We already concluded its your unacceptabe use of originate in hangup hook
right?
On Nov 1, 2009 7:45 PM, "Seven Du" wrote:
Just suspicious would be possible that happened on sqlite stage? I
manually deleted the channels from sqlite and nothing bad happend.
just FYI.
-- Forwarded message
Thanks! It works!
__Yehavi:
2009/11/1 Anthony Minessale
> try session.execute("hangup", "user_busy");
>
>
> On Sun, Nov 1, 2009 at 8:24 AM, Yehavi Bourvine <
> yehavi.bourv...@gmail.com> wrote:
>
>> Hello,
>>
>> We would like to handle an incoming call to a busy pho
FYI, Here is the log when I set sofia loglevel all 9
==
tport_wakeup_pri(00DFE3E8): events IN
tport_recv_event(00DFE3E8)
tport(00DFE3E8) msg 01B2E0C0 from (udp/MyIP:5060) has 1315 bytes, veclen
= 1
tport(00DFE3E8): msg 01B2E0C0 (1315 bytes) from udp/SSIP:5060/sip next=000
0
nta: rec
Hi all, I get a "415 Unsupported Media Type" when FS receiving call from a
softswitch. I captured some packets, It seems that the softswitch use SIP-I
protocol, does FS can handle SIP-I message?
===here is the invite messagefrom softswitch
INVITE sip:xx...@:5060;user=phone SIP/2.0
Contact:
M
Just suspicious would be possible that happened on sqlite stage? I
manually deleted the channels from sqlite and nothing bad happend.
just FYI.
-- Forwarded message --
From: Seven Du
Date: Sun, 1 Nov 2009 10:24:32 +0800
Subject: Fwd: [Freeswitch-users] Many CS_REPORTING state Zomb
Session-Expires: -1;refresher=uas
nta: 200 OK has fatal syntax errors
This is a know-bug in asterisk.
see: https://issues.asterisk.org/view.php?id=15621
On Sun, Nov 1, 2009 at 4:40 AM, Ivan C Myrvold wrote:
> No one have any idea why this is not working? I have combed through
> the log, but
if you set the variable to:
"originate foo bar baz"
"bgapi originate foo bar baz"
bgapi is itself an api command that takes the argument string and runs it in
a separate thread and returns instantly.
This will stop it from hanging at that point.
On Sun, Nov 1, 2009 at 1:53 PM, Dome Charoenyo
2009/11/2 Anthony Minessale :
> #4 makes no sense to me.
This solution call "miss callback". some contry cost for toll free
number is so high. but call to user cheaper. so when user want to
use international call. they call to my DID and my system callback to them.
> Are you just trying to create
you can make up your own variable and set whatever you want in there then
add it to the template.
On Fri, Oct 30, 2009 at 10:04 AM, DJB wrote:
> I wonder if I don't want to have b-leg in cdr csv, is there any variables
> that can give me the actual gateway ip address that is actually went out.
try session.execute("hangup", "user_busy");
On Sun, Nov 1, 2009 at 8:24 AM, Yehavi Bourvine
wrote:
> Hello,
>
> We would like to handle an incoming call to a busy phone according
> to user's prefference: Some want waiting call, some want to just reject the
> call, and others want to send the
#4 makes no sense to me.
Are you just trying to create a call that the channel does not participate
in?
Then for sure you want to use bgapi to cause the originate to happen in the
background.
Also your gcore report has to be taken while you have the stuck channels to
see why they are stuck.
I can
Meftah and Diego,
If you continue to argue on our mailing lists I will be forced to moderate
you to stop you from bothering other people.
I do not want to see one more reply to this thread from either of you.
Please do not reply to apologize, simply stop sending any more email to this
topic.
On
thank you
fixed in r15308
We do prefer jira for this kind of thing so we can track and make accurate
change logs.
On Sun, Nov 1, 2009 at 9:14 AM, Jonas Gauffin wrote:
> Same bug in switch_ivr_async.c, method switch_ivr_record_session.
>
>
> On Sun, Nov 1, 2009 at 4:06 PM, Jonas Gauffin wrote:
>
I never said your level is bad or anything, I just said that I don't
want people to involve me into that problem.
Diego
On Sun, Nov 1, 2009 at 5:09 PM, Diego Viola wrote:
> Hi Meftah,
>
> No, of course is not, and it will never be, I actually quite admire
> how you are able to do what you do.
