please follow the procedures http://wiki.freeswitch.org/wiki/Reporting_Bugs to
report bugs at http://jira.freeswitch.org. Also, you will need to provide far
more detail than in this email for anyone to be able to have a possibility of
fixing it. Please include details such as, what file is
Please retest this with current svn trunk fresh checkout.
Thanks
Mike
On Nov 23, 2009, at 9:47 PM, Brian West wrote:
Ok 32bit... we are currently working on that as I type.
/b
On Nov 23, 2009, at 8:44 PM, James Budge wrote:
2GHz Intel Core Duo
OS 10.6.2
Xcode 3.2.1
Updated to
This was fixed in trunk yesterday about 8 hrs before you sent this message.
(15619). Please update and try again.
Mike
On Nov 23, 2009, at 11:33 PM, John Platts wrote:
I was using revision 15586.
From: br...@freeswitch.org
Date: Mon, 23 Nov
Your not telling anything to call your callback.
On Nov 24, 2009, at 1:03 AM, Baskar wrote:
Hi,
I want to check value given to the javascript with conditions whether it is
voicefile, extension or mobile Number when i press the dtmf value.
Steps i need to check in javascript:
When i
async?
On Nov 24, 2009, at 2:22 AM, velusamy velu wrote:
Dear All,
I am using Perl ESL::IVR module to develop a simple IVR. I have
filtered DTMF events. I have also set playback_terminators to cut the
playback when giving the digits. I have faced problem that DTMF event has not
Yes, I am using async mode only..
On Tue, Nov 24, 2009 at 2:12 PM, Michael Jerris m...@jerris.com wrote:
async?
On Nov 24, 2009, at 2:22 AM, velusamy velu wrote:
Dear All,
I am using Perl ESL::IVR module to develop a simple IVR. I have
filtered DTMF events. I have also set
Hi,
thanks for the suggestion! In the end i updated freeswitch using lastest
source in the trunk and callee_id_number worked!
Best Regard,
Daniele
Michael Jerris ha scritto:
Try running the info app there and see if the info is anywhere in that output .
Mike
On Nov 23, 2009, at 5:36 AM,
Hi,
I'm trying to setup call transfer for a phone without a transfer button. I was
on IRC last night and got some pointers to how this is setup in dialplan.xml
and features.xml and what bind meta app does.
Once it became clear how the transfer is initiated and that the transfer, in
the
Yes Mr. Collins, I've tried with shed_api. But I was not able to control, if
the user reject the call.
I made a shed_api to originate a call to 1000 and If it is answered, I'll
transfer the call to 9097 (So it comes to my program, refer the dialplan in
my question). But what happens if the user
Hi Mike,
I understand. I just need to not use the session.answer().
Many thanks.
Michael Jerris wrote:
This is done automatically when you bridge 2 sessions together.
Mike
On Nov 23, 2009, at 6:45 AM, Oscav wrote:
How can we send the answer to the caller only when the callee
I've tried the following program as per the suggestion that you've told. But
it seems, no success. Once the connection is closed, I created a new
connection and I send originate to originate a new call. But it is not
working.
require ESL;
use IO::Socket::INET;
use Data::Dumper;
my $ip =
Hi Anthony,
Now it works very well. Thank you so much for your help. I'm having a lot of
fun with this platform.
Regards.
Anthony Minessale-2 wrote:
is that your exact code?
${uuid} will not be expanded by javascript
var uuid = session.getVariable(uuid);
Hello,
We have a mix of phones that support RTP encryption and those that do not.
I have to support both types in the meanwhile, and would like to have
encryption enabled on the relevant leg, even if the other leg does not
support it (why? one of our ATAs either must have it unencrypted or have
Hello Anthony,
Indeed I see the reference to this channel variable in the code, but when
trying to access it from the dial plan it is empty... I try to get the value
of ${sip_profile_name} and it is empty.
Thanks! __Yehavi:
2009/11/23 Anthony Minessale
Hi Jeff,
All is good, I have looked at the x64 related changes you made and will merge
them back to UniMRCP tree most probably during the next week.
Thanks,
Arsen.
From: Jeff Lenk jl...@frontiernet.net
To: freeswitch-users@lists.freeswitch.org
Sent: Mon,
Hi,
I have an Ubuntu box running FS1.0.4 which has been processing a good
volume of calls between local users with soft phones (Xlite) and GSM
handsets via a number or Portech gateways, this has worked very well
for some time and audio quality is very good.
