From max.bridgewater at gmail.com  Sat Jan  1 01:02:31 2011
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Fri, 31 Dec 2010 17:02:31 -0500
Subject: [Freeswitch-users] Issues with bidirectional early media
Message-ID: <AANLkTi=j5iu6i4zngZM6d39Eufdq-S9ETPM7TffagHph@mail.gmail.com>

Hi Folks,

A while ago, I tried the following code and it indeed worked with my
carriers here in Canada:

session.preAnswer();
dtmf = session.getDigits(4, "#", 10000);
console_log("got total digits: " + dtmf + "\n");

Eight months ago, I could read DTMF entered by the users while in
early media. It seems it's not working anymore. Has this behavior
changed in Freeswitch or all of the carriers turned it off?
I tried to test it with Xlite but the newest version seem not to send
DTMF at all.

I would highly appreciate if somebody could test this to confirm that
it's indeed related to my carriers.
Things work fine when I replace preAnswer() with answer(); but that's
not what i want to do.

Thanks,
Max.


From brian at freeswitch.org  Sat Jan  1 01:27:49 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 31 Dec 2010 16:27:49 -0600
Subject: [Freeswitch-users] Issues with bidirectional early media
In-Reply-To: <AANLkTi=j5iu6i4zngZM6d39Eufdq-S9ETPM7TffagHph@mail.gmail.com>
References: <AANLkTi=j5iu6i4zngZM6d39Eufdq-S9ETPM7TffagHph@mail.gmail.com>
Message-ID: <C96C53C9-2389-4AD5-BE20-934009E0A892@freeswitch.org>

Chances are they disabled it.

/b

On Dec 31, 2010, at 4:02 PM, Max Bridgewater wrote:

> Hi Folks,
> 
> A while ago, I tried the following code and it indeed worked with my
> carriers here in Canada:
> 
> session.preAnswer();
> dtmf = session.getDigits(4, "#", 10000);
> console_log("got total digits: " + dtmf + "\n");
> 
> Eight months ago, I could read DTMF entered by the users while in
> early media. It seems it's not working anymore. Has this behavior
> changed in Freeswitch or all of the carriers turned it off?
> I tried to test it with Xlite but the newest version seem not to send
> DTMF at all.
> 
> I would highly appreciate if somebody could test this to confirm that
> it's indeed related to my carriers.
> Things work fine when I replace preAnswer() with answer(); but that's
> not what i want to do.
> 
> Thanks,
> Max.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From steveayre at gmail.com  Sat Jan  1 01:48:44 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 31 Dec 2010 22:48:44 +0000
Subject: [Freeswitch-users] tone_detect and dinging
In-Reply-To: <C5A8430009F1461592CDA253B4F64DC9@e1705>
References: <C5A8430009F1461592CDA253B4F64DC9@e1705>
Message-ID: <23BB6F88-A7A8-4A84-A325-D4974B10BD41@gmail.com>

Is tone_detect answering the call?

Enable debug level log output and you'll be able to see what's going on.

Steve on iPhone

On 31 Dec 2010, at 20:51, "Madovsky" <infos at madovsky.org> wrote:

> I use this in my dialplan before a bridge
>  
> ...
> <action application="set" data="instant_ringback=true"/>
> <action application="ring_ready"/>
> <action application="tone_detect" data="fax 1100 r +5000 transfer 'receivefax XML features' 1" />
> ....
>  
> but no ring is back to the caller. if I remove tone_detect the ringback is working again.
> is anyone knows why ?
>  
> Thanks
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From joaocarlosleme at gmail.com  Sat Jan  1 02:06:37 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Fri, 31 Dec 2010 15:06:37 -0800
Subject: [Freeswitch-users] Can't build on Windows 7 64bit C++ Express!?
Message-ID: <AANLkTincSQSt-Pg00T4qVG+GXfVw2XpE6FU+2Ffe253y@mail.gmail.com>

When I open the project on C++ Express I get error messages:

Platform x64 referenced in the project file FreeSwitchConsole cannot be
found. Please make sure you have it installed under
%VCTargetsPath%\Platforms\x64

Then when I try to build I get:

========== Build: 69 succeeded, 72 failed, 0 up-to-date, 17 skipped
==========

I've attached the complete output. Any help is appreciated. I did all
following the instructions on
http://wiki.freeswitch.org/wiki/Installation_for_Windows and I did select
the Freeswitch.2010.express.sln

Thanks,
John

Resume:

LINK : fatal error LNK1104: cannot open file 'C:\Users\DEll
1\Documents\FreeSWITCH\freeswitch\libs\win32\pcre\Win32\Debug\libpcre.lib'

Many Warnings of each:

floating\bv32\bv32lspquan.c(156): warning C4244: '=' : conversion from 'int'
to 'int16_t', possible loss of data

..\..\flite-1.3.99\lang\cmu_time_awb\cmu_time_awb.c(81): warning C4090: '='
: different 'const' qualifiers

..\..\celt-0.7.1\libcelt\vq.c(63): warning C4244: '=' : conversion from
'double' to 'celt_word16', possible loss of data

..\..\libg722_1\src\tables.c(162): warning C4305: 'initializing' :
truncation from 'double' to 'const float'

and then Errors:

LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll
1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'

LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll
1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'

freeswitch_managed.cpp : fatal error C1192: #using failed on 'C:\Users\DEll
1\Documents\FreeSWITCH\freeswitch\Win32\Debug\mod\FreeSWITCH.Managed.dll'
          'The system cannot find the file specified.'

MORE
freeswitch_managed.cpp : fatal error C1192: #using failed on 'C:\Users\DEll
1\Documents\FreeSWITCH\freeswitch\Win32\Debug\mod\FreeSWITCH.Managed.dll'
          'The system cannot find the file specified.'
  freeswitch_wrap.2010.cxx
freeswitch_wrap.2010.cxx : fatal error C1192: #using failed on
'C:\Users\DEll
1\Documents\FreeSWITCH\freeswitch\Win32\Debug\mod\FreeSWITCH.Managed.dll'
          'The system cannot find the file specified.'
  mod_managed.cpp
mod_managed.cpp : fatal error C1192: #using failed on 'C:\Users\DEll
1\Documents\FreeSWITCH\freeswitch\Win32\Debug\mod\FreeSWITCH.Managed.dll'
          'The system cannot find the file specified.'
  Generating Code...
------ Build started: Project: mod_shout, Configuration: Debug Win32 ------
  mod_shout.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll
1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_loopback, Configuration: Debug Win32
------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis
warnings are not available in this edition of the compiler
  mod_loopback.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll
1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_vmd, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis
warnings are not available in this edition of the compiler
  mod_vmd.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll
1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_siren, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis
warnings are not available in this edition of the compiler
  mod_siren.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll
1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: fs_cli, Configuration: Debug Win32 ------
  getopt_long.c
  fs_cli.c
  Generating Code...
  fs_cli.2010.vcxproj -> C:\Users\DEll
1\Documents\FreeSWITCH\freeswitch\Win32\Debug\fs_cli.exe
------ Build started: Project: mod_easyroute, Configuration: Debug Win32
------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis
warnings are not available in this edition of the compiler
  mod_easyroute.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll
1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_lcr, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis
warnings are not available in this edition of the compiler
  mod_lcr.c
mod_lcr.c(637): warning C4244: '=' : conversion from 'float' to 'int',
possible loss of data
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll
1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'

*COMPLETE OUTPUT ATTACHED:*
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------ Build started: Project: make_modem_filter, Configuration: All Win32 ------
  make_modem_filter.c
  getopt.c
  filter_tools.c
  Generating Code...
  make_modem_filter.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\spandsp\src\msvc\Win32\All\make_modem_filter.exe
------ Build started: Project: libapr, Configuration: Debug Win32 ------
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_allocator.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_atomic.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_dso.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_env.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_errno.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_file_info.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_file_io.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_fnmatch.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_general.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_getopt.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_global_mutex.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_hash.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_inherit.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_lib.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_mmap.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_network_io.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_poll.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_pools.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_portable.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_proc_mutex.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_random.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_ring.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_shm.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_signal.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_strings.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_support.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_tables.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_thread_cond.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_thread_mutex.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_thread_proc.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_thread_rwlock.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_time.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_user.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_version.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr\..\..\apr\include\apr_want.h
  36 File(s) copied
  apr_atomic.c
  dso.c
  copy.c
  dir.c
  fileacc.c
  filedup.c
  filepath.c
  filepath_util.c
  filestat.c
  filesys.c
  flock.c
  fullrw.c
  mktemp.c
  open.c
  pipe.c
  readwrite.c
  seek.c
  tempdir.c
  proc_mutex.c
  thread_cond.c
  Generating Code...
  Compiling...
  thread_mutex.c
  thread_rwlock.c
  apr_pools.c
  charset.c
  env.c
  errorcodes.c
  getopt.c
  internal.c
  misc.c
  otherchild.c
  rand.c
  start.c
  utf8.c
  version.c
  common.c
  mmap.c
  inet_ntop.c
  inet_pton.c
  multicast.c
  select.c
  Generating Code...
  Compiling...
  sendrecv.c
  sockaddr.c
  sockets.c
  sockopt.c
  apr_getpass.c
  apr_random.c
  sha2.c
  sha2_glue.c
  shm.c
  apr_cpystrn.c
  apr_fnmatch.c
  apr_snprintf.c
  apr_strings.c
  apr_strnatcmp.c
  apr_strtok.c
  apr_hash.c
  apr_tables.c
  proc.c
  signals.c
  thread.c
  Generating Code...
  Compiling...
  threadpriv.c
  access.c
  time.c
  timestr.c
  groupinfo.c
  userinfo.c
  Generating Code...
     Creating library C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\libapr.lib and object C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\libapr.exp
  libapr.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\libapr.dll
------ Build started: Project: libtiff, Configuration: Debug Win32 ------
          1 file(s) copied.
          1 file(s) copied.
  tif_zip.c
  tif_write.c
  tif_warning.c
  tif_version.c
  tif_unix.c
  tif_tile.c
  tif_thunder.c
  tif_swab.c
  tif_strip.c
  tif_read.c
  tif_print.c
  tif_predict.c
  tif_pixarlog.c
  tif_packbits.c
  tif_open.c
  tif_ojpeg.c
  tif_next.c
  tif_lzw.c
  tif_luv.c
  tif_jpeg.c
  Generating Code...
  Compiling...
  tif_getimage.c
  tif_flush.c
  tif_fax3sm.c
  tif_fax3.c
  tif_extension.c
  tif_error.c
  tif_dumpmode.c
  tif_dirwrite.c
  tif_dirread.c
  tif_dirinfo.c
  tif_dir.c
  tif_compress.c
  tif_color.c
  tif_codec.c
  tif_close.c
  tif_aux.c
  Generating Code...
  libtiff.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\spandsp\src\Win32\Debug\libtiff.lib
------ Build started: Project: make_at_dictionary, Configuration: All Win32 ------
  make_at_dictionary.c
  make_at_dictionary.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\spandsp\src\msvc\Win32\All\make_at_dictionary.exe
------ Build started: Project: Download OPENSSL, Configuration: Debug Win32 ------
  Downloading OPENSSL.
  Downloading: http://files.freeswitch.org/downloads/win32/7za.exe
  Downloading: http://openssl.org/source/openssl-1.0.0a.tar.gz
  Extracting: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\openssl-1.0.0a.tar.gz
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\openssl-1.0.0a.tar.gz
  
  Extracting  openssl-1.0.0a.tar
  
  Everything is Ok
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\openssl-1.0.0a.tar
  