>
Hi Meftah,
No, of course is not, and it will never be, I actually quite admire
how you are able to do what you do.
I just wanted to say that I don't want people to put me into this
problem, because I don't have anything to do with it, I was just
curious about how pjsip compares to sofia, etc. Tha
=D
On Sun, Nov 1, 2009 at 4:04 PM, Giovanni Maruzzelli wrote:
> On Sun, Nov 1, 2009 at 3:37 PM, Diego Viola wrote:
>> Don't put me on the same leval as DelphiWorld please, I was just
>> curious about how this SIP stack compares to sofia.
>
> Smile Diego, smile. We're all just jocking! :)
>
> -g
hi diego,
what you mean?
so my level is nothing?
my level is bad?
my level is zero?
thank to gmaruzz/MikeJ that understand me quickly/easyly
Diego Viola a écrit :
Don't put me on the same leval as DelphiWorld please, I was just
curious about how this SIP stack compares to sofia.
Diego
On Sun, N
On Sun, Nov 1, 2009 at 3:37 PM, Diego Viola wrote:
> Don't put me on the same leval as DelphiWorld please, I was just
> curious about how this SIP stack compares to sofia.
Smile Diego, smile. We're all just jocking! :)
-gm
>
> Diego
>
> On Sun, Nov 1, 2009 at 2:24 AM, Chris Burns wrote:
>> M
Hi!
On Sun, Nov 1, 2009 at 01:21, Jason White wrote:
>
> http://wiki.freeswitch.org/ is the best there is.
>
> It is being improved by the community over time.
>
> You can also take advantage of the IRC channel, the FreeSWITCH conference
> and
> of course the mailing list.
>
>
>
Thank you very m
Same bug in switch_ivr_async.c, method switch_ivr_record_session.
On Sun, Nov 1, 2009 at 4:06 PM, Jonas Gauffin wrote:
> switch_ivr_play_say.c, line 486.
>
> file = switch_core_session_sprintf(session, "%s%s%s%s",
> switch_str_nil(tfile), tfile ? "]" : "", prefix, SWITCH_PATH_SEPARATOR,
> file);
switch_ivr_play_say.c, line 486.
file = switch_core_session_sprintf(session, "%s%s%s%s",
switch_str_nil(tfile), tfile ? "]" : "", prefix, SWITCH_PATH_SEPARATOR,
file);
There should be five %s, not four.
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FreeSWITCH-users
I am setting up fs in pfsense.
Following the multihomed tutorial (I also have a dedicated wan/lan int)
if I set directory/default.xml domain=10.0.0.1 (my lan int ip) it breaks
everything, but if I set
in conf/vars.xml I now get audio working correctly, * etc plays MOH.
Is there still somethin
Don't put me on the same leval as DelphiWorld please, I was just
curious about how this SIP stack compares to sofia.
Diego
On Sun, Nov 1, 2009 at 2:24 AM, Chris Burns wrote:
> My favorite part of this 'civilized' discussion on IRC was when DelphiWord and
> diegoviola sat around tryin to take the
Hello,
We would like to handle an incoming call to a busy phone according
to user's prefference: Some want waiting call, some want to just reject the
call, and others want to send the call to voicemail.
We have a small JavaScript which tests the status of the destination and
the user's will
No one have any idea why this is not working? I have combed through
the log, but couldn't find any clue there.
Incoming calls from my sip provider is working perfect, but for
outgoing calls it looks like Freeswitch is not letting the incoming
rtp to the local sip phone.
Ivan
On 30. okt. 200
and it works :P
On Sun, Nov 1, 2009 at 6:38 AM, Michael Jerris wrote:
> see rupa's explanation below.
>
>
> On Nov 1, 2009, at 1:24 AM, Michael Collins wrote:
>
> How would you do an expression like: if $x < 24 in a condition tag? Just
> curious. I would like to make sure that is properly docume
i tought so :PP
T.
On Sun, Nov 1, 2009 at 6:34 AM, Michael Jerris wrote:
> This is a non working module, just a shell for development.
>
> Mike
>
> On Oct 30, 2009, at 5:52 PM, Tihomir Culjaga wrote:
>
> > does anybody know how does it work and how to use it in a dialplan?
> >
>
>
> ___
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