Hi,
*After starting the freeswitch I try to dial a extension from console* *but
when i call extension 1002 from freeswitch console, call is connected to
extension 1002, but FS is aborted but call is established in1002. What shall
I do? What was the error?*
*
I have pasted the console events in pa
This appears to be a somewhat older version of svn trunk. Please re-
test with current svn trunk
Thanks
Mike
On Dec 2, 2008, at 5:57 AM, Baskar wrote:
Hi,
After starting the freeswitch I try to dial a extension from console
but when i call extension 1002 from freeswitch console, call is
*Hi,
This is the svn version i have installed before a month
FreeSWITCH Version 1.0.trunk (10130M)
*
--
*Warm Regards,
N.Baskar*
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Hi,
I am wondering if I am the only one getting this problem or not. When
sending in DTMF to freeswitch, freeswitch is not always capable of capturing
all the DTMF being sent. For instance, sending 1000 to freeswitch may end
up becoming 100 or 10003 becoming 1003. Am I the only one getting this
Hi Keith,
I was just writing a note along similar lines to Mike's. If you need a
hand getting
a packet capture or interpreting it, drop me a note off-list.
Cheers --
Dave
We generally are as good as possible on capturing dtmf reliably. If
you are seeing dropouts like that I would have to
Hi,
I am sorry again for sending another email to the group again. I am working
on embedding libfreeswitch to provide better monitoring. The first thing I
attempt to do is to run the sample code provided in the website:
#include
int main(int argc, char **argv)
{
switch_core_flag_t flags
My dialplan is pretty simple. I have a single trunk with a vonage softphone
DID (1303... we'll call it main) and a "virtual" DID (1816...) which rings
the softphone DID. All incoming calls show up as from softphone DID but the
sip_to_user holds the actual number dialed so I can enter the dialpla
we configured mod_shout and are able to record mp3. but if we start to
playback the file, it will only be played back to that point, which
was recorded, when we started the player.
we do this with "api uuid_record uuid start /var/www/test.mp3".
we are also able to playback a (radio-)stream to an u
Folks;
I've just taken the time to document the Sipura, err, Linksys, errr
Cisco SPA400 4 line FXO Analog Telephone Adapter in the Wiki.
http://wiki.freeswitch.org/wiki/SPA400_FreeSwitch_HowTo
If anyone uses these ATA's and has questions about it let me know and
I'll see if I can answer t
We generally are as good as possible on capturing dtmf reliably. If
you are seeing dropouts like that I would have to guess that this is a
very lossy line. Could you try and look at the packet capture of a
call that is missing digits and see if you are indeed dropping a lot
of packets. I
On Dec 2, 2008, at 5:55 AM, Woody Dickson wrote:
> Hi,
>
> I am just having a dumb question and hoping someone can help me. I
> am trying to run a c program with libfreeswitch embedded so I can
> use some external mechanism to keep track of freeswitch, but I am
> having problem while compi
Hi,
I am just having a dumb question and hoping someone can help me. I am
trying to run a c program with libfreeswitch embedded so I can use some
external mechanism to keep track of freeswitch, but I am having problem
while compiling:
[EMAIL PROTECTED] fs]# gcc switchnode.c -I/usr/local/freeswit
Can you show me the full XML for this extension including the regular
expression?
/b
On Dec 2, 2008, at 7:25 AM, ccav wrote:
> transfer to voicemail is as follows
>
>
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Are you on SVN trunk or what rev are you trying to use?
/b
On Dec 2, 2008, at 7:48 AM, Dennis wrote:
> it seems, that fs has to stream to recording file to a streaming
> server (like icecast), right? but if we do "api uuid_record uuid start
> shout://user:[EMAIL PROTECTED]:12345/" (and other com
What are you calling, sip I assume, this may be a case where the sip
signaling is sending a 180 ringing instead of a 183 and we are not
generating ringback in that case. Can you please confirm that and
test if setting the ringback channel variable before bridge fixes this
issue?
Mike
On
What revision of freeswitch is this? Can you please test this with
svn trunk?