  Extracting  openssl-1.0.0a\apps
  Extracting  openssl-1.0.0a\apps\app_rand.c
  Extracting  openssl-1.0.0a\apps\apps.c
  Extracting  openssl-1.0.0a\apps\apps.h
  Extracting  openssl-1.0.0a\apps\asn1pars.c
  Extracting  openssl-1.0.0a\apps\ca.c
  Extracting  openssl-1.0.0a\apps\ca-cert.srl
  Extracting  openssl-1.0.0a\apps\CA.com
  Extracting  openssl-1.0.0a\apps\ca-key.pem
  Extracting  openssl-1.0.0a\apps\CA.pl
  Extracting  openssl-1.0.0a\apps\CA.pl.in
  Extracting  openssl-1.0.0a\apps\ca-req.pem
  Extracting  openssl-1.0.0a\apps\CA.sh
  Extracting  openssl-1.0.0a\apps\cert.pem
  Extracting  openssl-1.0.0a\apps\ciphers.c
  Extracting  openssl-1.0.0a\apps\client.pem
  Extracting  openssl-1.0.0a\apps\cms.c
  Extracting  openssl-1.0.0a\apps\crl2p7.c
  Extracting  openssl-1.0.0a\apps\crl.c
  Extracting  openssl-1.0.0a\apps\demoCA
  Extracting  openssl-1.0.0a\apps\demoCA\cacert.pem
  Extracting  openssl-1.0.0a\apps\demoCA\index.txt
  Extracting  openssl-1.0.0a\apps\demoCA\private
  Extracting  openssl-1.0.0a\apps\demoCA\private\cakey.pem
  Extracting  openssl-1.0.0a\apps\demoCA\serial
  Extracting  openssl-1.0.0a\apps\dgst.c
  Extracting  openssl-1.0.0a\apps\dh1024.pem
  Extracting  openssl-1.0.0a\apps\dh2048.pem
  Extracting  openssl-1.0.0a\apps\dh4096.pem
  Extracting  openssl-1.0.0a\apps\dh512.pem
  Extracting  openssl-1.0.0a\apps\dh.c
  Extracting  openssl-1.0.0a\apps\dhparam.c
  Extracting  openssl-1.0.0a\apps\dsa1024.pem
  Extracting  openssl-1.0.0a\apps\dsa512.pem
  Extracting  openssl-1.0.0a\apps\dsa.c
  Extracting  openssl-1.0.0a\apps\dsa-ca.pem
  Extracting  openssl-1.0.0a\apps\dsaparam.c
  Extracting  openssl-1.0.0a\apps\dsa-pca.pem
  Extracting  openssl-1.0.0a\apps\dsap.pem
  Extracting  openssl-1.0.0a\apps\ec.c
  Extracting  openssl-1.0.0a\apps\ecparam.c
  Extracting  openssl-1.0.0a\apps\enc.c
  Extracting  openssl-1.0.0a\apps\engine.c
  Extracting  openssl-1.0.0a\apps\errstr.c
  Extracting  openssl-1.0.0a\apps\gendh.c
  Extracting  openssl-1.0.0a\apps\gendsa.c
  Extracting  openssl-1.0.0a\apps\genpkey.c
  Extracting  openssl-1.0.0a\apps\genrsa.c
  Extracting  openssl-1.0.0a\apps\install.com
  Extracting  openssl-1.0.0a\apps\makeapps.com
  Extracting  openssl-1.0.0a\apps\Makefile
  Extracting  openssl-1.0.0a\apps\md4.c
  Extracting  openssl-1.0.0a\apps\nseq.c
  Extracting  openssl-1.0.0a\apps\ocsp.c
  Extracting  openssl-1.0.0a\apps\oid.cnf
  Extracting  openssl-1.0.0a\apps\openssl.c
  Extracting  openssl-1.0.0a\apps\openssl.cnf
  Extracting  openssl-1.0.0a\apps\openssl-vms.cnf
  Extracting  openssl-1.0.0a\apps\passwd.c
  Extracting  openssl-1.0.0a\apps\pca-cert.srl
  Extracting  openssl-1.0.0a\apps\pca-key.pem
  Extracting  openssl-1.0.0a\apps\pca-req.pem
  Extracting  openssl-1.0.0a\apps\pkcs12.c
  Extracting  openssl-1.0.0a\apps\pkcs7.c
  Extracting  openssl-1.0.0a\apps\pkcs8.c
  Extracting  openssl-1.0.0a\apps\pkey.c
  Extracting  openssl-1.0.0a\apps\pkeyparam.c
  Extracting  openssl-1.0.0a\apps\pkeyutl.c
  Extracting  openssl-1.0.0a\apps\prime.c
  Extracting  openssl-1.0.0a\apps\privkey.pem
  Extracting  openssl-1.0.0a\apps\progs.h
  Extracting  openssl-1.0.0a\apps\progs.pl
  Extracting  openssl-1.0.0a\apps\rand.c
  Extracting  openssl-1.0.0a\apps\req.c
  Extracting  openssl-1.0.0a\apps\req.pem
  Extracting  openssl-1.0.0a\apps\rsa8192.pem
  Extracting  openssl-1.0.0a\apps\rsa.c
  Extracting  openssl-1.0.0a\apps\rsautl.c
  Extracting  openssl-1.0.0a\apps\s1024key.pem
  Extracting  openssl-1.0.0a\apps\s1024req.pem
  Extracting  openssl-1.0.0a\apps\s512-key.pem
  Extracting  openssl-1.0.0a\apps\s512-req.pem
  Extracting  openssl-1.0.0a\apps\s_apps.h
  Extracting  openssl-1.0.0a\apps\s_cb.c
  Extracting  openssl-1.0.0a\apps\s_client.c
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  Extracting  openssl-1.0.0a\crypto\dh
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  Extracting  openssl-1.0.0a\crypto\ebcdic.h
  Extracting  openssl-1.0.0a\crypto\ec
  Extracting  openssl-1.0.0a\crypto\ecdh
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  Extracting  openssl-1.0.0a\crypto\ec\ec_key.c
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  Extracting  openssl-1.0.0a\crypto\engine\tb_dh.c
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  Extracting  openssl-1.0.0a\crypto\err\err.h
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  Extracting  openssl-1.0.0a\crypto\err\openssl.ec
  Extracting  openssl-1.0.0a\crypto\evp
  Extracting  openssl-1.0.0a\crypto\evp\bio_b64.c
  Extracting  openssl-1.0.0a\crypto\evp\bio_enc.c
  Extracting  openssl-1.0.0a\crypto\evp\bio_md.c
  Extracting  openssl-1.0.0a\crypto\evp\bio_ok.c
  Extracting  openssl-1.0.0a\crypto\evp\c_all.c
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  Extracting  openssl-1.0.0a\crypto\evp\digest.c
  Extracting  openssl-1.0.0a\crypto\evp\e_aes.c
  Extracting  openssl-1.0.0a\crypto\evp\e_bf.c
  Extracting  openssl-1.0.0a\crypto\evp\e_camellia.c
  Extracting  openssl-1.0.0a\crypto\evp\e_cast.c
  Extracting  openssl-1.0.0a\crypto\evp\e_des3.c
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  Extracting  openssl-1.0.0a\crypto\evp\e_dsa.c
  Extracting  openssl-1.0.0a\crypto\evp\e_idea.c
  Extracting  openssl-1.0.0a\crypto\evp\encode.c
  Extracting  openssl-1.0.0a\crypto\evp\e_null.c
  Extracting  openssl-1.0.0a\crypto\evp\e_old.c
  Extracting  openssl-1.0.0a\crypto\evp\e_rc2.c
  Extracting  openssl-1.0.0a\crypto\evp\e_rc4.c
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  Extracting  openssl-1.0.0a\crypto\evp\evp_acnf.c
  Extracting  openssl-1.0.0a\crypto\evp\evp_enc.c
  Extracting  openssl-1.0.0a\crypto\evp\evp_err.c
  Extracting  openssl-1.0.0a\crypto\evp\evp.h
  Extracting  openssl-1.0.0a\crypto\evp\evp_key.c
  Extracting  openssl-1.0.0a\crypto\evp\evp_lib.c
  Extracting  openssl-1.0.0a\crypto\evp\evp_locl.h
  Extracting  openssl-1.0.0a\crypto\evp\evp_pbe.c
  Extracting  openssl-1.0.0a\crypto\evp\evp_pkey.c
  Extracting  openssl-1.0.0a\crypto\evp\evp_test.c
  Extracting  openssl-1.0.0a\crypto\evp\evptests.txt
  Extracting  openssl-1.0.0a\crypto\evp\e_xcbc_d.c
  Extracting  openssl-1.0.0a\crypto\evp\Makefile
  Extracting  openssl-1.0.0a\crypto\evp\m_dss1.c
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  Extracting  openssl-1.0.0a\crypto\evp\m_ecdsa.c
  Extracting  openssl-1.0.0a\crypto\evp\m_md2.c
  Extracting  openssl-1.0.0a\crypto\evp\m_md4.c
  Extracting  openssl-1.0.0a\crypto\evp\m_md5.c
  Extracting  openssl-1.0.0a\crypto\evp\m_mdc2.c
  Extracting  openssl-1.0.0a\crypto\evp\m_null.c
  Extracting  openssl-1.0.0a\crypto\evp\m_ripemd.c
  Extracting  openssl-1.0.0a\crypto\evp\m_sha1.c
  Extracting  openssl-1.0.0a\crypto\evp\m_sha.c
  Extracting  openssl-1.0.0a\crypto\evp\m_sigver.c
  Extracting  openssl-1.0.0a\crypto\evp\m_wp.c
  Extracting  openssl-1.0.0a\crypto\evp\names.c
  Extracting  openssl-1.0.0a\crypto\evp\openbsd_hw.c
  Extracting  openssl-1.0.0a\crypto\evp\p5_crpt2.c
  Extracting  openssl-1.0.0a\crypto\evp\p5_crpt.c
  Extracting  openssl-1.0.0a\crypto\evp\p_dec.c
  Extracting  openssl-1.0.0a\crypto\evp\p_enc.c
  Extracting  openssl-1.0.0a\crypto\evp\p_lib.c
  Extracting  openssl-1.0.0a\crypto\evp\pmeth_fn.c
  Extracting  openssl-1.0.0a\crypto\evp\pmeth_gn.c
  Extracting  openssl-1.0.0a\crypto\evp\pmeth_lib.c
  Extracting  openssl-1.0.0a\crypto\evp\p_open.c
  Extracting  openssl-1.0.0a\crypto\evp\p_seal.c
  Extracting  openssl-1.0.0a\crypto\evp\p_sign.c
  Extracting  openssl-1.0.0a\crypto\evp\p_verify.c
  Extracting  openssl-1.0.0a\crypto\ex_data.c
  Extracting  openssl-1.0.0a\crypto\hmac
  Extracting  openssl-1.0.0a\crypto\hmac\hmac.c
  Extracting  openssl-1.0.0a\crypto\hmac\hmac.h
  Extracting  openssl-1.0.0a\crypto\hmac\hmactest.c
  Extracting  openssl-1.0.0a\crypto\hmac\hm_ameth.c
  Extracting  openssl-1.0.0a\crypto\hmac\hm_pmeth.c
  Extracting  openssl-1.0.0a\crypto\hmac\Makefile
  Extracting  openssl-1.0.0a\crypto\ia64cpuid.S
  Extracting  openssl-1.0.0a\crypto\idea
  Extracting  openssl-1.0.0a\crypto\idea\i_cbc.c
  Extracting  openssl-1.0.0a\crypto\idea\i_cfb64.c
  Extracting  openssl-1.0.0a\crypto\idea\idea.h
  Extracting  openssl-1.0.0a\crypto\idea\idea_lcl.h
  Extracting  openssl-1.0.0a\crypto\idea\idea_spd.c
  Extracting  openssl-1.0.0a\crypto\idea\ideatest.c
  Extracting  openssl-1.0.0a\crypto\idea\i_ecb.c
  Extracting  openssl-1.0.0a\crypto\idea\i_ofb64.c
  Extracting  openssl-1.0.0a\crypto\idea\i_skey.c
  Extracting  openssl-1.0.0a\crypto\idea\Makefile
  Extracting  openssl-1.0.0a\crypto\idea\version
  Extracting  openssl-1.0.0a\crypto\install.com
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  Extracting  openssl-1.0.0a\demos\x509\mkcert.c
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  Extracting  openssl-1.0.0a\demos\x509\README
  Extracting  openssl-1.0.0a\doc
  Extracting  openssl-1.0.0a\doc\apps
  Extracting  openssl-1.0.0a\doc\apps\asn1parse.pod
  Extracting  openssl-1.0.0a\doc\apps\CA.pl.pod
  Extracting  openssl-1.0.0a\doc\apps\ca.pod
  Extracting  openssl-1.0.0a\doc\apps\ciphers.pod
  Extracting  openssl-1.0.0a\doc\apps\cms.pod
  Extracting  openssl-1.0.0a\doc\apps\config.pod
  Extracting  openssl-1.0.0a\doc\apps\crl2pkcs7.pod
  Extracting  openssl-1.0.0a\doc\apps\crl.pod
  Extracting  openssl-1.0.0a\doc\apps\dgst.pod
  Extracting  openssl-1.0.0a\doc\apps\dhparam.pod
  Extracting  openssl-1.0.0a\doc\apps\dsaparam.pod
  Extracting  openssl-1.0.0a\doc\apps\dsa.pod
  Extracting  openssl-1.0.0a\doc\apps\ecparam.pod
  Extracting  openssl-1.0.0a\doc\apps\ec.pod
  Extracting  openssl-1.0.0a\doc\apps\enc.pod
  Extracting  openssl-1.0.0a\doc\apps\errstr.pod
  Extracting  openssl-1.0.0a\doc\apps\gendsa.pod
  Extracting  openssl-1.0.0a\doc\apps\genpkey.pod
  Extracting  openssl-1.0.0a\doc\apps\genrsa.pod
  Extracting  openssl-1.0.0a\doc\apps\nseq.pod
  Extracting  openssl-1.0.0a\doc\apps\ocsp.pod
  Extracting  openssl-1.0.0a\doc\apps\openssl.pod
  Extracting  openssl-1.0.0a\doc\apps\passwd.pod
  Extracting  openssl-1.0.0a\doc\apps\pkcs12.pod
  Extracting  openssl-1.0.0a\doc\apps\pkcs7.pod
  Extracting  openssl-1.0.0a\doc\apps\pkcs8.pod
  Extracting  openssl-1.0.0a\doc\apps\pkeyparam.pod
  Extracting  openssl-1.0.0a\doc\apps\pkey.pod
  Extracting  openssl-1.0.0a\doc\apps\pkeyutl.pod
  Extracting  openssl-1.0.0a\doc\apps\rand.pod
  Extracting  openssl-1.0.0a\doc\apps\req.pod
  Extracting  openssl-1.0.0a\doc\apps\rsa.pod
  Extracting  openssl-1.0.0a\doc\apps\rsautl.pod
  Extracting  openssl-1.0.0a\doc\apps\s_client.pod
  Extracting  openssl-1.0.0a\doc\apps\sess_id.pod
  Extracting  openssl-1.0.0a\doc\apps\smime.pod
  Extracting  openssl-1.0.0a\doc\apps\speed.pod
  Extracting  openssl-1.0.0a\doc\apps\spkac.pod
  Extracting  openssl-1.0.0a\doc\apps\s_server.pod
  Extracting  openssl-1.0.0a\doc\apps\s_time.pod
  Extracting  openssl-1.0.0a\doc\apps\tsget.pod
  Extracting  openssl-1.0.0a\doc\apps\ts.pod
  Extracting  openssl-1.0.0a\doc\apps\verify.pod
  Extracting  openssl-1.0.0a\doc\apps\version.pod
  Extracting  openssl-1.0.0a\doc\apps\x509.pod
  Extracting  openssl-1.0.0a\doc\apps\x509v3_config.pod
  Extracting  openssl-1.0.0a\doc\c-indentation.el
  Extracting  openssl-1.0.0a\doc\crypto
  Extracting  openssl-1.0.0a\doc\crypto\ASN1_generate_nconf.pod
  Extracting  openssl-1.0.0a\doc\crypto\ASN1_OBJECT_new.pod
  Extracting  openssl-1.0.0a\doc\crypto\ASN1_STRING_length.pod
  Extracting  openssl-1.0.0a\doc\crypto\ASN1_STRING_new.pod
  Extracting  openssl-1.0.0a\doc\crypto\ASN1_STRING_print_ex.pod
  Extracting  openssl-1.0.0a\doc\crypto\BIO_ctrl.pod
  Extracting  openssl-1.0.0a\doc\crypto\BIO_f_base64.pod
  Extracting  openssl-1.0.0a\doc\crypto\BIO_f_buffer.pod
  Extracting  openssl-1.0.0a\doc\crypto\BIO_f_cipher.pod
  Extracting  openssl-1.0.0a\doc\crypto\BIO_find_type.pod
  Extracting  openssl-1.0.0a\doc\crypto\BIO_f_md.pod
  Extracting  openssl-1.0.0a\doc\crypto\BIO_f_null.pod
  Extracting  openssl-1.0.0a\doc\crypto\BIO_f_ssl.pod
  Extracting  openssl-1.0.0a\doc\crypto\BIO_new_CMS.pod
  Extracting  openssl-1.0.0a\doc\crypto\BIO_new.pod
  Extracting  openssl-1.0.0a\doc\crypto\bio.pod
  Extracting  openssl-1.0.0a\doc\crypto\BIO_push.pod
  Extracting  openssl-1.0.0a\doc\crypto\BIO_read.pod
  Extracting  openssl-1.0.0a\doc\crypto\BIO_s_accept.pod
  Extracting  openssl-1.0.0a\doc\crypto\BIO_s_bio.pod
  Extracting  openssl-1.0.0a\doc\crypto\BIO_s_connect.pod
  Extracting  openssl-1.0.0a\doc\crypto\BIO_set_callback.pod
  Extracting  openssl-1.0.0a\doc\crypto\BIO_s_fd.pod
  Extracting  openssl-1.0.0a\doc\crypto\BIO_s_file.pod
  Extracting  openssl-1.0.0a\doc\crypto\BIO_should_retry.pod
  Extracting  openssl-1.0.0a\doc\crypto\BIO_s_mem.pod
  Extracting  openssl-1.0.0a\doc\crypto\BIO_s_null.pod
  Extracting  openssl-1.0.0a\doc\crypto\BIO_s_socket.pod
  Extracting  openssl-1.0.0a\doc\crypto\blowfish.pod
  Extracting  openssl-1.0.0a\doc\crypto\BN_add.pod
  Extracting  openssl-1.0.0a\doc\crypto\BN_add_word.pod
  Extracting  openssl-1.0.0a\doc\crypto\BN_BLINDING_new.pod
  Extracting  openssl-1.0.0a\doc\crypto\BN_bn2bin.pod
  Extracting  openssl-1.0.0a\doc\crypto\BN_cmp.pod
  Extracting  openssl-1.0.0a\doc\crypto\BN_copy.pod
  Extracting  openssl-1.0.0a\doc\crypto\BN_CTX_new.pod
  Extracting  openssl-1.0.0a\doc\crypto\BN_CTX_start.pod
  Extracting  openssl-1.0.0a\doc\crypto\BN_generate_prime.pod
  Extracting  openssl-1.0.0a\doc\crypto\bn_internal.pod
  Extracting  openssl-1.0.0a\doc\crypto\BN_mod_inverse.pod
