Re: [Freeswitch-users] Secure RTP

2009-05-26 Thread Jason White
Jim Burke j...@evolutiontel.net wrote:
 
 extension name=On-Net_calls
   condition field=destination_number expression=^103$
 action application=set data=continue_on_fail=79/
 action application=set data=continue_on_fail=true/

Why are you setting the same variable twice? The result will be that only the
second of the above set commands takes effect, so you should delete the first
of them.


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Re: [Freeswitch-users] Secure RTP

2009-05-26 Thread Jason White
Jim Burke j...@evolutiontel.net wrote:
 If I understand your comment correctly, I did not have both of the
 above snippets in the dialplan at the same time.  The dialplan was
 modified continually to get the correct vars that worked for my
 situation and then reloadxml to get them working.

Right, you can't have
action application=set data=continue_on_fail=79/
action application=set data=continue_on_fail=true/
one after the other like this, because it is equivalent to
action application=set data=continue_on_fail=true/
since you're overriding the setting of the variable in the very next line.


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Re: [Freeswitch-users] Secure RTP

2009-05-26 Thread Jim Burke
Ahh right now I get what you are saying, I thought from the wiki that
I would have to set the feature on and then tell it what cause codes I
wanted to trap.  Will fix up my dialplan cause I don't want it to trap
other causes for this scenario.

Thanks!

On Tue, May 26, 2009 at 6:09 PM, Jason White ja...@jasonjgw.net wrote:
 Jim Burke j...@evolutiontel.net wrote:
 If I understand your comment correctly, I did not have both of the
 above snippets in the dialplan at the same time.  The dialplan was
 modified continually to get the correct vars that worked for my
 situation and then reloadxml to get them working.

 Right, you can't have
 action application=set data=continue_on_fail=79/
 action application=set data=continue_on_fail=true/
 one after the other like this, because it is equivalent to
 action application=set data=continue_on_fail=true/
 since you're overriding the setting of the variable in the very next line.


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Re: [Freeswitch-users] err after installling freeswitch

2009-05-26 Thread IPW
Hei!
Hvilken speditør?

Per Westrøm

Ing. Per Westrøm, Leiv Eirikssonsgt. 7, 0271 Oslo,
Norway Tlf. + 47 22444550, Fax + 47 22554964,
Mail p...@le7.no [mailto:p...@le7.no

-Opprinnelig melding-
Fra: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] På vegne av Michael
Jerris
Sendt: 26. mai 2009 03:18
Til: freeswitch-users@lists.freeswitch.org
Emne: Re: [Freeswitch-users] err after installling freeswitch

This looks like a permissions error on creating or opening files in  
the db directory.

Mike

On May 25, 2009, at 12:25 AM, mashudi wrote:

 Hi Guys,
 I have install Freeswitch with version : FreeSWITCH Version 1.0.4pre7
 (13238M) Started.
 I load the openzap module after install the wanpipe modul and  
 everything
 running , after I query the status of the Sangoma Card (A104D quad E1)
 we got error like this :
 2009-05-25 11:07:45 [ERR] switch_core_sqldb.c:95
 switch_core_db_persistant_execute_trans() SQL ERR [unable to open
 database file]
 2009-05-25 11:07:45 [ERR] switch_core_sqldb.c:95
 switch_core_db_persistant_execute_trans() SQL ERR [unable to open
 database file]
 2009-05-25 11:07:45 [ERR] switch_core_sqldb.c:95
 switch_core_db_persistant_execute_trans() SQL ERR [unable to open
 database file]


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Re: [Freeswitch-users] 407 Proxy Authentication Required

2009-05-26 Thread Jason White
Brad Tuan brad.t...@gmail.com wrote:
 
 But,the response message change from 407 Proxy Authentication Required to
 480 Temporarily Unavailable today.
 
 Anybody can tell me what happen??

Your SIP trace might give you a clue as to what happened.
sofia profile external siptrace on
(substituting the relevant profile for external in the above command, as
required, and repeating the command if more than one profile is involved).


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Re: [Freeswitch-users] uuid_transfer gets break

2009-05-26 Thread Anthony Minessale
This one happens to every new guy trying to make FS into a dialer app using
JS.

for every sessionX you create in js with the new Session constructor

sessionX.setAutoHangup(0);

This allows the channels to remain alive outside the script once they are
hungup/transferred etc.


On Mon, May 25, 2009 at 6:27 AM, Gopalakrishnan A.N sai...@gmail.comwrote:

 Hi,
   I had some discussion with the IRC regarding the uuid_transfer gets
 hang-up where the call is originated via javascript thru event socket. I was
 suggested to install latest SVN trunk. I did that again the same issue, the
 log is attached with here http://pastebin.freeswitch.org/9103

 My call flow like this,

 1. api jsrun fils.js arg1 arg2 arg3
 2. capture the uuid
 3. api uuid_transfer uuid -both ext no / conference no

 Both the leg gets hangedup.

  Someone can assist me where I am wrong.

 --
 Thank you  with regards,
 Gopal,


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Re: [Freeswitch-users] Unicast isn't working

2009-05-26 Thread David Knell
Hi Artem,

Please to see that some of the stuff I wrote is useful to someone..!

I've written an FS module which will send the audio over - it's more efficient 
than using unicast.  Let me know if you'd like a copy.

Cheers --

Dave
  - Original Message - 
  From: ?  
  To: freeswitch-users@lists.freeswitch.org 
  Sent: Tuesday, May 26, 2009 9:27 AM
  Subject: [Freeswitch-users] Unicast isn't working


  Hi,

  I am trying to setup ASR in FreeSwitch using Nuance ASR server and MRCP. Both 
FreeSwitch and Nuance installed on Windows Server 2003. FreeSwitch version is 
1.0.3 (12567M)

  I found an example in Perl at 
http://www.softivr.com/wiki/index.php/FreeSWITCH_MRCP_in_Perl and decided to do 
the same in C#.
  It establishes connection with Nuance and loads the grammar, everything works 
fine. The next step is to capture audio from FS and tramsmit it to ASR. This 
can be done with unicast. We must create an outbound socket and issue unicast 
command. Here it goes:

  private void SetupAudioTransmission()
  {
   EventSocket = new Socket(AddressFamily.InterNetwork, SocketType.Stream, 
ProtocolType.Tcp);
   EventSocket.SendTimeout = Config.Timeout;
   int ESPort = SearchESPort(EventSocket);

   Thread thrESListener = new Thread(new ThreadStart(ListenerThreadStart));
   thrESListener.Start();
   
   WriteLog(LogLevel.Info, Creating outbound event socket);
   Session.Execute(socket, 127.0.0.1: + ESPort); //main thread stops, 
listener thread listens for outbound socket connection.
   WriteLog(LogLevel.Info, Outbound event socket disconnected);

   EventSocket.Close();
  }

  //here we accept outbound socket and transmit unicast command through it
  private void ListenerThreadStart()
  {
   Socket sockHandler = EventSocket.Accept();
   WriteLog(LogLevel.Info, Incoming connection);
   sockHandler.Send(MessageEncoding.GetBytes(Connect\n\n));
 
   WriteLog(LogLevel.Info, GetServerResponse(sockHandler));

   int rtpPort = (RTPSocket.RemoteEndPoint as IPEndPoint).Port;
   string command = string.Format(sendmsg\r\ncall-command: 
unicast\r\nlocal-ip: {0}\r\nlocal-port: {1}\r\nremote-ip: {2}\r\n +
remote-port: {3}\r\ntransport: udp\r\nflags: native\r\n\r\n, 
Config.LocalIP, rtpPort + 1, Config.LocalIP, rtpPort);

   WriteLog(LogLevel.Info, command);

   sockHandler.Send(MessageEncoding.GetBytes(command));
   WriteLog(LogLevel.Info, GetServerResponse(sockHandler));
   
   sockHandler.Disconnect(false);  
   sockHandler.Close();  
  }

  After this, FS writes that unicast has been created on corresponding IPs and 
ports. It really creates an UDP socket, but doesn't transmit any data through 
it. I tested it with Wireshark and from my application, nothing was detected.
  Also, if we specify transport:tcp in unicast command, it uses UDP anyway, 
that's strange.

  Here is how I listen UDP packets.
  private void DetectSpeech()
  {
   WriteLog(LogLevel.Info, Reading audio);
   byte[] FSRecvBuf = new byte[2048];
   
   IPEndPoint epFS = new IPEndPoint(IPAddress.Loopback, 
(RTPSocket.RemoteEndPoint as IPEndPoint).Port);
   FSSocket = new Socket(AddressFamily.InterNetwork, SocketType.Dgram, 
ProtocolType.Udp);
   FSSocket.ReceiveTimeout = 10;
   FSSocket.SendTimeout = 10;
   WriteLog(LogLevel.Info, Binding socket to  + epFS.Port);

   FSSocket.Bind(epFS);
   FSSocket.Connect(new IPEndPoint(IPAddress.Loopback, epFS.Port + 1));
 
   while (Session.Ready())
   {
int recvCount = FSSocket.Receive(FSRecvBuf);
WriteLog(LogLevel.Info, Received bytes:  + recvCount);
   }
  }

  Can someone help me to solve this? Do I do something wrong or I forgot 
something or it doesn't work at all?

