However it does a lookup to the Directory of the
> target phone. So there may be something wrong with the bridge command?
>
> BTW: To be honest: I hav
On Sep 25, 2008, at 4:04 PM, Peter P GMX wrote:
> Hello Michael,
>
> thanks for the hint, but how shall a dial-string param look like? I
> looked up the internet but could not find an example.
> Can you provide an example?
>
> Best regards
> Peter
>
its just an originate string like you use with
They should all be there. Did you do "myevents" ?
Mike
On Sep 26, 2008, at 12:28 AM, "Adeel Ansari" <[EMAIL PROTECTED]>
wrote:
Thanks, its working like charm. Just curious, common events such as
CHANNEL_ANSWER, CHANNEL_BRIDGE, CHANNEL_HANGUP etc.. don't work with
mod_sofia?
On Thu, Se
ption on your ./configure
like so
./configure --enable-core-odbc-support && make && make install
-Ray
preetha Ayyappan wrote:
You mean i have to uncomment the line module="mod_spidermonkey_odbc"/>.?
yes.I have uncommented that line.eventhough the same error occurs
On Sep 29, 2008, at 3:14 AM, preetha Ayyappan wrote:
> Thanks.I have uncommented the line yoy specified and run the program
> sample.js from the freeswitch console.It shows the following error:
> [EMAIL PROTECTED]> 2008-09-29 18:13:03 [ERR] switch_odbc.c:160
> switch_odbc_handle_connect() STA
No one can say what your performance will be, what I can say is the
results you are getting are highly abnormal from what I have seen.
Try it for yourself and see.
Mike
On Sep 29, 2008, at 5:26 AM, Jon Bruel wrote:
The load on the CPU was after the calls were set up, this indicated
that’
if you rm all the sofia*.db from the db dir they will regenerate
properly.
Mike
On Sep 29, 2008, at 11:24 AM, Jim Flowers wrote:
> My FreeSWITCH Version 1.0.trunk (9609) with samples loaded
> continually throws
> an error when answering a call.
>
> There is no `profile_name` column in any ta
On Oct 2, 2008, at 3:00 PM, Jon Bruel wrote:
> I have made some further studies of the performance, and after I have
> removed and old PC running 10 Mb/s on the Ethernet, the performance
> has
> been drastically improved. So I tentatively think that there has
> been a
> bottleneck somehow in
On Oct 3, 2008, at 4:54 AM, Vito Andolini wrote:
Hi All,
I am familiar with Asterisk and doing some testing for my next
project. I have had some difficulties Asterisk, and now researching
FreeSwitch hoping that it has some out-of-the box answers for my
questions.
Basically I want to im
On Oct 4, 2008, at 8:39 AM, Ivan C Myrvold wrote:
>>
>
> Yes, after som more playing with it, I also have problems receiving
> calls. Originating is OK.
> I first thought this was an ordinary SIP client running on the iPhone,
> but I see now it isn't.
Like all iphone apps, its only running while
On Oct 4, 2008, at 8:22 AM, Vito Andolini wrote:
Let's say I am programatically initiating two calls and then
bridging them together. If I have the dialplan as
originate sofia/example/[EMAIL PROTECTED] &bridge(sofia/example/[EMAIL
PROTECTED])
and have the process_cdr set to "true" which is
On Oct 4, 2008, at 4:20 PM, Vito Andolini wrote:
Let's say I am programatically initiating two calls and then
bridging them together. If I have the dialplan as
originate sofia/example/[EMAIL PROTECTED] &bridge(sofia/example/[EMAIL
PROTECTED])
and have the process_cdr set to "true" which is
On Oct 4, 2008, at 4:45 PM, Vito Andolini wrote:
what do you mean by look at the cdr?
I checked these 2 wiki pages bu tthey provide only SOME of the
fields not all...
http://wiki.freeswitch.org/wiki/Mod_xml_cdr
http://wiki.freeswitch.org/wiki/Mod_cdr_csv
I also checked the API section but
We just swapped out the mp3 decoder library with a new one in order to
fix these problems last week. You might want to give trunk a try, I
think it should be much better.
Mike
On Oct 5, 2008, at 2:29 PM, Mark D. Anderson wrote:
> I apologize in advance that I can't characterize this better
Do you have autoconf installed on your system and in your path and is
it 2.59 or later?
