May be need
action application=set data=hangup_after_bridge=false/
before first bridge
Dome C.
2009/6/26 Edmar Cruz darklio...@yahoo.com:
include
extension name=multiple
condition field=destination_number expression=^(\d+)$
action application=set data=gate_site_id=1/
action
Dome Charoenyost d...@tel.co.th wrote:
May be need
action application=set data=hangup_after_bridge=false/
before first bridge
and also, reading this wiki page may help
http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate
(see the discussion of multiple destinations)
include
extension name=multiple
condition field=destination_number expression=^(\d+)$
action application=set data=gate_site_id=1/
action application=set
data=effective_caller_id_name=${effective_caller_id_name}/
action application=set
Dear All,
I'm asterisk developer(I have some code in Asterisk) .After 3
weeks with freeswich nothing to say. now i'm move all callingcard ,
wholesale platfrom to Freeswitch. I'm very happy with bridge ,
nibblebill. after finish this job i'll test FS PBX feature. i think
it's easy to do
Please try
include
extension name=multiple
condition field=destination_number expression=^(\d+)$
action application=set data=gate_site_id=1/
action application=set
data=effective_caller_id_name=${effective_caller_id_name}/
action application=set
Or Try pipe
action application=bridge data=sofia/default/$...@116.80.80.101 |
sofia/default/$...@116.80.80.102/
if you want to ring all. Try comma
action application=bridge data=sofia/default/$...@116.80.80.101 ,
sofia/default/$...@116.80.80.102/
2009/6/26 Edmar Cruz darklio...@yahoo.com:
you can combine the 2 gateways into one bridge app. pls see
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall
/nandy
On Fri, Jun 26, 2009 at 1:48 PM, Edmar Cruz darklio...@yahoo.com wrote:
include
extension name=multiple
condition field=destination_number
Yup your suggestions works... But I want my to have a prefix for the second
bridge
condition field=destination_number expression=^63819(\d+)$
Dome Charoenyost wrote:
Or Try pipe
action application=bridge data=sofia/default/$...@116.80.80.101 |
sofia/default/$...@116.80.80.102/
if you
Hi,
I receive an error message service not implemented sometimes when calling
a mobile phone number but sometimes it works. What maybe rhe cause of this
error? I already installed zfone, perfectly connect to two freeswitch and
the one issue I got today is these can you help me guys?
Thanks
hello all,
please i need a open source / free calling cart application to use with
my freeswitch
cool anyone chare any application with me?
thanks
__ Information from ESET NOD32 Antivirus, version of virus signature
database 4190 (20090626) __
The message was checked by ESET
[ Optimised G.729A 'Howlet' for FreeSWITCH Asterisk ]
Howler Technologies are proud to announce today the launch of
the first fully indemnified G.729A solution for FreeSWITCH.
This is the first in a series of products dubbed 'Howlets'
that add highly performant transcoding and signal
I can report that this problem (the failure of mod_portaudio.so to be linked
properly) still persists as of revision 13970.
The operating system is Debian Testing, and the difficulty began after
upgrading from Libtool 1 to Libtool 2.2.6a.
If anyone else can reproduce this or suggest a means of
Did you make bootstrap.sh and configure before compilation?
We have no issues on Ubuntu 9.04 (libtool 2.2.6a).
Regards,
Tamas
Jason White írta:
I can report that this problem (the failure of mod_portaudio.so to be linked
properly) still persists as of revision 13970.
The operating
hi,
we want to use uuid_displace with mux to playback a soundfile to a
bridged uuid, so that this uuid can hear the other side talk AND hear
the soundfile (whispering).
is there an option we can set, for defining the loudness level of the
soundfile? in our tests the soundfile was way to loud, so
Just add the prefix like -
action application=bridge data=sofia/default/$...@116.80.80.101 |
sofia/default/6381...@116.80.80.102/
BTW: Never give your real prefix. Anybody can use it to send traffic to your
trunk with that prefix and eat away your balance.
On Fri, Jun 26, 2009 at 3:05 AM, Edmar
Pedals! For countries with cheap labour, this can give endless power! Each
pedal connected to a dynamo. Workers come in shift after shift and keep
pedalling!
Its a solution for unemployement and electricity in those countries!
Check this out - http://www.msnbc.msn.com/id/26430304
On Fri, Jun 26,
[eap] EAP NAK
[eap] ERROR! Our request for tls was NAK'd with a request for tls. Skipping
the requested type.
Hi,
having a new voip pbx (OmniPCX Enterprise 9.0) from Alcatel-Lucent, I now try
to setup 802.1x with the phones, an Alcatel-Lucent IP Touch 4028 EE.
Freeradius is 2.x on a Debian
Sorry, wrong group :-) Freeswitch, Freeradius, both free
sorry :-)
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On Fri, 26 Jun 2009 00:51:34 -0400, Raymond Chandler
intralan...@freeswitch.org wrote:
windmills
Just set them up to be right next to the output of the system's fan!