  Best regards.
  Artem




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Re: [Freeswitch-users] Conference users hear MOH until leader enters?

2009-05-26 Thread Brian West
Jason,
There are many ways to accomplish this using FreeSWITCH.  All of  
which will require you to do a little bit of coding in js, lua or some  
other language.

1. Park all callers into a fifo.. (see mod_fifo)
2. When leader auths in your script then you uuid_transfer them all  
into the conference.

You could also do this via esl using perl, lua, python, php or ruby.

That should accomplish the same ending result.

/b



On May 26, 2009, at 9:38 AM, Jason Cromes wrote:

 This is easy in Asterisk because you can pop someone into MeetMe with
 different flags.  So, in my IVR, I prompt for the conference number
 (known to all) and then the password (known only to the owner/ 
 leader).
 If the proper password is entered, that user is sent to conference XYZ
 with the leader flag set.  If no password is entered, the user goes to
 conference XYZ, without the leader flag.  If anyone enters before the
 leader, they're told by MeetMe that the conference will begin when  
 the
 leader arrives and MeetMe provides MOH until that time.

 Help!  This is an absolute deal-breaker for my install...  How can I  
 do
 this in FreeSwitch?
 Thanks...


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Re: [Freeswitch-users] FS in Amazon EC2 for production?

2009-05-26 Thread Chris Danielson
There is a cloud computing company named 3Tera (AppLogic) that does have 
an international presence and will keep your FreeSWITCH instance running 
on a dedicated server using Xen and HA.  I spoke with one of their 
senior engineers about 1 month ago in regards to actually setting up an 
LCR scenario using their servers located in Europe and the United 
States.  These guys are a little different in the cloud computing world 
and I believe closer fit the needs of a telephony application.  As-is, 
there are companies using 3tera for their Asterisk installs.  So if you 
want cloud computing with dedicated hardware resources and a set 
geographic location, then these guys do it.  Kind of the best of both 
worlds.  Just a quick 2 cents...


Regards,
Chris


Brian West wrote:


On May 26, 2009, at 10:11 AM, Kristian Kielhofner wrote:


Hey Brian,

 FreeSWITCH in EC2 is a bit of a mystery to me...

 Call me old fashioned but in my mind VoIP and geography are linked
in %99 of scenarios.  Having VoIP services in a pure cloud
environment just doesn't sound like a good idea to me.

 Consider a hosted environment with clients registered to a
FreeSWITCH server.  One day your instance is physically running on
hardware in Seattle.  The next day it could (potentially) be running
in Chicago.  That's obviously a very different routing path for your
clients.  Even /if/ Amazon (or whomever) employs every routing/network
trick in the book you still won't be able to get over that change in
geography.


For some people this isn't a huge difference... now if it were to swap 
continents then yes it would be a problem.  But I haven't seen Amazon 
do this but I haven't left the instances up long enough to see.



 It's certainly possible a change like this may very well never
happen in practice.  I wouldn't know; I've never used EC2 and I don't
even know that much about it.  I'm just curious how well strictly,
practically speaking this will work in the long term.


There are other companies that do this stuff but personally me... I 
want my stuff running on real hardware.


Brian West
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Re: [Freeswitch-users] how to enable ESL for ruby?

2009-05-26 Thread Brian West
The makefile will have to be changed to work with OS X since the  
linking is done differently.


It would be very similar to this one http://www.bkw.org/esl.imac.diff

Below will get it to compile:

imac:esl brian$ svn diff
Index: ruby/Makefile
===
--- ruby/Makefile   (revision 13432)
+++ ruby/Makefile   (working copy)
@@ -1,6 +1,7 @@
 ## no idea how to simply ask ruby which one to use
 LOCAL_CFLAGS=-I$(shell ruby -e 'require rbconfig; puts  
RbConfig::CONFIG[topdir]')
 LOCAL_LDFLAGS=$(shell ruby -e 'require rbconfig; puts  
RbConfig::CONFIG[LIBRUBYARG]')

+LOCAL_LDFLAGS += -framework Ruby

 all: ESL.so

Index: ruby/esl_wrap.cpp
===
--- ruby/esl_wrap.cpp   (revision 13432)
+++ ruby/esl_wrap.cpp   (working copy)
@@ -823,7 +823,7 @@



-#include ruby.h
+#include ruby/ruby.h

 /* Remove global macros defined in Ruby's win32.h */
 #ifdef write
Index: Makefile
===
--- Makefile(revision 13432)
+++ Makefile(working copy)
@@ -12,7 +12,7 @@
 OBJS=src/esl.o src/esl_event.o src/esl_threadmutex.o src/esl_config.o
 SRC=src/esl.c src/esl_event.c src/esl_threadmutex.c src/esl_config.c  
src/esl_oop.cpp
 HEADERS=src/include/esl_config.h src/include/esl_event.h src/include/ 
esl.h src/include/esl_threadmutex.h src/include/esl_oop.h

-SOLINK=-shared -Xlinker -x
+SOLINK=-dynamiclib -Xlinker -x
 # comment the next line to disable c++ (no swig mods for you then)
 OBJS += src/esl_oop.o



/b



On May 26, 2009, at 10:22 AM, dujinfang wrote:


Hi,

Following the wiki: http://wiki.freeswitch.org/wiki/Event_Socket_Library
On MacOSX 10.5, I can't get ESL for ruby work. make throws error:

sevens-mac-pro:~/workspace/test/freeswitch/trunk/libs/esl$ make  
rubymod
make MYLIB=../libesl.a SOLINK=-shared -Xlinker -x CFLAGS=-I/ 
Users/

seven/workspace/test/freeswitch/trunk/libs/esl/src/include -
DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-
math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-
prototypes -Wmissing-prototypes CXXFLAGS=-I/Users/seven/workspace/
test/freeswitch/trunk/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -
I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable
CXX_CFLAGS= -C ruby
g++  -I/Users/seven/workspace/test/freeswitch/trunk/libs/esl/src/
include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall
-Werror -Wno-unused-variable -I/opt/local/lib/ruby/1.8/i686-darwin9 -c
esl_wrap.cpp -o esl_wrap.o
g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -lruby -o ESL.so -L.
Undefined symbols:
  _main, referenced from:
  start in crt1.10.5.o
ld: symbol(s) not found
collect2: ld returned 1 exit status
make[1]: *** [ESL.so] Error 1

Thanks for any help.


Brian West
br...@freeswitch.org

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Re: [Freeswitch-users] Conference users hear MOH until leader enters?

2009-05-26 Thread jcromes




Ok, that sounds doable... I have no problem banging around with some
code. Thanks for the advice...
I'm new to FIFOs and FreeSwitch in general, so please check my logic
here...

- When a user enters their conference number, I check to see if that
conference exists (because the conference shouldn't exist until the
leader enters). 
- If that conference exists, the user gets dropped straight into it.
- If that conference does not exist, I park that user into a FIFO, with
music. 

- When the leader enters their conference number and password, I put
them into that
conference (which creates the conference).
- Then check for the presence of
a FIFO with parked users.
- If the FIFO contains parked users, I transfer everyone from the FIFO
into
the conference.

Is that what you were thinking?


Brian West wrote:

  Jason,
	There are many ways to accomplish this using FreeSWITCH.  All of  
which will require you to do a little bit of coding in js, lua or some  
other language.

1. Park all callers into a fifo.. (see mod_fifo)
2. When leader auths in your script then you uuid_transfer them all  
into the conference.

You could also do this via esl using perl, lua, python, php or ruby.

That should accomplish the same ending result.

/b



On May 26, 2009, at 9:38 AM, Jason Cromes wrote:

  
  
This is easy in Asterisk because you can pop someone into MeetMe with
different flags.  So, in my IVR, I prompt for the "conference number"
(known to all) and then the "password" (known only to the owner/ 
leader).
If the proper password is entered, that user is sent to conference XYZ
with the leader flag set.  If no password is entered, the user goes to
conference XYZ, without the leader flag.  If anyone enters before the
leader, they're told by MeetMe that "the conference will begin when  
the
leader arrives" and MeetMe provides MOH until that time.

Help!  This is an absolute deal-breaker for my install...  How can I  
do
this in FreeSwitch?
Thanks...

  
  

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Re: [Freeswitch-users] Echo cancellation needed?

2009-05-26 Thread Niall Crosby
Thanks Brian.

2009/5/26 Brian West br...@freeswitch.org:

 On May 26, 2009, at 9:10 AM, Niall Crosby wrote:

 Dear FS users,

 If I am getting Sangoma hardware to connect Freeswitch to E1, should I
 get a card with echo cancellation or not?

 I would use hardware echo cancel if at all possible.

 Also, is there any advantage to PCIe over PCI? (I understand the
 difference between these and am gonig with PCIe, just checking encase
 there are legacy issues that make PCIe wrong choice)

 If its just one E1 it won't matter..



 Many thanks,
 Niall.


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Re: [Freeswitch-users] Echo cancellation needed?

2009-05-26 Thread Brian West

On May 26, 2009, at 9:10 AM, Niall Crosby wrote:

 Dear FS users,

 If I am getting Sangoma hardware to connect Freeswitch to E1, should I
 get a card with echo cancellation or not?

I would use hardware echo cancel if at all possible.