  Extracting  openssl-1.0.0a\doc\crypto\BN_mod_mul_montgomery.pod
  Extracting  openssl-1.0.0a\doc\crypto\BN_mod_mul_reciprocal.pod
  Extracting  openssl-1.0.0a\doc\crypto\BN_new.pod
  Extracting  openssl-1.0.0a\doc\crypto\BN_num_bytes.pod
  Extracting  openssl-1.0.0a\doc\crypto\bn.pod
  Extracting  openssl-1.0.0a\doc\crypto\BN_rand.pod
  Extracting  openssl-1.0.0a\doc\crypto\BN_set_bit.pod
  Extracting  openssl-1.0.0a\doc\crypto\BN_swap.pod
  Extracting  openssl-1.0.0a\doc\crypto\BN_zero.pod
  Extracting  openssl-1.0.0a\doc\crypto\buffer.pod
  Extracting  openssl-1.0.0a\doc\crypto\CMS_add0_cert.pod
  Extracting  openssl-1.0.0a\doc\crypto\CMS_add1_recipient_cert.pod
  Extracting  openssl-1.0.0a\doc\crypto\CMS_compress.pod
  Extracting  openssl-1.0.0a\doc\crypto\CMS_decrypt.pod
  Extracting  openssl-1.0.0a\doc\crypto\CMS_encrypt.pod
  Extracting  openssl-1.0.0a\doc\crypto\CMS_final.pod
  Extracting  openssl-1.0.0a\doc\crypto\CMS_get0_RecipientInfos.pod
  Extracting  openssl-1.0.0a\doc\crypto\CMS_get0_SignerInfos.pod
  Extracting  openssl-1.0.0a\doc\crypto\CMS_get0_type.pod
  Extracting  openssl-1.0.0a\doc\crypto\CMS_get1_ReceiptRequest.pod
  Extracting  openssl-1.0.0a\doc\crypto\CMS_sign_add1_signer.pod
  Extracting  openssl-1.0.0a\doc\crypto\CMS_sign.pod
  Extracting  openssl-1.0.0a\doc\crypto\CMS_sign_receipt.pod
  Extracting  openssl-1.0.0a\doc\crypto\CMS_uncompress.pod
  Extracting  openssl-1.0.0a\doc\crypto\CMS_verify.pod
  Extracting  openssl-1.0.0a\doc\crypto\CMS_verify_receipt.pod
  Extracting  openssl-1.0.0a\doc\crypto\CONF_modules_free.pod
  Extracting  openssl-1.0.0a\doc\crypto\CONF_modules_load_file.pod
  Extracting  openssl-1.0.0a\doc\crypto\crypto.pod
  Extracting  openssl-1.0.0a\doc\crypto\CRYPTO_set_ex_data.pod
  Extracting  openssl-1.0.0a\doc\crypto\d2i_ASN1_OBJECT.pod
  Extracting  openssl-1.0.0a\doc\crypto\d2i_DHparams.pod
  Extracting  openssl-1.0.0a\doc\crypto\d2i_DSAPublicKey.pod
  Extracting  openssl-1.0.0a\doc\crypto\d2i_PKCS8PrivateKey.pod
  Extracting  openssl-1.0.0a\doc\crypto\d2i_RSAPublicKey.pod
  Extracting  openssl-1.0.0a\doc\crypto\d2i_X509_ALGOR.pod
  Extracting  openssl-1.0.0a\doc\crypto\d2i_X509_CRL.pod
  Extracting  openssl-1.0.0a\doc\crypto\d2i_X509_NAME.pod
  Extracting  openssl-1.0.0a\doc\crypto\d2i_X509.pod
  Extracting  openssl-1.0.0a\doc\crypto\d2i_X509_REQ.pod
  Extracting  openssl-1.0.0a\doc\crypto\d2i_X509_SIG.pod
  Extracting  openssl-1.0.0a\doc\crypto\des_modes.pod
  Extracting  openssl-1.0.0a\doc\crypto\des.pod
  Extracting  openssl-1.0.0a\doc\crypto\DH_generate_key.pod
  Extracting  openssl-1.0.0a\doc\crypto\DH_generate_parameters.pod
  Extracting  openssl-1.0.0a\doc\crypto\DH_get_ex_new_index.pod
  Extracting  openssl-1.0.0a\doc\crypto\DH_new.pod
  Extracting  openssl-1.0.0a\doc\crypto\dh.pod
  Extracting  openssl-1.0.0a\doc\crypto\DH_set_method.pod
  Extracting  openssl-1.0.0a\doc\crypto\DH_size.pod
  Extracting  openssl-1.0.0a\doc\crypto\DSA_do_sign.pod
  Extracting  openssl-1.0.0a\doc\crypto\DSA_dup_DH.pod
  Extracting  openssl-1.0.0a\doc\crypto\DSA_generate_key.pod
  Extracting  openssl-1.0.0a\doc\crypto\DSA_generate_parameters.pod
  Extracting  openssl-1.0.0a\doc\crypto\DSA_get_ex_new_index.pod
  Extracting  openssl-1.0.0a\doc\crypto\DSA_new.pod
  Extracting  openssl-1.0.0a\doc\crypto\dsa.pod
  Extracting  openssl-1.0.0a\doc\crypto\DSA_set_method.pod
  Extracting  openssl-1.0.0a\doc\crypto\DSA_SIG_new.pod
  Extracting  openssl-1.0.0a\doc\crypto\DSA_sign.pod
  Extracting  openssl-1.0.0a\doc\crypto\DSA_size.pod
  Extracting  openssl-1.0.0a\doc\crypto\ecdsa.pod
  Extracting  openssl-1.0.0a\doc\crypto\engine.pod
  Extracting  openssl-1.0.0a\doc\crypto\ERR_clear_error.pod
  Extracting  openssl-1.0.0a\doc\crypto\ERR_error_string.pod
  Extracting  openssl-1.0.0a\doc\crypto\ERR_get_error.pod
  Extracting  openssl-1.0.0a\doc\crypto\ERR_GET_LIB.pod
  Extracting  openssl-1.0.0a\doc\crypto\ERR_load_crypto_strings.pod
  Extracting  openssl-1.0.0a\doc\crypto\ERR_load_strings.pod
  Extracting  openssl-1.0.0a\doc\crypto\err.pod
  Extracting  openssl-1.0.0a\doc\crypto\ERR_print_errors.pod
  Extracting  openssl-1.0.0a\doc\crypto\ERR_put_error.pod
  Extracting  openssl-1.0.0a\doc\crypto\ERR_remove_state.pod
  Extracting  openssl-1.0.0a\doc\crypto\ERR_set_mark.pod
  Extracting  openssl-1.0.0a\doc\crypto\EVP_BytesToKey.pod
  Extracting  openssl-1.0.0a\doc\crypto\EVP_DigestInit.pod
  Extracting  openssl-1.0.0a\doc\crypto\EVP_DigestSignInit.pod
  Extracting  openssl-1.0.0a\doc\crypto\EVP_DigestVerifyInit.pod
  Extracting  openssl-1.0.0a\doc\crypto\EVP_EncryptInit.pod
  Extracting  openssl-1.0.0a\doc\crypto\EVP_OpenInit.pod
  Extracting  openssl-1.0.0a\doc\crypto\EVP_PKEY_cmp.pod
  Extracting  openssl-1.0.0a\doc\crypto\EVP_PKEY_CTX_ctrl.pod
  Extracting  openssl-1.0.0a\doc\crypto\EVP_PKEY_CTX_new.pod
  Extracting  openssl-1.0.0a\doc\crypto\EVP_PKEY_decrypt.pod
  Extracting  openssl-1.0.0a\doc\crypto\EVP_PKEY_derive.pod
  Extracting  openssl-1.0.0a\doc\crypto\EVP_PKEY_encrypt.pod
  Extracting  openssl-1.0.0a\doc\crypto\EVP_PKEY_get_default_digest.pod
  Extracting  openssl-1.0.0a\doc\crypto\EVP_PKEY_keygen.pod
  Extracting  openssl-1.0.0a\doc\crypto\EVP_PKEY_new.pod
  Extracting  openssl-1.0.0a\doc\crypto\EVP_PKEY_print_private.pod
  Extracting  openssl-1.0.0a\doc\crypto\EVP_PKEY_set1_RSA.pod
  Extracting  openssl-1.0.0a\doc\crypto\EVP_PKEY_sign.pod
  Extracting  openssl-1.0.0a\doc\crypto\EVP_PKEY_verify.pod
  Extracting  openssl-1.0.0a\doc\crypto\EVP_PKEY_verifyrecover.pod
  Extracting  openssl-1.0.0a\doc\crypto\evp.pod
  Extracting  openssl-1.0.0a\doc\crypto\EVP_SealInit.pod
  Extracting  openssl-1.0.0a\doc\crypto\EVP_SignInit.pod
  Extracting  openssl-1.0.0a\doc\crypto\EVP_VerifyInit.pod
  Extracting  openssl-1.0.0a\doc\crypto\hmac.pod
  Extracting  openssl-1.0.0a\doc\crypto\i2d_CMS_bio_stream.pod
  Extracting  openssl-1.0.0a\doc\crypto\i2d_PKCS7_bio_stream.pod
  Extracting  openssl-1.0.0a\doc\crypto\lhash.pod
  Extracting  openssl-1.0.0a\doc\crypto\lh_stats.pod
  Extracting  openssl-1.0.0a\doc\crypto\md5.pod
  Extracting  openssl-1.0.0a\doc\crypto\mdc2.pod
  Extracting  openssl-1.0.0a\doc\crypto\OBJ_nid2obj.pod
  Extracting  openssl-1.0.0a\doc\crypto\OpenSSL_add_all_algorithms.pod
  Extracting  openssl-1.0.0a\doc\crypto\OPENSSL_Applink.pod
  Extracting  openssl-1.0.0a\doc\crypto\OPENSSL_config.pod
  Extracting  openssl-1.0.0a\doc\crypto\OPENSSL_ia32cap.pod
  Extracting  openssl-1.0.0a\doc\crypto\OPENSSL_load_builtin_modules.pod
  Extracting  openssl-1.0.0a\doc\crypto\OPENSSL_VERSION_NUMBER.pod
  Extracting  openssl-1.0.0a\doc\crypto\pem.pod
  Extracting  openssl-1.0.0a\doc\crypto\PEM_write_bio_CMS_stream.pod
  Extracting  openssl-1.0.0a\doc\crypto\PEM_write_bio_PKCS7_stream.pod
  Extracting  openssl-1.0.0a\doc\crypto\PKCS12_create.pod
  Extracting  openssl-1.0.0a\doc\crypto\PKCS12_parse.pod
  Extracting  openssl-1.0.0a\doc\crypto\PKCS7_decrypt.pod
  Extracting  openssl-1.0.0a\doc\crypto\PKCS7_encrypt.pod
  Extracting  openssl-1.0.0a\doc\crypto\PKCS7_sign_add_signer.pod
  Extracting  openssl-1.0.0a\doc\crypto\PKCS7_sign.pod
  Extracting  openssl-1.0.0a\doc\crypto\PKCS7_verify.pod
  Extracting  openssl-1.0.0a\doc\crypto\RAND_add.pod
  Extracting  openssl-1.0.0a\doc\crypto\RAND_bytes.pod
  Extracting  openssl-1.0.0a\doc\crypto\RAND_cleanup.pod
  Extracting  openssl-1.0.0a\doc\crypto\RAND_egd.pod
  Extracting  openssl-1.0.0a\doc\crypto\RAND_load_file.pod
  Extracting  openssl-1.0.0a\doc\crypto\rand.pod
  Extracting  openssl-1.0.0a\doc\crypto\RAND_set_rand_method.pod
  Extracting  openssl-1.0.0a\doc\crypto\rc4.pod
  Extracting  openssl-1.0.0a\doc\crypto\ripemd.pod
  Extracting  openssl-1.0.0a\doc\crypto\RSA_blinding_on.pod
  Extracting  openssl-1.0.0a\doc\crypto\RSA_check_key.pod
  Extracting  openssl-1.0.0a\doc\crypto\RSA_generate_key.pod
  Extracting  openssl-1.0.0a\doc\crypto\RSA_get_ex_new_index.pod
  Extracting  openssl-1.0.0a\doc\crypto\RSA_new.pod
  Extracting  openssl-1.0.0a\doc\crypto\RSA_padding_add_PKCS1_type_1.pod
  Extracting  openssl-1.0.0a\doc\crypto\rsa.pod
  Extracting  openssl-1.0.0a\doc\crypto\RSA_print.pod
  Extracting  openssl-1.0.0a\doc\crypto\RSA_private_encrypt.pod
  Extracting  openssl-1.0.0a\doc\crypto\RSA_public_encrypt.pod
  Extracting  openssl-1.0.0a\doc\crypto\RSA_set_method.pod
  Extracting  openssl-1.0.0a\doc\crypto\RSA_sign_ASN1_OCTET_STRING.pod
  Extracting  openssl-1.0.0a\doc\crypto\RSA_sign.pod
  Extracting  openssl-1.0.0a\doc\crypto\RSA_size.pod
  Extracting  openssl-1.0.0a\doc\crypto\sha.pod
  Extracting  openssl-1.0.0a\doc\crypto\SMIME_read_CMS.pod
  Extracting  openssl-1.0.0a\doc\crypto\SMIME_read_PKCS7.pod
  Extracting  openssl-1.0.0a\doc\crypto\SMIME_write_CMS.pod
  Extracting  openssl-1.0.0a\doc\crypto\SMIME_write_PKCS7.pod
  Extracting  openssl-1.0.0a\doc\crypto\threads.pod
  Extracting  openssl-1.0.0a\doc\crypto\ui_compat.pod
  Extracting  openssl-1.0.0a\doc\crypto\ui.pod
  Extracting  openssl-1.0.0a\doc\crypto\X509_NAME_add_entry_by_txt.pod
  Extracting  openssl-1.0.0a\doc\crypto\X509_NAME_ENTRY_get_object.pod
  Extracting  openssl-1.0.0a\doc\crypto\X509_NAME_get_index_by_NID.pod
  Extracting  openssl-1.0.0a\doc\crypto\X509_NAME_print_ex.pod
  Extracting  openssl-1.0.0a\doc\crypto\X509_new.pod
  Extracting  openssl-1.0.0a\doc\crypto\x509.pod
  Extracting  openssl-1.0.0a\doc\crypto\X509_STORE_CTX_get_error.pod
  Extracting  openssl-1.0.0a\doc\crypto\X509_STORE_CTX_get_ex_new_index.pod
  Extracting  openssl-1.0.0a\doc\crypto\X509_STORE_CTX_new.pod
  Extracting  openssl-1.0.0a\doc\crypto\X509_STORE_CTX_set_verify_cb.pod
  Extracting  openssl-1.0.0a\doc\crypto\X509_STORE_set_verify_cb_func.pod
  Extracting  openssl-1.0.0a\doc\crypto\X509_verify_cert.pod
  Extracting  openssl-1.0.0a\doc\crypto\X509_VERIFY_PARAM_set_flags.pod
  Extracting  openssl-1.0.0a\doc\fingerprints.txt
  Extracting  openssl-1.0.0a\doc\HOWTO
  Extracting  openssl-1.0.0a\doc\HOWTO\certificates.txt
  Extracting  openssl-1.0.0a\doc\HOWTO\keys.txt
  Extracting  openssl-1.0.0a\doc\HOWTO\proxy_certificates.txt
  Extracting  openssl-1.0.0a\doc\openssl_button.gif
  Extracting  openssl-1.0.0a\doc\openssl_button.html
  Extracting  openssl-1.0.0a\doc\openssl-shared.txt
  Extracting  openssl-1.0.0a\doc\openssl.txt
  Extracting  openssl-1.0.0a\doc\README
  Extracting  openssl-1.0.0a\doc\ssl
  Extracting  openssl-1.0.0a\doc\ssl\d2i_SSL_SESSION.pod
  Extracting  openssl-1.0.0a\doc\ssleay.txt
  Extracting  openssl-1.0.0a\doc\ssl\SSL_accept.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_alert_type_string.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CIPHER_get_name.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_clear.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_COMP_add_compression_method.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_connect.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_add_extra_chain_cert.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_add_session.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_ctrl.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_flush_sessions.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_free.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_get_ex_new_index.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_get_verify_mode.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_load_verify_locations.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_new.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_sessions.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_sess_number.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_sess_set_cache_size.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_sess_set_get_cb.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_set_cert_store.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_set_cert_verify_callback.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_set_cipher_list.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_set_client_CA_list.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_set_client_cert_cb.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_set_default_passwd_cb.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_set_generate_session_id.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_set_info_callback.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_set_max_cert_list.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_set_mode.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_set_msg_callback.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_set_options.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_set_psk_client_callback.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_set_quiet_shutdown.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_set_session_cache_mode.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_set_session_id_context.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_set_ssl_version.pod
  Extracting  openssl-1.0.0a\doc\ssl\SSL_CTX_set_timeout.pod
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  Everything is Ok
------ Build started: Project: Download PTHREAD, Configuration: Debug Win32 ------
  Downloading PTHREAD.
  Downloading: http://files.freeswitch.org/downloads/libs/pthreads-w32-2-7-0-release.tar.gz
  Extracting: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\pthreads-w32-2-7-0-release.tar.gz
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\pthreads-w32-2-7-0-release.tar.gz
  