I gotta get to the patent office, quick!
--Woof!
___
Freeswitch-users mailing
There doesn't seem to be any direct link between FreeSWITCH and Howler.
How does this benefit the project? Can we assume that Howler is giving
something back?
-Ray
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There doesn't seem to be any direct link between FreeSWITCH and Howler.
How does this benefit the project? Can we assume that Howler is giving
something back?
Well, they're providing additional functionality which a lot of folk
want, which requires substantial investment in money (visit
I can add a field to allow you to adjust the heartbeat on either channel if
that's necessary. Right now you are right, it's using the global setting
always.
Is this important?
-Original Message-
From: Dome Charoenyost [mailto:d...@tel.co.th]
Sent: Thursday, June 25, 2009 7:39 PM
To:
Congrats to them for taking the risk.
Just as an aside, just to let everyone know, I am taking the same risk. We
have invested in the same deal from sipro and will shortly be offering an
official FreeSWITCH g729 module in the very near future.
This, of course, does benifit the project directly,
FreeSWITCHers,
We have written an extensive guide on posting bugs to jira. Over the
past few weeks everyone has been a little lax in posting the correct
info the first time. The questions we ask are NOT optional to fill
out.. You must fill out every question on the list and doesn't
I have to agree with David here.
A G729 codec benefits the project immensely as it allows Freeswitch
users to use the most widespread commercial codec.
Best regards,
Vlasis Hatzistavrou
Kinetix Tele.com Hellas Ltd.
Monastiriou 9 Enotikon
54627
Thessaloniki
Greece
Tel.: +302310556134
Fax:
Brian, I would like to be one of the volunteers helping to report issues.
Chris
On Fri, Jun 26, 2009 at 11:13 AM, Brian West br...@freeswitch.org wrote:
FreeSWITCHers,
We have written an extensive guide on posting bugs to jira. Over the past
few weeks everyone has been a little lax in
2009/6/26 Harmeet Singh harm...@litatel.com:
Just add the prefix like -
action application=bridge data=sofia/default/$...@116.80.80.101 |
sofia/default/6381...@116.80.80.102/
BTW: Never give your real prefix. Anybody can use it to send traffic to your
trunk with that prefix and eat away
2009/6/26 Darren Schreiber d...@d-man.org:
I can add a field to allow you to adjust the heartbeat on either channel if
that's necessary. Right now you are right, it's using the global setting
always.
Is this important?
Yes important for me. and early media also :)
90% mobile in thailand use
Hey David,
There doesn't seem to be any direct link between FreeSWITCH and
Howler.
How does this benefit the project? Can we assume that Howler is
giving
something back?
Well, they're providing additional functionality which a lot of folk
want, which requires substantial investment
if you want to import multiple vars you can set import to a comma
separated list of vars. This should solve the problem in the short
term, the question for Darren is, do you think we should make the mod
read from the a or b channel for those vars.
Mike
On Jun 26, 2009, at 11:26 AM, Dome
Hey David,
There doesn't seem to be any direct link between FreeSWITCH and
Howler.
How does this benefit the project? Can we assume that Howler is
giving
something back?
Well, they're providing additional functionality which a lot of folk
want, which requires substantial investment in
most of the information about fifo is : http://wiki.freeswitch.org/wiki/Mod_fifo
Mike
On Jun 26, 2009, at 2:45 AM, Dome Charoenyost wrote:
Dear All,
I'm asterisk developer(I have some code in Asterisk) .After 3
weeks with freeswich nothing to say. now i'm move all callingcard ,
Brian West pisze:
FreeSWITCHers,
We have written an extensive guide on posting bugs to jira. Over the
past few weeks everyone has been a little lax in posting the correct
info the first time. The questions we ask are NOT optional to fill
out.. You must fill out every question on the list
Hello!
I got a problem with one way audio, symptoms are:
firstly play audio file to channel A (A is hears sound)
secondly bridge channel B with A (A doesn't hear B).
Environment:
- no NAT
- User Agents being used X-Lite, EyeBeam, SJphone - same result for all of
them- no audio, Wireshark shows
On Fri, Jun 26, 2009 at 9:05 AM, Michael Jerris m...@jerris.com wrote:
I would pre-adjust the volume of the soundfile with sox instead of
doing it real time.
Mike
I have to agree with Mike here. Sox is awesome for this kind of thing and
disk space is way more plentiful than CPU power.
-MC
Are you playing a stereo file by chance?
/b
On Jun 26, 2009, at 12:46 PM, Michael Collins wrote:
On Fri, Jun 26, 2009 at 9:05 AM, Michael Jerris m...@jerris.com
wrote:
I would pre-adjust the volume of the soundfile with sox instead of
doing it real time.
