Re: [Freeswitch-users] files.freeswitch.org resets connection.

2009-08-11 Thread Diego Viola
The address for me is: 69.174.57.101

On Tue, Aug 11, 2009 at 1:20 AM, Godson Gera godso...@gmail.com wrote:

 Ok here is the address 69.174.57.101 . Most of the time I use OpenDNS for
 name resolution. But the problem is still there even if I use my ISP's  DNS
 (mostly because its resolving to the same to IP address) .

 On Mon, Aug 10, 2009 at 7:01 PM, Brian West br...@freeswitch.org wrote:

 As MikeJ pointed out please report the IP address 
 files.freeswitch.orgresolves to.  We have that on a content delivery network 
 so that the files
 are closer to you geographically and you can download them faster but if
 you're having an issue I'll need the IP so I can report it correctly.
 Thanks,
 Brian

 On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:

 Hi FS Team,


 The files.freeswitch.org is resetting connection since 3 days. As
 a result I was not able to download latest release of FS. Got the trunk
 version from svn. But still it suffers from the lack of sound files.  When
 ever I do 'make uhd-sounds-install'http://files.freeswitch.org resets
 connection immediately wget tries 20 times and gives up. Other users on IRC
 also reported this issue.

 --
 Thanks  Regards,
 Godson Gera



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[Freeswitch-users] answer command

2009-08-11 Thread Maxim Tsvetov
Hello,

I found in WIKI list of supported commands
http://wiki.freeswitch.org/wiki/Mod_commands.

I need command answer but it doesn't exist (Freeswitch 1.04, Win32)

That's what Freeswitch replies for this command:

answer: Command not found!

Сould you please help?


Regards,
Maxim Tsvetov
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Re: [Freeswitch-users] ACL issue

2009-08-11 Thread Kevin Golding
Thanks Brian,

Well spotted with the domain/cidr :)

I changed the line param name=apply-inbound-acl value=domains/ in 
the internal.xml with a new value, and changed the list to match but I 
still get the same 'Rejected by acl domains' error.

And yes I reloaded the xml. :)

Kevin

Brian West wrote:
 First off edit the profile to use your own custom ACL... not the  
 domains one... secondly you'll have to use cidr= instead of domain=.
 
 Domain= will search the user directory building an ACL list from all  
 users with the cidr= attribute.
 
 /b
 
 
 On Aug 10, 2009, at 6:36 PM, Kevin Golding wrote:
 
 acl.conf.xml file, and have put the following in it:

 configuration name=acl.conf description=Network Lists
   network-lists
 list name=domains value=allow
 node type=allow domain=$${domain}/
 node type=allow domain=213.166.5.128/28/
 node type=allow domain=87.238.72.128/26/
 /list
   /network-lists
 /configuration


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Re: [Freeswitch-users] answer command

2009-08-11 Thread Szymon Olko
Maxim Tsvetov pisze:
 Hello,
 
 I found in WIKI list of supported commands
 http://wiki.freeswitch.org/wiki/Mod_commands.
 
 I need command answer but it doesn't exist (Freeswitch 1.04, Win32)
 
 That's what Freeswitch replies for this command:
 
 answer: Command not found!
 
 Сould you please help?

Answer is in dialplan commands.

http://wiki.freeswitch.org/wiki/Mod_dptools

Szymon

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Re: [Freeswitch-users] Error while creating object

2009-08-11 Thread lakshmanan ganapathy
Thanks for your replay.
I've tried that. But it says following error message.

2009-08-11 14:23:09 [ERR] mod_perl.c:69 Perl_safe_eval() [require
'/usr/local/freeswitch/conf/test.pl';]
Undefined subroutine freeswitch::DTMF called at
/usr/local/freeswitch/conf/test.pl line 6.Compilation failed in require at
(eval 2) line 1.

Please help me to solve this issue!!!

On Tue, Aug 11, 2009 at 4:59 AM, Michael Collins m...@freeswitch.org wrote:



 On Mon, Aug 10, 2009 at 3:00 AM, lakshmanan lakindi...@gmail.com wrote:


 Can any one please say what I did wrong here?


 Maybe this instead?
 my $sess=freeswitch::DTMF-new();

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Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86

2009-08-11 Thread Michal Bielicki

I'll retst it later today and give you a link with instructions

Am 10.08.2009 um 20:14 schrieb vmorales:


By ./compile I was referring to ./configure

Vladimir

-Original Message-
From: vmorales [mailto:email.list.subscri...@gmail.com]
Sent: Monday, August 10, 2009 11:49 AM
To: 'freeswitch-users@lists.freeswitch.org'
Subject: RE: [Freeswitch-users] Freeswitch pre-compiled for Solaris
10/x86

Thanks for the response(s):

I ran the ./compile script with a set PREFIX.  This took a few
attempts with errors before it was able to complete error-free, as I
had to install libtool.

Since then, I have tried running 'make', 'gmake', and
'/opt/gnu/bin/make', but each results with an error.  This is the
error when running 'make' or 'gmake':

snip
make: Fatal error: Command failed for target `all-recursive'
Current working directory /home/vmorales/freeswitch-1.0.4
*** Error code 1
make: Fatal error: Command failed for target `all'
/snip


This is the error when running '/opt/gnu/bin/make':

snip
make[5]: *** [mod_amr.so] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_amr-all] Error 1
make[2]: *** [all-recursive] Error 1
Making all in build
+ FreeSWITCH Build Complete ---+
+ FreeSWITCH has been successfully built.  +
+ Install by running:  +
+  +
+   /opt/gnu/bin/make install   +
+--+
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
/snip


I re-untar'd before each compile/make attempt.  Let me know if this is
something that I can resolve.

Vladimir


-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Jerris
Sent: Saturday, August 08, 2009 12:37 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
10/x86

This is not currently a supported platform, it only builds on 64 bit
right now I think on solaris.

Mike

On Aug 6, 2009, at 6:03 PM, vmorales wrote:


Hello,

Does anyone have, or know where to get, a pre-compiled copy of
FreeSwitch for Solaris 10/x86?



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Michal Bielicki
Leiter der Niederlassung
HaloKwadrat Sp. z o.o.
Niederlassung Kleinmachnow
Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
Ust.Id.: DE261885536
Geschaeftsfuehrer: Aleksander Wiercinski
Meiereifeld 2b, 14532 Kleinmachnow
t. +49 33203 263220
f. +49 33203 263229 sip. i...@halokwadrat.de
e. michal.bieli...@halokwadrat.de | w. www.halokwadrat.de
Hauptgeschäftsstelle:
Halo Kwadrat Sp. z o.o.
ul. Polna 46/14
00-644 Warszawa, Polen
EIngetragen im HRB Warszawa, KRS 153539



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Re: [Freeswitch-users] answer command

2009-08-11 Thread Maxim Tsvetov

Thank you.

Is it possible to use this command from command line or via event_socket
interface?



Szymon Olko wrote:
 
 Maxim Tsvetov pisze:
 Hello,
 
 I found in WIKI list of supported commands
 http://wiki.freeswitch.org/wiki/Mod_commands.
 
 I need command answer but it doesn't exist (Freeswitch 1.04, Win32)
 
 That's what Freeswitch replies for this command:
 
 answer: Command not found!
 
 Сould you please help?
 
 Answer is in dialplan commands.
 
 http://wiki.freeswitch.org/wiki/Mod_dptools
 
 Szymon
 
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Re: [Freeswitch-users] Spanish Prompts

2009-08-11 Thread samuel
I've taken a look at phrase_es.xml and would like other spanish native
speakers to check the following (i preferred to write down here my
impressions so others can contribute instead of sending a patch):

1) Centavos

prompt phrase=Centavos por minuto filename=cents-per-minute.wav/
prompt phrase=Centavos filename=cents.wav/

probably in european spanish the most apropiate word for cents is céntimo
but, if i'm not mistaken, in south american spanish is more common to use
centavos so this is more a localization issue...as you wish.
Since I live in Spain I would use the following:
prompt phrase=Céntimos por minuto filename=cents-per-minute.wav/
prompt phrase=Céntimos filename=cents.wav/

2) Minnor stress corrections:
instead of Dolar, you should use the stressed word Dólar:
-prompt phrase=Dolar filename=dollar.wav/
+prompt phrase=Dólar filename=dollar.wav/

instead of numero, you should use the stressed word número:
-prompt phrase=por favor ingrese su numero de usuario, seguido por
filename=vm-enter_id.wav/
+prompt phrase=por favor ingrese su número de usuario, seguido por
filename=vm-enter_id.wav/

3) en instead of a
-prompt phrase=La persona a la extensión filename=vm-person.wav/
+prompt phrase=La persona en la extensión filename=vm-person.wav/

4) femenin genre of contraseña changes the word seguido for seguida
-prompt phrase=por favor ingrese su contraseña, seguido por
filename=vm-enter_pass.wav/
+prompt phrase=por favor ingrese su contraseña, seguida por
filename=vm-enter_pass.wav/

5) I would propose the word almohadilla as the translation of pound so I
would change following sentences:
-prompt phrase=Signo de número filename=35.wav/
+prompt phrase=Almohadilla filename=35.wav/

-prompt phrase=Signo de número filename=pound.wav/
+prompt phrase=Almohadilla filename=pound.wav/

-prompt phrase=por favor ingrese el número de una extensión seguido del
signo de número filename=ivr-enter_ext.wav/
+prompt phrase=por favor ingrese el número de una extensión seguido de
almohadilla filename=ivr-enter_ext.wav/

-prompt phrase=Seguido del signo de número filename=followed.wav/
+prompt phrase=Seguido de almohadilla filename=followed.wav/

6)changing prepositions:
-prompt phrase=Un error ha ocurrido, por favor contacte al administrador
filename=error.wav/
+prompt phrase=Un error ha ocurrido, por favor contacte con el
administrador filename=error.wav/

7)instead of using ingrese, it sounds more natural to use introduzca when
interacting with an IVR asking for inserting data. So:
-prompt phrase=por favor ingrese su numero de usuario, seguido por
filename=vm-enter_id.wav/
+prompt phrase=por favor introduzca su número de usuario, seguido por
filename=vm-enter_id.wav/

-prompt phrase=por favor ingrese su contraseña, seguido por
filename=vm-enter_pass.wav/
+prompt phrase=por favor introduzca su contraseña, seguida por
filename=vm-enter_pass.wav/

-prompt phrase=Por favor ingrese el pin de la conferencia.
filename=conf-pin.wav/
+prompt phrase=Por favor introduzca el pin de la conferencia.
filename=conf-pin.wav/

-prompt phrase=por favor ingrese el número de una extensión seguido del
signo de número filename=ivr-enter_ext.wav/
+prompt phrase=por favor introduzca el número de una extensión seguido de
almohadilla filename=ivr-enter_ext.wav/



Comments?

Samuel


2009/8/11 Michael Collins m...@freeswitch.org

 Be sure to hop on IRC and speak with jmesquita because he's been working on
 this also. It would be very good to have Spanish-speaking users review the
 Spanish phrase_es.xml file for correctness. Also, we need Spanish-speaking
 programmers to assist with the mod_say application for Spanish - there are
 things that you have to do in Spanish that you don't have to do in English.

 Thanks,
 MC


 On Fri, Aug 7, 2009 at 1:57 PM, Luis F Urrea lfur...@gmail.com wrote:

 On linux I use Jack + Jamin + Ardour GTK2 for the takes and Rezound for
 edition .

 Record and edit at 48000Hz and then I use sox to downsample to 8000Hz for
 FS playback.

