check the sample config files for options to specify these:
http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/autoload_configs/lua.conf.xml?r=10747
On Aug 6, 2009, at 8:55 PM, Vladimir Rodionov wrote:
Good evening,
This is newbie question.
The FreeSWITCH lua module does not support
Hi,
I came across this thread
(http://lists.freeswitch.org/pipermail/freeswitch-users/2009-January/010172.html
) from January, and I'm having a similar problem, in that from the
moment an incoming analogue call starts ringing, it takes around 5-7
seconds before the dialplan gets executed.
Hi there,
Not sure whether this helps but test this without set bypass_media. In
my setup I have noticed the leg A session ends when bypass_media is
true. Call/bridge continue successfully.
Phillip Jones
On Thu, Aug 6, 2009 at 1:28 PM, Benedikt
Hi there,
I am trying to implement a scenario where I can terminate calls to
multiple destinations AND have termination carrier fail over.
Currently I can see how to do one or the other. But not both.
Multiple destinations is easy:
action application=bridge
is this just the timeout waiting for your max digits?
param name=max-digits value=11/
On Aug 8, 2009, at 4:58 AM, Merul Patel wrote:
Hi,
I came across this thread
(http://lists.freeswitch.org/pipermail/freeswitch-users/2009-January/010172.html
) from January, and I'm having a
Hi Phillip,
2009/8/8 Phillip Jones pjinthe...@gmail.com:
Not sure whether this helps but test this without set bypass_media. In
my setup I have noticed the leg A session ends when bypass_media is
true. Call/bridge continue successfully.
thx for that hint. unfortunately we can't do that due
FS is in the media path of an IVR call.
At the moment, the call is ulaw with DTMF in the audio I think coming
into FS and leaving FS.
The call is coming from an Asterisk server and going to an Asterisk
server.
Is there a way to disable FS from passing DTMF at some point in the
call? For
I'll see what I can do. Got a few sections of the PHP ESL to finish
but should be able to
write up fs_ivrd as well.
-- W
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
While any use of loopback can be considered abuse, that is how I solve this
issue. I use loopback for each dialed # and then in the dialplan use
mod_lcr to build a dialstring that does failover in order of cost.
So:
action application=bridge
You don't do make in the perl directory... You cd .. 'make permod'
from the top level dir.
/b
On Aug 8, 2009, at 12:28 AM, velusamy velu wrote:
Dear all,
When I do make for Perl ESL libraries in esl/perl directory I
have got the following error.
Mark Campbell-Smith wrote:
Hi Alan,
I hope you find your answers here as these are the sort of things that
are hard to find on the wiki, which is somewhat outdated in areas. If
you do find your answers, please post them back here for everyone
else.
I am new to FS also, so my comments
Examples exist in the scripts/lua folder called callback.lua. Also
please DO NOT hijack threads. You can't click reply, change the
subject as it messes up the mailing list threading and most mail
readers will thread it incorrectly.
Thanks,
Brian
On Aug 8, 2009, at 10:15 AM, Chad Phillips
Brian, I've read 8-9 of your emails asking to not hijack threads in threads
where this it didn't happen (like this). I see this as a new thread with a
single message (plus yours, of course) inside with [Freeswitch-users] how
to catch DTMF in Lua while a phrase macro is playing as the subject - and
On Aug 8, 2009, at 11:59 AM, Brian West wrote:
Examples exist in the scripts/lua folder called callback.lua.
i read this example code, but it doesn't seem to really address my
question. if you re-read the code and explanation that i posted
earlier, the input callback is working when i do
hi, during a call made with att_xfer, you can resume the user A on ring? Let
me explain better, with bind_meta_app and att_xfer can do this: during a
bridge if B dials *1 (for example), A is parked and B can call C. After C to
answer, B can to dial # and return in conversation with A. B could
Hi there,
Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re
INVITE) and only pass back the media to the network, or pass back
signaling also (SIP REFER)?
I know several suppliers who support SIP re INVITE but none that
support SIP REFER.
Check out
Thanks very much for that - very help.
Why would loopback be considered abuse? What would be the downsides
of doing this?
On Sat, Aug 8, 2009 at 7:18 AM, Rupa Schomakerr...@rupa.com wrote:
On Sat, Aug 8, 2009 at 9:16 AM, Rupa Schomaker r...@rupa.com wrote:
action application=bridge
His message was a direct reply to Alan Chandler's mail and Brian's mail client
is working perfectly fine
(as are both of my mail clients).
Look at the Message-Id and In-Reply-To headers of the mails if you still think
our clients are broken :)
stkn
Raffaele P. Guidi wrote:
Brian, I've read
http://wiki.freeswitch.org/wiki/PHP_ESL
Formatting could use a bit of help just exported my working document
to MediaWiki. Hope to add some more information to ESL as I continue
to do more with it.
I'm going to start a page on fs_ivrd as well. Feedback welcome.
-- W
IRC: wsuff on freenode
Yes, I need SIP-REFER (pass back media and signaling to PSTN). I am pretty
sure it is doable because voxeo offers this
option for their Voice XML customers but I am not interested in a hosted
solution at the time - it is quite expensive. As far as I understood, Voip
provider MUST have pstn call
Uh, oh, ok. Message-Id and In-Reply-To headers don't mean too much in a
gmail world (I think gmail does a better job than any other mail client).
Well, at least I understood what Brian was talking about, thanks.
Regards,
Raffaele
On Sat, Aug 8, 2009 at 20:01, Stefan Knoblich
Actually, this is what I need
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect
Will it work with PSTN? Can I redirect incoming PSTN call to another PSTN
number?