 Also, is there any advantage to PCIe over PCI? (I understand the
 difference between these and am gonig with PCIe, just checking encase
 there are legacy issues that make PCIe wrong choice)

If its just one E1 it won't matter..



 Many thanks,
 Niall.


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Re: [Freeswitch-users] FS in Amazon EC2 for production?

2009-05-26 Thread Kristian Kielhofner
On Tue, May 26, 2009 at 10:31 AM, Brian West br...@freeswitch.org wrote:
 Not with FreeSWITCH in our testing.  Now if you have stupid defaults
 in your virtualization env. it might act funny but I have run FS on
 EC2 without a problem.

 /b

Hey Brian,

  FreeSWITCH in EC2 is a bit of a mystery to me...

  Call me old fashioned but in my mind VoIP and geography are linked
in %99 of scenarios.  Having VoIP services in a pure cloud
environment just doesn't sound like a good idea to me.

  Consider a hosted environment with clients registered to a
FreeSWITCH server.  One day your instance is physically running on
hardware in Seattle.  The next day it could (potentially) be running
in Chicago.  That's obviously a very different routing path for your
clients.  Even /if/ Amazon (or whomever) employs every routing/network
trick in the book you still won't be able to get over that change in
geography.

  It's certainly possible a change like this may very well never
happen in practice.  I wouldn't know; I've never used EC2 and I don't
even know that much about it.  I'm just curious how well strictly,
practically speaking this will work in the long term.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com

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Re: [Freeswitch-users] FS in Amazon EC2 for production?

2009-05-26 Thread Brian West


On May 26, 2009, at 10:11 AM, Kristian Kielhofner wrote:


Hey Brian,

 FreeSWITCH in EC2 is a bit of a mystery to me...

 Call me old fashioned but in my mind VoIP and geography are linked
in %99 of scenarios.  Having VoIP services in a pure cloud
environment just doesn't sound like a good idea to me.

 Consider a hosted environment with clients registered to a
FreeSWITCH server.  One day your instance is physically running on
hardware in Seattle.  The next day it could (potentially) be running
in Chicago.  That's obviously a very different routing path for your
clients.  Even /if/ Amazon (or whomever) employs every routing/network
trick in the book you still won't be able to get over that change in
geography.


For some people this isn't a huge difference... now if it were to swap  
continents then yes it would be a problem.  But I haven't seen Amazon  
do this but I haven't left the instances up long enough to see.



 It's certainly possible a change like this may very well never
happen in practice.  I wouldn't know; I've never used EC2 and I don't
even know that much about it.  I'm just curious how well strictly,
practically speaking this will work in the long term.


There are other companies that do this stuff but personally me... I  
want my stuff running on real hardware.


Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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Re: [Freeswitch-users] how to disbale : switch_core_sqldb()

2009-05-26 Thread Brian West
I didn't see an error line maybe you have permission problems butyou  
just pasted only the INFO line.  You do know if you disable the sql  
you loose the ability to show calls and channels?

/b

On May 25, 2009, at 9:57 PM, mashudi wrote:

 Dear Muhammad Shahzad ,
 it is work and the err after has gone, thank you very much for your
 suggestion.


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[Freeswitch-users] how to enable ESL for ruby?

2009-05-26 Thread dujinfang
Hi,

Following the wiki: http://wiki.freeswitch.org/wiki/Event_Socket_Library
On MacOSX 10.5, I can't get ESL for ruby work. make throws error:

sevens-mac-pro:~/workspace/test/freeswitch/trunk/libs/esl$ make rubymod
make MYLIB=../libesl.a SOLINK=-shared -Xlinker -x CFLAGS=-I/Users/ 
seven/workspace/test/freeswitch/trunk/libs/esl/src/include - 
DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast- 
math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- 
prototypes -Wmissing-prototypes CXXFLAGS=-I/Users/seven/workspace/ 
test/freeswitch/trunk/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb - 
I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable  
CXX_CFLAGS= -C ruby
g++  -I/Users/seven/workspace/test/freeswitch/trunk/libs/esl/src/ 
include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall  
-Werror -Wno-unused-variable -I/opt/local/lib/ruby/1.8/i686-darwin9 -c  
esl_wrap.cpp -o esl_wrap.o
g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -lruby -o ESL.so -L.
Undefined symbols:
   _main, referenced from:
   start in crt1.10.5.o
ld: symbol(s) not found
collect2: ld returned 1 exit status
make[1]: *** [ESL.so] Error 1

Thanks for any help.

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[Freeswitch-users] IVR xml dialplan for outbound calls

2009-05-26 Thread Paul Li
There are a few examples in the wiki, showing how to configure IVR for
inbound calls. My question lies in whether it is possible to write a
dialplan in xml or scripts to configure IVR for outbound calls. Here is a
typical scenario (of usage):

(1) Call any extension or external endpoints (maybe PSTN numbers);
(2) Activate IVR menu once the recipient answers call;
(3) Take feedback (DTMF keystrokes or survey recordings).

Your assistance is highly appreciated!
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[Freeswitch-users] Unicast isn't working

2009-05-26 Thread Артем Васильев
Hi,

I am trying to setup ASR in FreeSwitch using Nuance ASR server and MRCP.
Both FreeSwitch and Nuance installed on Windows Server 2003. FreeSwitch
version is 1.0.3 (12567M)

I found an example in Perl at
http://www.softivr.com/wiki/index.php/FreeSWITCH_MRCP_in_Perl and decided to
do the same in C#.
It establishes connection with Nuance and loads the grammar, everything
works fine. The next step is to capture audio from FS and tramsmit it to
ASR. This can be done with unicast. We must create an outbound socket and
issue unicast command. Here it goes:

private void SetupAudioTransmission()
{
 EventSocket = new Socket(AddressFamily.InterNetwork, SocketType.Stream,
ProtocolType.Tcp);
 EventSocket.SendTimeout = Config.Timeout;
 int ESPort = SearchESPort(EventSocket);

 Thread thrESListener = new Thread(new ThreadStart(ListenerThreadStart));
 thrESListener.Start();

 WriteLog(LogLevel.Info, Creating outbound event socket);
 Session.Execute(socket, 127.0.0.1: + ESPort); //main thread stops,
listener thread listens for outbound socket connection.
 WriteLog(LogLevel.Info, Outbound event socket disconnected);

 EventSocket.Close();
}

//here we accept outbound socket and transmit unicast command through it
private void ListenerThreadStart()
{
 Socket sockHandler = EventSocket.Accept();
 WriteLog(LogLevel.Info, Incoming connection);
 sockHandler.Send(MessageEncoding.GetBytes(Connect\n\n));

 WriteLog(LogLevel.Info, GetServerResponse(sockHandler));

 int rtpPort = (RTPSocket.RemoteEndPoint as IPEndPoint).Port;
 string command = string.Format(sendmsg\r\ncall-command:
unicast\r\nlocal-ip: {0}\r\nlocal-port: {1}\r\nremote-ip: {2}\r\n +
  remote-port: {3}\r\ntransport: udp\r\nflags: native\r\n\r\n,
Config.LocalIP, rtpPort + 1, Config.LocalIP, rtpPort);

 WriteLog(LogLevel.Info, command);

 sockHandler.Send(MessageEncoding.GetBytes(command));
 WriteLog(LogLevel.Info, GetServerResponse(sockHandler));

 sockHandler.Disconnect(false);
 sockHandler.Close();
}

After this, FS writes that unicast has been created on corresponding IPs and
ports. It really creates an UDP socket, but doesn't transmit any data
through it. I tested it with Wireshark and from my application, nothing was
detected.
Also, if we specify transport:tcp in unicast command, it uses UDP anyway,
that's strange.

Here is how I listen UDP packets.
private void DetectSpeech()
{
 WriteLog(LogLevel.Info, Reading audio);
 byte[] FSRecvBuf = new byte[2048];

 IPEndPoint epFS = new IPEndPoint(IPAddress.Loopback,
(RTPSocket.RemoteEndPoint as IPEndPoint).Port);
 FSSocket = new Socket(AddressFamily.InterNetwork, SocketType.Dgram,
ProtocolType.Udp);
 FSSocket.ReceiveTimeout = 10;
 FSSocket.SendTimeout = 10;
 WriteLog(LogLevel.Info, Binding socket to  + epFS.Port);

 FSSocket.Bind(epFS);
 FSSocket.Connect(new IPEndPoint(IPAddress.Loopback, epFS.Port + 1));

 while (Session.Ready())
 {
  int recvCount = FSSocket.Receive(FSRecvBuf);
  WriteLog(LogLevel.Info, Received bytes:  + recvCount);
 }
}

Can someone help me to solve this? Do I do something wrong or I forgot
something or it doesn't work at all?

Best regards.
Artem
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Re: [Freeswitch-users] FS in Amazon EC2 for production?

2009-05-26 Thread Erik Wickstrom
I've been running a production FS app on EC2 since December.  It's been
really stable.  Same server/instance since day1.  We've haven't had any
complaints

Erik

On Tue, May 26, 2009 at 8:11 AM, Kristian Kielhofner 
kristian.kielhof...@gmail.com wrote:

 On Tue, May 26, 2009 at 10:31 AM, Brian West br...@freeswitch.org wrote:
  Not with FreeSWITCH in our testing.  Now if you have stupid defaults
  in your virtualization env. it might act funny but I have run FS on
  EC2 without a problem.
 