Mike
On Dec 2, 2008, at 2:27 AM, Baskar wrote:
Hi,
I have updated all the above events you told .It's working fine but
when i call extension 1002 from freeswitch console, call is
connected to extension 1002,
i am using the latest svn trunk from today.
2008/12/2 Brian West <[EMAIL PROTECTED]>:
> Are you on SVN trunk or what rev are you trying to use?
>
> /b
>
> On Dec 2, 2008, at 7:48 AM, Dennis wrote:
>
>> it seems, that fs has to stream to recording file to a streaming
>> server (like icecast), righ
And you have your shoutcast/icecast server set up and functional?
/b
On Dec 2, 2008, at 9:03 AM, Dennis wrote:
> i am using the latest svn trunk from today.
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no, not yet. i am still fiddling arround with icecast2.
we tried it with someone, who offers radiostreams. perhaps this just
works with icecast(2) and shoutcast?
2008/12/2 Brian West <[EMAIL PROTECTED]>:
> And you have your shoutcast/icecast server set up and functional?
>
> /b
>
> On Dec 2, 20
icecast2 is a known working server we have talked to before.
/b
On Dec 2, 2008, at 9:25 AM, Dennis wrote:
> no, not yet. i am still fiddling arround with icecast2.
>
> we tried it with someone, who offers radiostreams. perhaps this just
> works with icecast(2) and shoutcast?
__
Hello,
when using mod_portaudio for calling somebody I don't hear anything
until the other party answers the call. Is it possible to play a dialing
tone when the other party is ringing?
Best regards
René Pankratz
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I think the api changed a little bit for this. The easiest starting
point would be to just clone switch.c and chop out any of the stuff
you don't need, it's mostly argument handling code in there.
Mike
On Dec 2, 2008, at 7:05 AM, "Woody Dickson" <[EMAIL PROTECTED]>
wrote:
> Hi,
>
> I am s
If you can get it to break on linux I will ssh in and fix it for you.
If you cannot, i can try to fix it for you over rdp but that won't be very
fun.
We can think about reinstating mod_lumenvox as well as another windows based
asr
alternative. I deleted it for the same reason we will probably del
FreeSWITCH has an enterprise scale SIP UA. Not only can it listen on other
ports it can
listen and work on as many ip:port combos as you want simultaneously each
with it's own specific config.
If you have an affinity for port 5060 you can always bring up 2 IP on the
same box and give one to each
sorry, problem solved :-)
it works very good with icecast2.
2008/12/2 Brian West <[EMAIL PROTECTED]>:
> And you have your shoutcast/icecast server set up and functional?
>
> /b
>
> On Dec 2, 2008, at 9:03 AM, Dennis wrote:
>
>> i am using the latest svn trunk from today.
>
>
> _
from the source tree of FS please type
"make current"
when it completes, retest the call.
On Tue, Dec 2, 2008 at 5:07 AM, Baskar <[EMAIL PROTECTED]> wrote:
> *Hi,
>
> This is the svn version i have installed before a month
>
> FreeSWITCH Version 1.0.trunk (10130M)
> *
> --
> *Warm Regards,
>
Have you tried the latest msi build? It's based off svn 10564.
Carlos
On Sun, Nov 30, 2008 at 11:03 AM, Per Møller <[EMAIL PROTECTED]> wrote:
> I have installed FS 1.0.0 on a Mac using the precompiled .dmg and FS 1.0.1
> on a Windows Vista machine using the precompiled .msi - actually the same
>
All HFC-based cards supported by bristuffed Zaptel should work.
Stefan
Am Monday 01 December 2008 schrieb Michael Jerris:
> The bri support is still in development, basic calls on ptmp bri do
> appear to work, although I am not sure with what hardware.
>
> Mike
>
>
> On Dec 1, 2008, at 10:26
Mark and David,
I am willing to help some with testing here as well, if you need it.
Ping me directly or we can get on the IRC. I am on Mac OS, but have
readily available vm's with Debian, etc. I also have Prophecy.
I have a general interest in an ASR solution as well. Voxeo is great,
but
Naturally, either way is stupid.
