Hi All,
I'm just testing out mod_skypiax (great work Giovanni co!) and while it's
working and all, I find that when I call in from a Skype contact, it's
/very/ laggy. I would say something on the skype end and I would hear it on
the FS end quite a bit later. Funny thing is the audio going the
On Wed, Jul 1, 2009 at 6:56 PM, Alexey Lubimov lubi...@neolant.ru wrote:
grep -ir 111 *
default/bad.xml: user id=111
default/bad.xml: variable name=accountcode value=111/
default/bad.xml: variable name=effective_caller_id_number value=111/
default.xml: user id=111 type=pointer/
On Wed, Jan 7, 2009 at 6:39 PM, tester tato testeado...@gmail.com wrote:
Hello everyone,
Thanks in advance if anyone wants to help me in a couple of questions; I
set up my fs plant and included a grandstream fxo gateway gxw4104, the
gateway redirects all incomming calls from the PSTN to the
On Mon, Jan 5, 2009 at 2:47 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
if you do not have a proper jabber SRV record google will not send you the
requests.
I've got those setup. The jabber account I've setup on FS is a google hosted
domain with the right jabber SRV records. The
On Mon, Jan 5, 2009 at 11:14 AM, Michael Jerris m...@jerris.com wrote:
It already does
Mike
I've noticed it does for normal Jabber servers but not for Google's Talk
Server. It also has issues sending messages when using said server, but not
with other jabber servers I've tried.
- Gonzalo
On Mon, Jan 5, 2009 at 12:44 PM, Michael Jerris m...@jerris.com wrote:
There were some issues in earlier revisions that should now be fixed,
otherwise you may not have tls support compiled in which is required for
talking to googles servers.
I've just updated and same result. I fired up
On Fri, Dec 5, 2008 at 4:23 PM, Michael Collins [EMAIL PROTECTED] wrote:
On Fri, Dec 5, 2008 at 8:41 AM, Anthony Minessale
[EMAIL PROTECTED] wrote:
call_timeout is only used if you are bridging 2 channels where one or
both
of them is still unanswered.
what you want to use is
On Fri, Nov 28, 2008 at 8:45 AM, Faisal Maqsoodi
[EMAIL PROTECTED]wrote:
I m trying to play some basics on FreeSwitch plus twinkle softphone.
Following error msg is displayed on the softphone:
Failed to create a udp socket on port 5060. Address already in use
following on freeswitch window:
On Thu, Nov 27, 2008 at 8:54 AM, Tim Panton [EMAIL PROTECTED] wrote:
After some digging in the C I've added this feature (It's about 10
lines) .
How do I go about contributing the code back ?
I was looking at the code myself and I couldn't see how to specify more than
one URL.
You probably
to execute should be hangup but it never does.
Thanks
On Mon, Nov 17, 2008 at 12:22 PM, Gonzalo Servat [EMAIL PROTECTED] wrote:
Hi,
Nice work on mod_fax! I just tried sending myself a couple of faxes and
they were received successfully. I even tried using tone_detect which
correctly
Hi All,
We all like to be thanked and when someone does something for me, I like to
show my gratitude towards them and I think what Anthony did for me today
deserves a public show of gratitude. It goes like this:
I have a single port FXO card in my home machine running FS and I needed to
modify
Hi
1) I'm trying to get the hang of mod_dingaling and I noticed that when I
send a message from [EMAIL PROTECTED] to [EMAIL PROTECTED] (the latter being
the client profile setup on FS), mod_dingaling tries to send another message
again from the same sender/recipient which the google talk server
Hi,
I've got a single port FXO card which works really well with FS. I even got
Zaptel patched with OSLEC which eliminted the little echo I had. The only
trouble I'm left with is that when I call my house and let it ring twice
then hangup, FS would never detect that the remote side hanged up. I
On Fri, Nov 14, 2008 at 11:32 AM, Gonzalo Servat [EMAIL PROTECTED] wrote:
Hi,
I've got a single port FXO card which works really well with FS. I even got
Zaptel patched with OSLEC which eliminted the little echo I had. The only
trouble I'm left with is that when I call my house and let
Hi,
I just tried doing a make current (I was on 10231 and updated to 10416) and
mod_python failed to compile. I can't remember exactly what the error msg
was as it's beyond what my terminal's scroll buffer is configured to store
but it was something along the lines of cc1 will treat all warnings
On Wed, Nov 12, 2008 at 4:56 AM, Juan Backson [EMAIL PROTECTED] wrote:
Hi,
Here is the exact lua script:
digits = session:playAndGetDigits( 1, 1, 3, 3000, #,
phrase:enter_userid,phrase:invalid_input,[1|2|3|4|5|6])
If I repeatedly enter 9, here is what would happen:
[..snip..]