  Extracting  pthreads-w32-2-7-0-release.tar
  
  Everything is Ok
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\pthreads-w32-2-7-0-release.tar
  
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  Extracting  pthreads-w32-2-7-0-release\tests\README
  Extracting  pthreads-w32-2-7-0-release\tests\README.BENCHTESTS
  Extracting  pthreads-w32-2-7-0-release\tests\reuse1.c
  Extracting  pthreads-w32-2-7-0-release\tests\reuse2.c
  Extracting  pthreads-w32-2-7-0-release\tests\rwlock1.c
  Extracting  pthreads-w32-2-7-0-release\tests\rwlock2.c
  Extracting  pthreads-w32-2-7-0-release\tests\rwlock2_t.c
  Extracting  pthreads-w32-2-7-0-release\tests\rwlock3.c
  Extracting  pthreads-w32-2-7-0-release\tests\rwlock3_t.c
  Extracting  pthreads-w32-2-7-0-release\tests\rwlock4.c
  Extracting  pthreads-w32-2-7-0-release\tests\rwlock4_t.c
  Extracting  pthreads-w32-2-7-0-release\tests\rwlock5.c
  Extracting  pthreads-w32-2-7-0-release\tests\rwlock5_t.c
  Extracting  pthreads-w32-2-7-0-release\tests\rwlock6.c
  Extracting  pthreads-w32-2-7-0-release\tests\rwlock6_t.c
  Extracting  pthreads-w32-2-7-0-release\tests\rwlock6_t2.c
  Extracting  pthreads-w32-2-7-0-release\tests\rwlock7.c
  Extracting  pthreads-w32-2-7-0-release\tests\rwlock8.c
  Extracting  pthreads-w32-2-7-0-release\tests\self1.c
  Extracting  pthreads-w32-2-7-0-release\tests\self2.c
  Extracting  pthreads-w32-2-7-0-release\tests\semaphore1.c
  Extracting  pthreads-w32-2-7-0-release\tests\semaphore2.c
  Extracting  pthreads-w32-2-7-0-release\tests\semaphore3.c
  Extracting  pthreads-w32-2-7-0-release\tests\semaphore4.c
  Extracting  pthreads-w32-2-7-0-release\tests\semaphore4t.c
  Extracting  pthreads-w32-2-7-0-release\tests\sizes.c
  Extracting  pthreads-w32-2-7-0-release\tests\SIZES.GC
  Extracting  pthreads-w32-2-7-0-release\tests\SIZES.GCE
  Extracting  pthreads-w32-2-7-0-release\tests\SIZES.VC
  Extracting  pthreads-w32-2-7-0-release\tests\SIZES.VCE
  Extracting  pthreads-w32-2-7-0-release\tests\SIZES.VSE
  Extracting  pthreads-w32-2-7-0-release\tests\spin1.c
  Extracting  pthreads-w32-2-7-0-release\tests\spin2.c
  Extracting  pthreads-w32-2-7-0-release\tests\spin3.c
  Extracting  pthreads-w32-2-7-0-release\tests\spin4.c
  Extracting  pthreads-w32-2-7-0-release\tests\stress1.c
  Extracting  pthreads-w32-2-7-0-release\tests\test.h
  Extracting  pthreads-w32-2-7-0-release\tests\tryentercs.c
  Extracting  pthreads-w32-2-7-0-release\tests\tryentercs2.c
  Extracting  pthreads-w32-2-7-0-release\tests\tsd1.c
  Extracting  pthreads-w32-2-7-0-release\tests\tsd2.c
  Extracting  pthreads-w32-2-7-0-release\tests\valid1.c
  Extracting  pthreads-w32-2-7-0-release\tests\valid2.c
  Extracting  pthreads-w32-2-7-0-release\tests\Wmakefile
  Extracting  pthreads-w32-2-7-0-release\TODO
  Extracting  pthreads-w32-2-7-0-release\tsd.c
  Extracting  pthreads-w32-2-7-0-release\version.rc
  Extracting  pthreads-w32-2-7-0-release\w32_CancelableWait.c
  Extracting  pthreads-w32-2-7-0-release\WinCE-PORT
  