Mike
I have to agree with Mike
Well I'm going to try my best to reproduce issues on demand but
sometimes I just can't, other times the reporter stops responding and
I have no choice but to close the issue out.
:)
/b
On Jun 26, 2009, at 11:14 AM, Szymon Olko wrote:
I agree that those information are needed for
Edmar,
I strongly recommend you review the troubleshooting page on the wiki. No one
will be able to help you without more information about your issue.
http://wiki.freeswitch.org/wiki/Reporting_Bugs
Be sure to use pastebin and capture the debug output from the FS command
line and also paste your
It involves more than just reporting... it involves trying to
reproduce the issue or clarify the steps to reproduce an issue.. I
have been thru some issues that sound crazy hard to reproduce... like
stand on your head .. dial 1234, spin around dial 456, scream into the
phone... dance a
Updates:
1. One-way audio is in 95% tries. But how the rest 5% works??
2. Strange FS logging after the channels are bridged (user A talk to user B)
2009-06-26 02:16:07 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel
sofia/external/1...@192.168.147.1 entering state [ready]
2009-06-26
Can you try svn trunk... I know the remote FreeSWITCH isn't on trunk
also... due to the lines below.
/b
On Jun 26, 2009, at 12:59 PM, Artem Shiyanov wrote:
o=FreeSWITCH 3898736844884231 105343281763004076 IN IP4
192.168.147.130
s=FreeSWITCH
Hi all,
I'm just writing a perl script as dialplan to learn how to handle freeswitch.
Now I have the issue that my hangup hook won't be triggered after I intercepted
a call.
A normal hangup triggers the function.
Does anybody have a hint how to get the hangup hook triggered?
Thanks, Klaus
Depends what side of the call was the hangup hook on?
/b
On Jun 26, 2009, at 2:48 PM, Klaus Hochlehnert wrote:
Hi all,
I’m just writing a perl script as dialplan to learn how to handle
freeswitch.
Now I have the issue that my hangup hook won’t be triggered after I
intercepted a call.
Actually one of my first actions in the script is
$session-setHangupHook('on_hangup');
When a call comes in the hook is set and working.
The second time the script is called when I try to intercept. As it's the same
script there's also the function setHangupHook called.
That's what I've
well in your case I suspect your intercepting the leg of the call
without the hook on it.
/b
On Jun 26, 2009, at 3:22 PM, Klaus Hochlehnert wrote:
Actually one of my first actions in the script is
$session-setHangupHook('on_hangup');
When a call comes in the hook is set and working.
The
Ok, and how can I ask the hook to come with me?
If I understand this right...
When a call comes in the hook is set on the a-leg and it rings on the b-leg.
When I do an intercept I kill the ringing b-leg and the interceptor is now the
new b-leg, right?
I would assume that the old a-leg still has
Can you paste in your script so we can see what is going on?
-MC
On Fri, Jun 26, 2009 at 1:39 PM, Klaus Hochlehnert maili...@kh-dev.dewrote:
Ok, and how can I ask the hook to come with me?
If I understand this right...
When a call comes in the hook is set on the a-leg and it rings on the
How do you get a system variable from within a lua startup script?
Specifically I want domain_name from vars.xml ... normally I'd use
session:getVariable, however there is no session in this case.
-- John
-
| Feith Systems
You can execute global_getvar api call.
/b
On Jun 26, 2009, at 4:33 PM, John Wehle wrote:
How do you get a system variable from within a lua startup script?
Specifically I want domain_name from vars.xml ... normally I'd use
session:getVariable, however there is no session in this case.
You can execute global_getvar api call.
Thanks ... I've updated the wiki.
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com |
|John Wehle| Fax: 1-215-540-5495 |
John,
Also can you go over the few jira's you have left and see if we can
resolve/knock out some of them.. I'm wanting to roll pre9 this weekend.
Thanks,
Brian
On Jun 26, 2009, at 5:13 PM, John Wehle wrote:
You can execute global_getvar api call.
Thanks ... I've updated the wiki.
Brian told me to open a jira, what I did now.
But here's the script.
It basically writes incoming calls to a database and removes them after hangup.
When intercepting it also rewrites the destination of the call.
Thanks, Klaus
#!/usr/bin/perl
use strict;
use DBI;
use POSIX qw(strftime);
our
Please open the jira and attach all this and the xml dialplan to
execute it .. also the schema for the db would be helpful also.
Thanks,
/b
On Jun 26, 2009, at 6:16 PM, Klaus Hochlehnert wrote:
Brian told me to open a jira, what I did now.
___
I used the Project FS Scripts. Hope that's ok.
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West
Sent: Saturday, June 27, 2009 1:28 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] hangup
Its more FSCORE but i'll move it into the right place ;)
/b
On Jun 26, 2009, at 6:36 PM, Klaus Hochlehnert wrote:
I used the Project „FS Scripts“. Hope that’s ok.