 Here's a guide that has been put together for reference on what to record.

 http://fisheye.freeswitch.org/browse/FreeSWITCH/docs/phrase/phrase_es.xml

 Regards,




 On Fri, Aug 7, 2009 at 9:21 AM, bakko asannu...@gmail.com wrote:

 I'd like to begin record spanish prompts for FS.

 Do you know any software/hardware to make it?

 Thank you

 BR


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Re: [Freeswitch-users] Spanish Prompts

2009-08-11 Thread bakko
Maybe the best solution is make two spanish translation. 

One for american y one for España.

I'm italian but i know this little differences beacouse i live in Colombia and 
i have studied in Spain.

Maybe i can help with italian prompts but i don't understand how.

BR

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Re: [Freeswitch-users] answer command

2009-08-11 Thread Milena
sendmsg
call-command: execute
execute-app-name: answer\n\n

the example is in this wiki page:
http://wiki.freeswitch.org/wiki/Event_socket_outbound
http://wiki.freeswitch.org/wiki/Event_socket_outbound
2009/8/11 Maxim Tsvetov maxim.tsve...@gmail.com


 Thank you.

 Is it possible to use this command from command line or via event_socket
 interface?



 Szymon Olko wrote:
 
  Maxim Tsvetov pisze:
  Hello,
 
  I found in WIKI list of supported commands
  http://wiki.freeswitch.org/wiki/Mod_commands.
 
  I need command answer but it doesn't exist (Freeswitch 1.04, Win32)
 
  That's what Freeswitch replies for this command:
 
  answer: Command not found!
 
  Сould you please help?
 
  Answer is in dialplan commands.
 
  http://wiki.freeswitch.org/wiki/Mod_dptools
 
  Szymon
 
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Re: [Freeswitch-users] VoiceMail transcription

2009-08-11 Thread Darren Wiebe
We've been using mycaption for a number of months and have been very 
happy with the results.  I'd suggest giving them a try.

Darren Wiebe
dar...@aleph-com.net

Pete Mueller wrote:
 I apologize, I should have been more clear.  We will be using humans 
 to scan the translated results.  But we are looking for a system to 
 perform the first pass on the audio to hopefully help the human type 
 less.

 Although the question has been raised if it's faster to have a human 
 just transcribe the whole thing, or fix up what the computer spit 
 out.  If you have any insights on this, that would be great.

 -pete

  Original Message 
 Subject: Re: [Freeswitch-users] VoiceMail transcription
 From: David Knell d...@3c.co.uk
 Date: Mon, August 10, 2009 11:51 am
 To: freeswitch-users@lists.freeswitch.org

 Good evening Pete,

 The only way to do this is, I'm afraid, to use a human. We use
 Amazon's
 Mechanical Turk to good effect.

 Cheers --

 Dave

  Good morning all,
 
  I realize this is slightly off the FS topic, but I am wondering if
  anyone out there has experience with software packages designed for
  the transcription of voicemails to text. I've used pocketsphinx with
  FS to handle IVR menus, but now have the task of figuring out how to
  convert recorded phone conversations (voicemails mostly) to text.
 
  This does not have to be a real-time process, I can store the audio
  files and process them over time. This would need to be a software
  (preferable open source) solution. ASPs like VoiceCloud would not
  work for this application.
 
  Thanks for any help
  -pete
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Re: [Freeswitch-users] answer command

2009-08-11 Thread daqiang wang
why not use:
session:answer()


2009/8/11 Maxim Tsvetov maxim.tsve...@gmail.com


 Thank you.

 Is it possible to use this command from command line or via event_socket
 interface?



 Szymon Olko wrote:
 
  Maxim Tsvetov pisze:
  Hello,
 
  I found in WIKI list of supported commands
  http://wiki.freeswitch.org/wiki/Mod_commands.
 
  I need command answer but it doesn't exist (Freeswitch 1.04, Win32)
 
  That's what Freeswitch replies for this command:
 
  answer: Command not found!
 
  Сould you please help?
 
  Answer is in dialplan commands.
 
  http://wiki.freeswitch.org/wiki/Mod_dptools
 
  Szymon
 
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Re: [Freeswitch-users] answer command

2009-08-11 Thread Maxim Tsvetov

I've tried all this command from FS console 
and all of them return Unknown command.



Milena-6 wrote:
 
 sendmsg
 call-command: execute
 execute-app-name: answer\n\n
 
 the example is in this wiki page:
 http://wiki.freeswitch.org/wiki/Event_socket_outbound
 http://wiki.freeswitch.org/wiki/Event_socket_outbound
 2009/8/11 Maxim Tsvetov maxim.tsve...@gmail.com
 

 Thank you.

 Is it possible to use this command from command line or via event_socket
 interface?



 Szymon Olko wrote:
 
  Maxim Tsvetov pisze:
  Hello,
 
  I found in WIKI list of supported commands
  http://wiki.freeswitch.org/wiki/Mod_commands.
 
  I need command answer but it doesn't exist (Freeswitch 1.04, Win32)
 
  That's what Freeswitch replies for this command:
 
  answer: Command not found!
 
  Сould you please help?
 
  Answer is in dialplan commands.
 
  http://wiki.freeswitch.org/wiki/Mod_dptools
 
  Szymon
 
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Re: [Freeswitch-users] answer command

2009-08-11 Thread Maxim Tsvetov

I've tried all this command from FS console 
and all of them return Unknown command


daqiang wang wrote:
 
 why not use:
 session:answer()
 
 
 2009/8/11 Maxim Tsvetov maxim.tsve...@gmail.com
 

 Thank you.

 Is it possible to use this command from command line or via event_socket
 interface?



 Szymon Olko wrote:
 
  Maxim Tsvetov pisze:
  Hello,
 
  I found in WIKI list of supported commands
  http://wiki.freeswitch.org/wiki/Mod_commands.
 
  I need command answer but it doesn't exist (Freeswitch 1.04, Win32)
 
  That's what Freeswitch replies for this command:
 
  answer: Command not found!
 
  Сould you please help?
 
  Answer is in dialplan commands.
 
  http://wiki.freeswitch.org/wiki/Mod_dptools
 
  Szymon
 
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Re: [Freeswitch-users] VoiceMail transcription

2009-08-11 Thread David Knell
Hi Pete,

I'm afraid that the answer's still the same: use a human.  Here's an
article describing the state of the art:
http://www.theregister.co.uk/2009/08/05/spinvox_demo_day/
- the links to previous stories at the bottom provide good background.

--Dave

 I apologize, I should have been more clear.  We will be using humans
 to scan the translated results.  But we are looking for a system to
 perform the first pass on the audio to hopefully help the human type
 less.
 
 Although the question has been raised if it's faster to have a human
 just transcribe the whole thing, or fix up what the computer spit out.
 If you have any insights on this, that would be great.
 
 -pete
 
  Original Message 
 Subject: Re: [Freeswitch-users] VoiceMail transcription
 From: David Knell d...@3c.co.uk
 Date: Mon, August 10, 2009 11:51 am
 To: freeswitch-users@lists.freeswitch.org
 
 Good evening Pete,
 
 The only way to do this is, I'm afraid, to use a human. We use
 Amazon's
 Mechanical Turk to good effect.
 
 Cheers --
 
 Dave
 
  Good morning all,
  
  I realize this is slightly off the FS topic, but I am
 wondering if
  anyone out there has experience with software packages
 designed for
  the transcription of voicemails to text. I've used
 pocketsphinx with
  FS to handle IVR menus, but now have the task of figuring
 out how to
  convert recorded phone conversations (voicemails mostly) to
 text.
  
  This does not have to be a real-time process, I can store
 the audio
  files and process them over time. This would need to be a
 software
  (preferable open source) solution. ASPs like VoiceCloud
 would not
  work for this application.
  
  Thanks for any help
  -pete
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Re: [Freeswitch-users] answer command

2009-08-11 Thread Seven Du
answer only works on outbound event socket. why you don't answer in a  
dialplan? what's scenario you use this?

On Aug 11, 2009, at 9:31 PM, Maxim Tsvetov wrote:

 I've tried all this command from FS console
 and all of them return Unknown command


 daqiang wang wrote:

 why not use:
session:answer()


 2009/8/11 Maxim Tsvetov maxim.tsve...@gmail.com


 Thank you.

 Is it possible to use this command from command line or via  
 event_socket
 interface?



 Szymon Olko wrote:

 Maxim Tsvetov pisze:
 Hello,

 I found in WIKI list of supported commands
 http://wiki.freeswitch.org/wiki/Mod_commands.

 I need command answer but it doesn't exist (Freeswitch 1.04,  
 Win32)

 That's what Freeswitch replies for this command:

 answer: Command not found!

 Сould you please help?

 Answer is in dialplan commands.

 http://wiki.freeswitch.org/wiki/Mod_dptools

 Szymon

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Re: [Freeswitch-users] answer command

2009-08-11 Thread Milena
you can send the answer on the console too;this is *how to* use
*sendmsg*from the console:
http://wiki.freeswitch.org/wiki/Event_Socket#sendmsg

http://wiki.freeswitch.org/wiki/Event_Socket#sendmsgbtw, the previous
command i gave you works just fine on event socket, it will work for you if
you replicate the Netcat example.

2009/8/11 Maxim Tsvetov maxim.tsve...@gmail.com


 I've tried all this command from FS console
 and all of them return Unknown command


 daqiang wang wrote:
 
  why not use:
  session:answer()
 
 
  2009/8/11 Maxim Tsvetov maxim.tsve...@gmail.com
 
 
  Thank you.
 
  Is it possible to use this command from command line or via event_socket
  interface?
 
 
 
  Szymon Olko wrote:
  
   Maxim Tsvetov pisze:
   Hello,
  
   I found in WIKI list of supported commands
   http://wiki.freeswitch.org/wiki/Mod_commands.
  
   I need command answer but it doesn't exist (Freeswitch 1.04, Win32)
  
   That's what Freeswitch replies for this command:
  
   answer: Command not found!
  
   Сould you please help?
  
   Answer is in dialplan commands.
  
   http://wiki.freeswitch.org/wiki/Mod_dptools
  
   Szymon
  
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Re: [Freeswitch-users] answer command

2009-08-11 Thread Brian West
Its not an api command the docs are wrong and should be fixed.

/b

On Aug 11, 2009, at 8:31 AM, Maxim Tsvetov wrote:


 I've tried all this command from FS console
 and all of them return Unknown command



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Re: [Freeswitch-users] Loopback and bypass_media

2009-08-11 Thread Phillip Jones
It occurred to me that I better check that this still occurs using lua
and that this is not a mod_manged issue.

The lua script:

session:answer();
session:execute(set, bypass_media_after_bridge=true);
session:execute(bridge, loopback/6095553828/default);

does produce the same issue. The bridge fails.

I am wondering whether this is something I should just put straight into jira.

On Mon, Aug 10, 2009 at 4:35 PM, Phillip Jonespjinthe...@gmail.com wrote:
 Hi there,

 I am attempting to do a simple bridge. Leg A is an inbound DID. Leg B
 is terminated through a SIP carrier.

 I am unable to use loopback AND bypass_media_after_bridge=true. The
 bridge fails.

 Here is my three line application:

 Session.Answer();

 Session.Execute(set, bypass_media_after_bridge=true);

 Session.Execute(bridge, loopback/6095553828/default);

 FreeSWITCH log: http://pastebin.freeswitch.org/9949

 I believe the bridge fails as the SIP-REINVITE is issued and fails.

 Using loopback and bypass_media_after_bridge=false works fine. As does
 using bypass_media_after_bridge=true and sending call directly through
 the gateway to which the loopback points.

 e.g.  http://pastebin.freeswitch.org/9948

 Has anyone else encountered this? Am I missing something?