-Vladimir Rodionov
On Sat, Aug 8, 2009 at 11:39 AM, Vladimir Rodionov
vladrodio...@gmail.comwrote:
Yes, I need
Are your calls coming in on TDM or SIP trunks? Are your calls answered
by FreeSWITCH before you need to redirect them?
On Sat, Aug 8, 2009 at 11:52 AM, Vladimir
Rodionovvladrodio...@gmail.com wrote:
Actually, this is what I need
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect
Damn, looks like we need to learn some formatting =D.
I made some changes on that article to fix the formatting a bit, lets take
care of the wiki as we care about the software :).
On Sat, Aug 8, 2009 at 2:34 PM, William Suffill
william.suff...@gmail.comwrote:
does anybody else find it amusing that my thread about Lua scripting
was hijacked by people talking about hijacking threads?? ;)
/me waits patiently for discussion on the topic
On Aug 8, 2009, at 2:48 PM, Raffaele P. Guidi wrote:
Uh, oh, ok. Message-Id and In-Reply-To headers don't mean
A call is coming on SIP trunk. From PSTN. I does not need to be answered,
actually - I need to do some logic before redirecting call but I can answer
call as well It won't break the app logic.
-Vladimir Rodionov
On Sat, Aug 8, 2009 at 12:00 PM, Phillip Jones pjinthe...@gmail.com wrote:
Are
Thank you all, but I decided to go with ESL and Java.
On Fri, Aug 7, 2009 at 11:11 PM, Michael Jerris m...@jerris.com wrote:
check the sample config files for options to specify these:
http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/autoload_configs/lua.conf.xml?r=10747
On Aug 6,
I did that. And I do find it quite amusing :D
On Sat, Aug 8, 2009 at 21:09, Chad Phillips -- Apartment Lines
c...@apartmentlines.com wrote:
does anybody else find it amusing that my thread about Lua scripting was
hijacked by people talking about hijacking threads?? ;)
/me waits patiently
how many does it stop at? is it the same number each time?
Mike
On Aug 8, 2009, at 7:53 AM, Benedikt Fraunhofer wrote:
Hi Phillip,
2009/8/8 Phillip Jones pjinthe...@gmail.com:
Not sure whether this helps but test this without set bypass_media.
In
my setup I have noticed the leg A
loopback ends up using extra threads which we are only able to drop
later in certain situations so it will decrease your total amount of
calls you can do if your not careful with them.
Mike
On Aug 8, 2009, at 1:44 PM, Phillip Jones wrote:
Thanks very much for that - very help.
Why would
use the SayPhrase method of sessin instead of executing the phrase
application then your input callback should work as expected.
Mike
On Aug 8, 2009, at 11:15 AM, Chad Phillips -- Apartment Lines wrote:
in a lua script, i've tried using session:setInputCallback() to catch
DTMF tones while a
What do the debug logs on fs say when you try to put the call on hold?
Mike
On Aug 6, 2009, at 1:01 PM, Kozak Vladimir wrote:
The scenario is the following:
FS User A dial an extension
Extention opens outbound socket channel to my application
My application bridges the call to FS User B
The
typically when these questions are asked we find people really want to
be doing it the way these signals are automatically passed across.
Can you describe what your trying to do a bit more?
Mike
On Aug 7, 2009, at 7:44 PM, Max Bridgewater wrote:
Hi,
using javascript, i do originate the
Mike/Rupa ,
Thanks for your help on this. So I am correct that summarizing that
FreeSWITCH does not really support fail over and multiple call
destinations because the same mechanism is used to achieve both? And
that loopback as a solution is possible but not recommended?
Is there any other
I think that summary is totally wrong. Loopback should be used
here, and this should work to do what you want, just be aware of what
that means.
Mike
On Aug 8, 2009, at 4:24 PM, Phillip Jones wrote:
Mike/Rupa ,
Thanks for your help on this. So I am correct that summarizing that
Ok - that's great. I will build this out - thanks both for your help
on this. Much appreciated.
On Sat, Aug 8, 2009 at 1:32 PM, Michael Jerrism...@jerris.com wrote:
I think that summary is totally wrong. Loopback should be used
here, and this should work to do what you want, just be aware of
Hello Mike,
2009/8/8 Michael Jerris m...@jerris.com:
how many does it stop at? is it the same number each time?
i tried to express that non-wisdom using the words
This only works fine if we've few concurrent calls. There is no magic
borderline where it starts to refuse work.
this is surely
/freeswitch-users
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please open a bug on jira.freeswitch.org with the details of exactly
how to re-create this issue
Mike
On Aug 8, 2009, at 5:21 PM, Benedikt Fraunhofer wrote:
Hello Mike,
2009/8/8 Michael Jerris m...@jerris.com:
how many does it stop at? is it the same number each time?
i tried to
You should be able to use PHP with mod_managed just fine. You'll just need
Phalanger:
http://www.codeplex.com/Phalanger
http://www.php-compiler.net/doku.php
I don't have any use for PHP, but if anyone is interested in this, I'd be happy
to help out if it doesn't just work out of the box. It
Yes it happens 100% of the time I say not to do it..
Example this thread started with:
In-Reply-To: 4a7be35b.8010...@chandlerfamily.org.uk
References: 4a7be35b.8010...@chandlerfamily.org.uk
The hijacked thread has:
References: 4a7be35b.8010...@chandlerfamily.org.uk
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