  Everything is Ok
------ Build started: Project: libteletone, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  libteletone_generate.c
  libteletone_detect.c
  Generating Code...
     Creating library C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\libteletone.lib and object C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\libteletone.exp
  libteletone.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\libteletone.dll
------ Build started: Project: libspeexdsp, Configuration: Debug Win32 ------
  smallft.c
  resample.c
  preprocess.c
  mdf.c
  kiss_fftr.c
  kiss_fft.c
  jitter.c
  filterbank.c
  fftwrap.c
  buffer.c
  Generating Code...
  libspeexdsp.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\speex\win32\VS2008\libspeexdsp\Win32\Debug\libspeexdsp.lib
------ Build started: Project: libsrtp, Configuration: Debug Win32 ------
  stat.c
  datatypes.c
  ut_sim.c
  rdbx.c
  rdb.c
  sha1.c
  null_auth.c
  hmac.c
  auth.c
  null_cipher.c
  cipher.c
  aes_icm.c
  aes_cbc.c
  aes.c
  rand_source.c
  prng.c
  key.c
  err.c
  ctr_prng.c
  crypto_kernel.c
  Generating Code...
  Compiling...
  alloc.c
  srtp.c
  Generating Code...
  libsrtp.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\srtp\Win32\Debug\libsrtp.lib
------ Build started: Project: libspandsp, Configuration: Debug Win32 ------
  Copying C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\spandsp\src\msvc\spandsp.h to C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\spandsp\src\spandsp.h
          1 file(s) copied.
  gettimeofday.c
  vector_int.c
  vector_float.c
  v8.c
  v42bis.c
  v42.c
  v29tx.c
  v29rx.c
  v27ter_tx.c
  v27ter_rx.c
  v22bis_tx.c
  v22bis_rx.c
  v18.c
  v17tx.c
  v17rx.c
  tone_generate.c
  tone_detect.c
  timezone.c
  time_scale.c
  testcpuid.c
  Generating Code...
  Compiling...
  t38_terminal.c
  t38_non_ecm_buffer.c
  t38_gateway.c
  t38_core.c
  t35.c
  t31.c
  t30_logging.c
  t30_api.c
  t30.c
  t4_tx.c
  t4_rx.c
  swept_tone.c
  super_tone_tx.c
  super_tone_rx.c
  silence_gen.c
  sig_tone.c
  schedule.c
  queue.c
  power_meter.c
  plc.c
  Generating Code...
  Compiling...
  playout.c
  oki_adpcm.c
  noise.c
  modem_connect_tones.c
  modem_echo.c
  lpc10_voicing.c
  lpc10_placev.c
  lpc10_encode.c
  lpc10_decode.c
  lpc10_analyse.c
  logging.c
  ima_adpcm.c
  hdlc.c
  gsm0610_short_term.c
  gsm0610_rpe.c
  gsm0610_preprocess.c
  gsm0610_lpc.c
  gsm0610_long_term.c
  gsm0610_encode.c
  gsm0610_decode.c
  Generating Code...
  Compiling...
  g726.c
  g722.c
  g711.c
  fsk.c
  fax_modems.c
  fax.c
  echo.c
  dtmf.c
  dds_int.c
  dds_float.c
  crc.c
  complex_vector_int.c
  complex_vector_float.c
  complex_filters.c
  bitstream.c
  bit_operations.c
  bert.c
  bell_r2_mf.c
  awgn.c
  at_interpreter.c
  Generating Code...
  Compiling...
  async.c
  adsi.c
  Generating Code...
     Creating library C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\libspandsp.lib and object C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\libspandsp.exp
     Creating library C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\libspandsp.lib and object C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\libspandsp.exp
  libspandsp.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\libspandsp.dll
------ Build started: Project: libsqlite, Configuration: Debug Win32 ------
  where.c
  vtab.c
  vdbemem.c
  vdbefifo.c
  vdbeaux.c
  vdbeapi.c
  vdbe.c
  vacuum.c
  util.c
  utf.c
  update.c
  trigger.c
  tokenize.c
  table.c
  shell.c
  select.c
  random.c
  printf.c
  prepare.c
  pragma.c
  Generating Code...
  Compiling...
  parse.c
  pager.c
  os_win.c
  os.c
  opcodes.c
  main.c
  loadext.c
  legacy.c
  insert.c
  hash.c
  func.c
  expr.c
  delete.c
  date.c
  complete.c
  callback.c
  build.c
  btree.c
  auth.c
  attach.c
  Generating Code...
  Compiling...
  analyze.c
  alter.c
  Generating Code...
  sqlite.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\sqlite\Win32\Debug\libsqlite.lib
------ Build started: Project: libaprutil, Configuration: Debug Win32 ------
  The system cannot find the file specified.
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_anylock.h
  The system cannot find the file specified.
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_base64.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_buckets.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_date.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_dbd.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_dbm.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_hooks.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_ldap.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_ldap_init.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_ldap_option.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_ldap_url.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_md4.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_md5.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_optional.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_optional_hooks.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_queue.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_reslist.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_rmm.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_sdbm.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_sha1.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_strmatch.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_uri.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_uuid.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_xlate.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_xml.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apu.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apu_config.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apu_select_dbm.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apu_version.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apu_want.h
  30 File(s) copied
  Creating apr_ldap.h from apr_ldap.hw
  Creating apu.h from apu.hw
  Creating apu_config.h from apu_config.hw
  Creating apu_select_dbm.h from apu_select_dbm.hw
  Creating apu_want.h from apu_want.hw
  apr_md4.c
  apr_md5.c
  apr_sha1.c
  getuuid.c
  uuid.c
  apr_base64.c
  apr_queue.c
  xlate.c
  Generating Code...
     Creating library C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\libaprutil.lib and object C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\libaprutil.exp
  libaprutil.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\libaprutil.dll
------ Build started: Project: Download OGG, Configuration: Debug Win32 ------
  Downloading OGG.
  Downloading: http://downloads.xiph.org/releases/ogg/libogg-1.1.3.tar.gz
  Extracting: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\libogg-1.1.3.tar.gz
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\libogg-1.1.3.tar.gz
  
  Extracting  libogg-1.1.3.tar
  
  Everything is Ok
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\libogg-1.1.3.tar
  
  Extracting  libogg-1.1.3
  Extracting  libogg-1.1.3\doc
  Extracting  libogg-1.1.3\doc\white-ogg.png
  Extracting  libogg-1.1.3\doc\framing.html
  Extracting  libogg-1.1.3\doc\index.html
  Extracting  libogg-1.1.3\doc\ogg-multiplex.html
  Extracting  libogg-1.1.3\doc\Makefile.am
  Extracting  libogg-1.1.3\doc\Makefile.in
  Extracting  libogg-1.1.3\doc\white-xifish.png
  Extracting  libogg-1.1.3\doc\libogg
  Extracting  libogg-1.1.3\doc\libogg\ogg_page_continued.html
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  Extracting  libogg-1.1.3\doc\libogg\ogg_page_pageno.html
  Extracting  libogg-1.1.3\doc\libogg\encoding.html
  Extracting  libogg-1.1.3\doc\libogg\datastructures.html
  Extracting  libogg-1.1.3\doc\libogg\ogg_stream_packetin.html
  Extracting  libogg-1.1.3\doc\libogg\oggpack_look1.html
  Extracting  libogg-1.1.3\doc\libogg\oggpack_writeclear.html
  Extracting  libogg-1.1.3\doc\libogg\oggpack_read.html
  Extracting  libogg-1.1.3\doc\libogg\style.css
  Extracting  libogg-1.1.3\doc\libogg\ogg_stream_eos.html
  Extracting  libogg-1.1.3\doc\libogg\oggpack_write.html
  Extracting  libogg-1.1.3\doc\libogg\ogg_stream_reset_serialno.html
  Extracting  libogg-1.1.3\doc\libogg\oggpack_adv1.html
  Extracting  libogg-1.1.3\doc\libogg\ogg_page_eos.html
  Extracting  libogg-1.1.3\doc\libogg\vorbis_info.html
  Extracting  libogg-1.1.3\doc\libogg\general.html
  Extracting  libogg-1.1.3\doc\libogg\oggpack_readinit.html
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  Extracting  libogg-1.1.3\doc\libogg\ogg_page_packets.html
  Extracting  libogg-1.1.3\doc\libogg\ogg_sync_reset.html
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  Extracting  libogg-1.1.3\doc\libogg\oggpack_writetrunc.html
  Extracting  libogg-1.1.3\doc\libogg\oggpack_bits.html
  Extracting  libogg-1.1.3\doc\libogg\ogg_packet.html
  Extracting  libogg-1.1.3\doc\libogg\index.html
  Extracting  libogg-1.1.3\doc\libogg\ogg_stream_flush.html
  Extracting  libogg-1.1.3\doc\libogg\ogg_packet_clear.html
  Extracting  libogg-1.1.3\doc\libogg\oggpack_writeinit.html
  Extracting  libogg-1.1.3\doc\libogg\ogg_stream_reset.html
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  Extracting  libogg-1.1.3\doc\libogg\ogg_stream_state.html
  Extracting  libogg-1.1.3\doc\libogg\ogg_page_checksum_set.html
  Extracting  libogg-1.1.3\doc\libogg\ogg_page_serialno.html
  Extracting  libogg-1.1.3\doc\libogg\ogg_page_bos.html
  Extracting  libogg-1.1.3\doc\libogg\overview.html
  Extracting  libogg-1.1.3\doc\libogg\oggpack_adv.html
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  Extracting  libogg-1.1.3\doc\libogg\oggpack_bytes.html
  Extracting  libogg-1.1.3\doc\libogg\Makefile.am
  Extracting  libogg-1.1.3\doc\libogg\Makefile.in
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  Extracting  libogg-1.1.3\doc\libogg\ogg_page_granulepos.html
  Extracting  libogg-1.1.3\doc\libogg\oggpack_look.html
  Extracting  libogg-1.1.3\doc\libogg\bitpacking.html
  Extracting  libogg-1.1.3\doc\libogg\ogg_page.html
  Extracting  libogg-1.1.3\doc\libogg\ogg_stream_packetout.html
  Extracting  libogg-1.1.3\doc\libogg\decoding.html
  Extracting  libogg-1.1.3\doc\libogg\ogg_sync_init.html
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  Extracting  libogg-1.1.3\doc\libogg\vorbis_comment.html
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  Extracting  libogg-1.1.3\doc\libogg\ogg_stream_pagein.html
  Extracting  libogg-1.1.3\doc\libogg\ogg_page_version.html
  Extracting  libogg-1.1.3\doc\libogg\ogg_sync_destroy.html
  Extracting  libogg-1.1.3\doc\libogg\oggpack_buffer.html
  Extracting  libogg-1.1.3\doc\libogg\ogg_stream_init.html
  Extracting  libogg-1.1.3\doc\vorbisword2.png
  Extracting  libogg-1.1.3\doc\stream.png
  Extracting  libogg-1.1.3\doc\oggstream.html
  Extracting  libogg-1.1.3\doc\rfc3533.txt
  Extracting  libogg-1.1.3\doc\rfc3534.txt
  Extracting  libogg-1.1.3\src
  Extracting  libogg-1.1.3\src\framing.c
  Extracting  libogg-1.1.3\src\bitwise.c
  Extracting  libogg-1.1.3\src\Makefile.am
  Extracting  libogg-1.1.3\src\Makefile.in
  Extracting  libogg-1.1.3\compile
  Extracting  libogg-1.1.3\depcomp
  Extracting  libogg-1.1.3\aclocal.m4
  Extracting  libogg-1.1.3\macos
  Extracting  libogg-1.1.3\macos\libogg.mcp
  Extracting  libogg-1.1.3\macos\libogg.mcp.exp
  Extracting  libogg-1.1.3\macos\compat
  Extracting  libogg-1.1.3\macos\compat\sys
  Extracting  libogg-1.1.3\macos\compat\sys\types.h
  Extracting  libogg-1.1.3\macos\compat\strdup.c
  Extracting  libogg-1.1.3\win32
  Extracting  libogg-1.1.3\win32\ogg_dynamic.dsp
  Extracting  libogg-1.1.3\win32\build_ogg_dynamic_debug.bat
  Extracting  libogg-1.1.3\win32\build_ogg_dynamic.bat
  Extracting  libogg-1.1.3\win32\Makefile.am
  Extracting  libogg-1.1.3\win32\Makefile.in
  Extracting  libogg-1.1.3\win32\build_ogg_static_debug.bat
  Extracting  libogg-1.1.3\win32\ogg_static.dsp
  Extracting  libogg-1.1.3\win32\ogg.def
  Extracting  libogg-1.1.3\win32\ogg.dsw
  Extracting  libogg-1.1.3\win32\build_ogg_static.bat
  Extracting  libogg-1.1.3\README
  Extracting  libogg-1.1.3\ltmain.sh
  Extracting  libogg-1.1.3\configure
  Extracting  libogg-1.1.3\configure.in
  Extracting  libogg-1.1.3\config.guess
  Extracting  libogg-1.1.3\install-sh
  Extracting  libogg-1.1.3\config.sub
  Extracting  libogg-1.1.3\missing
  Extracting  libogg-1.1.3\debian
  Extracting  libogg-1.1.3\debian\control
  Extracting  libogg-1.1.3\debian\libogg-dev.docs
  Extracting  libogg-1.1.3\debian\rules
  Extracting  libogg-1.1.3\debian\watch
  Extracting  libogg-1.1.3\debian\changelog
  Extracting  libogg-1.1.3\debian\libogg0.README.Debian
  Extracting  libogg-1.1.3\debian\libogg-dev.install
  Extracting  libogg-1.1.3\debian\copyright
  Extracting  libogg-1.1.3\debian\libogg0.install
  Extracting  libogg-1.1.3\debian\.cvsignore
  Extracting  libogg-1.1.3\libogg.spec.in
  Extracting  libogg-1.1.3\ogg.pc.in
  Extracting  libogg-1.1.3\Makefile.am
  Extracting  libogg-1.1.3\Makefile.in
  Extracting  libogg-1.1.3\macosx
  Extracting  libogg-1.1.3\macosx\Ogg.xcodeproj
  Extracting  libogg-1.1.3\macosx\Ogg.xcodeproj\project.pbxproj
  Extracting  libogg-1.1.3\macosx\Ogg_Prefix.pch
  Extracting  libogg-1.1.3\macosx\English.lproj
  Extracting  libogg-1.1.3\macosx\English.lproj\InfoPlist.strings
  Extracting  libogg-1.1.3\macosx\Info.plist
  Extracting  libogg-1.1.3\config.h.in
  Extracting  libogg-1.1.3\ogg-uninstalled.pc.in
  Extracting  libogg-1.1.3\ogg.m4
  Extracting  libogg-1.1.3\AUTHORS
  Extracting  libogg-1.1.3\CHANGES
  Extracting  libogg-1.1.3\include
  Extracting  libogg-1.1.3\include\ogg
  Extracting  libogg-1.1.3\include\ogg\ogg.h
  Extracting  libogg-1.1.3\include\ogg\Makefile.am
  Extracting  libogg-1.1.3\include\ogg\Makefile.in
  Extracting  libogg-1.1.3\include\ogg\config_types.h.in
  Extracting  libogg-1.1.3\include\ogg\os_types.h
  Extracting  libogg-1.1.3\include\Makefile.am
  Extracting  libogg-1.1.3\include\Makefile.in
  Extracting  libogg-1.1.3\COPYING
  Extracting  libogg-1.1.3\libogg.spec
  
  Everything is Ok
------ Build started: Project: libpcre, Configuration: Debug Win32 ------
  pcre_xclass.c
  pcre_version.c
  pcre_valid_utf8.c
  pcre_ucp_searchfuncs.c
..\..\pcre\pcre_ucp_searchfuncs.c(158): warning C4018: '<' : signed/unsigned mismatch
..\..\pcre\pcre_ucp_searchfuncs.c(163): warning C4018: '<=' : signed/unsigned mismatch
  pcre_ucd.c
  pcre_newline.c
  pcre_try_flipped.c
  pcre_tables.c
  pcre_study.c
  pcre_refcount.c
  pcre_ord2utf8.c
  pcre_maketables.c
  pcre_info.c
  pcre_globals.c
  pcre_get.c
  pcre_fullinfo.c
  pcre_exec.c
  pcre_dfa_exec.c
  pcre_config.c
  pcre_compile.c
  Generating Code...
  Compiling...
  pcre_chartables.c
c1 : fatal error C1083: Cannot open source file: 'pcre_chartables.c': No such file or directory
  Generating Code...
------ Build started: Project: Download JSON, Configuration: Debug Win32 ------
  Downloading JSON.
  Downloading: http://files.freeswitch.org/downloads/libs/json-c-0.9.tar.gz
  Extracting: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\json-c-0.9.tar.gz
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\json-c-0.9.tar.gz
  