  /b

 Hey Brian,

  FreeSWITCH in EC2 is a bit of a mystery to me...

  Call me old fashioned but in my mind VoIP and geography are linked
 in %99 of scenarios.  Having VoIP services in a pure cloud
 environment just doesn't sound like a good idea to me.

  Consider a hosted environment with clients registered to a
 FreeSWITCH server.  One day your instance is physically running on
 hardware in Seattle.  The next day it could (potentially) be running
 in Chicago.  That's obviously a very different routing path for your
 clients.  Even /if/ Amazon (or whomever) employs every routing/network
 trick in the book you still won't be able to get over that change in
 geography.

  It's certainly possible a change like this may very well never
 happen in practice.  I wouldn't know; I've never used EC2 and I don't
 even know that much about it.  I'm just curious how well strictly,
 practically speaking this will work in the long term.

 --
 Kristian Kielhofner
 http://www.astlinux.org
 http://blog.krisk.org
 http://www.star2star.com
 http://www.submityoursip.com
 http://www.voalte.com

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Re: [Freeswitch-users] how to enable ESL for ruby?

2009-05-26 Thread dujinfang
Thanks Brain. Got ESL.so, however on my Mac it is #include ruby.h  
instead of ruby/ruby.h.


But it can't find the ESL when I require 'ESL' in ruby. Even I put  
ESL.so in one of the dir of $:


Any clue for me?

se...@du-sevens-mac-pro:~/workspace/test/freeswitch/oldtrunk/libs/esl$  
make rubymod
cc  -I/Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/ 
include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 - 
ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- 
prototypes -Wmissing-prototypes -c src/esl.c -o src/esl.o
cc  -I/Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/ 
include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 - 
ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- 
prototypes -Wmissing-prototypes -c src/esl_event.c -o src/esl_event.o
cc  -I/Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/ 
include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 - 
ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- 
prototypes -Wmissing-prototypes -c src/esl_threadmutex.c -o src/ 
esl_threadmutex.o
cc  -I/Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/ 
include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 - 
ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- 
prototypes -Wmissing-prototypes -c src/esl_config.c -o src/esl_config.o
g++  -I/Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/ 
include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall  
-Werror -Wno-unused-variable -c src/esl_oop.cpp -o src/esl_oop.o
ar rcs libesl.a src/esl.o src/esl_event.o src/esl_threadmutex.o src/ 
esl_config.o src/esl_oop.o

ranlib libesl.a
make MYLIB=../libesl.a SOLINK=-dynamiclib -Xlinker -x CFLAGS=-I/ 
Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/include - 
DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast- 
math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- 
prototypes -Wmissing-prototypes CXXFLAGS=-I/Users/seven/workspace/ 
test/freeswitch/oldtrunk/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb  
-I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable  
CXX_CFLAGS= -C ruby
g++  -I/Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/ 
include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall  
-Werror -Wno-unused-variable -I/opt/local/lib/ruby/1.8/i686-darwin9 -c  
esl_wrap.cpp -o esl_wrap.o
g++ -dynamiclib -Xlinker -x esl_wrap.o ../libesl.a -lruby -framework  
Ruby -o ESL.so -L.
se...@du-sevens-mac-pro:~/workspace/test/freeswitch/oldtrunk/libs/esl$  
cd ruby/
se...@du-sevens-mac-pro:~/workspace/test/freeswitch/oldtrunk/libs/esl/ 
ruby$ ruby single_command.rb
single_command.rb:3:in `require': no such file to load -- ESL  
(LoadError)

from single_command.rb:3
se...@du-sevens-mac-pro:~/workspace/test/freeswitch/oldtrunk/libs/esl/ 
ruby$ irb

irb(main):001:0 $:
= [/opt/local/lib/ruby/site_ruby/1.8, /opt/local/lib/ruby/ 
site_ruby/1.8/i686-darwin9, /opt/local/lib/ruby/site_ruby, /opt/ 
local/lib/ruby/vendor_ruby/1.8, /opt/local/lib/ruby/vendor_ruby/1.8/ 
i686-darwin9, /opt/local/lib/ruby/vendor_ruby, /opt/local/lib/ruby/ 
1.8, /opt/local/lib/ruby/1.8/i686-darwin9, .]

irb(main):002:0
On May 26, 2009, at 11:27 PM, Brian West wrote:

The makefile will have to be changed to work with OS X since the  
linking is done differently.


It would be very similar to this one http://www.bkw.org/esl.imac.diff

Below will get it to compile:

imac:esl brian$ svn diff
Index: ruby/Makefile
===
--- ruby/Makefile   (revision 13432)
+++ ruby/Makefile   (working copy)
@@ -1,6 +1,7 @@
 ## no idea how to simply ask ruby which one to use
 LOCAL_CFLAGS=-I$(shell ruby -e 'require rbconfig; puts  
RbConfig::CONFIG[topdir]')
 LOCAL_LDFLAGS=$(shell ruby -e 'require rbconfig; puts  
RbConfig::CONFIG[LIBRUBYARG]')

+LOCAL_LDFLAGS += -framework Ruby

 all: ESL.so

Index: ruby/esl_wrap.cpp
===
--- ruby/esl_wrap.cpp   (revision 13432)
+++ ruby/esl_wrap.cpp   (working copy)
@@ -823,7 +823,7 @@



-#include ruby.h
+#include ruby/ruby.h

 /* Remove global macros defined in Ruby's win32.h */
 #ifdef write
Index: Makefile
===
--- Makefile(revision 13432)
+++ Makefile(working copy)
@@ -12,7 +12,7 @@
 OBJS=src/esl.o src/esl_event.o src/esl_threadmutex.o src/esl_config.o
 SRC=src/esl.c src/esl_event.c src/esl_threadmutex.c src/ 
esl_config.c src/esl_oop.cpp
 HEADERS=src/include/esl_config.h src/include/esl_event.h src/ 
include/esl.h src/include/esl_threadmutex.h src/include/esl_oop.h

-SOLINK=-shared -Xlinker -x
+SOLINK=-dynamiclib -Xlinker -x
 # comment the next line to disable c++ (no swig mods for you then)
 OBJS += src/esl_oop.o



/b



On May 26, 2009, at 10:22 AM, 

[Freeswitch-users] Conference users hear MOH until leader enters?

2009-05-26 Thread Jason Cromes
First off, I apologize if this has been sent multiple times, the mailing 
list won't cooperate with me...  Hopefully that is resolved now.

I'm attempting to replicate the behavior of an Asterisk conferencing 
system and I need a feature that, I'm surprised to say, doesn't seem to 
be supported (easily) in FreeSwitch.

I've setup my dialplan so that when a specific extension is hit, it 
calls out to some javascript which acts like an IVR to handle the 
prompts and conferencing setup.  (Similar to this: 
 http://wiki.freeswitch.org/wiki/Examples_confcall_js but with my own 
improvements.)  Anyway, the conferences are stored permanently in a 
database table, but I need them disabled until the conference owner 
dials in and enables his conference.  If other users have entered the 
conference prior to the owner, they should hear music-on-hold until the 
leader enters.  (The purpose of this is to prevent abuse of conferences 
that are, basically, always available.)

This is easy in Asterisk because you can pop someone into MeetMe with 
different flags.  So, in my IVR, I prompt for the conference number 
(known to all) and then the password (known only to the owner/leader). 
 If the proper password is entered, that user is sent to conference XYZ 
with the leader flag set.  If no password is entered, the user goes to 
conference XYZ, without the leader flag.  If anyone enters before the 
leader, they're told by MeetMe that the conference will begin when the 
leader arrives and MeetMe provides MOH until that time.

Help!  This is an absolute deal-breaker for my install...  How can I do 
this in FreeSwitch?  
Thanks...

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[Freeswitch-users] Conference users hear MOH until leader enters?

2009-05-26 Thread j3flight

I'm attempting to replicate the behavior of an Asterisk system with
FreeSwitch and I need a feature that, I'm surprised to say, doesn't seem to
be supported (easily).

Ok, so I've setup my dialplan so that when a specific extension is hit, it
calls out to some javascript which acts like an IVR to handle the conference
setup.  (Similar to this: 
http://wiki.freeswitch.org/wiki/Examples_confcall_js but with my own
improvements.)  Anyway, the conferences are stored permanently in a
database, but I want them protected by their owner so they can only be
used when that conference owner dials.  If other users have entered the
conference prior to the owner, they should hear music-on-hold until the
leader enters.

This is easy in Asterisk because you can pop someone into MeetMe with
different flags.  So, in my IVR, I prompt for the conference number (known
to all) and then the password (known only to the owner/leader).  If the
proper password is entered, the user is sent to conference XYZ with the
leader flag set.  If no password is entered, the user goes to conference
XYZ, without the leader flag.  If anyone enters before the leader, they're
told by MeetMe that the conference will begin when the leader arrives and
MeetMe provides MOH until that time.

Help!  This is an absolute deal-breaker for my install...  How can I do this
in FreeSwitch?  
Thanks...
-- 
View this message in context: 
http://www.nabble.com/Conference-users-hear-MOH-until-leader-enters--tp23715721p23715721.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


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Re: [Freeswitch-users] Secure RTP

2009-05-26 Thread Brian West
I know this but you don't have to have it in x-lite or eyebeam  
directly.  You just need the zfone application along with Eyebeam or X- 
Lite right now.