Mike
On Oct 6, 2008, at 12:41 AM, John Lum-Wah wrote:
> Hi, I'm new to Freeswitch. I tried installing Freeswitch using the
> commands from the wiki
>
>
> svn checkout http://svn.freeswitch.org/svn/freeswitc
Your missing enable the module in spidermonkey.conf.xml
On Oct 6, 2008, at 9:42 AM, Gayatri Kulkarni wrote:
connect to a remote database using javascript.
When i searched the WIki I got this page:
http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc#SpiderMonkey_ODBC
I did the steps what are
Check out http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/languages/mod_python/python_example.py
for more information on how you need to structure your python
scripts. Please note that the information on the wiki at :
http://wiki.freeswitch.org/wiki/Mod_python
is out of date. If som
On Oct 7, 2008, at 9:34 AM, Gayatri Kulkarni wrote:
> If i have to run my script on Windows m/c, will i still need unixODBC?
> Is unixODBC available for windows???
On windows we use the native odbc interfaces and it builds by default
(no need to do anything special to enable it)
Mike
__
On Oct 8, 2008, at 3:21 PM, Nicholas Amorim wrote:
>
> I'm building a web interface with Python/Django.
>
> Freeswitch will run on a separate server and fetches the
> information using
> xml_curl. That's working fine.
>
> What I want to do is:
>
> I want that for every voicemail received, free
Voicemail metadata is already stored in a database (of your choice via
odbc) and if you store the files on some remotely mountable location
you should get the same effect. I'll try to throw an event in today
but I think some of what your trying to do is already done for you.
Mike
On Oct 8
On Oct 9, 2008, at 7:33 AM, Peter P GMX wrote:
> Hello,
>
> is it possible to play special ring tones or a wav file before
> answering
> the call? I do not really believe so, but maybe there is a chance?
If you playback prior to calling the answer application it will
playback in early media.
On Oct 9, 2008, at 8:01 AM, Gopal krishnan wrote:
> Hi,
>
>I am trying to record thru telnet with sendevent record and also
> tried sendevent record_session but I cant able to record. Is there
> any command to record thru telnet?
http://wiki.freeswitch.org/wiki/Event_Socket#SendMsg
http:
On Oct 9, 2008, at 7:53 AM, Peter P GMX wrote:
> I have another 3 Questions. I know I had already 2 before within the
> last 15 minutes, but I need to qualify whether we can build this
> special
> app with freeswitch or not.
>
> The questions are:
> 1.)
> When I do a uuid_playback I want to be
This is now fixed in trunk:
http://fisheye.freeswitch.org/changelog/FreeSWITCH/?cs=9917
Mike
On Oct 6, 2008, at 10:25 PM, David Knell wrote:
> Going back a step, to where Jon was seeing more packets than there
> should have been, I've just encountered a similar issue having
> upgraded
> to th
On Oct 10, 2008, at 10:54 AM, Alfred Richmond wrote:
Hello,
I am attempting to generate a message to convert to speech and send
it out to my users. I am a newbie but I am just not getting it after
reading through the documentation. In testing it works fine when
sending to my voip connecte
As there seems to be quite some interest in G.729, those interested
who can commit to purchasing licenses, please update this wiki page:
http://wiki.freeswitch.org/wiki/Bounty#G729_Licensing_Bounty
Mike
On Oct 10, 2008, at 11:55 PM, Alex Vostrikov wrote:
> Mitul Limbani wrote:
>
> yeah yeah,
On Oct 11, 2008, at 4:00 PM, Cesar Cepeda wrote:
> How much is a license in dollars?
This is not anything we have set in stone, but you can expect it to be
competitive with other solutions out there.
Mike
___
Freeswitch-users mailing list
Freeswi
This issue is fixed as of svn r9881.
Mike
On Oct 12, 2008, at 1:14 PM, "Mark D. Anderson" <[EMAIL PROTECTED]>
wrote:
>
> On Sun, 12 Oct 2008 11:52:28 -0500, "Brian West"
> <[EMAIL PROTECTED]> said:
>> I think you also need to clarify you're on a 32bit platform?
>
> yes, linux 2.6.24 32-bit s
There are some example extensions in the default configs that do just
this:
http://wiki.freeswitch.org/wiki/Call_Groups
On Oct 13, 2008, at 10:49 AM, Meftah Tayeb wrote:
hi,
please ho to add a number for a call queue ?
i want to this number to by a call queue:
ringing in all selected users
On Oct 13, 2008, at 11:39 AM, Jon Bruel wrote:
Brian, there is a reason for the same address: it's connection to an
Asterisk server. My original question remains: Does the originate
command work as it should ("originate user/[EMAIL PROTECTED],user/[EMAIL PROTECTED]
206 XML internalpreparat
On Oct 14, 2008, at 9:17 AM, gary wrote:
1) Do I need to reload directory into FS after I added new users in
XML file? How? reloadxml?
correct
2) Can I use alias for user? I'd like to register the user with 10
digits DID and also assign 4 digits extension to the same user so
people ca
Every time I have set this up in the past with the avaya I have used
ip auth and an ip trusted peer on the avaya side.