From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Brian
Chris Chen chris.chen2...@gmail.com wrote:
Brian, I would like to be one of the volunteers helping to report issues.
That's great. We need more volunteers.
For some FreeSWITCH users (of whom I am one), the user interface of Jira is an
obstacle to reporting bugs via that mechanism, for
Tamas jal...@gmail.com wrote:
Did you make bootstrap.sh and configure before compilation?
Yes. This was a clean export from svn, built by running the Debian debuild
tool, as in
svn export to a temporary directory, followed by debuild (after changing the
version number to make the package
Jason, In your case we'll gladly accept your bug reports via the mailing
list for that exact reason. ;)
/b
On Fri, Jun 26, 2009 at 7:02 PM, Jason White ja...@jasonjgw.net wrote:
Chris Chen chris.chen2...@gmail.com wrote:
Brian, I would like to be one of the volunteers helping to report
what are the error messages?
/b
On Fri, Jun 26, 2009 at 7:05 PM, Jason White ja...@jasonjgw.net wrote:
Tamas jal...@gmail.com wrote:
Did you make bootstrap.sh and configure before compilation?
Yes. This was a clean export from svn, built by running the Debian debuild
tool, as in
svn
Brian West br...@freeswitch.org wrote:
what are the error messages?
There aren't any. The build completes without error, but the module doesn't
load due to the undefined symbols.
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Hi, I recently tried upgrading FreeSWITCH from 1.0.2 to 1.0.4pre8 on my
server, but the application I wrote uses custom channel variables which I
create using setVariable in Python. In version 1.0.2, I am able to
retrieve that variable within the event xml (it prepends the variable name
with
On Wed, Jun 24, 2009 at 12:53:23AM -0400, Michael Jerris wrote:
Can you post a bug to Jira.freeswitch.org with all these warnings,
even better with patches to fix it.
OK. I think I have narrowed the problem down to 3 issues.
1. The build system is treating even a single warning as a
On Thu, Jun 25, 2009 at 06:06:04PM -0400, Michael Jerris wrote:
Is there a log file somewhere that contains the actual compile
commands and error output so you can find out what happened when
there is a error? Or perhaps a configuration to enable it to come
out on the console?
Execute the app verbose_events
/b
On Jun 26, 2009, at 9:10 PM, Adam Wilt wrote:
Hi, I recently tried upgrading FreeSWITCH from 1.0.2 to 1.0.4pre8 on
my server, but the application I wrote uses custom channel variables
which I create using setVariable in Python. In version 1.0.2, I am
Sorry - I misread the question.
The error is:
freeswi...@default load mod_portaudio
-ERR [module load file routine returned an error]
2009-06-27 12:30:00.740316 [CRIT] switch_loadable_module.c:871 Error Loading
mod
ule /opt/freeswitch/mod/mod_portaudio.so
**/opt/freeswitch/mod/mod_portaudio.so:
In the latest SVN this seem to be true for CHANNEL_HANGUP, which appends the
following after the normal line breaks.
Disconnected, goodbye.\n
See you at ClueCon! http://www.cluecon.com/;
Perhaps it can be changed to conform the chunked data structure?
variable_disconnect_message: Disconnected,
The protocol has a content-length header.. you read exactly that many
bytes and you won't be confused by this disconnect message.
/b
On Jun 26, 2009, at 9:50 PM, Lon Baker wrote:
In the latest SVN this seem to be true for CHANNEL_HANGUP, which
appends the following after the normal line
Brian,
Thanks! I was just about to post a never mind after further reading
source.
Lon
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On Jun 26, 2009, at 10:17 PM, Vincent Stemen vince.freeswi...@hightek.org
wrote:
On Wed, Jun 24, 2009 at 12:53:23AM -0400, Michael Jerris wrote:
Can you post a bug to Jira.freeswitch.org with all these warnings,
even better with patches to fix it.
OK. I think I have narrowed the
I'm trying to use xml_rpc to initiate an att_xfer on channel A (which is
already bridged to channel B), but I'm running into some issues.
I know the uuid's from both channel A and B, but the documentation I found
on att_xfer only seems to indicate a way to do this from DMTF
presses occurring on
Not sure what you want to do is doable via XML RPC. That app is to be
run on an existing session. The other solution is to take both legs
and park them.. Then execute bridge on one leg to the target transfer
person. Once that call is up.. you can them park both of those..
uuid_bridge
can you 3 way with uuid_bridge?
--matt
On Fri, Jun 26, 2009 at 9:08 PM, Brian West br...@freeswitch.org wrote:
Not sure what you want to do is doable via XML RPC. That app is to be
run on an existing session. The other solution is to take both legs
and park them.. Then execute bridge on one
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