/b

On May 26, 2009, at 12:52 AM, Jim Burke wrote:


 FYI...According to Counterpath ZRTP is not added to the retail
 versions of Eyebeam or X-Lite, you need to purchase a bulk order.


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Re: [Freeswitch-users] FS in Amazon EC2 for production?

2009-05-26 Thread Brian West
Not with FreeSWITCH in our testing.  Now if you have stupid defaults  
in your virtualization env. it might act funny but I have run FS on  
EC2 without a problem.

/b

On May 26, 2009, at 1:05 AM, John Nicholson wrote:

 Virtualization has issues with timing in my experiance.

 Sent from my iPhone


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Re: [Freeswitch-users] how to enable ESL for ruby?

2009-05-26 Thread Brian West


On May 26, 2009, at 11:20 AM, dujinfang wrote:

Thanks Brain. Got ESL.so, however on my Mac it is #include ruby.h  
instead of ruby/ruby.h.


Actually since we do -framework Ruby it should be ruby/ruby but I  
think the line above the -framework Ruby should be removed since  
you're doing i tthe Mac way.


/b




But it can't find the ESL when I require 'ESL' in ruby. Even I put  
ESL.so in one of the dir of $:


Any clue for me?

se...@du-sevens-mac-pro:~/workspace/test/freeswitch/oldtrunk/libs/esl 
$ make rubymod
cc  -I/Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/ 
include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2  
-ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- 
prototypes -Wmissing-prototypes -c src/esl.c -o src/esl.o
cc  -I/Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/ 
include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2  
-ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- 
prototypes -Wmissing-prototypes -c src/esl_event.c -o src/esl_event.o
cc  -I/Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/ 
include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2  
-ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- 
prototypes -Wmissing-prototypes -c src/esl_threadmutex.c -o src/ 
esl_threadmutex.o
cc  -I/Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/ 
include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2  
-ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- 
prototypes -Wmissing-prototypes -c src/esl_config.c -o src/ 
esl_config.o
g++  -I/Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/ 
include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC - 
Wall -Werror -Wno-unused-variable -c src/esl_oop.cpp -o src/esl_oop.o
ar rcs libesl.a src/esl.o src/esl_event.o src/esl_threadmutex.o src/ 
esl_config.o src/esl_oop.o

ranlib libesl.a
make MYLIB=../libesl.a SOLINK=-dynamiclib -Xlinker -x CFLAGS=-I/ 
Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/include - 
DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast- 
math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- 
prototypes -Wmissing-prototypes CXXFLAGS=-I/Users/seven/workspace/ 
test/freeswitch/oldtrunk/libs/esl/src/include -DHAVE_EDITLINE -g - 
ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused- 
variable CXX_CFLAGS= -C ruby
g++  -I/Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/ 
include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC - 
Wall -Werror -Wno-unused-variable -I/opt/local/lib/ruby/1.8/i686- 
darwin9 -c esl_wrap.cpp -o esl_wrap.o
g++ -dynamiclib -Xlinker -x esl_wrap.o ../libesl.a -lruby -framework  
Ruby -o ESL.so -L.
se...@du-sevens-mac-pro:~/workspace/test/freeswitch/oldtrunk/libs/esl 
$ cd ruby/
se...@du-sevens-mac-pro:~/workspace/test/freeswitch/oldtrunk/libs/ 
esl/ruby$ ruby single_command.rb
single_command.rb:3:in `require': no such file to load -- ESL  
(LoadError)

from single_command.rb:3
se...@du-sevens-mac-pro:~/workspace/test/freeswitch/oldtrunk/libs/ 
esl/ruby$ irb

irb(main):001:0 $:
= [/opt/local/lib/ruby/site_ruby/1.8, /opt/local/lib/ruby/ 
site_ruby/1.8/i686-darwin9, /opt/local/lib/ruby/site_ruby, /opt/ 
local/lib/ruby/vendor_ruby/1.8, /opt/local/lib/ruby/vendor_ruby/ 
1.8/i686-darwin9, /opt/local/lib/ruby/vendor_ruby, /opt/local/ 
lib/ruby/1.8, /opt/local/lib/ruby/1.8/i686-darwin9, .]

irb(main):002:0


Brian West
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Re: [Freeswitch-users] FS in Amazon EC2 for production?

2009-05-26 Thread Gregory Boehnlein
I can say, from having met with and talked to the CEO and founder of
Applogic that these guys are really revolutionary in their approach to cloud
computing. I spoke on a panel w/ the founder at an ISPcon several years ago,
and their approach is that of a utility company, treating computing
resources like that of a  power company. Cool stuff!

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Chris
Danielson
Sent: Tuesday, May 26, 2009 11:39 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS in Amazon EC2 for production?

 

There is a cloud computing company named 3Tera (AppLogic) that does have an
international presence and will keep your FreeSWITCH instance running on a
dedicated server using Xen and HA.  I spoke with one of their senior
engineers about 1 month ago in regards to actually setting up an LCR
scenario using their servers located in Europe and the United States.  These
guys are a little different in the cloud computing world and I believe
closer fit the needs of a telephony application.  As-is, there are companies
using 3tera for their Asterisk installs.  So if you want cloud computing
with dedicated hardware resources and a set geographic location, then these
guys do it.  Kind of the best of both worlds.  Just a quick 2 cents...

Regards,
Chris


Brian West wrote: 

 

On May 26, 2009, at 10:11 AM, Kristian Kielhofner wrote:





Hey Brian,

 FreeSWITCH in EC2 is a bit of a mystery to me...

 Call me old fashioned but in my mind VoIP and geography are linked
in %99 of scenarios.  Having VoIP services in a pure cloud
environment just doesn't sound like a good idea to me.

 Consider a hosted environment with clients registered to a
FreeSWITCH server.  One day your instance is physically running on
hardware in Seattle.  The next day it could (potentially) be running
in Chicago.  That's obviously a very different routing path for your
clients.  Even /if/ Amazon (or whomever) employs every routing/network
trick in the book you still won't be able to get over that change in
geography.



 

For some people this isn't a huge difference... now if it were to swap
continents then yes it would be a problem.  But I haven't seen Amazon do
this but I haven't left the instances up long enough to see.

 

 It's certainly possible a change like this may very well never
happen in practice.  I wouldn't know; I've never used EC2 and I don't
even know that much about it.  I'm just curious how well strictly,
practically speaking this will work in the long term.



 

There are other companies that do this stuff but personally me... I want my
stuff running on real hardware.

 

Brian West

br...@freeswitch.org

 

-- Meet us at ClueCon!  http://www.cluecon.com http://www.cluecon.com/ 

 

 

 





 



  _  



 
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Re: [Freeswitch-users] Help understanding call_timeout in two dialplans

2009-05-26 Thread Brian West

last one set will win!

/b

On May 26, 2009, at 11:49 AM, Larry Marshall wrote:

On inbound calls made to 5551212, which call_timeout will be active,  
15 or 20? Is it the last hit?


Is there a URL which describes all the applications, for example,  
export in the default.xml?


Thanks for this amazing software, Lars

In conf/dialplan/public/00_inbound.xml:
include
  extension name=Inbound-5551212
 condition field=destination_number  
expression=^1323(5551212)$

action application=ring_ready /
action application=set data=call_timeout=15/
action application=set data=domain_name=$${domain}/
action application=transfer data=1000 XML default/
 /condition
  /extension
/include


Brian West
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Re: [Freeswitch-users] 407 Proxy Authentication Required

2009-05-26 Thread Brian West
Looks like his outbound call is failing now.

/b

On May 26, 2009, at 6:07 AM, Jason White wrote:

 Your SIP trace might give you a clue as to what happened.
 sofia profile external siptrace on
 (substituting the relevant profile for external in the above  
 command, as
 required, and repeating the command if more than one profile is  
 involved).


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Re: [Freeswitch-users] Sounds order

2009-05-26 Thread François Delawarde
Hi Brian,

Is FreeSWITCH going to have Spanish/French/... sounds as well or do
those need to be?

Thanks,
François.

On Tue, 2009-05-26 at 11:36 -0500, Brian West wrote:

 I'm getting ready for the next sound file order for FreeSWITCH.  I
 have a rather large set of files to be recorded for the zRTP
 integration if anyone wants to help out.  ;)  Please contact me off
 list.  I would like everyone to update and try out voicemail and
 nitpick anything that you feel is wrong there too and let me know so I
 can have them corrected in this order also.
 
 
 
 Thanks,
 
 
 Brian West
 br...@freeswitch.org
 
 
 
 -- Meet us at ClueCon!  http://www.cluecon.com
 
 
 
 
 
 
 
 
 
 
 
 
 
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Re: [Freeswitch-users] Sounds order

2009-05-26 Thread Brian West
At this time we only have english it will take $1200-$2000 to record  
each language.


/b

On May 26, 2009, at 12:06 PM, François Delawarde wrote:


Hi Brian,

Is FreeSWITCH going to have Spanish/French/... sounds as well or do  
those need to be?


Thanks,
François.