 Any help would be much appreciated!

 Thanks


 Phillip Jones


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Re: [Freeswitch-users] answer command

2009-08-11 Thread Milena
Hello Brian,

I wanna fix the wiki, but to make sure i got it right, does it only work on
outbound event socket? or is there any other scenario where it would work.

Thank you.


2009/8/11 Brian West br...@freeswitch.org

 Its not an api command the docs are wrong and should be fixed.

 /b

 On Aug 11, 2009, at 8:31 AM, Maxim Tsvetov wrote:

 
  I've tried all this command from FS console
  and all of them return Unknown command
 


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Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86

2009-08-11 Thread vmorales
Would truly appreciate that.

Thanks Michal!


-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michal Bielicki
Sent: Tuesday, August 11, 2009 5:33 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
10/x86

I'll retst it later today and give you a link with instructions

Am 10.08.2009 um 20:14 schrieb vmorales:

 By ./compile I was referring to ./configure

 Vladimir

 -Original Message-
 From: vmorales [mailto:email.list.subscri...@gmail.com]
 Sent: Monday, August 10, 2009 11:49 AM
 To: 'freeswitch-users@lists.freeswitch.org'
 Subject: RE: [Freeswitch-users] Freeswitch pre-compiled for Solaris
 10/x86

 Thanks for the response(s):

 I ran the ./compile script with a set PREFIX.  This took a few
 attempts with errors before it was able to complete error-free, as I
 had to install libtool.

 Since then, I have tried running 'make', 'gmake', and
 '/opt/gnu/bin/make', but each results with an error.  This is the
 error when running 'make' or 'gmake':

 snip
 make: Fatal error: Command failed for target `all-recursive'
 Current working directory /home/vmorales/freeswitch-1.0.4
 *** Error code 1
 make: Fatal error: Command failed for target `all'
 /snip


 This is the error when running '/opt/gnu/bin/make':

 snip
 make[5]: *** [mod_amr.so] Error 1
 make[4]: *** [all] Error 1
 make[3]: *** [mod_amr-all] Error 1
 make[2]: *** [all-recursive] Error 1
 Making all in build
 + FreeSWITCH Build Complete ---+
 + FreeSWITCH has been successfully built.  +
 + Install by running:  +
 +  +
 +   /opt/gnu/bin/make install   +
 +--+
 make[1]: *** [all-recursive] Error 1
 make: *** [all] Error 2
 /snip


 I re-untar'd before each compile/make attempt.  Let me know if this
is
 something that I can resolve.

 Vladimir


 -Original Message-
 From: freeswitch-users-boun...@lists.freeswitch.org
 [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
 Michael Jerris
 Sent: Saturday, August 08, 2009 12:37 AM
 To: freeswitch-users@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
 10/x86

 This is not currently a supported platform, it only builds on 64 bit
 right now I think on solaris.

 Mike

 On Aug 6, 2009, at 6:03 PM, vmorales wrote:

 Hello,

 Does anyone have, or know where to get, a pre-compiled copy of
 FreeSwitch for Solaris 10/x86?


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Michal Bielicki
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Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
Ust.Id.: DE261885536
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f. +49 33203 263229 sip. i...@halokwadrat.de
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Hauptgeschäftsstelle:
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ul. Polna 46/14
00-644 Warszawa, Polen
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Re: [Freeswitch-users] VoiceMail transcription

2009-08-11 Thread Kirk Bateman
I'm still interested in getting pocketsphinx to attempt speech recognition
on an audio file.

To be honest, most of the problem is that at 8Khz (mobile phone call rate),
speech detection is NOT very accurate, at 16Khz it IS significantly better.

I'm planning to have a play with the speechtools module and mod_pocketsphinx
etc to try and get an audio file parsed, spare time permitting.

Will let the list know if I get anywhere.

Regards

Kirk Bateman


2009/8/11 David Knell d...@3c.co.uk

 Hi Pete,

 I'm afraid that the answer's still the same: use a human.  Here's an
 article describing the state of the art:
 http://www.theregister.co.uk/2009/08/05/spinvox_demo_day/
 - the links to previous stories at the bottom provide good background.

 --Dave

  I apologize, I should have been more clear.  We will be using humans
  to scan the translated results.  But we are looking for a system to
  perform the first pass on the audio to hopefully help the human type
  less.
 
  Although the question has been raised if it's faster to have a human
  just transcribe the whole thing, or fix up what the computer spit out.
  If you have any insights on this, that would be great.
 
  -pete
 
   Original Message 
  Subject: Re: [Freeswitch-users] VoiceMail transcription
  From: David Knell d...@3c.co.uk
  Date: Mon, August 10, 2009 11:51 am
  To: freeswitch-users@lists.freeswitch.org
 
  Good evening Pete,
 
  The only way to do this is, I'm afraid, to use a human. We use
  Amazon's
  Mechanical Turk to good effect.
 
  Cheers --
 
  Dave
 
   Good morning all,
  
   I realize this is slightly off the FS topic, but I am
  wondering if
   anyone out there has experience with software packages
  designed for
   the transcription of voicemails to text. I've used
  pocketsphinx with
   FS to handle IVR menus, but now have the task of figuring
  out how to
   convert recorded phone conversations (voicemails mostly) to
  text.
  
   This does not have to be a real-time process, I can store
  the audio
   files and process them over time. This would need to be a
  software
   (preferable open source) solution. ASPs like VoiceCloud
  would not
   work for this application.
  
   Thanks for any help
   -pete
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Re: [Freeswitch-users] Fwd: ALARM signal in esl libraries

2009-08-11 Thread Brian West
Replying again asking for help on the same thing over and over won't  
get you far on this list.  It'll usually get you ignored.  Can you  
explain what you're trying to do better?  Why are you needing an  
alarm? what purpose?


/b

On Aug 10, 2009, at 11:29 PM, velusamy velu wrote:


Dear Greats,
Could you please help me to solve this problem?

-- Forwarded message --
From: velusamy velu velu.techni...@gmail.com
Date: Mon, Aug 10, 2009 at 12:08 PM
Subject: ALARM signal in esl libraries
To: freeswitch-users@lists.freeswitch.org


Dear All,
   I have registered  ALARM signal  in my perl program to handle  
the DTMF digit timeout.
When ALARM signal generated  the  connection with  ESL  is   
automatically  closed.


I have checked the connection with connected: function, it returns 0.

Why the connection was closed?
Is there any idea to alive the connection after ALARM signal  
generation??


Please help me.

Thanks,
Velusamy

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Re: [Freeswitch-users] Spanish Prompts

2009-08-11 Thread samuel
I'm also for different spanish localization if it's not too complicated. It
was also for me the first time I see signo de número for pound ;)


Samuel.

2009/8/11 Nicolas Brenner nico...@medularis.com

 I think bakko is right, I have a few observations about the changes
 proposed by samuel, and I think it makes a lot of sense to have different
 localizations.


 On Tue, Aug 11, 2009 at 3:49 AM, samuel sam...@gmail.com wrote:

 I've taken a look at phrase_es.xml and would like other spanish native
 speakers to check the following (i preferred to write down here my
 impressions so others can contribute instead of sending a patch):

 1) Centavos

 prompt phrase=Centavos por minuto filename=cents-per-minute.wav/
 prompt phrase=Centavos filename=cents.wav/


 probably in european spanish the most apropiate word for cents is
 céntimo but, if i'm not mistaken, in south american spanish is more common
 to use centavos so this is more a localization issue...as you wish.
 Since I live in Spain I would use the following:
 prompt phrase=Céntimos por minuto filename=cents-per-minute.wav/
 prompt phrase=Céntimos filename=cents.wav/


 You are right, in South America we use Centavos.



 2) Minnor stress corrections:
 instead of Dolar, you should use the stressed word Dólar:
 -prompt phrase=Dolar filename=dollar.wav/
 +prompt phrase=Dólar filename=dollar.wav/

 instead of numero, you should use the stressed word número:
 -prompt phrase=por favor ingrese su numero de usuario, seguido por
 filename=vm-enter_id.wav/
 +prompt phrase=por favor ingrese su número de usuario, seguido por
 filename=vm-enter_id.wav/

 3) en instead of a
 -prompt phrase=La persona a la extensión filename=vm-person.wav/
 +prompt phrase=La persona en la extensión filename=vm-person.wav/

 4) femenin genre of contraseña changes the word seguido for seguida
 -prompt phrase=por favor ingrese su contraseña, seguido por
 filename=vm-enter_pass.wav/
 +prompt phrase=por favor ingrese su contraseña, seguida por
 filename=vm-enter_pass.wav/

 5) I would propose the word almohadilla as the translation of pound so I
 would change following sentences:
 -prompt phrase=Signo de número filename=35.wav/
 +prompt phrase=Almohadilla filename=35.wav/

 -prompt phrase=Signo de número filename=pound.wav/
 +prompt phrase=Almohadilla filename=pound.wav/


 I don't know about the rest of the countries in South America, but at least
 in Chile it would me much clearer Signo de número instead of Almohadilla
 (first time I've heard that name for the pound sign).



 -prompt phrase=por favor ingrese el número de una extensión seguido del
 signo de número filename=ivr-enter_ext.wav/
 +prompt phrase=por favor ingrese el número de una extensión seguido de
 almohadilla filename=ivr-enter_ext.wav/

 -prompt phrase=Seguido del signo de número filename=followed.wav/
 +prompt phrase=Seguido de almohadilla filename=followed.wav/

 6)changing prepositions:
 -prompt phrase=Un error ha ocurrido, por favor contacte al
 administrador filename=error.wav/
 +prompt phrase=Un error ha ocurrido, por favor contacte con el
 administrador filename=error.wav/


 Here it would be:
 prompt phrase=Ha ocurrido un error, por favor contacte al administrador
 filename=error.wav/


 7)instead of using ingrese, it sounds more natural to use introduzca when
 interacting with an IVR asking for inserting data. So:
 -prompt phrase=por favor ingrese su numero de usuario, seguido por
 filename=vm-enter_id.wav/
 +prompt phrase=por favor introduzca su número de usuario, seguido por
 filename=vm-enter_id.wav/

 -prompt phrase=por favor ingrese su contraseña, seguido por
 filename=vm-enter_pass.wav/
 +prompt phrase=por favor introduzca su contraseña, seguida por
 filename=vm-enter_pass.wav/

 -prompt phrase=Por favor ingrese el pin de la conferencia.
 filename=conf-pin.wav/
 +prompt phrase=Por favor introduzca el pin de la conferencia.
 filename=conf-pin.wav/

 -prompt phrase=por favor ingrese el número de una extensión seguido del
 signo de número filename=ivr-enter_ext.wav/
 +prompt phrase=por favor introduzca el número de una extensión seguido
 de almohadilla filename=ivr-enter_ext.wav/



 Comments?


 I am not very convinced by the use of stresses in the words, I know it's
 more correct, but sometimes it's a pain, especially if you are working on
 a remote server and don't have the right keyboard setting or something like
 that. I've also had a few headaches with file encoding, so depending on that
 and changes to the encoding you can screw your config files really easily.
 For simplification I'd leave all stresses out.