Hi Juan,
On Sat, Nov 8, 2008 at 10:30 AM, Juan Backson [EMAIL PROTECTED] wrote:
Hi Gonzalo,
I tried changing to digits = session:playAndGetDigits( 1, 1, 3, 3000,
[#*], phrase:admin_menu, phrase:invalid_input, [1-6] )
Freeswitch still returns 92 for if I pressed 9 the first time and the
2 for the
On Fri, Nov 7, 2008 at 9:44 AM, Juan Backson [EMAIL PROTECTED] wrote:
Hi Gonzalo,
Here is the lua code I am using
1.
2. digits = session:playAndGetDigits(1, 1, 3, 3000, #*,
phrase:admin_menu, phrase:invalid_input, 1|2|3|4|5|6)
Try with:
digits = session:playAndGetDigits( 1,
On Fri, Nov 7, 2008 at 6:08 AM, Juan Backson [EMAIL PROTECTED] wrote:
Hi,
I am encountering an issue when using playAndGetDigit. I have the regex
set as 1|2|3|4|5|6, and then pressed 9, it gives error as expected. Then, I
pressed 2. This time it passes, but the digit being returned by
.
{execute_on_answer=playback\s/some/file.wav}sofia/profile/[EMAIL PROTECTED]
On Sun, Nov 2, 2008 at 8:59 PM, Gonzalo Servat [EMAIL PROTECTED] wrote:
Hi All,
This is basically what I'm trying to do:
- Caller dials in and FS runs the Lua script I'm writing
- Main menu is played to the caller
- Caller
On Fri, Oct 3, 2008 at 1:05 PM, Brian West [EMAIL PROTECTED] wrote:
Don't forget today is the day you can call into
sip:[EMAIL PROTECTED] [EMAIL PROTECTED]
or 213-799-1400 and ask questions and interact with Mike, Anthony
and Myself and other FreeSWITCHer's that might happen to be in the
On Sat, Aug 30, 2008 at 7:38 AM, Sunil Singh [EMAIL PROTECTED]wrote:
Hi,
[..snip..] (the above is all that I agree with)
It's quite clear Sunil is just a frustrated user who couldn't get FS
configured how he wanted it in a few minutes and decided to troll on the
list to vent his frustration.
On Wed, Aug 6, 2008 at 1:00 AM, Roberto Hernandez [EMAIL PROTECTED] wrote:
Anyone have suggestions on how to the enable the following scenario? Allow
internal but control external per user.
More importantly how to disable or enable PSTN access for a specific user
in a domain (profile)?
On Wed, Jul 23, 2008 at 4:34 PM, Ashutosh [EMAIL PROTECTED] wrote:
[..snip..]
Salutes to the never ending spirit of man, thats called Anthony!!
.. and the entire FS team!
- Gonzalo
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On Sun, Jul 13, 2008 at 1:31 PM, Adnan Barakat [EMAIL PROTECTED]
wrote:
Hi All,
I'm looking for a way to play an audio file to the end destination before
bridging the call, eg. below is an example in Asterisk which I'm trying to
replicate in FS.
exten =
On Tue, Jul 8, 2008 at 11:50 AM, Brian West [EMAIL PROTECTED] wrote:
Doh OFF BY ONE ERROR!!! its the 9th not the 9th.
Off by one again!!! 9th not the 8th you mean? :-)
- GS
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Freeswitch-users@lists.freeswitch.org
On Tue, Jul 1, 2008 at 3:37 PM, Henk Oegema [EMAIL PROTECTED]
wrote:
[..snip..]
When I press F8, I see:
[EMAIL PROTECTED]
API CALL [console(loglevel 7)] output:
+OK console log level set to DEBUG
[EMAIL PROTECTED]
I'm pretty sure Brian meant Press F8, try to dial again and see what it
On Tue, Jul 1, 2008 at 4:07 PM, Henk Oegema [EMAIL PROTECTED]
wrote:
[..snip..]
2008-07-01 21:05:04 [INFO] mod_dialplan_xml.c:222 dialplan_hunt()
Processing
1000-[EMAIL PROTECTED]
2008-07-01 21:05:04 [DEBUG] mod_dialplan_xml.c:107 parse_exten() test
conditions
On Mon, Jun 23, 2008 at 4:51 PM, Jeremy Kiffiak [EMAIL PROTECTED]
wrote:
Michael,
I have confirmed the issue with search myself and double-checked it
against the search functionality of my own MediaWiki installation and
that of Wikipedia. Something is definitely borked. It would appear
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