  Extracting  json-c-0.9.tar
  
  Everything is Ok
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\json-c-0.9.tar
  
  Extracting  json-c-0.9
  Extracting  json-c-0.9\aclocal.m4
  Extracting  json-c-0.9\arraylist.c
  Extracting  json-c-0.9\arraylist.h
  Extracting  json-c-0.9\AUTHORS
  Extracting  json-c-0.9\bits.h
  Extracting  json-c-0.9\ChangeLog
  Extracting  json-c-0.9\config.guess
  Extracting  json-c-0.9\config.h.in
  Extracting  json-c-0.9\config.h.win32
  Extracting  json-c-0.9\config.sub
  Extracting  json-c-0.9\configure
  Extracting  json-c-0.9\configure.in
  Extracting  json-c-0.9\COPYING
  Extracting  json-c-0.9\debug.c
  Extracting  json-c-0.9\debug.h
  Extracting  json-c-0.9\depcomp
  Extracting  json-c-0.9\doc
  Extracting  json-c-0.9\doc\html
  Extracting  json-c-0.9\doc\html\annotated.html
  Extracting  json-c-0.9\doc\html\arraylist_8h.html
  Extracting  json-c-0.9\doc\html\bits_8h.html
  Extracting  json-c-0.9\doc\html\classes.html
  Extracting  json-c-0.9\doc\html\config_8h.html
  Extracting  json-c-0.9\doc\html\debug_8h.html
  Extracting  json-c-0.9\doc\html\doxygen.css
  Extracting  json-c-0.9\doc\html\doxygen.png
  Extracting  json-c-0.9\doc\html\files.html
  Extracting  json-c-0.9\doc\html\functions.html
  Extracting  json-c-0.9\doc\html\functions_vars.html
  Extracting  json-c-0.9\doc\html\globals.html
  Extracting  json-c-0.9\doc\html\globals_defs.html
  Extracting  json-c-0.9\doc\html\globals_enum.html
  Extracting  json-c-0.9\doc\html\globals_eval.html
  Extracting  json-c-0.9\doc\html\globals_func.html
  Extracting  json-c-0.9\doc\html\globals_type.html
  Extracting  json-c-0.9\doc\html\globals_vars.html
  Extracting  json-c-0.9\doc\html\index.html
  Extracting  json-c-0.9\doc\html\json_8h.html
  Extracting  json-c-0.9\doc\html\json__object_8h.html
  Extracting  json-c-0.9\doc\html\json__object__private_8h.html
  Extracting  json-c-0.9\doc\html\json__tokener_8h.html
  Extracting  json-c-0.9\doc\html\json__util_8h.html
  Extracting  json-c-0.9\doc\html\linkhash_8h.html
  Extracting  json-c-0.9\doc\html\printbuf_8h.html
  Extracting  json-c-0.9\doc\html\structarray__list.html
  Extracting  json-c-0.9\doc\html\structjson__object.html
  Extracting  json-c-0.9\doc\html\structjson__object__iter.html
  Extracting  json-c-0.9\doc\html\structjson__tokener.html
  Extracting  json-c-0.9\doc\html\structjson__tokener__srec.html
  Extracting  json-c-0.9\doc\html\structlh__entry.html
  Extracting  json-c-0.9\doc\html\structlh__table.html
  Extracting  json-c-0.9\doc\html\structprintbuf.html
  Extracting  json-c-0.9\doc\html\tabs.css
  Extracting  json-c-0.9\doc\html\tab_b.gif
  Extracting  json-c-0.9\doc\html\tab_l.gif
  Extracting  json-c-0.9\doc\html\tab_r.gif
  Extracting  json-c-0.9\doc\html\unionjson__object_1_1data.html
  Extracting  json-c-0.9\INSTALL
  Extracting  json-c-0.9\install-sh
  Extracting  json-c-0.9\json.h
  Extracting  json-c-0.9\json.pc.in
  Extracting  json-c-0.9\json_object.c
  Extracting  json-c-0.9\json_object.h
  Extracting  json-c-0.9\json_object_private.h
  Extracting  json-c-0.9\json_tokener.c
  Extracting  json-c-0.9\json_tokener.h
  Extracting  json-c-0.9\json_util.c
  Extracting  json-c-0.9\json_util.h
  Extracting  json-c-0.9\linkhash.c
  Extracting  json-c-0.9\linkhash.h
  Extracting  json-c-0.9\ltmain.sh
  Extracting  json-c-0.9\Makefile.am
  Extracting  json-c-0.9\Makefile.in
  Extracting  json-c-0.9\missing
  Extracting  json-c-0.9\NEWS
  Extracting  json-c-0.9\printbuf.c
  Extracting  json-c-0.9\printbuf.h
  Extracting  json-c-0.9\README
  Extracting  json-c-0.9\README-WIN32.html
  Extracting  json-c-0.9\README.html
  Extracting  json-c-0.9\test1.c
  Extracting  json-c-0.9\test2.c
  Extracting  json-c-0.9\test3.c
  
  Everything is Ok
------ Build started: Project: Download FLITE, Configuration: Debug Win32 ------
  Downloading Flite.
  Downloading: http://files.freeswitch.org/downloads/libs/flite-1.3.99-latest.tar.gz
  Extracting: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\flite-1.3.99-latest.tar.gz
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\flite-1.3.99-latest.tar.gz
  
  Extracting  flite-1.3.99-latest.tar
  
  Everything is Ok
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\flite-1.3.99-latest.tar
  
  Extracting  flite-1.3.99
  Extracting  flite-1.3.99\configure
  Extracting  flite-1.3.99\config.sub
  Extracting  flite-1.3.99\include
  Extracting  flite-1.3.99\include\cst_utt_utils.h
  Extracting  flite-1.3.99\include\cst_diphone.h
  Extracting  flite-1.3.99\include\cst_synth.h
  Extracting  flite-1.3.99\include\cst_error.h
  Extracting  flite-1.3.99\include\cst_sts.h
  Extracting  flite-1.3.99\include\cst_track.h
  Extracting  flite-1.3.99\include\cst_math.h
  Extracting  flite-1.3.99\include\cst_features.h
  Extracting  flite-1.3.99\include\cst_lts_rewrites.h
  Extracting  flite-1.3.99\include\cst_alloc.h
  Extracting  flite-1.3.99\include\cst_hrg.h
  Extracting  flite-1.3.99\include\cst_lts.h
  Extracting  flite-1.3.99\include\cst_cart.h
  Extracting  flite-1.3.99\include\cst_ss.h
  Extracting  flite-1.3.99\include\cst_audio.h
  Extracting  flite-1.3.99\include\cst_string.h
  Extracting  flite-1.3.99\include\cst_tokenstream.h
  Extracting  flite-1.3.99\include\cst_item.h
  Extracting  flite-1.3.99\include\cst_units.h
  Extracting  flite-1.3.99\include\cst_relation.h
  Extracting  flite-1.3.99\include\cst_val_const.h
  Extracting  flite-1.3.99\include\cst_val.h
  Extracting  flite-1.3.99\include\cst_phoneset.h
  Extracting  flite-1.3.99\include\cst_file.h
  Extracting  flite-1.3.99\include\cst_cg.h
  Extracting  flite-1.3.99\include\cst_lexicon.h
  Extracting  flite-1.3.99\include\cst_wchar.h
  Extracting  flite-1.3.99\include\cst_args.h
  Extracting  flite-1.3.99\include\flite.h
  Extracting  flite-1.3.99\include\cst_utterance.h
  Extracting  flite-1.3.99\include\cst_val_defs.h
  Extracting  flite-1.3.99\include\cst_sigpr.h
  Extracting  flite-1.3.99\include\cst_clunits.h
  Extracting  flite-1.3.99\include\cst_endian.h
  Extracting  flite-1.3.99\include\cst_socket.h
  Extracting  flite-1.3.99\include\cst_regex.h
  Extracting  flite-1.3.99\include\cst_ffeatures.h
  Extracting  flite-1.3.99\include\cst_viterbi.h
  Extracting  flite-1.3.99\include\cst_voice.h
  Extracting  flite-1.3.99\include\Makefile
  Extracting  flite-1.3.99\include\cst_wave.h
  Extracting  flite-1.3.99\wince
  Extracting  flite-1.3.99\wince\flowm.h
  Extracting  flite-1.3.99\wince\flowm.rc
  Extracting  flite-1.3.99\wince\flowm_flite.c
  Extracting  flite-1.3.99\wince\flowm.bmp
  Extracting  flite-1.3.99\wince\flowm.notes
  Extracting  flite-1.3.99\wince\flowm_main.c
  Extracting  flite-1.3.99\wince\flowm.ico
  Extracting  flite-1.3.99\wince\Makefile
  Extracting  flite-1.3.99\sapi
  Extracting  flite-1.3.99\sapi\usenglish
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  Extracting  flite-1.3.99\sapi\FliteCMUKalDiphone\FliteCMUKalDiphoneps.mk
  Extracting  flite-1.3.99\sapi\FliteCMUKalDiphone\FliteCMUKalDiphone.def
  Extracting  flite-1.3.99\sapi\FliteCMUKalDiphone\FliteCMUKalDiphoneObj.h
  Extracting  flite-1.3.99\sapi\FliteCMUKalDiphone\FliteCMUKalDiphone.dsp
  Extracting  flite-1.3.99\sapi\FliteCMUKalDiphone\FliteCMUKalDiphoneps.def
  Extracting  flite-1.3.99\sapi\FliteCMUKalDiphone\FliteCMUKalDiphoneObj.rgs
  Extracting  flite-1.3.99\sapi\FliteCMUKalDiphone\StdAfx.cpp
  Extracting  flite-1.3.99\sapi\FliteCMUKalDiphone\StdAfx.h
  Extracting  flite-1.3.99\sapi\FliteCMUKalDiphone\FliteCMUKalDiphoneObj.cpp
  Extracting  flite-1.3.99\sapi\FliteCMUKalDiphone\register_vox
  Extracting  flite-1.3.99\sapi\FliteCMUKalDiphone\register_vox\register_vox.dsp
  Extracting  flite-1.3.99\sapi\FliteCMUKalDiphone\register_vox\register-vox.cpp
  Extracting  flite-1.3.99\sapi\FliteCMUKalDiphone\register_vox\Makefile
  Extracting  flite-1.3.99\sapi\FliteCMUKalDiphone\resource.h
  Extracting  flite-1.3.99\sapi\FliteCMUKalDiphone\FliteCMUKalDiphone.rc
  Extracting  flite-1.3.99\sapi\FliteCMUKalDiphone\FliteCMUKalDiphone.cpp
  Extracting  flite-1.3.99\sapi\FliteCMUKalDiphone\FliteCMUKalDiphone.idl
  Extracting  flite-1.3.99\sapi\FliteCMUKalDiphone\Makefile
  Extracting  flite-1.3.99\sapi\flite_sapi.dsw
  Extracting  flite-1.3.99\sapi\cmu_us_kal
  Extracting  flite-1.3.99\sapi\cmu_us_kal\cmu_us_kal.dsp
  Extracting  flite-1.3.99\sapi\cmu_us_kal\Makefile
  Extracting  flite-1.3.99\sapi\FliteTTSEngineObj
  Extracting  flite-1.3.99\sapi\FliteTTSEngineObj\FliteTTSEngineObj.h
  Extracting  flite-1.3.99\sapi\FliteTTSEngineObj\FliteTTSEngineObj.cpp
  Extracting  flite-1.3.99\sapi\FliteTTSEngineObj\flite_sapi_usenglish.c
  Extracting  flite-1.3.99\sapi\FliteTTSEngineObj\FliteTTSEngineObj.dsp
  Extracting  flite-1.3.99\sapi\FliteTTSEngineObj\flite_sapi_usenglish.h
  Extracting  flite-1.3.99\sapi\FliteTTSEngineObj\Makefile
  Extracting  flite-1.3.99\sapi\cmulex
  Extracting  flite-1.3.99\sapi\cmulex\cmulex.dsp
  Extracting  flite-1.3.99\sapi\cmulex\Makefile
  Extracting  flite-1.3.99\sapi\README
  Extracting  flite-1.3.99\sapi\Makefile
  Extracting  flite-1.3.99\lang
  Extracting  flite-1.3.99\lang\cmu_us_awb
  Extracting  flite-1.3.99\lang\cmu_us_awb\cmu_us_awb_cg_durmodel.c
  Extracting  flite-1.3.99\lang\cmu_us_awb\cmu_us_awb_cg_mcep_trees.c
  Extracting  flite-1.3.99\lang\cmu_us_awb\cmu_us_awb_cg_f0_trees.c
  Extracting  flite-1.3.99\lang\cmu_us_awb\cmu_us_awb_cg_durmodel.h
  Extracting  flite-1.3.99\lang\cmu_us_awb\cmu_us_awb.c
  Extracting  flite-1.3.99\lang\cmu_us_awb\cmu_us_awb_cg_f0_trees.h
  Extracting  flite-1.3.99\lang\cmu_us_awb\cmu_us_awb_cg_phonestate.c
  Extracting  flite-1.3.99\lang\cmu_us_awb\cmu_us_awb_cg_mcep_trees.h
  Extracting  flite-1.3.99\lang\cmu_us_awb\voxdefs.h
  Extracting  flite-1.3.99\lang\cmu_us_awb\cmu_us_awb_cg_params.c
  Extracting  flite-1.3.99\lang\cmu_us_awb\cmu_us_awb_cg.c
  Extracting  flite-1.3.99\lang\cmu_us_awb\Makefile
  Extracting  flite-1.3.99\lang\usenglish
  Extracting  flite-1.3.99\lang\usenglish\us_int_accent_cart.h
  Extracting  flite-1.3.99\lang\usenglish\usenglish.c
  Extracting  flite-1.3.99\lang\usenglish\us_nums_cart.c
  Extracting  flite-1.3.99\lang\usenglish\us_phrasing_cart.c
  Extracting  flite-1.3.99\lang\usenglish\us_durz_cart.h
  Extracting  flite-1.3.99\lang\usenglish\make_us_regexes
  Extracting  flite-1.3.99\lang\usenglish\us_text.h
  Extracting  flite-1.3.99\lang\usenglish\us_f0lr.c
  Extracting  flite-1.3.99\lang\usenglish\usenglish.h
  Extracting  flite-1.3.99\lang\usenglish\us_regexes.h
  Extracting  flite-1.3.99\lang\usenglish\us_text.c
  Extracting  flite-1.3.99\lang\usenglish\us_int_accent_cart.c
  Extracting  flite-1.3.99\lang\usenglish\us_expand.c
  Extracting  flite-1.3.99\lang\usenglish\us_f0.h
  Extracting  flite-1.3.99\lang\usenglish\us_dur_stats.c
  Extracting  flite-1.3.99\lang\usenglish\us_int_tone_cart.h
  Extracting  flite-1.3.99\lang\usenglish\us_phrasing_cart.h
  Extracting  flite-1.3.99\lang\usenglish\us_phoneset.c
  Extracting  flite-1.3.99\lang\usenglish\us_f0_model.c
  Extracting  flite-1.3.99\lang\usenglish\us_int_tone_cart.c
  Extracting  flite-1.3.99\lang\usenglish\us_ffeatures.c
  Extracting  flite-1.3.99\lang\usenglish\us_gpos.c
  Extracting  flite-1.3.99\lang\usenglish\us_nums_cart.h
  Extracting  flite-1.3.99\lang\usenglish\us_ffeatures.h
  Extracting  flite-1.3.99\lang\usenglish\us_aswd.c
  Extracting  flite-1.3.99\lang\usenglish\us_durz_cart.c
  Extracting  flite-1.3.99\lang\usenglish\Makefile
  Extracting  flite-1.3.99\lang\cmu_us_slt
  Extracting  flite-1.3.99\lang\cmu_us_slt\cmu_us_slt.c
  Extracting  flite-1.3.99\lang\cmu_us_slt\cmu_us_slt_cg_phonestate.c
  Extracting  flite-1.3.99\lang\cmu_us_slt\cmu_us_slt_cg.c
  Extracting  flite-1.3.99\lang\cmu_us_slt\cmu_us_slt_cg_mcep_trees.h
  Extracting  flite-1.3.99\lang\cmu_us_slt\cmu_us_slt_cg_f0_trees.h
  Extracting  flite-1.3.99\lang\cmu_us_slt\cmu_us_slt_cg_durmodel.h
  Extracting  flite-1.3.99\lang\cmu_us_slt\cmu_us_slt_cg_durmodel.c
  Extracting  flite-1.3.99\lang\cmu_us_slt\voxdefs.h
  Extracting  flite-1.3.99\lang\cmu_us_slt\cmu_us_slt_cg_mcep_trees.c
  Extracting  flite-1.3.99\lang\cmu_us_slt\cmu_us_slt_cg_f0_trees.c
  Extracting  flite-1.3.99\lang\cmu_us_slt\cmu_us_slt_cg_params.c
  Extracting  flite-1.3.99\lang\cmu_us_slt\Makefile
  Extracting  flite-1.3.99\lang\cmu_time_awb
  Extracting  flite-1.3.99\lang\cmu_time_awb\cmu_time_awb_clunits.c
  Extracting  flite-1.3.99\lang\cmu_time_awb\cmu_time_awb_lpc.c
  Extracting  flite-1.3.99\lang\cmu_time_awb\cmu_time_awb_lex_entry.c
  Extracting  flite-1.3.99\lang\cmu_time_awb\cmu_time_awb_mcep.c
  Extracting  flite-1.3.99\lang\cmu_time_awb\cmu_time_awb_cart.c
  Extracting  flite-1.3.99\lang\cmu_time_awb\cmu_time_awb.c
  Extracting  flite-1.3.99\lang\cmu_time_awb\voxdefs.h
  Extracting  flite-1.3.99\lang\cmu_time_awb\Makefile
  Extracting  flite-1.3.99\lang\cmu_us_rms
  Extracting  flite-1.3.99\lang\cmu_us_rms\cmu_us_rms_cg_phonestate.c
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  Everything is Ok
------ Build started: Project: Download LAME, Configuration: Debug Win32 ------
  Downloading Lame.
  Downloading: http://files.freeswitch.org/downloads/libs/lame-3.97.tar.gz
  Extracting: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\lame-3.97.tar.gz
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\lame-3.97.tar.gz
  