 Samuel


 2009/8/11 Michael Collins m...@freeswitch.org

 Be sure to hop on IRC and speak with jmesquita because he's been working
 on this also. It would be very good to have Spanish-speaking users review
 the Spanish phrase_es.xml file for correctness. Also, we need
 Spanish-speaking programmers to assist with the mod_say application for
 Spanish - there are things that you have to do in 

Re: [Freeswitch-users] Loopback and bypass_media

2009-08-11 Thread Brian West
This can all be accomplished with standard bridge lines stacked one on  
top of the other with continue_on_fail set to the list of codes you  
wish to fail over with.  You would also want to set  
hangup_after_bridge=true.  Loopback should NEVER be used in this case.


http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
http://wiki.freeswitch.org/wiki/Channel_Variables#hangup_after_bridge

/b

On Aug 11, 2009, at 9:46 AM, Phillip Jones wrote:


Thanks for the reply Brian.

I went through this fairly carefully with Mike Jerris - please see
http://www.nabble.com/Calling-Multiple-Destinations-with-Failover-td24877157.html

I am very open to any other mechanism that allows the calling multiple
destinations with carrier failover support. I would have thought this
is a fairly common requirement and is a fundamental requirement in my
application.

As always - your help is appreciated.


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Re: [Freeswitch-users] OpenZAP/Sangoma in Windows

2009-08-11 Thread Moises Silva
On Tue, Aug 11, 2009 at 2:16 AM, Peter Olsson 
peter.ols...@visionutveckling.se wrote:

 Hi, I'm trying to evaluate the OpenZAP/Sangoma-support in Windows, using
 PRI E1 connections.


Thanks for testing this :-)

I have been meaning to install FreeSWITCH on Windows but just could not find
the time.


 1. Has anyone tested it in Windows at all? I know the build-files for
 Visual Studio has only been checked in for a couple of months, so that's why
 I'm asking.


The drivers have been tested quite well but not using FreeSWITCH.


 2. Does anyone have any directions how to configure the driver within
 Windows? Should I use BitStream or HDLC, and how should the channel groups
 be configured?


You should use HDLC for the D-channel and Bitstream for the B-Channels.
Typically you would create 2 groups, one with channels 1-23 and the other
with just channel 24. The first group would work in bitstream and
TDM_CHAN_VOICE_API operational mode and the second in HDLC/API mode.

You can find me in #openzap, #freeswitch or #freeswitch-dev as moy if you
have more questions.

-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. m...@sangoma.com
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Re: [Freeswitch-users] Spanish Prompts

2009-08-11 Thread Diego Toro
Hi all,
 
I have some suggested code to support currency and number pronounced in spanish,
 
http://jira.freeswitch.org/browse/MODAPP-317?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel
 

Diego 
http://lacarretade.blogspot.com
 
 


--- On Tue, 8/11/09, Nicolas Brenner nico...@medularis.com wrote:


From: Nicolas Brenner nico...@medularis.com
Subject: Re: [Freeswitch-users] Spanish Prompts
To: freeswitch-users@lists.freeswitch.org
Date: Tuesday, August 11, 2009, 9:27 AM


I think bakko is right, I have a few observations about the changes proposed by 
samuel, and I think it makes a lot of sense to have different localizations.



On Tue, Aug 11, 2009 at 3:49 AM, samuel sam...@gmail.com wrote:

I've taken a look at phrase_es.xml and would like other spanish native speakers 
to check the following (i preferred to write down here my impressions so others 
can contribute instead of sending a patch):

1) Centavos

prompt phrase=Centavos por minuto filename=cents-per-minute.wav/
prompt phrase=Centavos filename=cents.wav/ 

probably in european spanish the most apropiate word for cents is céntimo 
but, if i'm not mistaken, in south american spanish is more common to use 
centavos so this is more a localization issue...as you wish.
Since I live in Spain I would use the following:
prompt phrase=Céntimos por minuto filename=cents-per-minute.wav/
prompt phrase=Céntimos filename=cents.wav/


You are right, in South America we use Centavos.
 

2) Minnor stress corrections:
instead of Dolar, you should use the stressed word Dólar:
-prompt phrase=Dolar filename=dollar.wav/
+prompt phrase=Dólar filename=dollar.wav/

instead of numero, you should use the stressed word número:
-prompt phrase=por favor ingrese su numero de usuario, seguido por 
filename=vm-enter_id.wav/
+prompt phrase=por favor ingrese su número de usuario, seguido por 
filename=vm-enter_id.wav/

3) en instead of a
-prompt phrase=La persona a la extensión filename=vm-person.wav/
+prompt phrase=La persona en la extensión filename=vm-person.wav/

4) femenin genre of contraseña changes the word seguido for seguida
-prompt phrase=por favor ingrese su contraseña, seguido por 
filename=vm-enter_pass.wav/
+prompt phrase=por favor ingrese su contraseña, seguida por 
filename=vm-enter_pass.wav/

5) I would propose the word almohadilla as the translation of pound so I 
would change following sentences:
-prompt phrase=Signo de número filename=35.wav/
+prompt phrase=Almohadilla filename=35.wav/

-prompt phrase=Signo de número filename=pound.wav/
+prompt phrase=Almohadilla filename=pound.wav/


I don't know about the rest of the countries in South America, but at least in 
Chile it would me much clearer Signo de número instead of Almohadilla (first 
time I've heard that name for the pound sign).
 

-prompt phrase=por favor ingrese el número de una extensión seguido del signo 
de número filename=ivr-enter_ext.wav/
+prompt phrase=por favor ingrese el número de una extensión seguido de 
almohadilla filename=ivr-enter_ext.wav/

-prompt phrase=Seguido del signo de número filename=followed.wav/
+prompt phrase=Seguido de almohadilla filename=followed.wav/

6)changing prepositions:
-prompt phrase=Un error ha ocurrido, por favor contacte al administrador 
filename=error.wav/
+prompt phrase=Un error ha ocurrido, por favor contacte con el administrador 
filename=error.wav/


Here it would be:
prompt phrase=Ha ocurrido un error, por favor contacte al administrador 
filename=error.wav/ 



7)instead of using ingrese, it sounds more natural to use introduzca when 
interacting with an IVR asking for inserting data. So:
-prompt phrase=por favor ingrese su numero de usuario, seguido por 
filename=vm-enter_id.wav/
+prompt phrase=por favor introduzca su número de usuario, seguido por 
filename=vm-enter_id.wav/

-prompt phrase=por favor ingrese su contraseña, seguido por 
filename=vm-enter_pass.wav/
+prompt phrase=por favor introduzca su contraseña, seguida por 
filename=vm-enter_pass.wav/

-prompt phrase=Por favor ingrese el pin de la conferencia. 
filename=conf-pin.wav/
+prompt phrase=Por favor introduzca el pin de la conferencia. 
filename=conf-pin.wav/

-prompt phrase=por favor ingrese el número de una extensión seguido del signo 
de número filename=ivr-enter_ext.wav/
+prompt phrase=por favor introduzca el número de una extensión seguido de 
almohadilla filename=ivr-enter_ext.wav/



Comments?


I am not very convinced by the use of stresses in the words, I know it's more 
correct, but sometimes it's a pain, especially if you are working on a remote 
server and don't have the right keyboard setting or something like that. I've 
also had a few headaches with file encoding, so depending on that and changes 
to the encoding you can screw your config files really easily. For 
simplification I'd leave all stresses out.
 

Samuel



2009/8/11 Michael Collins m...@freeswitch.org




Be sure to hop on IRC and speak with jmesquita because he's been working on 
this also. It 

Re: [Freeswitch-users] Loopback and bypass_media

2009-08-11 Thread Phillip Jones
Can you use this method to make simultaneous calls though while
preserving carrier order? i.e. each call must try carrier 1 first then
carrier 2 (because each carrier terminates a different subset of
numbers)

So - if I did NOT want to support failover I would use:

session:execute(bridge,
sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828);
  call 6095553828 and 7325553828 at the same time

If I were to use failover and not need multiple destinations then I would use:

session:execute(bridge,
sofia/gateway/broadvox/6095553828|sofia/gateway/quest/6095553828);
 call using broadvox first and quest it that fails

I need to combine:

Call 6095553828  and 7325553828 simultaneously and each trying
broadvox then on fail quest.

session:execute(bridge,
(sofia/gateway/broadvox/6095553828|sofia/gateway/quest/6095553828),(sofia/gateway/broadvox/7325553828,sofia/gateway/quest/7325553828));

But of course that syntax does not work.

I can not see another syntax that would achieve this?


On Tue, Aug 11, 2009 at 10:58 AM, Brian Westbr...@freeswitch.org wrote:
 This can all be accomplished with standard bridge lines stacked one on top
 of the other with continue_on_fail set to the list of codes you wish to fail
 over with.  You would also want to set hangup_after_bridge=true.  Loopback
 should NEVER be used in this case.
 http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
 http://wiki.freeswitch.org/wiki/Channel_Variables#hangup_after_bridge
 /b
 On Aug 11, 2009, at 9:46 AM, Phillip Jones wrote:

 Thanks for the reply Brian.

 I went through this fairly carefully with Mike Jerris - please see
 http://www.nabble.com/Calling-Multiple-Destinations-with-Failover-td24877157.html

 I am very open to any other mechanism that allows the calling multiple
 destinations with carrier failover support. I would have thought this
 is a fairly common requirement and is a fundamental requirement in my
 application.

 As always - your help is appreciated.

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Re: [Freeswitch-users] VoiceMail transcription

2009-08-11 Thread João Mesquita
I am sorry for the ignorance on the matter, but how does google voice does?
Do they also have humans?

jmesquita

On Tue, Aug 11, 2009 at 11:17 AM, Kirk Bateman kirk.bate...@gmail.comwrote:

 I'm still interested in getting pocketsphinx to attempt speech recognition
 on an audio file.

 To be honest, most of the problem is that at 8Khz (mobile phone call rate),
 speech detection is NOT very accurate, at 16Khz it IS significantly better.

 I'm planning to have a play with the speechtools module and
 mod_pocketsphinx etc to try and get an audio file parsed, spare time
 permitting.

 Will let the list know if I get anywhere.

 Regards

 Kirk Bateman


 2009/8/11 David Knell d...@3c.co.uk

 Hi Pete,

 I'm afraid that the answer's still the same: use a human.  Here's an
 article describing the state of the art:
 http://www.theregister.co.uk/2009/08/05/spinvox_demo_day/
 - the links to previous stories at the bottom provide good background.

 --Dave

  I apologize, I should have been more clear.  We will be using humans
  to scan the translated results.  But we are looking for a system to
  perform the first pass on the audio to hopefully help the human type
  less.
 
  Although the question has been raised if it's faster to have a human
  just transcribe the whole thing, or fix up what the computer spit out.
  If you have any insights on this, that would be great.
 
  -pete
 
   Original Message 
  Subject: Re: [Freeswitch-users] VoiceMail transcription
  From: David Knell d...@3c.co.uk
  Date: Mon, August 10, 2009 11:51 am
  To: freeswitch-users@lists.freeswitch.org
 
  Good evening Pete,
 
  The only way to do this is, I'm afraid, to use a human. We use
  Amazon's
  Mechanical Turk to good effect.
 
  Cheers --
 
  Dave
 
   Good morning all,
  
   I realize this is slightly off the FS topic, but I am
  wondering if
   anyone out there has experience with software packages
  designed for
   the transcription of voicemails to text. I've used
  pocketsphinx with
   FS to handle IVR menus, but now have the task of figuring
  out how to
   convert recorded phone conversations (voicemails mostly) to
  text.
  
   This does not have to be a real-time process, I can store
  the audio
   files and process them over time. This would need to be a
  software
   (preferable open source) solution. ASPs like VoiceCloud
  would not
   work for this application.
  