The whole idea of putting the transport in a uri param is equally stupid to
using 2 different protocol names but since SIP is the descendant of http it
they decided to stick with the stupidity of http/https and have sip/sips
which is almost as if it was designed to
from build root:
svn co -r8809
http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/asr_tts/mod_lumenvoxsrc/mod/asr_tts/mod_lumenvox
They did seem to express an interest in granting some dev licenses when they
realized we took the code out of tree but I have not actually dealt with the
issue y
Ok
I have a ping in with Lumenvox about dev licensing, and pulled the
mod. Not sure where this will go, but will take a peek at things.
Balancing the effort against something like getting unimcrp going and/
or openmrcp tested and stable.
Thanks.
Andy
On Dec 2, 2008, at 11:43 AM, Anthony
hi,
because we do not get tired of testing and playing a lot with the
beloved fs, we now arrived at the fax feature :-)
i am not sure if the docs are up to date or if there was a lot of
development in the meantime. therefore i would like to ask, what is
possible and what will come in the near fut
They contacted us shortly thereafter and asked if we want to have them sell
you the license for 50 bucks.
hmm, i wonder why i deleted the module.
I will tell them that if they give you a developer license you will work on
getting it back into trunk.
On Tue, Dec 2, 2008 at 11:27 AM, Andrew Gil
Cool. Thanks for the answer.
> All HFC-based cards supported by bristuffed Zaptel should work.
>
> Stefan
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On 12/2/08, Anthony Minessale <[EMAIL PROTECTED]> wrote:
> Naturally, either way is stupid.
Word.
> The whole idea of putting the transport in a uri param is equally stupid to
> using 2 different protocol names but since SIP is the descendant of http it
> they decided to stick with the stupidit
-- filename "dialplan/extensions/13033253678.xml" -- This is the primary
DID assigned.
-- filename "dialplan/extensions/18162565804.xml" -- This is the primary
DID assigned.
Bring on SNAP, baby!
On Tue, Dec 2, 2008 at 11:03 AM, Kristian Kielhofner <
[EMAIL PROTECTED]> wrote:
> On 12/2/08, Anthony Minessale <[EMAIL PROTECTED]> wrote:
> > Naturally, either way is stupid.
>
> Word.
>
> > The whole idea of putting the transport in a uri param is equally stupid
> to
> >
We'll schedule a round table with the topic
SIP OMFG STFU
At the next ClueCon aug 4th-6th 2009 to stir things up a bit =D
On Tue, Dec 2, 2008 at 1:03 PM, Kristian Kielhofner <
[EMAIL PROTECTED]> wrote:
> On 12/2/08, Anthony Minessale <[EMAIL PROTECTED]> wrote:
> > Naturally, either way is stup
On 12/2/08, Anthony Minessale <[EMAIL PROTECTED]> wrote:
> We'll schedule a round table with the topic
>
> SIP OMFG STFU
>
> At the next ClueCon aug 4th-6th 2009 to stir things up a bit =D
>
Heh. I've been trying to make it back these last couple of years. I
just might make it in '09!
--
Krist
Note: while reading up on regex, I see that the ',' in ([0,1]) is superflous,
has been removed. regex is now:
^([01]?)(8162565804)$
Didn't fix the problem but I'm a perfectionist, had to be changed. :D
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I checked out the trunk version, and it's still slow. However I found one
improvement - it does not crash on shutdown anymore.
Could anymore give me some pointers on how to try to debug this on the
Windows platform?
// Per
Fra: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] På vegne af Carlos
Talb
Can you do a console loglevel debug, then send all the output around that time?
Apart from that, the quickest way might just to attach a debugger, then break
all when it pauses and see where the threads are :).
-Michael
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED
Hi all,
Is it possible to use mod_spidermonkey_odbc with a Windows installation of
FreeSWITCH at the moment? If so does anyone have any pointers? I get:
2008-12-02 14:23:57 [DEBUG] switch_odbc.c:145 switch_odbc_handle_connect()
Connecting ivr_test
2008-12-02 14:23:57 [ERR] switch_odbc.c:160 switc
is it stun timeout ?
do you have one of the ip set to stun:foo ?
On Tue, Dec 2, 2008 at 1:33 PM, Michael Giagnocavo <[EMAIL PROTECTED]>wrote:
> Can you do a console loglevel debug, then send all the output around that
> time?
