Anthony,
I just tried to print the variable with the log app, with read it prints,
with play_and_get_digits doesn't.
I'm using latest SVN rev:
FreeSWITCH Version 1.0.trunk (13012M)
Thanks,
Diego
On Tue, Apr 14, 2009 at 1:53 AM, Diego Viola wrote:
> I remember playAndGetDigits had a bug like
Hi,
I have tried google talk integration with FS and is working fine. Great Work!
Is it possible to have multiple concurrent incoming calls on the same
gmail account?
--
Regards,
Moiz Chinoy.
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Freeswitch-users mailing list
Freeswitch-users@lists.fr
I remember playAndGetDigits had a bug like this too.
Anthony, please help me.
On Tue, Apr 14, 2009 at 1:42 AM, Diego Viola wrote:
> It works if I use "read" and do this:
>
>
>
>
>
>
>
>
>
>
> But I need play_and_get_digits to work like that too, please.
>
> Diego
>
>
It works if I use "read" and do this:
But I need play_and_get_digits to work like that too, please.
Diego
On Tue, Apr 14, 2009 at 1:36 AM, Diego Viola wrote:
> Hi everyone,
>
> I have a question... I have this on my dialplan:
>
>
>
>
>
>
>
>
> Wha
Hi everyone,
I have a question... I have this on my dialplan:
What I want to do is play and read some digits and as soon as I get those
digits, transfer to that extension... but this never happens, even if I
terminate with a #.
I do the same thing with Lua and it works with Lu
Brian West wrote:
> user/ uses the dial-string in the domain to find the user on the
> internal profile by default.. so to call someone registered via ipv6
> you'll need to put a dial-string param in the user to find them on the
> internal-ipv6 profile.
Oh okay, that works, thanks a lot. T
see how it works in latest trunk 13011
nontheless you can just say
session.execute("bridge", "loopback/5000");
and get the same result without touching that other channel.
when the call fails, you will have an originate_disposition variable in
session you can check.
On Mon, Apr 13, 2009 at 1
I highly recommend that you set aside this endeavor for the time being and
use the default configuration. Once you get familiar with the default config
then you'll realize how to make changes to registered users and to the
dialplan. Don't let the size of the default configuration scare you off. It
user/ uses the dial-string in the domain to find the user on the
internal profile by default.. so to call someone registered via ipv6
you'll need to put a dial-string param in the user to find them on the
internal-ipv6 profile.
/b
On Apr 13, 2009, at 11:35 AM, Bernhard Schmidt wrote:
Hi,
Are both of these phones on the same LAN as FreeSWITCH? What kind of phones
are they? Also, can you reverse the dialing order and reproduce the
symptoms? Just curious if it's always the same phone causing the noise or if
it is always the second phone connected, regardless of which phone.
-MC
On M
Hi,
probably a pretty easy problem, but I can't figure it out nevertheless.
I'm still experimenting with FreeSwitch (SVN trunk, about two weeks
old), mainly due to the IPv6 SIP support. I'm pretty much running the
default configuration included in the sourcetree. Now I've hit the
following proble
Hello.
I'm trying to use freeswitch, was able to compile it without problems,
which is very nice. Then studying the configurations etc., I managed
to set up SIP accounts those register properly. But now, if I want to
call one registered account from the other one, I get error 404 - not
found. I tr
1. The latest trunk I've tried with is 13008. Since I'm not doing anything
for production yet (just testing/evaluating), so I tend to update as soon as
there is new version available..
2. Yep, you will find it below. In javascript - my sample for .NET does
basically the same thing, with the
1) When you say latest, which rev does that mean? we change revs pretty
often.
2) Do you have a minimal script that reproduces your issue.
3) is there a reason you cannot just session.execute("bridge", dest);
instead of doing it manually (which is a process not for the faint at
heart)?
On Mo
Anthony,
Please read my comments on Jira cases MODOPAL-3 and MODOPAL-5.
In MODOPAL-3 I've also attached a patch to get DTMF working better (handle both
Tone and String input), and also increasing the stability of mod_opal.
But the problem also is connected to MODOPAL-5, which I created just now
You can store gateway xml in both a user and in the sofia.conf, we don't
enforce which ones who can use because that would be a function of the
dialplan but you can certainly store them in your configs that way and set
the names of a user's specifc gateways in a variable that would be present
on ea
I have two problems that I haven't been able to solve. I've done the same tests
in both javascript, and in .NET.
The two scripts are pretty simple, they just answer an incomming call, creates
a new session, wait for an answer on the second call leg, and then bridge the
two channels together.
I
which rev was "latest" for you?
I have confirmation that it is indeed working.
On Sun, Apr 12, 2009 at 6:28 AM, Peter Olsson <
peter.ols...@visionutveckling.se> wrote:
> Thanks,
>
> I tried latest trunk, but still no success.. :(
>
> Peter
>
>
> On 09-04-11 18.30, "Anthony Minessale"
> wrote:
>
The only part I see fit for integration with CouchDB is for storing
CDR documents.
This kind of database is imho best-used for storing large sets of data,
in a document structure.
I don't think the FS config fits this description, since the amount of
config documents are typically not "large".
You should update to SVN trunk and try again.
/b
On Apr 13, 2009, at 8:09 AM, Baskar wrote:
Hi,
I have loaded the java module in freeswitch. But when i run
freeswitch in console i get this error.
2009-04-13 17:38:33 [NOTICE] modjava.c:244 mod_java_load() Java
Framework Loading...
20
Hi,
I have loaded the java module in freeswitch. But when i run freeswitch
in console i get this error.
2009-04-13 17:38:33 [NOTICE] modjava.c:244 mod_java_load() Java Framework
Loading...
2009-04-13 17:38:33 [ERR] modjava.c:133 load_config() Error loading
/usr/java/jdk1.6.0/jre/lib/i386/clie
Great work on Skypiax, Giovanni.
We've tested it in our lab for sometime and it works very well.
Unfortunately, when we tried deploying it on a production environment
(running Win2K3 server farm), we ran into a barrier:
1. FS is running as terminal server console application (to be easil
Hello,
I use below to realize call out conference:
conference testconf bgdial
{originate_timeout=30}sofia/default/1...@192.168.0.72 1234567890
FreeSWITCH_Conference"
conference testconf bgdial
{originate_timeout=30}sofia/default/1...@192.168.0.170 1234567890
FreeSWITCH_Conference"
but when t
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