   Thanks for any help
   -pete
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 E: d...@3c.co.uk
 W: http://www.3c.co.uk


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Re: [Freeswitch-users] ACL issue

2009-08-11 Thread Michael Collins
On Tue, Aug 11, 2009 at 2:38 AM, Kevin Golding ke...@kgolding.co.uk wrote:

 Thanks Brian,

 Well spotted with the domain/cidr :)

 I changed the line param name=apply-inbound-acl value=domains/ in
 the internal.xml with a new value, and changed the list to match but I
 still get the same 'Rejected by acl domains' error.


You need to apply the name of your new ACL in the sip profile. In the file
internal.xml look for this line:
param name=apply-inbound-acl value=domains/

And change domains to whatever your new ACL is named:
param name=apply-inbound-acl value=YOUR_NEW_ACL_NAME_HERE/

Then restart FS or go to the command line and execute:
reloadacl reloadxml

Should be good after that!
-MC


 And yes I reloaded the xml. :)

 Kevin

 Brian West wrote:
  First off edit the profile to use your own custom ACL... not the
  domains one... secondly you'll have to use cidr= instead of domain=.
 
  Domain= will search the user directory building an ACL list from all
  users with the cidr= attribute.
 
  /b
 
 
  On Aug 10, 2009, at 6:36 PM, Kevin Golding wrote:
 
  acl.conf.xml file, and have put the following in it:
 
  configuration name=acl.conf description=Network Lists
network-lists
  list name=domains value=allow
  node type=allow domain=$${domain}/
  node type=allow domain=213.166.5.128/28/
  node type=allow domain=87.238.72.128/26/
  /list
/network-lists
  /configuration


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[Freeswitch-users] Multiple phones and single extensions

2009-08-11 Thread bsnipes

I've tried to do this a couple of ways in the past but can't quite seem to
get it right. The scenario is this:

attorney at extension 326 with 2 phones
secretary at extension 327 with 1 phone and a blf that monitors 326

Initially I was simply adding the attorney's extension to the secretary's
phone as a second registration but the issue comes up that the monitor light
for the 326 line lights up as if the attorney is on the phone and to other
people it shows the attorney's extension in use as if he were on the phone. 
Ok says I... I'll change the dialstring and have it ring both his and her
extension so that if she picks up it will be on her extension and it will
still show her with her blf if he is on his 326 extension. That works but
the blf light stays lit a lot after she picks up on a call destined for his
extension.  I use the following dialstring lines:

action application=set data=pcaller_id_name=ATTY:${caller_id_name}/
action application=set
data=dialstring=${sofia_contact(internal/${dialed_e...@192.168.5.4)},[origination_caller_id_name=${pcaller_id_name}]sofia/default/327/

The phones are SNOM 370s and what  I am not sure of is if the light
constantly blinking is a firmware issue on the phone or a notification issue
from FS. I've noticed that the same constant blinking happens with groups
also. If I monitor a phone that is a member of a group and the group gets
called the light for that monitor key and the message light will blink even
if no one is on the phone. Doesn't happen every time but multiple times a
day it does.

Any suggestions?
Brian

-- 
View this message in context: 
http://www.nabble.com/Multiple-phones-and-single-extensions-tp24921557p24921557.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


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Re: [Freeswitch-users] answer command

2009-08-11 Thread Michael Collins
On Tue, Aug 11, 2009 at 9:05 AM, Milena testeado...@gmail.com wrote:


 Hello Brian,

 I wanna fix the wiki, but to make sure i got it right, does it only work on
 outbound event socket? or is there any other scenario where it would work.


FYI, Diego Viola fixed the wiki. (Thanks Diego!)
-MC
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Re: [Freeswitch-users] Spanish Prompts

2009-08-11 Thread Luis F Urrea
On Tue, Aug 11, 2009 at 9:38 AM, Michael Jerris m...@jerris.com wrote:

 Correct me if I am wrong, but can't all these differences be done in a
 different soundset without modifying any code or other config?


Particularly the changes in discussion only refer to different ways to say
things.
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Re: [Freeswitch-users] ACL issue

2009-08-11 Thread Brian West
you have to now apply the ACL correctly to the sofia profile.

/b

On Aug 11, 2009, at 2:38 AM, Kevin Golding wrote:

 Thanks Brian,

 Well spotted with the domain/cidr :)

 I changed the line param name=apply-inbound-acl value=domains/  
 in
 the internal.xml with a new value, and changed the list to match but I
 still get the same 'Rejected by acl domains' error.

 And yes I reloaded the xml. :)

 Kevin


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Re: [Freeswitch-users] answer command

2009-08-11 Thread Diego Viola
Michael, you're welcome :).

Milena, answer is a mod_dptools command, you can use it from the XML
dialplan or from the event socket outbound. mod_commands API are APIs that
you execute from the socket, event socket inbound, etc. But you can also
execute them from event socket outbound using the api command.

I hope that makes sense, correct me if I'm wrong =D.

On Tue, Aug 11, 2009 at 1:09 PM, Michael Collins m...@freeswitch.org wrote:



 On Tue, Aug 11, 2009 at 9:05 AM, Milena testeado...@gmail.com wrote:


 Hello Brian,

 I wanna fix the wiki, but to make sure i got it right, does it only work
 on outbound event socket? or is there any other scenario where it would
 work.


 FYI, Diego Viola fixed the wiki. (Thanks Diego!)
 -MC


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Re: [Freeswitch-users] answer command

2009-08-11 Thread Diego Viola
I suggest that you learn the differences between mod_commands commands and
mod_dptools applications, and also the interfaces where you can access and
use them.

As said before, mod_dptools is accessible from dialplan, event socket
outbound, etc. and mod_commands is accessible from the CLI, event socket
(inbound/outbound), XML RPC, etc.

That's all described in the wiki I think.

Let us know if you have any questions =D.

On Tue, Aug 11, 2009 at 2:10 PM, Diego Viola diego.vi...@gmail.com wrote:

 Michael, you're welcome :).

 Milena, answer is a mod_dptools command, you can use it from the XML
 dialplan or from the event socket outbound. mod_commands API are APIs that
 you execute from the socket, event socket inbound, etc. But you can also
 execute them from event socket outbound using the api command.

 I hope that makes sense, correct me if I'm wrong =D.

 On Tue, Aug 11, 2009 at 1:09 PM, Michael Collins m...@freeswitch.orgwrote:



 On Tue, Aug 11, 2009 at 9:05 AM, Milena testeado...@gmail.com wrote:


 Hello Brian,

 I wanna fix the wiki, but to make sure i got it right, does it only work
 on outbound event socket? or is there any other scenario where it would
 work.


 FYI, Diego Viola fixed the wiki. (Thanks Diego!)
 -MC


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Re: [Freeswitch-users] answer command

2009-08-11 Thread Michael Collins
On Tue, Aug 11, 2009 at 1:10 PM, Diego Viola diego.vi...@gmail.com wrote:

 Michael, you're welcome :).

 Milena, answer is a mod_dptools command, you can use it from the XML
 dialplan or from the event socket outbound. mod_commands API are APIs that
 you execute from the socket, event socket inbound, etc. But you can also
 execute them from event socket outbound using the api command.

 I hope that makes sense, correct me if I'm wrong =D.


A simple way to remember it is like this:

mod_dptools are dialplan tools that go in the dialplan, aka dialplan apps
mod_commands are APIs that are typed at the FS command line

Of course, there's some crossover, but as long as you separate the idea of
an API and an app you'll be fine.
-MC
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[Freeswitch-users] hangup_after_bridge=false only works if bypass_media=false

2009-08-11 Thread Jeremiah Johnson
This is an integral part of my application.  I need to have FreeSWITCH
outside of the media path as well as be able to do multiple bridges for the
same A leg.
/*WORKS*/
action application=set data=hangup_after_bridge=false/
action application=bridge
data=sofia/gateway/${mygateway}/1${client_one}/
action application=bridge
data=sofia/gateway/${mygateway}/1${client_two}/

/*DOES NOT WORK*/
action application=set data=hangup_after_bridge=false/
action application=set data=bypass_media=true/
action application=bridge
data=sofia/gateway/${mygateway}/1${client_one}/
action application=bridge
data=sofia/gateway/${mygateway}/1${client_two}/

In the DOES NOT WORK example, the A leg hangs up as soon as the leg for
client_one hangs up.
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Re: [Freeswitch-users] Loopback and bypass_media

2009-08-11 Thread Phillip Jones
Hi Michael,

Thanks for trying to help with this.

 action application=set data=continue_on_fail=true/
 action application=bridge
 data=sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828/
 !-- if either of the above dialstrings is successful then this next line
 won't execute... --
 action application=bridge data=sofia/gateway/quest/7325553828/

In the dial plan above, consider that target is sitting at their
6095553828 phone. And that broadvox does not route to 6095553828  and
passes me a NO_ROUTE_DESTINATION.

The target will never get the call.

So if I extend this to:

action application=bridge  data=sofia/gateway/broadvox/6095553828,
sofia/gateway/broadvox/7325553828/
action application=bridge data=sofia/gateway/quest/6095553828.
sofia/gateway/quest/7325553828/

In this plan you are not calling the numbers simultaneously if the
broadvox is not able to route both calls. I can not see that you can
stack this way and get the ability to call multiple destinations, so I
must be missing something.

Call blast or Followme is basically the functionality I am trying
to implement. Separately we use multiple carriers. This is fairly
common requirement.

I am using .NET by the way - but point taken.

Any other ideas on how to achieve multiple destinations with carrier
failover would great!

Thanks again.


On Tue, Aug 11, 2009 at 2:07 PM, Michael Collinsm...@freeswitch.org wrote:


 On Tue, Aug 11, 2009 at 11:08 AM, Phillip Jones pjinthe...@gmail.com
 wrote:

 Can you use this method to make simultaneous calls though while
 preserving carrier order? i.e. each call must try carrier 1 first then
 carrier 2 (because each carrier terminates a different subset of
 numbers)

 So - if I did NOT want to support failover I would use:

 session:execute(bridge,
 sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828);
   call 6095553828 and 7325553828 at the same time

 If I were to use failover and not need multiple destinations then I would
 use:

 session:execute(bridge,
 sofia/gateway/broadvox/6095553828|sofia/gateway/quest/6095553828);
  call using broadvox first and quest it that fails

 I need to combine:

 Call 6095553828  and 7325553828 simultaneously and each trying
 broadvox then on fail quest.

 session:execute(bridge,

 (sofia/gateway/broadvox/6095553828|sofia/gateway/quest/6095553828),(sofia/gateway/broadvox/7325553828,sofia/gateway/quest/7325553828));

 But of course that syntax does not work.

 I can not see another syntax that would achieve this?

 Just to make sure I'm reading you correctly I want to clarify...

 You are trying to maintain carrier order and also trying to dial
 simultaneously. If I understand what you mean, you have two separate
 dialstrings that you want to dial simultaneously and if BOTH fail, THEN you
 want to dial the third dialstring?

 If the above is correct then this dialplan snippet SHOULD work:

 action application=set data=continue_on_fail=true/
 action application=bridge
 data=sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828/
 !-- if either of the above dialstrings is successful then this next line
 won't execute... --
 action application=bridge data=sofia/gateway/quest/7325553828/

 I also noticed that you're using js in your example... I hope you will
 consider using the dialplan instead. The DP is quick and clean. At the very
 least, if you really need js for some logic then let the js do the logic and
 then transfer back out to the dialplan. That way you will use fewer system
 resources and your project can scale more easily.

 -MC

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Re: [Freeswitch-users] hangup_after_bridge=false only works if bypass_media=false

2009-08-11 Thread Phillip Jones
Hi there,

I the 'does not work' example your media stream is back up at the
carrier as the called party hangs up.

You would have to cause another SIP REINVITE and re capture the media.