Mike
On Oct 14, 2008, at 10:18 AM, Gerry Hull wrote:
On Mon, Oct 13, 2008 at 12:59 PM, Brian West <[EMAIL PROTECTED]>
wrote:
You need to add otherwise
the register contac
On Oct 14, 2008, at 11:20 AM, Kristian Kielhofner wrote:
> Hello everyone,
>
> I'm looking to build a distributed voicemail platform with
> FreeSWITCH. What is the current recommended way to store messages,
> recordings, etc in a distributed manner? Would a simple NFS share
> work? Not that I
For this we should be looking at perpetual sounds, not moh.
Mike
On Oct 14, 2008, at 12:42 PM, Sheeju Alex wrote:
> huh..might be I sense a small gap before MOH is played again after
> this command.
>
>
> On Tue, Oct 14, 2008 at 9:55 PM, Brian West <[EMAIL PROTECTED]>
> wrote:
>> I would bet
can you record the audio and see if there is a gap in the middle of
the 6 digit?
Mike
On Oct 14, 2008, at 1:10 AM, Alex Vostrikov wrote:
> hi guys,
>
> here is ivr i'm trying to use. but i've got a problem that first
> dtmf digit detected twice.
> maybe i'm doing something wrong? as a worka
On Oct 14, 2008, at 1:13 PM, Birgit Arkesteijn wrote:
> Hi all,
>
> I'm trying to manage a conference call in js. I found an example on
> the
> FS wiki:
> http://wiki.freeswitch.org/wiki/Examples_confcall_js
>
> I know that using
> session.execute("conference", route + "@default");
> I can drop
On Oct 14, 2008, at 11:51 AM, Sheeju Alex wrote:
> Also I think it would be good option if we could control moh through
> api
>
> say,
> conference moh stop
> conference moh start
>
> Sheeju
I think "stop" will make it stop, I thought there was a way to start
it but on quick review I didn'
On Oct 14, 2008, at 11:15 AM, Kristian Kielhofner wrote:
> Hello everyone,
>
> When you enter a conference without any other users, you hear the
> "you are the only person in this conference" recording with MOH in the
> background. Is there a way to make the MOH play after the other
> recording
at 12:14 PM, Sheeju Alex wrote:
> Michael, No this doesn't stop MOH
>
> [EMAIL PROTECTED]> conference 7 stop all
> API CALL [conference(7 stop all)] output:
> Stopped 1 files.
>
> I see Stopped 1 files but MOH is still running.
>
> Sheeju
>
> On Tu
On Oct 14, 2008, at 11:47 AM, Kristian Kielhofner wrote:
> On Tue, Oct 14, 2008 at 11:32 AM, Michael Jerris <[EMAIL PROTECTED]>
> wrote:
>>
>>
>> I'll take a look and see how hard it is to hold off the music
>> starting. Can you pass me a bug on http:/
Tony's patch.. my email :D
On Oct 14, 2008, at 4:18 PM, Kristian Kielhofner wrote:
> On Tue, Oct 14, 2008 at 3:40 PM, Michael Jerris <[EMAIL PROTECTED]>
> wrote:
>>
>> Fixed in svn revision 10015.
>>
>
> Te
On Oct 14, 2008, at 12:14 PM, Sheeju Alex wrote:
> Michael, No this doesn't stop MOH
>
> [EMAIL PROTECTED]> conference 7 stop all
> API CALL [conference(7 stop all)] output:
> Stopped 1 files.
>
> I see Stopped 1 files but MOH is still running.
>
> Sheeju
Hmmm.. I swear we had a way to d
For the first call, if all you need to do is send it into conference,
you can just use the originate api command with apiExecute.
Mike
On Oct 15, 2008, at 10:12 AM, Birgit Arkesteijn wrote:
> Hi Mike,
>
> Thanks again for your answer.
> I found the page:
> http://wiki.freeswitch.org/wiki/ApiEx
On Oct 14, 2008, at 1:24 PM, Gerry Hull wrote:
> Mike,
>
> We have the freeswitch box as a trusted host. Still getting the
> 407. Any other ideas?