Brian West
br...@freeswitch.org

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[Freeswitch-users] NOTIFY Messages

2009-05-26 Thread Andrey Nepomnyaschih
Hello everyone,

Does anyone know is it possible to disable NOTIFY messages coming out of 
FreeSwitch for particular endpoint? The reason I'm asking it is because I have 
a gateway (D-Link DVG-7111S) that reboots when it receives such a packet.

Kind Regards,
Andrey Nepomnyaschih
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Re: [Freeswitch-users] NOTIFY Messages

2009-05-26 Thread Brian West

You can set this param on a profile

param name=send-message-query-on-register value=false/

This should take care of it but really that gateway is broken if it  
reboots on a MWI notify.


/b


On May 26, 2009, at 12:19 PM, Andrey Nepomnyaschih wrote:


Hello everyone,

Does anyone know is it possible to disable NOTIFY messages coming  
out of FreeSwitch for particular endpoint? The reason I’m asking it  
is because I have a gateway (D-Link DVG-7111S) that reboots when it  
receives such a packet.


Kind Regards,
Andrey Nepomnyaschih
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[Freeswitch-users] STUN and rport on Polycom phones

2009-05-26 Thread Brian West
It has come to my attention that Polycom hasn't had a business case to  
support rport and STUN.  If you can please kindly email marek.dutkiew...@polycom.com 
 and let him know you would like to see STUN and rport support in the  
polycom products.  Its really one of the last things missing in the  
phone to make it easy to deploy in a NAT env.


Thanks,

Brian West
br...@freeswitch.org

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Re: [Freeswitch-users] NOTIFY Messages

2009-05-26 Thread Andrey Nepomnyaschih
Well, that didn't help.

I've tried to put it into endpoint (conf/directory/default/*.xml) configuration 
or profile (conf/sip_profiles/internal.xml) without a success.

Is there any other option?

From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West
Sent: Tuesday, May 26, 2009 9:28 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] NOTIFY Messages

You can set this param on a profile

param name=send-message-query-on-register value=false/

This should take care of it but really that gateway is broken if it reboots on 
a MWI notify.

/b


On May 26, 2009, at 12:19 PM, Andrey Nepomnyaschih wrote:


Hello everyone,

Does anyone know is it possible to disable NOTIFY messages coming out of 
FreeSwitch for particular endpoint? The reason I'm asking it is because I have 
a gateway (D-Link DVG-7111S) that reboots when it receives such a packet.

Kind Regards,
Andrey Nepomnyaschih
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Re: [Freeswitch-users] How to work with the debug logfile?

2009-05-26 Thread Michael Collins
On Mon, May 25, 2009 at 8:19 AM, Dennis oderm...@googlemail.com wrote:

 hi,

 we encounter some small problems withing the past 2 days and we are
 trying to find out more about the problems. for this we downloaded the
 debug logfiles written by fs, but we do not manage to filter all
 log-entries for one single special call.

 we are using socket outbound and would like to see all entries
 (inbound/outbound) of one call.

 is this possible?

 kind regards
 dennis


This isn't the easiest thing to do but it is possible. In the past I've used
a combination of uuid and channel name with grep. Also, I recommend rotating
log files using fsctl send_sighup which helps keep you from getting a 10GB
freeswitch.log file. :) The first thing I do is find the call's uuid, either
by manually searching through the log file or by using a CDR. (Use whichever
works best for you.) Then use grep to find all the line numbers for that
particular uuid. Here's a sample from my Mac where I used portaudio to call
the main conference line:

grep -n b1f9e5ce-29a6-497a-a628-71f1bdefa0cb freeswitch.log

output snipped

In this case I found out my channel name is portaudio/
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org so
I grep for it:

grep 
portaudio/sip:8...@conference.freeswitch.orgsip%3a...@conference.freeswitch.org
freeswitch.log

output snipped

It's not perfect but it helps narrow down your call.

-MC
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Re: [Freeswitch-users] IVR xml dialplan for outbound calls

2009-05-26 Thread Michael Collins
This is totally possible. You need to look at the originate command. Also,
if you have an IVR or a dialplan extension that does what you want - plays
prompts, accepts DTMF digits from caller, etc. - then you can use it with
inbound or outbound calls. For an outbound call just route it to the dp
extension. For example to route an outbound call to the demo IVR do this:

originate sofia/gateway/mygatewayname/18005551212 5000

Have fun.
-MC

On Mon, May 25, 2009 at 10:03 PM, Paul Li plite2...@gmail.com wrote:

 There are a few examples in the wiki, showing how to configure IVR for
 inbound calls. My question lies in whether it is possible to write a
 dialplan in xml or scripts to configure IVR for outbound calls. Here is a
 typical scenario (of usage):

 (1) Call any extension or external endpoints (maybe PSTN numbers);
 (2) Activate IVR menu once the recipient answers call;
 (3) Take feedback (DTMF keystrokes or survey recordings).

 Your assistance is highly appreciated!

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Re: [Freeswitch-users] Sounds order

2009-05-26 Thread Michael Collins
On Tue, May 26, 2009 at 10:06 AM, François Delawarde 
fdelawa...@wirelessmundi.com wrote:

  Hi Brian,

 Is FreeSWITCH going to have Spanish/French/... sounds as well or do those
 need to be?


We have a volunteer who is going to record a set of Spanish prompts. These
would probably be classified as Spanish Latin-America as opposed to
Castillian. We could use a volunteer for French prompts. Or if there is a
very generous benefactor who is willing to donate we could have GM Voices do
high-quality French prompts.

-MC




 Thanks,
 François.

 On Tue, 2009-05-26 at 11:36 -0500, Brian West wrote:

 I'm getting ready for the next sound file order for FreeSWITCH.  I have a
 rather large set of files to be recorded for the zRTP integration if anyone
 wants to help out.  ;)  Please contact me off list.  I would like everyone
 to update and try out voicemail and nitpick anything that you feel is wrong
 there too and let me know so I can have them corrected in this order also.



  Thanks,


  Brian West

  br...@freeswitch.org



  -- Meet us at ClueCon!  http://www.cluecon.com









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[Freeswitch-users] Pre8 Release on Digg

2009-05-26 Thread Brian West

Dear FreeSWITCHers,
	Now I'm gonna take a moment here to guilt each and everyone of you  
into checking out the story about Pre8 on Digg.  We have all worked  
long and hard to get to 1.0.4 and we still have a little bit to go.   
So everyone out there that asks What can I do to help the project?,  
this is your chance to do so.  Help us to promote the project, which  
brings more people to help in supporting the community, the project  
and you the end user.


Also looking for people to help manage jira, test bugs, ask the right  
questions and line up the bugs so we can knock them out! Please email  
me if you're interested.


Here is the link for you to help out http://digg.com/search?s=FreeSWITCH+Pre8

Brian West
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Re: [Freeswitch-users] Pre8 Release on Digg

2009-05-26 Thread Diego Viola
Digged.

On Tue, May 26, 2009 at 3:47 PM, Brian West br...@freeswitch.org wrote:
 Dear FreeSWITCHers,
 Now I'm gonna take a moment here to guilt each and everyone of you into
 checking out the story about Pre8 on Digg.  We have all worked long and hard
 to get to 1.0.4 and we still have a little bit to go.  So everyone out there
 that asks What can I do to help the project?, this is your chance to do
 so.  Help us to promote the project, which brings more people to help in
 supporting the community, the project and you the end user.
 Also looking for people to help manage jira, test bugs, ask the right
 questions and line up the bugs so we can knock them out! Please email me if
 you're interested.
 Here is the link for you to help
 out http://digg.com/search?s=FreeSWITCH+Pre8
 Brian West
 br...@freeswitch.org
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[Freeswitch-users] Double-dtmf detection in IVR when a call is routed through FreeSWITCH

2009-05-26 Thread Drew Ozier
I've got a configuration where I receive inbound calls and dial out to a
pre-determined 800-number based on the DNIS of the call. I set action
application=start_dtmf/ and have everything set up so that DTMF only
comes to me via inband. When I'm providing DTMF data to the IVR, it will
recognize a single keypress as a double-tap. My FreeSWITCH logs only contain
one DTMF entry, but when I listen to the receiving end of the call, I can
hear a hiccup in the DTMF tone that is getting played. When I do not use
'start_dtmf', this problem goes away. I need inband DTMF detection, but I
can't have it messing up the audio stream. Any thoughts?

-Drew Ozier
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Re: [Freeswitch-users] Pre8 Release on Digg

2009-05-26 Thread Even André Fiskvik
Diggedy dug!

On 26. mai. 2009, at 22.03, Diego Viola wrote:

 Digged.

 On Tue, May 26, 2009 at 3:47 PM, Brian West br...@freeswitch.org  
 wrote:
 Dear FreeSWITCHers,
 Now I'm gonna take a moment here to guilt each and everyone of you  
 into
 checking out the story about Pre8 on Digg.  We have all worked long  
 and hard
 to get to 1.0.4 and we still have a little bit to go.  So everyone  
 out there
 that asks What can I do to help the project?, this is your chance  
 to do
 so.  Help us to promote the project, which brings more people to  
 help in
 supporting the community, the project and you the end user.
 Also looking for people to help manage jira, test bugs, ask the right
 questions and line up the bugs so we can knock them out! Please  
 email me if
 you're interested.
 Here is the link for you to help
 out http://digg.com/search?s=FreeSWITCH+Pre8
 Brian West
 br...@freeswitch.org
 -- Meet us at ClueCon!  http://www.cluecon.com





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Re: [Freeswitch-users] Pre8 Release on Digg

2009-05-26 Thread Even André Fiskvik

...or be followed with 10 yrs of bad luck and hardware failures!

