http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
You can configure odbc as normal on the box to setup the dsn's then
follow the instructions at that link. FreeSWITCH will create the
tables it needs for you.
Mike
On Nov 12, 2008, at 3:54 PM, gary wrote:
I am tring to configure FS
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 there
How about in the response, typically the server would challenge for
auth creds.
Mike
On Nov 10, 2008, at 12:17 PM, MEHDi CHAABOUNi wrote:
I've just analyzed the header of the HTTP request made by
Freeswitch, there's no Authorization section like
Authorization: Basic
Not without modifications to the c code.
Mike
On Nov 10, 2008, at 7:55 PM, Shelby Ramsey [EMAIL PROTECTED]
wrote:
Hello,
Any way to disable sending / using RPID in the dialplan and not at
the SIP profile level?
Thanks!
SR
___
Is this with current svn trunk? I think we have fixed this issue now
in trunk, could you please test and let us know.
Mike
On Nov 10, 2008, at 7:54 PM, Shelby Ramsey [EMAIL PROTECTED]
wrote:
Hello FS'ers,
I'm having an issue where I get a call and send it via a profile ...
but if
action application=set data=hangup_after_bridge=true/
change to
action application=set data=hangup_after_bridge=false/
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.
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
guys
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 938:
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,
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 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
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 reference
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
to 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 wrote:
Hi,
The answer is pretty simply
, 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 Cseke wrote:
Hello,
I have
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
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
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.
Here
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?
___
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:
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 the
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 : error
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.
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 scenario I
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 sure
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 phones and
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
___
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 logs of
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 conf-name dial user1 ?
Call it or originate a call to it:
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
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 the
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
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
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:
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 that
(info, after put cSession in conf);
}
(end of script)
On the console, I run the script as:
jsrun conference1.js.
I only get the second 'console_log' until hangup.
Cheers, Birgit
On 14/10/08 18:38, Michael Jerris wrote:
On Oct 14, 2008, at 1:13 PM, Birgit Arkesteijn wrote:
Hi all
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
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 param name=extension
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 that
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
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 conf_name moh stop
conference conf_name moh start
Sheeju
I think stop will make it stop, I thought there was a way to start
it but on quick
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
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 Tue, Oct 14, 2008 at 9:32 PM, Michael Jerris [EMAIL PROTECTED
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://jira.freeswitch.org so I
don't forget.
Mike
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.
Tested, and works. Once again Mike - MONEY
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 do it, but
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
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 single cpu,
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
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 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
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 the
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, freeswitch
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
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
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
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
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 its
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
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
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 the
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() STATE:
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
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 table
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
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:
3303
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 bridge
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, Sep 25,
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
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
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
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
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 what may be causing the problem and how
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
extension name=public_did
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
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
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
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 getting
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 creating
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
Make current will not work from release tarballs but if you grab the
new version it should install without wiping out your config.
Mike
On Sep 20, 2008, at 6:10 PM, Michael S Collins [EMAIL PROTECTED]
wrote:
Yes make current will update you without breaking your existing
configs. Even if
On Sep 19, 2008, at 5:48 AM, Rocco Lucente wrote:
Hello,
mod_sofia don't start correctly when there isn't a internet
connection.
We can use fs only in lan (without internet connection)? Is there a
particolar configuration for this situation?
Regards,
The default configuration includes
On Sep 19, 2008, at 9:07 AM, Jon Bruel wrote:
I have tested the option of adding channel variables to the bridge
string, and it does not work.
This dialplan works:
extension name=External calls
condition field=destination_number expression=^(\d{8})$
action application=set
On Sep 19, 2008, at 9:18 AM, Gopal krishnan wrote:
Hi,
Since I am not able to make the outbound call, when I use this
command oz dump 1 a in the console, I used to get the all the 31
channels , for a refrence I am posing one channel block here,
span_id: 1
chan_id: 31
physical_span_id:
On Sep 19, 2008, at 10:46 AM, xbipin wrote:
the FS site says FS supports TLS etc so wouldnt it be good if the
windows
binary were compiled with TLS as by default they r not so guys like
me who
actually download the msi and install and run it can atleast have TLS
support by default on
On Sep 19, 2008, at 12:19 PM, Luke Graybill wrote:
On Thu, Sep 18, 2008 at 3:49 PM, Luke Graybill [EMAIL PROTECTED]
wrote:
My suggested solution is to apply the job-id concept from bgapi to
messages as well, and to go a step further; borrow the Asterisk idea
of transmitting an
Your dialplan is fixed now, this is an issue with openzap now, can you
clarify your openzap configuration and what kind of line it is hooked
too please?
Mike
On Sep 18, 2008, at 3:55 AM, Gopal krishnan wrote:
Hi Brian,
I tried as you suggested, but still my outbound is not yet thru,
On Sep 18, 2008, at 4:53 AM, Jon Bruel wrote:
Thank you for your replies. In checking the SIP messaging sequence,
I saw that you are right about the source of the tag to the To-
header: it is the phone, which as a response to the INVITE, adds the
tag (and not, as I wrote, the FreeSWITCH).
On Sep 18, 2008, at 12:00 PM, xbipin wrote:
are u interested in a paid implementation as im willing to post a
bounty for
it if there r others like us willing to use it with freeswitch.
wikipbx is
also good for a start but one thing i dont understand is if
freeswitch is
made for
euro should work fine with the q931 dialect.
Miike
On Sep 18, 2008, at 12:04 PM, Evgeniy Zolotov wrote:
Thanks for all. But I am in perplexity now - how we can khow, what
any certain dialect is carried out?
Look for this example:
If we'll specify for nonexistent dialects:
pri_spans
my guess would be that it just fails to load. what does the debug say?
Mike
On Sep 18, 2008, at 1:19 PM, Evgeniy Zolotov wrote:
Thanks, Mike. But what dialect works when we make
param name=dialect value=ABC /
or
param name=dialect value=XYZ /
How we can define it
On Sep 18, 2008, at 3:08 PM, Robert Clayton wrote:
All,
Due to the limited audio abilities implemented thus far in FreeSwitch
I am needing to handle the audio streams outside of FreeSwitch.
So can I use Lua to pass the audio back and forth to an outside
library as streamRecord and FilePlay
Do you have any performance tests to show the difference?
Mike
On Sep 18, 2008, at 6:50 PM, Cristian Talle wrote:
I gave it a try though... and it works nicely (at least for
directory entries for now). I've added 3 more commands to xml_curl:
cache_on, cache_off and cache_delete_key.
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