>
> Gerry
If your still getting a 407 then either your trusted host is setup
wrong on the avaya side or possibly your sending to an extension t
t; westhawk/0662");
> while (cSession.ready()) {
> console_log("info", "put cSession in conf");
> cSession.execute("conference", "[EMAIL PROTECTED]");
> console_log("info", "after put cSession in conf");
> }
If you are seeing nothing at all on the console with all that set,
then the packets are never getting to FreeSWITCH. My first guess
would be either firewall or bound to the wrong ip/port.
Mike
On Oct 16, 2008, at 9:27 AM, Gavin Henry wrote:
> Hi All,
>
> I'm trying to get a SIP forwarded ca
We need to be in media path to do the ringback in an attended
transfer. There are some new params in trunk that will make it pop
back out of the media path on the completion of transfer.
Mike
On Oct 16, 2008, at 11:41 AM, Ruchir Brahmbhatt wrote:
If you want to do transfer then fs should
2008-10-16 15:54:29 [NOTICE] switch_ivr.c:1116
switch_ivr_session_transfer()
Transfer sofia/internal/[EMAIL PROTECTED] to [EMAIL PROTECTED]
2008-10-16 15:54:31 [INFO] switch_core_state_machine.c:114
switch_core_standard_on_routing() No Route, Aborting
Your routing to enum for extension an
this error means you still have it in modules.conf.xml. Remove it.
On Oct 17, 2008, at 5:30 AM, Baskar wrote:
Hi,
I did that too but still same error
please correct me where i am wrong
On Fri, Oct 17, 2008 at 2:20 PM, Brian West <[EMAIL PROTECTED]>
wrote:
mod_spidermonkey_odbc is loaded
I think I see what is going on here, will try to get it set up to test.
Mike
On Oct 17, 2008, at 7:49 AM, Leon de Rooij wrote:
> Hi,
>
> Sorry it took a while, but I wanted to see for myself whether I could
> find out what the problem is, though I still don't see it..
>
> Part of sofia.conf on t
It is possible you have some providers who do not support rfc2833
which is the default. You would need to look at the sdp's exchanged
to know for sure.
Mike
On Oct 17, 2008, at 3:20 PM, Noah Silverman wrote:
> Hi,
>
> Weird situation.
>
> I am now running my office phone through freeswitch.
You don't have the c++ compiler or other things necessary to compile c+
+ code installed. Install it, configure again.
Mike
On Oct 19, 2008, at 11:46 PM, Woody Dickson wrote:
> Hi,
>
> I am installing the latest Freeswitch build from SVN, but the
> following errors are encountered:
>
> gcc -
On Oct 20, 2008, at 4:26 AM, HM Kias wrote:
> Hi ,
> Has anyone figured to create a call file , so that we can auto-
> dialoot . I dont see a spool directory for FS.
There is no support for call files, but there is a number of different
ways to run the fsapi commands such as originate:
http:
On Oct 20, 2008, at 8:42 AM, jay binks wrote:
> Is there an easy way to define a separate log file per SIP Profile ??
No.
> Im trying to emulate the apache function of ( optionally ) having
> separate logs per virtual server.
MIke
___
Freeswitch-
On Oct 20, 2008, at 9:16 AM, jay binks wrote:
> would this be a feature request worthy of addition ?
> I certainly think so.
>
> I have a bunch of customers, each with their own internal sip
> profile ( separate private networks to my sip cluster )
> having this means I can easily watch the log
On Oct 20, 2008, at 9:21 AM, Gayatri Kulkarni wrote:
> Hi guys,
> I read through mod_conference page on wiki
> but i am still not clear about how do i initiate a conference after
> i do the configuration
>
> conference dial user1 ?
Call it or originate a call to it:
http://wiki.freeswitch.o
you can just as the last exten in your context do a transfer to the
other context.
Mike
On Oct 20, 2008, at 5:11 PM, Michael Collins wrote:
>> Sure:
>>
>> [include]
>> exten => 727,1,DoStuff
>> exten => 800,1,DoOtherStuff
>>
>>
>> [default]
>> include => include
>>
>> I know it's a simple cas
just added vm-a1-hash as well that you can use to override the
standard a1 hash for voicemail use only.