See http://wiki.freeswitch.org/wiki/Bypass_Media and the How to
disable/enable it on the fly? section as a start.




On Tue, Aug 11, 2009 at 2:29 PM, Jeremiah Johnsonboone...@gmail.com wrote:
 This is an integral part of my application.  I need to have FreeSWITCH
 outside of the media path as well as be able to do multiple bridges for the
 same A leg.
 /*WORKS*/
 action application=set data=hangup_after_bridge=false/
 action application=bridge
 data=sofia/gateway/${mygateway}/1${client_one}/
 action application=bridge
 data=sofia/gateway/${mygateway}/1${client_two}/
 /*DOES NOT WORK*/
 action application=set data=hangup_after_bridge=false/
 action application=set data=bypass_media=true/
 action application=bridge
 data=sofia/gateway/${mygateway}/1${client_one}/
 action application=bridge
 data=sofia/gateway/${mygateway}/1${client_two}/
 In the DOES NOT WORK example, the A leg hangs up as soon as the leg for
 client_one hangs up.
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Re: [Freeswitch-users] Spanish Prompts

2009-08-11 Thread Luis F Urrea
The patch submitted by Diego Toro, however would fix the actual
pronunciation of currency and dates  from 20th - 29th which are not working
with the english implementation of mod_say_es.c

In regards the dates pronunciation for voicemail for example we used the
following changes to mod_say_es.c

372,375c412,427
-   say_file(time/day-%d.wav, tm.tm_wday);
-   say_file(time/mon-%d.wav, tm.tm_mon);
-   say_num(tm.tm_mday, SSM_COUNTED);
-   say_num(tm.tm_year + 1900, SSM_PRONOUNCED);
---
+   if (tm.tm_mday == 1) {
+   say_file(time/day-%d.wav, tm.tm_wday);
+   say_num(tm.tm_mday, SSM_COUNTED);
+   say_file(time/de.wav);
+   say_file(time/mon-%d.wav, tm.tm_mon);
+   say_file(time/de.wav);
+   say_num(tm.tm_year + 1900, SSM_PRONOUNCED);
+
+   } else {
+   say_file(time/day-%d.wav, tm.tm_wday);
+   say_num(tm.tm_mday, SSM_PRONOUNCED);
+   say_file(time/de.wav);
+   say_file(time/mon-%d.wav, tm.tm_mon);
+   say_file(time/de.wav);
+   say_num(tm.tm_year + 1900, SSM_PRONOUNCED);
+   }
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[Freeswitch-users] Call transfer issues

2009-08-11 Thread Shameem Shiek
Hello,

Good work guys. I am having good fun using freeswitch so far. Currently, I
am having a serious issue on making a call transfer happen. The scenario is
simple.

1. Caller A arrives on extension 1 and is waiting on a fifo queue.
2. Caller B arrives on  extension 2 and dial plan bridges the call to Caller
C.

Now after a certain period of time, I want to hang up Caller B and put
Caller C in the fifo queue where Caller A is waiting. I know the UUIDs of
the all 3 calls. Naively, I tried the following and failed.

1. I put the bridged call to Caller C on park .
2. I did a hang up to caller B.
3. I did fifo in for Caller C into fifo queue of caller A.

This did not work. What is the right way to do this? Do I have make sure
uuid_transfer? I am using Event socket to do all of this.

-Shameem.
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Re: [Freeswitch-users] Call transfer issues

2009-08-11 Thread Brian West
OK I try so hard to follow what you're doing but it makes little sense  
to me.

I'm guessing you'll need to set/export hangup_after_bridge=false to  
prevent the hangup from taking place on B and C!

The other option is to uuid_transfer both B and C to park using the - 
both then hangup on C and then uuid_bridge A and C.  Because once you  
break the bridge on B and C by hanging up on B you left C hanging so  
its naturally going to hangup.

-USAGE: uuid [-bleg|-both] dest-exten [dialplan] [context]

/b

On Aug 11, 2009, at 2:10 PM, Shameem Shiek wrote:

 Hello,

 Good work guys. I am having good fun using freeswitch so far.  
 Currently, I am having a serious issue on making a call transfer  
 happen. The scenario is simple.

 1. Caller A arrives on extension 1 and is waiting on a fifo queue.
 2. Caller B arrives on  extension 2 and dial plan bridges the call  
 to Caller C.

 Now after a certain period of time, I want to hang up Caller B and  
 put Caller C in the fifo queue where Caller A is waiting. I know the  
 UUIDs of the all 3 calls. Naively, I tried the following and failed.

 1. I put the bridged call to Caller C on park .
 2. I did a hang up to caller B.
 3. I did fifo in for Caller C into fifo queue of caller A.

 This did not work. What is the right way to do this? Do I have make  
 sure uuid_transfer? I am using Event socket to do all of this.

 -Shameem.
 ___


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Re: [Freeswitch-users] Loopback and bypass_media

2009-08-11 Thread Michael Collins
On Tue, Aug 11, 2009 at 1:43 PM, Phillip Jones pjinthe...@gmail.com wrote:

 Hi Michael,

 Thanks for trying to help with this.

  action application=set data=continue_on_fail=true/
  action application=bridge
 
 data=sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828/
  !-- if either of the above dialstrings is successful then this next line
  won't execute... --
  action application=bridge data=sofia/gateway/quest/7325553828/

 In the dial plan above, consider that target is sitting at their
 6095553828 phone. And that broadvox does not route to 6095553828  and
 passes me a NO_ROUTE_DESTINATION.

 The target will never get the call.

 So if I extend this to:

 action application=bridge  data=sofia/gateway/broadvox/6095553828,
 sofia/gateway/broadvox/7325553828/
 action application=bridge data=sofia/gateway/quest/6095553828.
 sofia/gateway/quest/7325553828/

 In this plan you are not calling the numbers simultaneously if the
 broadvox is not able to route both calls. I can not see that you can
 stack this way and get the ability to call multiple destinations, so I
 must be missing something.


When you say call multiple destinations what exactly do you mean? If the
Broadvox is unable to connect either call then the dialplan moves on to the
Quest dial attempts. If that isn't what you need then can you clarify? Under
what condition(s) would the dialplan need to attempt to call via Quest?



 Call blast or Followme is basically the functionality I am trying
 to implement. Separately we use multiple carriers. This is fairly
 common requirement.


Yep, makes sense. Could you maybe write it out in pseudo code? Something
like this:
#1 Call phone numbers ABC and XYZ simultaneously via Broadvox
#2 If BOTH ABC AND XYZ fail, then call ABC and XYZ via Quest



 I am using .NET by the way - but point taken.


My mistake. :)



 Any other ideas on how to achieve multiple destinations with carrier
 failover would great!


I'm 100% certain that this is possible with FreeSWITCH. You might just need
a different approach, depending upon the circumstances with your carriers.
Try the above suggestions first and let's see what happens. We'll take it
from there.
-MC
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Re: [Freeswitch-users] Spanish Prompts

2009-08-11 Thread Alan Chandler
samuel wrote:
 I'm also for different spanish localization if it's not too complicated. 
 It was also for me the first time I see signo de número for pound ;)
 


I just idly noticed this - so just a comment from a Brit who iteracts 
quite a bit with Americans.

The # symbol in the UK is not called pound because (I don't know if 
this will come out on your screens correctly) we use £ for our currency.

I would refer to the # symbol as hash or just possibly the number 
symbol.


-- 
Alan Chandler
http://www.chandlerfamily.org.uk



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Re: [Freeswitch-users] Call transfer issues

2009-08-11 Thread Shameem Shiek
Thanks for the explanation. I was trying to find the correct way to do this
as I saw several ways of doing this. I will try to explain how I did on the
asterisk land.

In asterisk land, Caller A will be parked at extension 701 and Caller B
transfers the call to extension 701 and and Caller A and C are connected
and Caller B is hung up right after transfer.

In Freeswitch, Can Caller B do a transfer and connect caller A and C? How
can I do this? Since Freeswitch does not have a concept of parked
extensions.

Thank you for the help.


On Tue, Aug 11, 2009 at 3:19 PM, Brian West br...@freeswitch.org wrote:

 OK I try so hard to follow what you're doing but it makes little sense
 to me.

 I'm guessing you'll need to set/export hangup_after_bridge=false to
 prevent the hangup from taking place on B and C!

 The other option is to uuid_transfer both B and C to park using the -
 both then hangup on C and then uuid_bridge A and C.  Because once you
 break the bridge on B and C by hanging up on B you left C hanging so
 its naturally going to hangup.

 -USAGE: uuid [-bleg|-both] dest-exten [dialplan] [context]

 /b

 On Aug 11, 2009, at 2:10 PM, Shameem Shiek wrote:

  Hello,
 
  Good work guys. I am having good fun using freeswitch so far.
  Currently, I am having a serious issue on making a call transfer
  happen. The scenario is simple.
 
  1. Caller A arrives on extension 1 and is waiting on a fifo queue.
  2. Caller B arrives on  extension 2 and dial plan bridges the call
  to Caller C.
 
  Now after a certain period of time, I want to hang up Caller B and
  put Caller C in the fifo queue where Caller A is waiting. I know the
  UUIDs of the all 3 calls. Naively, I tried the following and failed.
 
  1. I put the bridged call to Caller C on park .
  2. I did a hang up to caller B.
  3. I did fifo in for Caller C into fifo queue of caller A.
 
  This did not work. What is the right way to do this? Do I have make
  sure uuid_transfer? I am using Event socket to do all of this.
 
  -Shameem.
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Re: [Freeswitch-users] Call transfer issues

2009-08-11 Thread Brian West

On Aug 11, 2009, at 2:33 PM, Shameem Shiek wrote:

 Thanks for the explanation. I was trying to find the correct way to  
 do this as I saw several ways of doing this. I will try to explain  
 how I did on the asterisk land.

 In asterisk land, Caller A will be parked at extension 701 and  
 Caller B transfers the call to extension 701 and and Caller A and  
 C are connected and Caller B is hung up right after transfer.

 In Freeswitch, Can Caller B do a transfer and connect caller A and  
 C? How can I do this? Since Freeswitch does not have a concept of  
 parked extensions.

This is where you're wrong... you can park people with the park  
application just transfer them to it... But you have to take off the  
asterisk hat.. I told you in the last email how to do it correctly  
using nothing but API commands.

/b



 Thank you for the help.


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Re: [Freeswitch-users] Spanish Prompts

2009-08-11 Thread Michael Jerris
again, this issue should be addressed when you do a sound set for that  
dialect, we are attempting to keep the c code common for all dialects  
within a language, we will see if this works unless anyone can point  
to a place this will not work.

Mike

On Aug 11, 2009, at 3:25 PM, Alan Chandler wrote:

 samuel wrote:
 I'm also for different spanish localization if it's not too  
 complicated.
 It was also for me the first time I see signo de número for  
 pound ;)



 I just idly noticed this - so just a comment from a Brit who iteracts
 quite a bit with Americans.

 The # symbol in the UK is not called pound because (I don't know if
 this will come out on your screens correctly) we use £ for our  
 currency.

 I would refer to the # symbol as hash or just possibly the number
 symbol.


 -- 
 Alan Chandler
 http://www.chandlerfamily.org.uk



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Re: [Freeswitch-users] Spanish Prompts

2009-08-11 Thread João Mesquita
Mike, the gender thing will eventually have to change code, I guess. I have
not yet looked at the say code, so I am just imagining here.



On Tue, Aug 11, 2009 at 4:45 PM, Michael Jerris m...@jerris.com wrote:

 again, this issue should be addressed when you do a sound set for that
 dialect, we are attempting to keep the c code common for all dialects
 within a language, we will see if this works unless anyone can point
 to a place this will not work.