  Extracting  lame-3.97.tar
  
  Everything is Ok
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\lame-3.97.tar
  
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  Everything is Ok
------ Build started: Project: Download CELT, Configuration: Debug Win32 ------
  Downloading CELT.
  Downloading: http://files.freeswitch.org/downloads/libs/celt-0.7.1.tar.gz
  Extracting: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\celt-0.7.1.tar.gz
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\celt-0.7.1.tar.gz
  
  Extracting  celt-0.7.1.tar
  
  Everything is Ok
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\celt-0.7.1.tar
  
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  Extracting  celt-0.7.1\libcelt\kiss_fft.c
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  Extracting  celt-0.7.1\libcelt\Makefile.in
  Extracting  celt-0.7.1\libcelt\plc.c
  Extracting  celt-0.7.1\COPYING
  Extracting  celt-0.7.1\NEWS
  Extracting  celt-0.7.1\install-sh
  Extracting  celt-0.7.1\Doxyfile.devel
  Extracting  celt-0.7.1\ltmain.sh
  Extracting  celt-0.7.1\ChangeLog
  Extracting  celt-0.7.1\config.guess
  Extracting  celt-0.7.1\acinclude.m4
  Extracting  celt-0.7.1\tools
  Extracting  celt-0.7.1\tools\Makefile.am
  Extracting  celt-0.7.1\tools\getopt_win.h
  Extracting  celt-0.7.1\tools\skeleton.h
  Extracting  celt-0.7.1\tools\getopt1.c
  Extracting  celt-0.7.1\tools\celtenc.c
  Extracting  celt-0.7.1\tools\wave_out.h
  Extracting  celt-0.7.1\tools\wav_io.c
  Extracting  celt-0.7.1\tools\getopt.c
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  Extracting  celt-0.7.1\tools\celtdec.c
  Extracting  celt-0.7.1\tools\wave_out.c
  Extracting  celt-0.7.1\tools\Makefile.in
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  Extracting  celt-0.7.1\configure
  Extracting  celt-0.7.1\aclocal.m4
  Extracting  celt-0.7.1\missing
  Extracting  celt-0.7.1\Makefile.in
  
  Everything is Ok
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\\celt\config.h
  1 File(s) copied
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\\celt\float_cast.h
  1 File(s) copied
------ Build started: Project: iksemel, Configuration: Debug Win32 ------
  utility.c
  stream.c
  sha.c
  sax.c
  md5.c
  jabber.c
  io-posix.c
  ikstack.c
  iks.c
  filter.c
  dom.c
  base64.c
  Generating Code...
  iksemel.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\iksemel\Win32\Debug\iksemel.lib
------ Build started: Project: Download LIBSHOUT, Configuration: Debug Win32 ------
  Downloading Flite.
  Downloading: http://files.freeswitch.org/downloads/libs/libshout-2.2.2.tar.gz
  Extracting: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\libshout-2.2.2.tar.gz
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\libshout-2.2.2.tar.gz
  
  Extracting  libshout-2.2.2.tar
  
  Everything is Ok
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\libshout-2.2.2.tar
  
  Extracting  libshout-2.2.2
  Extracting  libshout-2.2.2\m4
  Extracting  libshout-2.2.2\m4\vorbis.m4
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  Extracting  libshout-2.2.2\win32\Makefile.in
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  Extracting  libshout-2.2.2\win32\libshout.dsw
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  Extracting  libshout-2.2.2\depcomp
  