>
> Apart from that, the quickest way might just to attach a debugger
Right now this page is up-to-date with the latest info:
http://wiki.freeswitch.org/wiki/Mod_fax
T.38 is not (yet) supported.
-MC
On Tue, Dec 2, 2008 at 9:40 AM, Dennis <[EMAIL PROTECTED]> wrote:
> hi,
>
> because we do not get tired of testing and playing a lot with the
> beloved fs, we now arr
On Tue, Dec 2, 2008 at 3:32 PM, Michael Collins <[EMAIL PROTECTED]> wrote:
> Right now this page is up-to-date with the latest info:
> http://wiki.freeswitch.org/wiki/Mod_fax
>
> T.38 is not (yet) supported.
>
> -MC
>
Can you (or someone) elaborate on this? Maybe the answer really is
no, but what
T.38 passthrough IS supported, T.38 endpoint and gateway are not yet
supported.
Mike
On Dec 2, 2008, at 4:28 PM, Kristian Kielhofner wrote:
> On Tue, Dec 2, 2008 at 3:32 PM, Michael Collins <[EMAIL PROTECTED]>
> wrote:
>> Right now this page is up-to-date with the latest info:
>> http://wiki
Yes, it should work fine. As the error message says it didn't find
the data source name you specified. You need to setup your odbc data
source on the system
Mike
On Dec 2, 2008, at 9:29 AM, Joe Bain wrote:
> Hi all,
>
> Is it possible to use mod_spidermonkey_odbc with a Windows
> install
After you set ${dialed_user}=$2 try using ${dialed_user} everywhere
instead of $2 just to test.
/b
On Dec 2, 2008, at 1:29 PM, ccav wrote:
>
> Note: while reading up on regex, I see that the ',' in ([0,1]) is
> superflous,
> has been removed. regex is now:
> ^([01]?)(8162565804)$
> Didn't f
Made the change, no joy.
Do I need to set sip_req_user to the updated DID?
Also, I misspoke in my first post, apparently the bridge is NOT going
through either. Is there some var/param I can set with $2 so I can see it
in the "info"?
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Okay, I found out who the culprit is, but I still want to find a fix so the
dialplan works like I want.
The http://www.nabble.com/Wrong---in-voicemail-tp20791453p20804247.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.
___
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On Tue, Dec 2, 2008 at 1:28 PM, Kristian Kielhofner <
[EMAIL PROTECTED]> wrote:
> On Tue, Dec 2, 2008 at 3:32 PM, Michael Collins <[EMAIL PROTECTED]>
> wrote:
> > Right now this page is up-to-date with the latest info:
> > http://wiki.freeswitch.org/wiki/Mod_fax
> >
> > T.38 is not (yet) supported
RESOLVED.
Duh, I'm sposed to use ringback, not playback...
Someone should write a book on this... Maybe I will.
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Kristian,
Are you on the IRC channel by any chance?
-MC (IRC: mercutioviz)
On Tue, Dec 2, 2008 at 1:28 PM, Kristian Kielhofner <
[EMAIL PROTECTED]> wrote:
> On Tue, Dec 2, 2008 at 3:32 PM, Michael Collins <[EMAIL PROTECTED]>
> wrote:
> > Right now this page is up-to-date with the latest info:
>
hehe, careful what you wish for...
On Tue, Dec 2, 2008 at 5:11 PM, ccav <[EMAIL PROTECTED]> wrote:
>
> RESOLVED.
>
> Duh, I'm sposed to use ringback, not playback...
>
> Someone should write a book on this... Maybe I will.
> --
> View this message in context:
> http://www.nabble.com/Wrong---in-v
Hi Folks,
so far i could understand how to bridge calls with Javascript. I'm trying to do
the same with Java via the Socket Interface. My first trials weren't
successful. maybe you can help me understand what is goin on.
What i want to do is to bridge an existing leg (Unique-ID is known) to a
You probably have several options depending upon your needs. Could you
elaborate a bit on what the big picture is? Also, what exactly were you
doing when you established the second call leg? Did the second call let get
created and a valid uuid assigned, etc.? Just checking.
Let us know,
MC
On Tue
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