 Mike

 On Aug 11, 2009, at 3:25 PM, Alan Chandler wrote:

  samuel wrote:
  I'm also for different spanish localization if it's not too
  complicated.
  It was also for me the first time I see signo de número for
  pound ;)
 
 
 
  I just idly noticed this - so just a comment from a Brit who iteracts
  quite a bit with Americans.
 
  The # symbol in the UK is not called pound because (I don't know if
  this will come out on your screens correctly) we use £ for our
  currency.
 
  I would refer to the # symbol as hash or just possibly the number
  symbol.
 
 
  --
  Alan Chandler
  http://www.chandlerfamily.org.uk
 
 
 
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Re: [Freeswitch-users] Ivr and variables

2009-08-11 Thread Michael Collins
What value is in the variable ${some} ?
-MC

On Mon, Aug 10, 2009 at 1:43 PM, bakko asannu...@gmail.com wrote:

 I'm tryng to put a variable in a IVR digits line like this:

 entry action=menu-exec-app digits=${some} param=transfer 1001 XML
 default/

 but don't work

 Is It possible tu use this solution?

 thank you

 BR



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Re: [Freeswitch-users] Spanish Prompts

2009-08-11 Thread Michael Collins
2009/8/11 João Mesquita jmesqu...@gmail.com

 Mike, the gender thing will eventually have to change code, I guess. I have
 not yet looked at the say code, so I am just imagining here.


Are there gender differences between dialects of the same language?
-MC
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Re: [Freeswitch-users] Ivr and variables

2009-08-11 Thread Brian West
What he is wanting is not possible.

/b

On Aug 11, 2009, at 2:54 PM, Michael Collins wrote:

 What value is in the variable ${some} ?
 -MC


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Re: [Freeswitch-users] Spanish Prompts

2009-08-11 Thread João Mesquita
Oops, I thought you were saying different languages. Sorry about that.

jmesquita

On Tue, Aug 11, 2009 at 4:54 PM, Michael Collins m...@freeswitch.org wrote:



 2009/8/11 João Mesquita jmesqu...@gmail.com

 Mike, the gender thing will eventually have to change code, I guess. I
 have not yet looked at the say code, so I am just imagining here.


 Are there gender differences between dialects of the same language?
 -MC


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Re: [Freeswitch-users] Loopback and bypass_media

2009-08-11 Thread Phillip Jones
Thanks Michael,

pseudo code would look this.

1) Call 609555 and 732555 simultaneously.
2a) As you dial 609555 go to the next carrier if you receive
NO_ROUTE_DESTINATION.
2b) As you dial 732555 go to the next carrier if you receive
NO_ROUTE_DESTINATION
3) The first number to answer and accept the call - bridge to leg a.
Drop the other.


Brian maintains that this can be done without loopback and loopback
should never be used. Mike Jerris believes that loop back handles this
and should be used. I am happy for all the help, but I am a bit
confused at this stage.

If this can not be done in the dailplan then I will need to pursue
loopback and see how far I get with that. At the moment it works but
not with bypass_media.

As I said call followme and carrier failover are fairly common
requirements so I do feel I am missing something. This must be
implemented elsewhere.

Again - thank you all for your help. You do a great job of getting
getting your heads around all this stuff day in day out.

Phillip Jones


On Tue, Aug 11, 2009 at 3:19 PM, Michael Collinsm...@freeswitch.org wrote:


 On Tue, Aug 11, 2009 at 1:43 PM, Phillip Jones pjinthe...@gmail.com wrote:

 Hi Michael,

 Thanks for trying to help with this.

  action application=set data=continue_on_fail=true/
  action application=bridge
 
  data=sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828/
  !-- if either of the above dialstrings is successful then this next
  line
  won't execute... --
  action application=bridge data=sofia/gateway/quest/7325553828/

 In the dial plan above, consider that target is sitting at their
 6095553828 phone. And that broadvox does not route to 6095553828  and
 passes me a NO_ROUTE_DESTINATION.

 The target will never get the call.

 So if I extend this to:

 action application=bridge  data=sofia/gateway/broadvox/6095553828,
 sofia/gateway/broadvox/7325553828/
 action application=bridge data=sofia/gateway/quest/6095553828.
 sofia/gateway/quest/7325553828/

 In this plan you are not calling the numbers simultaneously if the
 broadvox is not able to route both calls. I can not see that you can
 stack this way and get the ability to call multiple destinations, so I
 must be missing something.

 When you say call multiple destinations what exactly do you mean? If the
 Broadvox is unable to connect either call then the dialplan moves on to the
 Quest dial attempts. If that isn't what you need then can you clarify? Under
 what condition(s) would the dialplan need to attempt to call via Quest?



 Call blast or Followme is basically the functionality I am trying
 to implement. Separately we use multiple carriers. This is fairly
 common requirement.

 Yep, makes sense. Could you maybe write it out in pseudo code? Something
 like this:
 #1 Call phone numbers ABC and XYZ simultaneously via Broadvox
 #2 If BOTH ABC AND XYZ fail, then call ABC and XYZ via Quest


 I am using .NET by the way - but point taken.

 My mistake. :)


 Any other ideas on how to achieve multiple destinations with carrier
 failover would great!

 I'm 100% certain that this is possible with FreeSWITCH. You might just need
 a different approach, depending upon the circumstances with your carriers.
 Try the above suggestions first and let's see what happens. We'll take it
 from there.
 -MC


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Re: [Freeswitch-users] Loopback and bypass_media

2009-08-11 Thread David Knell
Just to add my $0.02-worth (if you're feeling generous..) - I don't
think that the dialplan is expressive enough to do what's needed here,
and that's where the trouble's coming from.  It's not enormously tricky
to build a generic dial this set of numbers according to these rules
service using something hanging off the event socket - there's a writeup
here: http://www.softivr.com/wiki/index.php/Find_me showing how it could
be done on SoftIVR.

To roll something similar yourself using the event socket, you'd need to
map the dial function to 'originate', bridge to (IIRC) 'uuid_bridge',
and have some way of passing messages around between the threads
handling the different call legs, assuming that you're using one thread
per leg.

--Dave


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Re: [Freeswitch-users] files.freeswitch.org resets connection.

2009-08-11 Thread Diego Viola
It resolves fine for me already.

On Tue, Aug 11, 2009 at 2:54 AM, Diego Viola diego.vi...@gmail.com wrote:

 The address for me is: 69.174.57.101


 On Tue, Aug 11, 2009 at 1:20 AM, Godson Gera godso...@gmail.com wrote:

 Ok here is the address 69.174.57.101 . Most of the time I use OpenDNS for
 name resolution. But the problem is still there even if I use my ISP's  DNS
 (mostly because its resolving to the same to IP address) .

 On Mon, Aug 10, 2009 at 7:01 PM, Brian West br...@freeswitch.org wrote:

 As MikeJ pointed out please report the IP address 
 files.freeswitch.orgresolves to.  We have that on a content delivery 
 network so that the files
 are closer to you geographically and you can download them faster but if
 you're having an issue I'll need the IP so I can report it correctly.
 Thanks,
 Brian

 On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:

 Hi FS Team,


 The files.freeswitch.org is resetting connection since 3 days.
 As a result I was not able to download latest release of FS. Got the trunk
 version from svn. But still it suffers from the lack of sound files.  When
 ever I do 'make uhd-sounds-install'http://files.freeswitch.org resets
 connection immediately wget tries 20 times and gives up. Other users on IRC
 also reported this issue.

 --
 Thanks  Regards,
 Godson Gera



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Re: [Freeswitch-users] files.freeswitch.org resets connection.

2009-08-11 Thread Brian West

Does it still give you connection refused?

/b

On Aug 11, 2009, at 2:59 PM, Diego Viola wrote:


It resolves fine for me already.

On Tue, Aug 11, 2009 at 2:54 AM, Diego Viola diego.vi...@gmail.com  
wrote:

The address for me is: 69.174.57.101


On Tue, Aug 11, 2009 at 1:20 AM, Godson Gera godso...@gmail.com  
wrote:
Ok here is the address 69.174.57.101 . Most of the time I use  
OpenDNS for name resolution. But the problem is still there even if  
I use my ISP's  DNS (mostly because its resolving to the same to IP  
address) .


On Mon, Aug 10, 2009 at 7:01 PM, Brian West br...@freeswitch.org  
wrote:
As MikeJ pointed out please report the IP address  
files.freeswitch.org resolves to.  We have that on a content  
delivery network so that the files are closer to you geographically  
and you can download them faster but if you're having an issue I'll  
need the IP so I can report it correctly.


Thanks,
Brian

On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:


Hi FS Team,


The files.freeswitch.org is resetting connection since 3  
days. As a result I was not able to download latest release of FS.  
Got the trunk version from svn. But still it suffers from the lack  
of sound files.  When ever I do 'make uhd-sounds-install'http://files.freeswitch.org 
 resets connection immediately wget tries 20 times and gives up.  
Other users on IRC also reported this issue.


--
Thanks  Regards,
Godson Gera



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Godson Gera
http://godson.in


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Re: [Freeswitch-users] files.freeswitch.org resets connection.

2009-08-11 Thread Diego Viola
Nope.

On Tue, Aug 11, 2009 at 4:01 PM, Brian West br...@freeswitch.org wrote:

 Does it still give you connection refused?
 /b

 On Aug 11, 2009, at 2:59 PM, Diego Viola wrote:

 It resolves fine for me already.

 On Tue, Aug 11, 2009 at 2:54 AM, Diego Viola diego.vi...@gmail.comwrote:

 The address for me is: 69.174.57.101


 On Tue, Aug 11, 2009 at 1:20 AM, Godson Gera godso...@gmail.com wrote:

 Ok here is the address 69.174.57.101 . Most of the time I use OpenDNS for
 name resolution. But the problem is still there even if I use my ISP's  DNS
 (mostly because its resolving to the same to IP address) .

 On Mon, Aug 10, 2009 at 7:01 PM, Brian West br...@freeswitch.orgwrote:

 As MikeJ pointed out please report the IP address 
 files.freeswitch.orgresolves to.  We have that on a content delivery 
 network so that the files
 are closer to you geographically and you can download them faster but if
 you're having an issue I'll need the IP so I can report it correctly.
 Thanks,
 Brian

 On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:

 Hi FS Team,


 The files.freeswitch.org is resetting connection since 3 days.
 As a result I was not able to download latest release of FS. Got the trunk
 version from svn. But still it suffers from the lack of sound files.  When
 ever I do 'make uhd-sounds-install'http://files.freeswitch.org resets
 connection immediately wget tries 20 times and gives up. Other users on IRC
 also reported this issue.

 --
 Thanks  Regards,
 Godson Gera



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 --
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 Godson Gera
 http://godson.in


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Re: [Freeswitch-users] hangup_after_bridge=false only works if bypass_media=false

2009-08-11 Thread Jeremiah Johnson
I guess that makes sense, but that's not the answer I wanted to hear.  I
don't necessarily know when the B leg is going to hangup so I may not be
able to do the re-invite beforehand.  Is there some sort of workaround?

On Tue, Aug 11, 2009 at 12:50 PM, Phillip Jones pjinthe...@gmail.comwrote:

 Hi there,

 I the 'does not work' example your media stream is back up at the
 carrier as the called party hangs up.

 You would have to cause another SIP REINVITE and re capture the media.

 See http://wiki.freeswitch.org/wiki/Bypass_Media and the How to
 disable/enable it on the fly? section as a start.