On 26. mai. 2009, at 22.42, Brian West wrote:


Thank you... Now please tell 10 of your friends about FreeSWITCH ;)

/b

On May 26, 2009, at 3:40 PM, Even André Fiskvik wrote:


Diggedy dug!

On 26. mai. 2009, at 22.03, Diego Viola wrote:


Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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Re: [Freeswitch-users] Pre8 Release on Digg

2009-05-26 Thread Brian West
Remember Save the cheerlead, save the world?  In this case Digg the  
story, get your bugs fixed?


/b

On May 26, 2009, at 4:11 PM, Even André Fiskvik wrote:


...or be followed with 10 yrs of bad luck and hardware failures!

On 26. mai. 2009, at 22.42, Brian West wrote:


Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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[Freeswitch-users] PBX on a (Sheeva|Pogo)Plug?

2009-05-26 Thread Fred-145

Hello

I was wondering if someone had successfully ran a PBX on those tiny, $100
devices that run Linux?

www.pogoplug.com
www.plugcomputer.org

I'm thinking of hooking up a Sangoma U100
(http://wiki.sangoma.com/sangoma-wanpipe-usbfxo) to the USB port so it can
handle a couple of analog lines. And installing a compact LAMP server so the
user can manage the PBX server from a web browser.

What do you think?

Thank you.
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Re: [Freeswitch-users] Double-dtmf detection in IVR when a call isrouted through FreeSWITCH

2009-05-26 Thread David Knell
 Hi Drew,

When you say that the problem goes away if you don't use start_dtmf, do you 
mean that you get one tone recognised per tone or none?  If the former, then 
you've got DTMF being signalled out of band as well; in that case, why do you 
need inband detection?

--Dave
  - Original Message - 
  From: Drew Ozier 
  To: freeswitch-users@lists.freeswitch.org 
  Sent: Tuesday, May 26, 2009 9:36 PM
  Subject: [Freeswitch-users] Double-dtmf detection in IVR when a call isrouted 
through FreeSWITCH


  I've got a configuration where I receive inbound calls and dial out to a 
pre-determined 800-number based on the DNIS of the call. I set action 
application=start_dtmf/ and have everything set up so that DTMF only comes 
to me via inband. When I'm providing DTMF data to the IVR, it will recognize a 
single keypress as a double-tap. My FreeSWITCH logs only contain one DTMF 
entry, but when I listen to the receiving end of the call, I can hear a hiccup 
in the DTMF tone that is getting played. When I do not use 'start_dtmf', this 
problem goes away. I need inband DTMF detection, but I can't have it messing up 
the audio stream. Any thoughts?

  -Drew Ozier



--


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Re: [Freeswitch-users] Conference users hear MOH until leader enters?

2009-05-26 Thread Anthony Minessale
the easiest way would be the new feature I added to 13442

in the conference profile add

param name=conference-flags value=wait-mod/

to your profile

and in your dialplan

action application=set data=conference_member_flags=moderator/
action application=conference data=1...@wideband/

or

action application=conference data=1...@wideband+flags{moderator}/


Don't forget the wishlist and donate button on the main site




On Tue, May 26, 2009 at 10:46 AM, jcro...@gmail.com wrote:

  Ok, that sounds doable...  I have no problem banging around with some
 code.  Thanks for the advice...
 I'm new to FIFOs and FreeSwitch in general, so please check my logic
 here...

 - When a user enters their conference number, I check to see if that
 conference exists (because the conference shouldn't exist until the leader
 enters).
 - If that conference exists, the user gets dropped straight into it.
 - If that conference does not exist, I park that user into a FIFO, with
 music.

 - When the leader enters their conference number and password, I put them
 into that conference (which creates the conference).
 - Then check for the presence of a FIFO with parked users.
 - If the FIFO contains parked users, I transfer everyone from the FIFO into
 the conference.

 Is that what you were thinking?



 Brian West wrote:

 Jason,
   There are many ways to accomplish this using FreeSWITCH.  All of
 which will require you to do a little bit of coding in js, lua or some
 other language.

 1. Park all callers into a fifo.. (see mod_fifo)
 2. When leader auths in your script then you uuid_transfer them all
 into the conference.

 You could also do this via esl using perl, lua, python, php or ruby.

 That should accomplish the same ending result.

 /b



 On May 26, 2009, at 9:38 AM, Jason Cromes wrote:



  This is easy in Asterisk because you can pop someone into MeetMe with
 different flags.  So, in my IVR, I prompt for the conference number
 (known to all) and then the password (known only to the owner/
 leader).
 If the proper password is entered, that user is sent to conference XYZ
 with the leader flag set.  If no password is entered, the user goes to
 conference XYZ, without the leader flag.  If anyone enters before the
 leader, they're told by MeetMe that the conference will begin when
 the
 leader arrives and MeetMe provides MOH until that time.

 Help!  This is an absolute deal-breaker for my install...  How can I
 do
 this in FreeSwitch?
 Thanks...


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-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
pstn:213-799-1400
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[Freeswitch-users] Questions on build 13441

2009-05-26 Thread Lars Zeb
I followed Michael Collin's instruction in an earlier email about building
pre8. During the ./bootstrap.h, I encountered the following error:

 

Entering directory /usr/src/freeswitch.trunk/libs/libsndfile

Creating aclocal.m4

Running libtoolize...

Putting files in AC_CONFIG_AUX_DIR, `Cfg'.

Creating configure

configure.ac:278: error: possibly undefined macro: AC_OCTAVE_BUILD

  If this token and others are legitimate, please use m4_pattern_allow.

  See the Autoconf documentation.

configure.ac:478: error: possibly undefined macro: AC_GCC_VERSION

 

Do I need to worry about this?

 

Also, in general, do the XML files in config get updated during the 'make
install', or are they left as they were from the previous builds?

 

Thanks, Lars

 

 

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Re: [Freeswitch-users] Questions on build 13441

2009-05-26 Thread Brian West
First off pre8 came prebootstrapped no need to run it.  So please try  
again.


/b

On May 26, 2009, at 7:03 PM, Lars Zeb wrote:

I followed Michael Collin’s instruction in an earlier email about  
building pre8. During the ./bootstrap.h, I encountered the following  
error:


Entering directory /usr/src/freeswitch.trunk/libs/libsndfile
Creating aclocal.m4
Running libtoolize...
Putting files in AC_CONFIG_AUX_DIR, `Cfg'.
Creating configure
configure.ac:278: error: possibly undefined macro: AC_OCTAVE_BUILD
  If this token and others are legitimate, please use  
m4_pattern_allow.

  See the Autoconf documentation.
configure.ac:478: error: possibly undefined macro: AC_GCC_VERSION

Do I need to worry about this?

Also, in general, do the XML files in config get updated during the  
‘make install’, or are they left as they were from the previous  
builds?


Thanks, Lars


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Re: [Freeswitch-users] Conference users hear MOH until leader enters?

2009-05-26 Thread Michael Collins
On Tue, May 26, 2009 at 4:56 PM, Anthony Minessale 
anthony.miness...@gmail.com wrote:

 the easiest way would be the new feature I added to 13442

 in the conference profile add

 param name=conference-flags value=wait-mod/

 to your profile

 and in your dialplan

 action application=set data=conference_member_flags=moderator/
 action application=conference data=1...@wideband/

 or

 action application=conference data=1...@wideband+flags{moderator}/


 Don't forget the wishlist and donate button on the main site


And the wiki tax if you feel comfortable adding this to the wiki. If not let
me know and I'll add it.
-MC
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Re: [Freeswitch-users] Questions on build 13441

2009-05-26 Thread Michael Collins
 Also, in general, do the XML files in config get updated during the ‘make
 install’, or are they left as they were from the previous builds?


Running make install or make samples will not overwrite your existing
configuration files. NOTE: This means that when the default configuration
changes you will need to update your files manually and integrate the new
changes. This is one reason why I recommend making as few edits as possible
to the default configuration files.
-MC
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Re: [Freeswitch-users] mod_nibblebill question

2009-05-26 Thread Diego Viola
Hi,

Darren just added this today, in case if someone is interested.

http://wiki.freeswitch.org/wiki/Mod_nibblebill#Hangup_the_call_when_cash_is_depleted

Thanks Darren :).

Regards,

Diego

On Sun, May 10, 2009 at 3:06 PM, Diego Viola diego.vi...@gmail.com wrote:
 Darren Schreiber to me
 That won't work. The code isn't written yet. Give mea week or two to
 finish that.


 On Fri, May 8, 2009 at 3:08 PM, Diego Viola diego.vi...@gmail.com wrote:
 I have set these actions:

 param name=nobal_amt value=0/
 param name=nobal_action value=hangup/

 But when it reaches 0 cash it doesn't hangup :(.