I've now added a Sangoma A200 with
Hi everyone, I'm setting up FS cluster In my application, I plan to use
two FS server as front and four FS as backend, the incoming calls first
send to the front FS, then the front FS forward the call to backend FS
server by return 302 to invite message. The front FS need to known the
Hi
Can we enable passive recording in freeswitch ,wanpipe ,openzap , we
are using a sangoma tapping system with freeswitch.
Thanks
Imthiyaz
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FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
Hello,
I have been going through FreeSWITCH for quite sometime now. I would like to
develop my own SIP Client soft-phone in Java/etc., how do I start?. Will I
get any SDK/APIs for this.
Please assist.
Thanks,
Rex
--
View this message in context:
Hi to all,
I am very new this platform . I just downloaded freeswitch to my windows xp
machine , compiled successfully and run. After that I dont have any idea
what to do :( . I am not finding simple kind of tutorial on the net. could
you please suggest me, how I have to proceed. My requirement
Hi,
I tried to use the variable remote_media_ip from within dialplan, but it is
not returning anything.
Does anyone know when this variable gets set and how to have this variable
to be set as soon as an INVITE hit freeswitch?
Thanks,
jb
___
Suggestion: Be one the first to integrate QuteCOm
-E
Gpro.ws
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FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
What does this have to do with uniMRCP?
Mike
On Nov 24, 2009, at 9:54 AM, Imthiyaz Ahmed wrote:
Hi
Can we enable passive recording in freeswitch ,wanpipe ,openzap , we
are using a sangoma tapping system with freeswitch.
___
FreeSWITCH-users
It gets set whenever the codec is negotiated. So it'll be NULL until
(pre_)answer if you have late-negotiation on.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 24-Nov-09, at 10:22 AM, Juan Backson wrote:
Hi,
I tried
Hi,
In the case of proxy_media=true, does it gets set at all then?
thanks,
jb
On Tue, Nov 24, 2009 at 11:46 PM, Mathieu Rene mrene_li...@avgs.ca wrote:
It gets set whenever the codec is negotiated. So it'll be NULL until
(pre_)answer if you have late-negotiation on.
Mathieu Rene
1) Did you ever supply a log of your problem?
2) Are you using ESL lib or did you make your own event socket client, (if
you did maybe you implemented the protocol client wrong)
You are not supplying any specific information like traces of the connection
or the version of the code you are using,
I have modified sofia.c in mod_sofia so that I can define gateways without
having to specify the password parameter. This is because I am using a SIP
gateway that does not require SIP registration. The modified version still
requires the password to be set on any gateway for which register is
John,
If the remote end doesn't require a username or password then you
don't need to create a gateway to send a call to that endpoint. You
can simply do sofia/profile/num...@remoteip and it'll work.
Also can you put the patch on jira via http://jira.freeswitch.org
/b
On Nov 24,
1. can you supply a trace of this esl communications.
2. is it inband or rfc2833 dtmf ?
MIke
On Nov 24, 2009, at 3:59 AM, velusamy velu wrote:
Yes, I am using async mode only..
On Tue, Nov 24, 2009 at 2:12 PM, Michael Jerris m...@jerris.com wrote:
async?
On Nov 24, 2009, at 2:22 AM,
On Nov 24, 2009, at 5:29 AM, Dave Stevenson wrote:
Hi,
I'm trying to setup call transfer for a phone without a transfer button. I
was on IRC last night and got some pointers to how this is setup in
dialplan.xml and features.xml and what bind meta app does.
Once it became clear how
Hello,
I have a similar problem with Freeswitch behind OpenSIPS as a load balancer:
When registering, Freeeswitch does not send a MWI NOTIFY message for a
Phone which has voicemails. Even after recording a new voicemail there
is no NOTIFY message sent. And there are no error messages on the
I was having trouble doing call forwarding from my SIP phone that is connected
to FreeSWITCH. It turns out that my SIP phone is actually sending 302 Moved
Temporarily responses, but my SIP gateway does not support 302 Moved
Temporarily or SIP REFER messages. How do I get FreeSWITCH to forward
I actually checked out revision 15654 today, and I was still getting problems
with proxy media and bypass media in FreeSWITCH.
From: m...@jerris.com
Date: Tue, 24 Nov 2009 03:39:16 -0500
To: freeswitch-users@lists.freeswitch.org
Subject: Re:
You'll have to hairpin the media thru your machine usually if they
won't accept either of those.
/b
On Nov 24, 2009, at 3:05 PM, John Platts wrote:
How do I get FreeSWITCH to forward calls without sending 302 Moved
Temporarily or SIP REFER messages?
Are you sure you did a make current? and can you outline the issue in
more detail?