Mike
On Oct 20, 2008, at 7:27 PM, Anthony Minessale wrote:
> if you want to test latest trunk i added code that *should* let you
> auth the vm using the same
> a1-hash also we added an "m
On Oct 21, 2008, at 10:56 AM, Arturo Díaz Almagro wrote:
> Hi all,
>
> I am new in the FreeSWITCH world and I am not been able to discover
> how FS can acts as a pure SIP forwarder, sorry. My problem is that I
> need a kind of B2BUA "in the middle" of the signaling path between
> my SIP pho
can you post debug logs of this output?
Mike
On Oct 23, 2008, at 6:43 AM, Birgit Arkesteijn wrote:
> Hi,
>
> Thanks Anthony for your response.
> Unfortunately removing the 'kick' didn't make a difference.
>
> The result is also different if I change the order of the UUIDs
> around.
> In one sce
On Oct 23, 2008, at 11:23 AM, Birgit Arkesteijn wrote:
> Hi Anthony,
>
> Well, that definitely improved the situation, thanks!
> I'm running 498:8901, installed on 2008-10-08.
revision 8901 is from july 7 of this year. I would say you would have
to update to trunk and try this again to be sur
There is some work towards earlier drafts in the sip stack, but I
would doubt we are 100% in sync with the latest drafts.
Mike
On Oct 24, 2008, at 3:09 PM, paulo leonardo wrote:
Hi list,
i would like to know if freeswitch implements draft sip-outbound or
i can active this draft.
http://t
This should work on a fresh checkout.
Mike
On Oct 27, 2008, at 10:23 AM, Tamas Cseke wrote:
> Hello,
>
> I have a problem with windows build:
>
> Microsoft Visual C++ 2005
> Revision: 10158
>
> Creating library Debug/FreeSwitchCore.lib and object
> Debug/FreeSwitchCore.exp
> switch_apr.obj : e
Unsure at this time. There has been some work on mod_cdr_odbc. We
generally advise against direct to db cdr methods without a very
robust backup method for when the db is down.
On Oct 29, 2008, at 9:57 AM, "[EMAIL PROTECTED]" <[EMAIL PROTECTED]> wrote:
> Hi,
>
>I saw in the wiki that th
They should already be on the initial events. Take a look at the raw
output, you probably were taking them out of a later event.
Mike
On Oct 29, 2008, at 2:14 PM, Andy Spitzer wrote:
> Woof!
>
> On Wed, 29 Oct 2008 09:10:32 -0400, Anthony Minessale
> <[EMAIL PROTECTED]> wrote:
>
>> by default
, Tamas Cseke <[EMAIL PROTECTED]
> wrote:
Hello,
I still have problem after a fresh checkout.
I tried with MSVC++ 2008 express and I got the same errors too.
Tamas
Michael Jerris írta:
> This should work on a fresh checkout.
>
> Mike
>
> On Oct 27, 2008, at 10:23 AM, Tamas Csek
did you try 2008 with the freeswitch.sln file?
On Oct 30, 2008, at 9:10 AM, Tamas Cseke wrote:
> Hi,
>
> The answer is pretty simply for me: I have only 2008 express.
> But I tried 2008 express too as I said earlier and got the same.
> Maybe it is not an issue with 2008 professional.
>
> Best r
The built in setup has turned out to be too limiting. In tree there
is an advanced installer config file at :
http://svn.freeswitch.org/svn/freeswitch/trunk/w32/Setup/freeswitch.aip
That you can use to build a msi.
I am proposing to remove the 2005 support from tree completely. I need
to co
What svn revision was this?
Mike
On Oct 30, 2008, at 2:00 PM, Marc Lewis wrote:
> Got my second wedge of the day. Last time the channel state showed
> in my lua autoattendant script. This time the channel state showed
> in voicemail -- which is where it usually shows when its wedged.
>
> H
There is currently no OSP support, although it would be interesting to
add it.
Mike
On Oct 30, 2008, at 8:16 PM, Gregory Boehnlein wrote:
> Anyone know if Freeswitch can work w/ OSP and use something like
> Transnexus
> for CDR/Rating?
___
Freesw
http://svn.freeswitch.org/svn/freeswitch/trunk/scripts/ps_pizza.js
http://svn.freeswitch.org/svn/freeswitch/trunk/scripts/js_modules/SpeechTools.jm
On Oct 31, 2008, at 12:44 AM, [EMAIL PROTECTED] wrote:
It looks like I'll also need a file like speechTools.jm
-Original Message-
From:
2008 as 2005 will eventually be removed.