 On Tue, Aug 11, 2009 at 2:29 PM, Jeremiah Johnsonboone...@gmail.com
 wrote:
  This is an integral part of my application.  I need to have FreeSWITCH
  outside of the media path as well as be able to do multiple bridges for
 the
  same A leg.
  /*WORKS*/
  action application=set data=hangup_after_bridge=false/
  action application=bridge
  data=sofia/gateway/${mygateway}/1${client_one}/
  action application=bridge
  data=sofia/gateway/${mygateway}/1${client_two}/
  /*DOES NOT WORK*/
  action application=set data=hangup_after_bridge=false/
  action application=set data=bypass_media=true/
  action application=bridge
  data=sofia/gateway/${mygateway}/1${client_one}/
  action application=bridge
  data=sofia/gateway/${mygateway}/1${client_two}/
  In the DOES NOT WORK example, the A leg hangs up as soon as the leg
 for
  client_one hangs up.
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Re: [Freeswitch-users] hangup_after_bridge=false only works if bypass_media=false

2009-08-11 Thread Jeremiah Johnson
Also, if I understand bypass_media, the SIP signaling is going through
FreeSWITCH and the audio is endpoint to endpoint, so I would think the
manner in which the B leg hangup is handled is controlled by FreeSWITCH.
 That is what appears to be happening in the debug.

On Tue, Aug 11, 2009 at 2:13 PM, Jeremiah Johnson boone...@gmail.comwrote:

 I guess that makes sense, but that's not the answer I wanted to hear.  I
 don't necessarily know when the B leg is going to hangup so I may not be
 able to do the re-invite beforehand.  Is there some sort of workaround?


 On Tue, Aug 11, 2009 at 12:50 PM, Phillip Jones pjinthe...@gmail.comwrote:

 Hi there,

 I the 'does not work' example your media stream is back up at the
 carrier as the called party hangs up.

 You would have to cause another SIP REINVITE and re capture the media.

 See http://wiki.freeswitch.org/wiki/Bypass_Media and the How to
 disable/enable it on the fly? section as a start.




 On Tue, Aug 11, 2009 at 2:29 PM, Jeremiah Johnsonboone...@gmail.com
 wrote:
  This is an integral part of my application.  I need to have FreeSWITCH
  outside of the media path as well as be able to do multiple bridges for
 the
  same A leg.
  /*WORKS*/
  action application=set data=hangup_after_bridge=false/
  action application=bridge
  data=sofia/gateway/${mygateway}/1${client_one}/
  action application=bridge
  data=sofia/gateway/${mygateway}/1${client_two}/
  /*DOES NOT WORK*/
  action application=set data=hangup_after_bridge=false/
  action application=set data=bypass_media=true/
  action application=bridge
  data=sofia/gateway/${mygateway}/1${client_one}/
  action application=bridge
  data=sofia/gateway/${mygateway}/1${client_two}/
  In the DOES NOT WORK example, the A leg hangs up as soon as the leg
 for
  client_one hangs up.
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Re: [Freeswitch-users] Call transfer issues

2009-08-11 Thread Shameem Shiek
OK. I am little confused here. You say I can transfer to a parked call
using uuid_transfer . I do not see any options there where would or how
would I specify parked call in the options.

Can I do:

uuid_transfer uuid -both park

Where park is the destination?


On Tue, Aug 11, 2009 at 3:40 PM, Brian West br...@freeswitch.org wrote:


 On Aug 11, 2009, at 2:33 PM, Shameem Shiek wrote:

  Thanks for the explanation. I was trying to find the correct way to
  do this as I saw several ways of doing this. I will try to explain
  how I did on the asterisk land.
 
  In asterisk land, Caller A will be parked at extension 701 and
  Caller B transfers the call to extension 701 and and Caller A and
  C are connected and Caller B is hung up right after transfer.
 
  In Freeswitch, Can Caller B do a transfer and connect caller A and
  C? How can I do this? Since Freeswitch does not have a concept of
  parked extensions.

 This is where you're wrong... you can park people with the park
 application just transfer them to it... But you have to take off the
 asterisk hat.. I told you in the last email how to do it correctly
 using nothing but API commands.

 /b


 
  Thank you for the help.


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Re: [Freeswitch-users] Call transfer issues

2009-08-11 Thread Brian West
You also have uuid_park... but uuid_transfer to the dialplan were the  
extension calls the park application.

/b

On Aug 11, 2009, at 3:27 PM, Shameem Shiek wrote:

 OK. I am little confused here. You say I can transfer to a parked  
 call using uuid_transfer . I do not see any options there where  
 would or how would I specify parked call in the options.

 Can I do:

 uuid_transfer uuid -both park

 Where park is the destination?


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Re: [Freeswitch-users] Spanish Prompts

2009-08-11 Thread Michael Jerris
We have a plan to address this already, I can't recall if we added the  
gender types in code yet.


Mike

On Aug 11, 2009, at 3:56 PM, João Mesquita wrote:


Oops, I thought you were saying different languages. Sorry about that.

jmesquita

On Tue, Aug 11, 2009 at 4:54 PM, Michael Collins  
m...@freeswitch.org wrote:



2009/8/11 João Mesquita jmesqu...@gmail.com

Mike, the gender thing will eventually have to change code, I guess.  
I have not yet looked at the say code, so I am just imagining here.


Are there gender differences between dialects of the same language?
-MC



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Re: [Freeswitch-users] Ivr and variables

2009-08-11 Thread bakko
The value of  ${some} is 1234.

Thank you.

BR

 What value is in the variable ${some} ?
 -MC
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Re: [Freeswitch-users] ACL issue

2009-08-11 Thread Kevin Golding
Sorry, but could you give me a pointer on what this involves or where to 
read up on it please?

Brian West wrote:
 you have to now apply the ACL correctly to the sofia profile.
 
 /b
 
 On Aug 11, 2009, at 2:38 AM, Kevin Golding wrote:
 
 Thanks Brian,

 Well spotted with the domain/cidr :)

 I changed the line param name=apply-inbound-acl value=domains/  
 in
 the internal.xml with a new value, and changed the list to match but I
 still get the same 'Rejected by acl domains' error.

 And yes I reloaded the xml. :)

 Kevin
 
 
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Re: [Freeswitch-users] ACL issue

2009-08-11 Thread Brian West
I would guess in conf/sip_profiles/internal.xml where it applies the  
domains ACL to the profile... you can change that or add additional  
lines to apply more ACL's to the profile.

/b

On Aug 11, 2009, at 4:21 PM, Kevin Golding wrote:

 Sorry, but could you give me a pointer on what this involves or  
 where to
 read up on it please?

 Brian West wrote:
 you have to now apply the ACL correctly to the sofia profile.

 /b



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Re: [Freeswitch-users] Ivr and variables

2009-08-11 Thread Michael Collins
As Brian mentioned earlier, what you are trying to do is not possible in the
ivr.conf.xml file. Variables are not interpolated in the digits param. If
you need this dynamic functionality you will need to investigate
mod_xml_curl which allows you to create dynamic dialplans, etc.

-MC

On Tue, Aug 11, 2009 at 4:00 PM, bakko asannu...@gmail.com wrote:

  The value of  ${some} is 1234.

 Thank you.

 BR

  What value is in the variable ${some} ?
  -MC

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Re: [Freeswitch-users] ACL issue

2009-08-11 Thread Kevin Golding
Thank you Brian. Have got the ACL issue sorted with your help - thank 
you. See new thread for my next problem :)

kevin

Brian West wrote:
 I would guess in conf/sip_profiles/internal.xml where it applies the  
 domains ACL to the profile... you can change that or add additional  
 lines to apply more ACL's to the profile.
 
 /b
 
 On Aug 11, 2009, at 4:21 PM, Kevin Golding wrote:
 
 Sorry, but could you give me a pointer on what this involves or  
 where to
 read up on it please?

 Brian West wrote:
 you have to now apply the ACL correctly to the sofia profile.

 /b
 
 
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Re: [Freeswitch-users] Loopback and bypass_media

2009-08-11 Thread Michael Giagnocavo
It's also simple enough to write a plugin in one of the scripting languages to 
add an app to do exactly what you want...

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of David Knell
Sent: Tuesday, August 11, 2009 1:55 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Loopback and bypass_media

Just to add my $0.02-worth (if you're feeling generous..) - I don't
think that the dialplan is expressive enough to do what's needed here,
and that's where the trouble's coming from.  It's not enormously tricky
to build a generic dial this set of numbers according to these rules
service using something hanging off the event socket - there's a writeup
here: http://www.softivr.com/wiki/index.php/Find_me showing how it could
be done on SoftIVR.

To roll something similar yourself using the event socket, you'd need to
map the dial function to 'originate', bridge to (IIRC) 'uuid_bridge',
and have some way of passing messages around between the threads
handling the different call legs, assuming that you're using one thread
per leg.

--Dave


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Re: [Freeswitch-users] files.freeswitch.org resets connection.

2009-08-11 Thread Diego Viola
http://files-sync.freeswitch.org/ works fine.

On Wed, Aug 12, 2009 at 12:56 AM, Diego Viola diego.vi...@gmail.com wrote:

 Resolving files.freeswitch.org... failed: Temporary failure in name
 resolution.

 Again...


 On Tue, Aug 11, 2009 at 4:03 PM, Diego Viola diego.vi...@gmail.comwrote:

 Nope.


 On Tue, Aug 11, 2009 at 4:01 PM, Brian West br...@freeswitch.org wrote:

 Does it still give you connection refused?
 /b

 On Aug 11, 2009, at 2:59 PM, Diego Viola wrote:

 It resolves fine for me already.

 On Tue, Aug 11, 2009 at 2:54 AM, Diego Viola diego.vi...@gmail.comwrote:

 The address for me is: 69.174.57.101


 On Tue, Aug 11, 2009 at 1:20 AM, Godson Gera godso...@gmail.comwrote:

 Ok here is the address 69.174.57.101 . Most of the time I use OpenDNS
 for name resolution. But the problem is still there even if I use my ISP's
 DNS (mostly because its resolving to the same to IP address) .

 On Mon, Aug 10, 2009 at 7:01 PM, Brian West br...@freeswitch.orgwrote:

 As MikeJ pointed out please report the IP address
 files.freeswitch.org resolves to.  We have that on a content delivery
 network so that the files are closer to you geographically and you can
 download them faster but if you're having an issue I'll need the IP so I 
 can
 report it correctly.
 Thanks,
 Brian

 On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:

 Hi FS Team,


 The files.freeswitch.org is resetting connection since 3
 days. As a result I was not able to download latest release of FS. Got 
 the
 trunk version from svn. But still it suffers from the lack of sound 
 files.
 When ever I do 'make uhd-sounds-install'http://files.freeswitch.org 
 resets
 connection immediately wget tries 20 times and gives up. Other users on 
 IRC
 also reported this issue.

 --
 Thanks  Regards,
 Godson Gera



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 --
 Thanks  Regards,
 Godson Gera
 http://godson.in


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Re: [Freeswitch-users] files.freeswitch.org resets connection.

2009-08-11 Thread Jason White
Diego Viola diego.vi...@gmail.com wrote:
 Resolving files.freeswitch.org... failed: Temporary failure in name
 resolution.

It must be a problem at your end.

ja...@jdc:~$ host files.freeswitch.org
files.freeswitch.org is an alias for filessync.freeswitch.netdna-cdn.com.
filessync.freeswitch.netdna-cdn.com has address 69.174.57.101

Note that I am running my own Bind 9 daemon on this host. The record wasn't
cached, as it has been a few weeks since I last upgraded FreeSWITCH.


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