/b
On Nov 24, 2009, at 3:28 PM, John Platts wrote:
I actually checked out revision 15654 today, and I was still getting
problems with proxy media and bypass media in FreeSWITCH.
connect to FS with fs_cli
Issue the command:
/events MESSAGE_QUERY MESSAGE_WAITING
then leave some voice mails
probably you have a mis-configuration where the user/domain/profile cannot
be resolved to the correct
sofia profile to send the notify
The event starts out as a freeswitch event and
I have considered writing JavaScript code to bridge two calls together.
However, I would like to perform custom handling of the 302 Moved Temporarily
response. How do I handle the 302 Moved Temporarily response if I use
JavaScript?
you may want to try the latest release of both wanpipe and FS openzap is
still a moving target since its in constant development from both the
hardware and software end
On Tue, Nov 24, 2009 at 7:25 AM, Steven Brown st...@justfone.com wrote:
Hi,
I have an Ubuntu box running FS1.0.4 which has
Hi Mike,
thanks for the reply. I am using the pre-compiled Windows binary - is there a
1.0.5 pre-release of that yet ?
FreeSwitch reports its version as 1.0.4 (14460) but this is not correct, I was
sure that I had previously loaded a later SVN Version, but just did it again,
unless I'm not
Anthony, thanks for the hint,
I receive events like the following
RECV EVENT
Event-Name: MESSAGE_WAITING
Core-UUID: e71632c8-d948-11de-942b-0138c6269e37
FreeSWITCH-Hostname: sip11.mydomain.com
FreeSWITCH-IPv4: 192.168.178.200
FreeSWITCH-IPv6: ::1
Event-Date-Local: 2009-11-24 23:33:13
Is there any way to tell FreeSWITCH to do the following when 302 Moved
Temporarily is sent to FreeSWITCH:
- End the session between FreeSWITCH and the phone
- Bridge the original session with the number that the call is forwarded to
From:
Hi members,
I'm controlling freeswitch with the conference module via xmlrpc.
Is it desired that the kick command can only kick users that are connected
to the conference?
Is there no chance abort an invitation?
The kick command has no effect until the person I invited with the dial
Hi again folks,
I have posted a dump into the Pastebin (11276), could someone have a look and
perhaps suggest where the problem might be please ?
I'm sure you'll be able to work it out, but the log is for a call where :-
incoming on PSTN Line (ext 1000)
Group exts, 111, 1001, 1001
Answered on
you can do this in follow steps:
1.edit default.xml diaplan config file in your fs config
directory(FS/conf/dialplan/default.xml), and section
extension name=ivr_demo2
condition field=destination_number expression=^\*114$
action application=lua data=../ivr/test.lua/
I do not see the meta app getting added in your log
-
Dialplan: sofia/internal/1...@192.168.1.50 Action bind_meta_app(*
Without this no meta actions will occur
Dave Stevenson wrote:
Hi again folks,
I have posted a dump into the Pastebin (11276), could someone have a look
and perhaps
People commonly use 60 sec registration refreshes to keep NAT routers happy
Phillip Jones-2 wrote:
hi there,
I have set up some cisco 7960 up with fs. They work fine - but the only
way
I can keep them registered is to set the timer_register_expires in the
Cisco cfg file to something
The example script is there in the following link
http://pastebin.com/f332f2fda
In the previous post I have attached it. But it was not shown.
2009/11/25 Thangappan.M thangappan...@gmail.com
FreeSWITCH version: freeswitch 1.0.4
I am using ESL library
I attached the example Perl script
you should use execute_complete events to tell when a command you tried to
execute has finished and not poll the channel for a variable to be set because
FreeSWITCH is an asynchronous application in the mode you are describing and
you can never be sure of the timing.
You are STILL polling for
Hi .
Could you please tell me, How to connect sip phone (which one is more
friendly with freeswitch) to freeswitch. How I can check whether connection
is properly established or not?
--
If you have come to help me, you are wasting your time.
If you have come to because your liberation is
http://wiki.freeswitch.org/wiki/Getting_Started_Guide
http://wiki.freeswitch.org/wiki/Interop_List
On Nov 25, 2009, at 1:36 AM, ovvenkat wrote:
Hi .
Could you please tell me, How to connect sip phone (which one is more
friendly with freeswitch) to freeswitch. How I can check whether
Hi there Itamar,
Does the SPA3102 support TLS or only SRTP? And what about Brians
comments that 'It uses a sick twisted method of doing SRTP'. Do you
have it working using SRTP together with FS? What about TLS?
Otherwise are there any other ATA's that support TLS SRTP?
On Sun, Nov 22, 2009
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