Mike
On Oct 30, 2008, at 9:27 AM, Tamas Cseke wrote:
> no, freeswitch.2008.sln or whatever is it
> Michael Jerris írta:
>> did you try 2008 with the freeswitch.sln file?
>>
>>
>> On Oct 30, 2008, at 9:10 AM, Tamas Cseke wr
It wasn't being well maintained and the advanced installer file in
tree should be used instead.
Mike
p.s. Carlos, do we have anything we need to add to the ais file in tree?
On Nov 1, 2008, at 2:05 AM, "UV" <[EMAIL PROTECTED]> wrote:
Mike,
I just noticed that you have removed the refere
My best suggestion is to run the debug build from msvc and if it
faults it will give you a stack trace of where to help us understand
why.
Mike
On Nov 1, 2008, at 2:43 PM, [EMAIL PROTECTED] wrote:
The "pizza demo" understands my pizza order (i.e. "take out",
"delivery") but then Windows
On Nov 3, 2008, at 3:39 PM, Klaus Teller wrote:
> Hi,
>
> I have Freeswitch running on a CentOS 5 box. From this box, i can
> resolve all domain names without problem. Yet Freeswitch is not able
> to originate to addresses when the domain name is specified. When i
> use the IP address ever
fixed in svn revision 10238.
Mike
On Nov 4, 2008, at 9:51 AM, Brian Wood wrote:
> SOX v14.0.1. I'm running FreeSWITCH trunk 10208.
>
>
> Brian West wrote:
>> I use that every now and then it works fine... I'll have to retest
>> this shortly and see... what sox version are you using?
>>
>> /b
>>
On Nov 4, 2008, at 11:47 AM, [EMAIL PROTECTED] wrote:
> Is it compulsory that I use different ports for different profiles?
> What if I want to use the same ports for my authenticated users and
> the
> non-authenticated ones?
Each sip profile is its own ip/port bindings, so yes, you must use
On Nov 5, 2008, at 9:15 AM, Birgit Arkesteijn wrote:
> Hi Anthony,
>
> Thanks for your reply!
>
> Does mod_xml_cdr use curl as well?
Yes.
> I seem to remember that I had a problem a couple of months ago with
> xml_cdr. It wouldn't work without curl-devel.
>
This seems wrong, it should use our
Fixed in svn revision 10251.
Cheers
Mike
On Nov 5, 2008, at 12:04 PM, Birgit Arkesteijn wrote:
> Hi,
>
> A bit more digging ...
> As far as I can tell, this has been added in revision 10239.
>
> I've deleted the semicolons and fortunately it compiles again.
>
> File ./include/switch_core.h, line
Please open a bug on http://jira.freeswitch.org . Please include a
full debug output from the freeswitch console with TPORT_LOG enabled
(info on the sofia page on the wiki).
Mike
On Nov 5, 2008, at 11:39 PM, David Aldworth wrote:
> Brian, we updated the acl to:
>
>
>
>
>
>
On Nov 6, 2008, at 10:08 AM, Klaus Teller wrote:
> OK. I updated and tried flushing the DTMFs before playing the
> commands and it works. Thanks.
>
> Now, i feel there is a more general issue of scalability around DTMF
> (both inband as well as RFC2833) handling in Freeswitch. What do you
>
change to
also, if your serving up from xml_curl, you can do the conditions on
your cgi and just have a blank condition tag, no reason to have the
switch do the regex as well.
Mike
On Nov 7, 2008, at 1:35 PM, Shelby Ramsey wrote:
Hello,
I have a question re: xml_curl ... if I reply
We don't currently have any tdm card support on windows.
Mike
On Nov 7, 2008, at 7:41 PM, [EMAIL PROTECTED] wrote:
> I would appreciate some recommendations on ATA (analogue telephony
> adapters) of either external (e.g. Sipura 3000 which has been
> updated to a Linksys box) or internal (Dig
http://wiki.freeswitch.org/wiki/Sofia#Modifying_the_To:_header
On Nov 10, 2008, at 5:33 AM, Helmut Kuper wrote:
> is there a way in dialplan or in application api to set the host part
> only the firstline in SIP-INVITE-Requests without setting the hostpart
> in To header field?
>
>
> If not, is
On Nov 10, 2008, at 7:25 AM, Woody Dickson wrote:
> Hi,
>
> I am trying to write an event listener that can record the time when
> a FIFO consumer rejoins the queue after the caller hangs up.
> Tracing through all the event traffic, I notice that there is FIFO-
> Action= consumer_start, con
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