 On Fri, May 8, 2009 at 2:34 PM, Diego Viola diego.vi...@gmail.com wrote:
 Oh I see that it has it already :D

 !--
  By default, warn a caller when their balance is at $5.00. You can set
 this to a negative number.
 --
 param name=lowbal_amt value=5/
 param name=lowbal_action value=play ding/
 -
 !--
  By default, terminate a caller when their balance hits $0.00. You can
 set this to a negative number.
 --
 param name=nobal_amt value=0/
 param name=nobal_action value=hangup/

 Diego

 On Fri, May 8, 2009 at 1:20 PM, Diego Viola diego.vi...@gmail.com wrote:
 Hi everyone,

 I'm currently developing a calling card application that uses event
 socket and mod_nibblebill to bill calls. Well, the question is: can
 mod_nibblebill disconnect a call when the balance is depleted, or when
 it reaches 0 cash?

 The wiki says:

 Allow for disconnecting or re-routing calls when balance is depleted.

 But then at the bottom in future goals it says:

 We don't yet warn or terminate calls when they've reached their limits.

 So I'm confused, I also don't see any options in how to set that. In
 case that mod_nibblebill doesn't support this yet, how hard would it
 be to add this? I'm willing to do it, I guess it's a variable on the
 XML config and then read that from the mod_nibblebill.c, and do some
 logic there.

 Just wondering if anyone had some experience with this, and if the
 feature is already there.

 Thanks,

 Diego





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Re: [Freeswitch-users] Pre8 Release on Digg

2009-05-26 Thread Jason White
Brian West br...@freeswitch.org wrote:
 Thank you... Now please tell 10 of your friends about FreeSWITCH ;)

Also, if you're a member of a Linux user's group or similar organization, now
might be a good opportunity to raise FreeSWITCH awareness on their mailing
list or at a meeting.


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Re: [Freeswitch-users] Conference users hear MOH until leader enters?

2009-05-26 Thread Stephen Crosby
Michael,

Pardon me for hopping on this thread, but can you explain more about this
new feature? I've been wanting something like this to apply different
behaviors for different conference members. Can this be used to provide a
'moderator' with different behaviors bound to DTMF keys than regular
callers?

--Stephen

On Tue, May 26, 2009 at 5:55 PM, Michael Collins m...@freeswitch.org wrote:



 On Tue, May 26, 2009 at 4:56 PM, Anthony Minessale 
 anthony.miness...@gmail.com wrote:

 the easiest way would be the new feature I added to 13442

 in the conference profile add

 param name=conference-flags value=wait-mod/

 to your profile

 and in your dialplan

 action application=set data=conference_member_flags=moderator/
 action application=conference data=1...@wideband/

 or

 action application=conference data=1...@wideband+flags{moderator}/


 Don't forget the wishlist and donate button on the main site


 And the wiki tax if you feel comfortable adding this to the wiki. If not
 let me know and I'll add it.
 -MC



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[Freeswitch-users] XML config error

2009-05-26 Thread Diego Viola
Hi, I have downloaded the latest freeswitch trunk, and when I do
reloadxml I get this.

Error [unterminated ${var}] in line
/usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xml
line 12

Any ideas? I haven't edited that file myself.

Thanks,

Diego

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Re: [Freeswitch-users] Questions on build 13441

2009-05-26 Thread Lars Zeb
So your saying instead of:

 

cd /usr/src/freeswitch.trunk
make clean
svn up
../bootstrap.sh
../configure
make install



do:

 

cd /usr/src/freeswitch.trunk
make clean
svn up
../configure
make install



?

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, May 26, 2009 5:18 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Questions on build 13441

 

First off pre8 came prebootstrapped no need to run it.  So please try again.

 

/b

 

On May 26, 2009, at 7:03 PM, Lars Zeb wrote:





I followed Michael Collin's instruction in an earlier email about building
pre8. During the ./bootstrap.h, I encountered the following error:

 

Entering directory /usr/src/freeswitch.trunk/libs/libsndfile

Creating aclocal.m4

Running libtoolize...

Putting files in AC_CONFIG_AUX_DIR, `Cfg'.

Creating configure

configure.ac:278: error: possibly undefined macro: AC_OCTAVE_BUILD

  If this token and others are legitimate, please use m4_pattern_allow.

  See the Autoconf documentation.

configure.ac:478: error: possibly undefined macro: AC_GCC_VERSION

 

Do I need to worry about this?

 

Also, in general, do the XML files in config get updated during the 'make
install', or are they left as they were from the previous builds?

 

Thanks, Lars

 

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Re: [Freeswitch-users] XML config error

2009-05-26 Thread Jason White
Diego Viola diego.vi...@gmail.com wrote:
 Hi, I have downloaded the latest freeswitch trunk, and when I do
 reloadxml I get this.
 
 Error [unterminated ${var}] in line
 /usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xml
 line 12
 
 Any ideas? I haven't edited that file myself.

Have a look in that file to see whether there's a syntax error there. If so,
and if it's in the source tree, fix it and submit a patch.


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Re: [Freeswitch-users] Conference users hear MOH until leader enters?

2009-05-26 Thread j3flight

Anthony - unbelievable!  Thank you so much for implementing that!  I kept
going through the possibilities of using a FIFO, putting the javascript in a
polling loop, or having everyone enter the conference muted and manually
playing MOH.  This feature absolutely makes my code a snap...  Yes, I will
remember the wish-list and I will update the wiki (probably tomorrow
though).

Stephen:  This new feature doesn't provide the ability you're referring
to...  It gives me the ability to make normal users wait on a moderator to
open the conference.

However, the capability you're asking for already exists in FreeSwitch by
choosing different profiles for your conferences.  Within the
mod_conference config file you can create different profiles which use
different caller controls a.k.a. in-conference dtmf features.  So,
create one profile for your normal users and another identical profile
(except for the caller controls) for your moderators/admins.

When you put them into the conference, you can select your profile like so:
action application=conference data=1...@profile_for_users/
action application=conference data=1...@profile_for_moderators/

I hope that's all correct - again, I'm new.

Anthony, I will try this out first thing tomorrow!


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Re: [Freeswitch-users] XML config error

2009-05-26 Thread Diego Viola
I just updated it, it was a bug that got fixed already.

22:19 @bkw__ diegoviola: already fixed
22:19 @bkw__ update
22:19 @bkw__ close the }
22:20 @bkw__ it was a bug we fixed already this morning that catches
unclosed global preprocess vars

Thanks,

Diego

On Tue, May 26, 2009 at 10:16 PM, Jason White ja...@jasonjgw.net wrote:
 Diego Viola diego.vi...@gmail.com wrote:
 Hi, I have downloaded the latest freeswitch trunk, and when I do
 reloadxml I get this.

 Error [unterminated ${var}] in line
 /usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xml
 line 12

 Any ideas? I haven't edited that file myself.

 Have a look in that file to see whether there's a syntax error there. If so,
 and if it's in the source tree, fix it and submit a patch.


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Re: [Freeswitch-users] mod_nibblebill question

2009-05-26 Thread Diego Viola
Sorry, this is the link.

http://wiki.freeswitch.org/wiki/Mod_nibblebill#Hangup_the_call_when_the_balance_is_depleted

Diego

On Tue, May 26, 2009 at 9:06 PM, Diego Viola diego.vi...@gmail.com wrote:
 Hi,

 Darren just added this today, in case if someone is interested.

 http://wiki.freeswitch.org/wiki/Mod_nibblebill#Hangup_the_call_when_cash_is_depleted

 Thanks Darren :).

 Regards,

 Diego

 On Sun, May 10, 2009 at 3:06 PM, Diego Viola diego.vi...@gmail.com wrote:
 Darren Schreiber to me
 That won't work. The code isn't written yet. Give mea week or two to
 finish that.


 On Fri, May 8, 2009 at 3:08 PM, Diego Viola diego.vi...@gmail.com wrote:
 I have set these actions:

 param name=nobal_amt value=0/
 param name=nobal_action value=hangup/

 But when it reaches 0 cash it doesn't hangup :(.

 On Fri, May 8, 2009 at 2:34 PM, Diego Viola diego.vi...@gmail.com wrote:
 Oh I see that it has it already :D

 !--
  By default, warn a caller when their balance is at $5.00. You can set
 this to a negative number.
 --
 param name=lowbal_amt value=5/
 param name=lowbal_action value=play ding/
 -
 !--
  By default, terminate a caller when their balance hits $0.00. You can
 set this to a negative number.
 --
 param name=nobal_amt value=0/
 param name=nobal_action value=hangup/

 Diego

 On Fri, May 8, 2009 at 1:20 PM, Diego Viola diego.vi...@gmail.com wrote:
 Hi everyone,

 I'm currently developing a calling card application that uses event
 socket and mod_nibblebill to bill calls. Well, the question is: can
 mod_nibblebill disconnect a call when the balance is depleted, or when
 it reaches 0 cash?

 The wiki says:

 Allow for disconnecting or re-routing calls when balance is depleted.

 But then at the bottom in future goals it says:

 We don't yet warn or terminate calls when they've reached their limits.

 So I'm confused, I also don't see any options in how to set that. In
 case that mod_nibblebill doesn't support this yet, how hard would it
 be to add this? I'm willing to do it, I guess it's a variable on the
 XML config and then read that from the mod_nibblebill.c, and do some
 logic there.

 Just wondering if anyone had some experience with this, and if the
 feature is already there.

 Thanks,

 Diego






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