  Everything is Ok
------ Build started: Project: curllib, Configuration: Debug Win32 ------
  version.c
  url.c
  transfer.c
  timeval.c
  tftp.c
  telnet.c
  strtoofft.c
  strtok.c
  strerror.c
  strequal.c
  ssluse.c
  sslgen.c
  splay.c
  speedcheck.c
  socks.c
  share.c
  sendf.c
  select.c
  security.c
  progress.c
  Generating Code...
  Compiling...
  parsedate.c
  netrc.c
  multi.c
  mprintf.c
  memdebug.c
  md5.c
  llist.c
  ldap.c
  krb4.c
  inet_pton.c
  inet_ntop.c
  if2ip.c
  http_ntlm.c
  http_negotiate.c
  http_digest.c
  http_chunks.c
  http.c
  hostthre.c
  hostsyn.c
  hostip6.c
  Generating Code...
  Compiling...
  hostip4.c
  hostip.c
  hostasyn.c
  hostares.c
  hash.c
  gtls.c
  getinfo.c
  getenv.c
  ftp.c
  formdata.c
  file.c
  escape.c
  easy.c
  dict.c
  cookie.c
  content_encoding.c
  connect.c
  base64.c
  Generating Code...
  curllib.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\curl\Win32\Debug\curllib.lib
------ Build started: Project: js, Configuration: Debug Win32 ------
  win32_errors.c
  w32poll.c
  pruthr.c
  prustack.c
  prulock.c
  prucv.c
  prucpu.c
  prtpd.c
  prtime.c
  prseg.c
  prprf.c
  prosdep.c
  prolock.c
  prmwait.c
  prmmap.c
  prmem.c
  prlog.c
  prlayer.c
  priometh.c
  prio.c
  Generating Code...
  Compiling...
  prinrval.c
  prinit.c
  prfile.c
  prfdcach.c
  prerror.c
  prdir.c
  prcthr.c
  pratom.c
  ntthread.c
  ntsec.c
  ntmisc.c
  ntio.c
  ntinrval.c
  w_sqrt.c
  w_sinh.c
  w_scalb.c
  w_remainder.c
  w_pow.c
  w_log10.c
  w_log.c
  Generating Code...
  Compiling...
  w_lgamma_r.c
  w_lgamma.c
  w_jn.c
  w_j1.c
  w_j0.c
  w_hypot.c
  w_gamma_r.c
  w_gamma.c
  w_fmod.c
  w_exp.c
  w_cosh.c
  w_atanh.c
  w_atan2.c
  w_asin.c
  w_acosh.c
  w_acos.c
  s_tanh.c
  s_tan.c
  s_sin.c
  s_significand.c
  Generating Code...
  Compiling...
  s_signgam.c
  s_scalbn.c
  s_rint.c
  s_nextafter.c
  s_modf.c
  s_matherr.c
  s_logb.c
  s_log1p.c
  s_lib_version.c
  s_ldexp.c
  s_isnan.c
  s_ilogb.c
  s_frexp.c
  s_floor.c
  s_finite.c
  s_fabs.c
  s_expm1.c
  s_erf.c
  s_cos.c
  s_copysign.c
  Generating Code...
  Compiling...
  s_ceil.c
  s_cbrt.c
  s_atan.c
  s_asinh.c
  prmjtime.c
  k_tan.c
  k_standard.c
  k_sin.c
  k_rem_pio2.c
  k_cos.c
  jsxml.c
  jsxdrapi.c
  jsutil.c
  jsstr.c
  jsscript.c
  jsscope.c
  jsscan.c
  jsregexp.c
  jsprf.c
  jsparse.c
  Generating Code...
  Compiling...
  jsopcode.c
  jsobj.c
  jsnum.c
  jsmath.c
  jslong.c
  jslog2.c
  jslock.c
  jsinterp.c
  jshash.c
  jsgc.c
  jsfun.c
  jsfile.c
  jsexn.c
  jsemit.c
  jsdtoa.c
  jsdso.c
  jsdhash.c
  jsdbgapi.c
  jsdate.c
  jscntxt.c
  Generating Code...
  Compiling...
  jsbool.c
  jsatom.c
  jsarray.c
  jsarena.c
  jsapi.c
  e_sqrt.c
  e_sinh.c
  e_scalb.c
  e_remainder.c
  e_rem_pio2.c
  e_pow.c
  e_log10.c
  e_log.c
  e_lgamma_r.c
  e_lgamma.c
  e_jn.c
  e_j1.c
  e_j0.c
  e_hypot.c
  e_gamma_r.c
  Generating Code...
  Compiling...
  e_gamma.c
  e_fmod.c
  e_exp.c
  e_cosh.c
  e_atanh.c
  e_atan2.c
  e_asin.c
  e_acosh.c
  e_acos.c
  Generating Code...
     Creating library C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\js.lib and object C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\js.exp
  js.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\js.dll
------ Build started: Project: libeay32, Configuration: Debug Win32 ------
          1 file(s) copied.
          1 file(s) copied.
          1 file(s) copied.
          1 file(s) copied.
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\aes.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\asn1.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\asn1t.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\asn1_mac.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\bio.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\blowfish.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\bn.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\buffer.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\camellia.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\cast.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\cms.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\comp.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\conf.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\conf_api.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\crypto.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\des.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\des_old.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\dh.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\dsa.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\dso.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\dtls1.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\ebcdic.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\ec.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\ecdh.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\ecdsa.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\engine.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\err.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\evp.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\e_os.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\e_os2.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\hmac.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\idea.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\krb5_asn.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\kssl.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\lhash.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\md2.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\md4.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\md5.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\mdc2.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\modes.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\objects.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\obj_mac.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\ocsp.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\opensslv.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\ossl_typ.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\o_dir.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\o_str.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\pem.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\pem2.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\pkcs12.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\pkcs7.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\pqueue.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\pq_compat.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\rand.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\rc2.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\rc4.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\rc5.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\ripemd.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\rsa.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\safestack.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\seed.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\sha.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\ssl.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\ssl2.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\ssl23.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\ssl3.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\stack.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\store.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\symhacks.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\tls1.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\tmdiff.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\ts.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\txt_db.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\ui.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\ui_compat.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\whrlpool.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\x509.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\x509v3.h
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\openssl\\include\x509_vfy.h
  79 File(s) copied
  cpt_err.c
  cryptlib.c
  cversion.c
  ex_data.c
  mem.c
  mem_clr.c
  mem_dbg.c
  o_dir.c
  o_time.c
  uid.c
  rc2_cbc.c
  rc2_ecb.c
  rc2_skey.c
  rc2cfb64.c
  rc2ofb64.c
  rc4_enc.c
  rc4_skey.c
  i_cbc.c
  i_cfb64.c
  i_ecb.c
  Generating Code...
  Compiling...
  i_ofb64.c
  i_skey.c
  bf_cfb64.c
  bf_ecb.c
  bf_enc.c
  bf_ofb64.c
  bf_skey.c
  c_cfb64.c
  c_ecb.c
  c_enc.c
  c_ofb64.c
  c_skey.c
  rmd_dgst.c
  rmd_one.c
  cbc_cksm.c
  cbc_enc.c
  cfb64ede.c
  cfb64enc.c
  cfb_enc.c
  des_enc.c
  Generating Code...
  Compiling...
  des_old.c
  des_old2.c
  ecb3_enc.c
  ecb_enc.c
  ede_cbcm_enc.c
  enc_read.c
  enc_writ.c
  fcrypt.c
  fcrypt_b.c
  ofb64ede.c
  ofb64enc.c
  ofb_enc.c
  pcbc_enc.c
  qud_cksm.c
  rand_key.c
  read2pwd.c
  rpc_enc.c
  set_key.c
  str2key.c
  xcbc_enc.c
  Generating Code...
  Compiling...
  aes_cbc.c
  aes_cfb.c
  aes_core.c
  aes_ctr.c
  aes_ecb.c
  aes_ige.c
  aes_misc.c
  aes_ofb.c
  aes_wrap.c
  camellia.c
  cmll_cbc.c
  cmll_cfb.c
  cmll_ctr.c
  cmll_ecb.c
  cmll_misc.c
  cmll_ofb.c
  seed.c
  seed_cbc.c
  seed_cfb.c
  seed_ecb.c
  Generating Code...
  Compiling...
  seed_ofb.c
  cbc128.c
  cfb128.c
  ctr128.c
  cts128.c
  ofb128.c
  bn_add.c
  bn_asm.c
  bn_blind.c
  bn_const.c
  bn_ctx.c
  bn_depr.c
  bn_div.c
  bn_err.c
  bn_exp.c
  bn_exp2.c
  bn_gcd.c
  bn_gf2m.c
  bn_kron.c
  bn_lib.c
  Generating Code...
  Compiling...
  bn_mod.c
  bn_mont.c
  bn_mpi.c
  bn_mul.c
  bn_nist.c
  bn_prime.c
  bn_print.c
  bn_rand.c
  bn_recp.c
  bn_shift.c
  bn_sqr.c
  bn_sqrt.c
  bn_word.c
  rsa_ameth.c
  rsa_asn1.c
  rsa_chk.c
  rsa_depr.c
  rsa_eay.c
  rsa_err.c
  rsa_gen.c
  Generating Code...
  Compiling...
  rsa_lib.c
  rsa_none.c
  rsa_null.c
  rsa_oaep.c
  rsa_pk1.c
  rsa_pmeth.c
  rsa_prn.c
  rsa_pss.c
  rsa_saos.c
  rsa_sign.c
  rsa_ssl.c
  rsa_x931.c
  dsa_ameth.c
  dsa_asn1.c
  dsa_depr.c
  dsa_err.c
  dsa_gen.c
  dsa_key.c
  dsa_lib.c
  dsa_ossl.c
  Generating Code...
  Compiling...
  dsa_pmeth.c
  dsa_prn.c
  dsa_sign.c
  dsa_vrf.c
  md_rand.c
  rand_egd.c
  rand_err.c
  rand_lib.c
  rand_nw.c
  rand_os2.c
  rand_unix.c
  rand_win.c
  randfile.c
  b_dump.c
  b_print.c
  b_sock.c
  bf_buff.c
  bf_nbio.c
  bf_null.c
  bio_cb.c
  Generating Code...
  Compiling...
  bio_err.c
  bio_lib.c
  bss_acpt.c
  bss_bio.c
  bss_conn.c
  bss_dgram.c
  bss_fd.c
  bss_file.c
  bss_log.c
  bss_mem.c
  bss_null.c
  bss_sock.c
  err.c
  err_all.c
  err_prn.c
  ui_compat.c
  ui_err.c
  ui_lib.c
  ui_openssl.c
  ui_util.c
  Generating Code...
  Compiling...
  by_dir.c
  by_file.c
  x509_att.c
  x509_cmp.c
  x509_d2.c
  x509_def.c
  x509_err.c
  x509_ext.c
  x509_lu.c
  x509_obj.c
  x509_r2x.c
  x509_req.c
  x509_set.c
  x509_trs.c
  x509_txt.c
  x509_v3.c
  x509_vfy.c
  x509_vpm.c
  x509cset.c
  x509name.c
  Generating Code...
  Compiling...
  x509rset.c
  x509spki.c
  x509type.c
  x_all.c
  a_bitstr.c
  a_bool.c
  a_bytes.c
  a_d2i_fp.c
  a_digest.c
  a_dup.c
  a_enum.c
  a_gentm.c
  a_i2d_fp.c
  a_int.c
  a_mbstr.c
  a_object.c
  a_octet.c
  a_print.c
  a_set.c
  a_sign.c
  Generating Code...
  Compiling...
  a_strex.c
  a_strnid.c
  a_time.c
  a_type.c
  a_utctm.c
  a_utf8.c
  a_verify.c
  ameth_lib.c
  asn1_err.c
  asn1_gen.c
  asn1_lib.c
  asn1_par.c
  asn_mime.c
  asn_moid.c
  asn_pack.c
  bio_asn1.c
  bio_ndef.c
  d2i_pr.c
  d2i_pu.c
  evp_asn1.c
  Generating Code...
  Compiling...
  f_enum.c
  f_int.c
  f_string.c
  i2d_pr.c
  i2d_pu.c
  n_pkey.c
  nsseq.c
  p5_pbe.c
  p5_pbev2.c
  p8_pkey.c
  t_bitst.c
  t_crl.c
  t_pkey.c
  t_req.c
  t_spki.c
  t_x509.c
  t_x509a.c
  tasn_dec.c
  tasn_enc.c
  tasn_fre.c
  Generating Code...
  Compiling...
  tasn_new.c
  tasn_prn.c
  tasn_typ.c
  tasn_utl.c
  x_algor.c
  x_attrib.c
  x_bignum.c
  x_crl.c
  x_exten.c
  x_info.c
  x_long.c
  x_name.c
  x_nx509.c
  x_pkey.c
  x_pubkey.c
  x_req.c
  x_sig.c
  x_spki.c
  x_val.c
  x_x509.c
  Generating Code...
  Compiling...
  x_x509a.c
  o_names.c
  obj_dat.c
  obj_err.c
  obj_lib.c
  obj_xref.c
  bio_b64.c
  bio_enc.c
  bio_md.c
  bio_ok.c
  c_all.c
  c_allc.c
  c_alld.c
  digest.c
  e_aes.c
  e_bf.c
  e_camellia.c
  e_cast.c
  e_des.c
  e_des3.c
  Generating Code...
  Compiling...
  e_idea.c
  e_null.c
  e_old.c
  e_rc2.c
  e_rc4.c
  e_rc5.c
  e_seed.c
  e_xcbc_d.c
  encode.c
  evp_acnf.c
  evp_enc.c
  evp_err.c
  evp_key.c
  evp_lib.c
  evp_pbe.c
  evp_pkey.c
  m_dss.c
  m_dss1.c
  m_ecdsa.c
  m_md2.c
  Generating Code...
  Compiling...
  m_md4.c
  m_md5.c
  m_mdc2.c
  m_null.c
  m_ripemd.c
  m_sha.c
  m_sha1.c
  m_sigver.c
  m_wp.c
  names.c
  openbsd_hw.c
  p5_crpt.c
  p5_crpt2.c
  p_dec.c
  p_enc.c
  p_lib.c
  p_open.c
  p_seal.c
  p_sign.c
  p_verify.c
  Generating Code...
  Compiling...
  pmeth_fn.c
  pmeth_gn.c
  pmeth_lib.c
  buf_err.c
  buffer.c
  cms_asn1.c
  cms_att.c
  cms_cd.c
  cms_dd.c
  cms_enc.c
  cms_env.c
  cms_err.c
  cms_ess.c
  cms_io.c
  cms_lib.c
  cms_sd.c
  cms_smime.c
  eng_all.c
  eng_cnf.c
  eng_cryptodev.c
  Generating Code...
  Compiling...
  eng_ctrl.c
  eng_dyn.c
  eng_err.c
  eng_fat.c
  eng_init.c
  eng_lib.c
  eng_list.c
  eng_openssl.c
  eng_pkey.c
  eng_table.c
  tb_asnmth.c
  tb_cipher.c
  tb_dh.c
  tb_digest.c
  tb_dsa.c
  tb_ecdh.c
  tb_ecdsa.c
  tb_pkmeth.c
  tb_rand.c
  tb_rsa.c
  Generating Code...
  Compiling...
  tb_store.c
  stack.c
  bio_pk7.c
  pk7_asn1.c
  pk7_attr.c
  pk7_doit.c
  pk7_lib.c
  pk7_mime.c
  pk7_smime.c
  pkcs7err.c
  dh_ameth.c
  dh_asn1.c
  dh_check.c
  dh_depr.c
  dh_err.c
  dh_gen.c
  dh_key.c
  dh_lib.c
  dh_pmeth.c
  dh_prn.c
  Generating Code...
  Compiling...
  ocsp_asn.c
  ocsp_cl.c
  ocsp_err.c
  ocsp_ext.c
  ocsp_ht.c
  ocsp_lib.c
  ocsp_prn.c
  ocsp_srv.c
  ocsp_vfy.c
  pcy_cache.c
  pcy_data.c
  pcy_lib.c
  pcy_map.c
  pcy_node.c
  pcy_tree.c
  v3_addr.c
  v3_akey.c
  v3_akeya.c
  v3_alt.c
  v3_asid.c
  Generating Code...
  Compiling...
  v3_bcons.c
  v3_bitst.c
  v3_conf.c
  v3_cpols.c
  v3_crld.c
  v3_enum.c
  v3_extku.c
  v3_genn.c
  v3_ia5.c
  v3_info.c
  v3_int.c
  v3_lib.c
  v3_ncons.c
  v3_ocsp.c
  v3_pci.c
  v3_pcia.c
  v3_pcons.c
  v3_pku.c
  v3_pmaps.c
  v3_prn.c
  Generating Code...
  Compiling...
  v3_purp.c
  v3_skey.c
  v3_sxnet.c
  v3_utl.c
  v3err.c
  ts_asn1.c
  ts_conf.c
  ts_err.c
  ts_lib.c
  ts_req_print.c
  ts_req_utils.c
  ts_rsp_print.c
  ts_rsp_sign.c
  ts_rsp_utils.c
  ts_rsp_verify.c
  ts_verify_ctx.c
  conf_api.c
  conf_def.c
  conf_err.c
  conf_lib.c
  Generating Code...
  Compiling...
  conf_mall.c
  conf_mod.c
  conf_sap.c
  ec2_mult.c
  ec2_smpl.c
  ec_ameth.c
  ec_asn1.c
  ec_check.c
  ec_curve.c
  ec_cvt.c
  ec_err.c
  ec_key.c
  ec_lib.c
  ec_mult.c
  ec_pmeth.c
  ec_print.c
  eck_prn.c
  ecp_mont.c
  ecp_nist.c
  ecp_smpl.c
  Generating Code...
  Compiling...
  ech_err.c
  ech_key.c
  ech_lib.c
  ech_ossl.c
  ecs_asn1.c
  ecs_err.c
  ecs_lib.c
  ecs_ossl.c
  ecs_sign.c
  ecs_vrf.c
  md5_dgst.c
  md5_one.c
  md4_dgst.c
  md4_one.c
  lh_stats.c
  lhash.c
  dso_beos.c
  dso_dl.c
  dso_dlfcn.c
  dso_err.c
  Generating Code...
  Compiling...
  dso_lib.c
  dso_null.c
  dso_openssl.c
  dso_vms.c
  dso_win32.c
  p12_add.c
  p12_asn.c
  p12_attr.c
  p12_crpt.c
  p12_crt.c
  p12_decr.c
  p12_init.c
  p12_key.c
  p12_kiss.c
  p12_mutl.c
  p12_npas.c
  p12_p8d.c
  p12_p8e.c
  p12_utl.c
  pk12err.c
  Generating Code...
  Compiling...
  hm_ameth.c
  hm_pmeth.c
  hmac.c
  sha1_one.c
  sha1dgst.c
  sha256.c
  sha512.c
  sha512t.c
  sha_dgst.c
  sha_one.c
  c_rle.c
  c_zlib.c
  comp_err.c
  comp_lib.c
  pem_all.c
  pem_err.c
  pem_info.c
  pem_lib.c
  pem_oth.c
  pem_pk8.c
  Generating Code...
  Compiling...
  pem_pkey.c
  pem_seal.c
  pem_sign.c
  pem_x509.c
  pem_xaux.c
  pvkfmt.c
  wp_block.c
  wp_dgst.c
  mdc2_one.c
  mdc2dgst.c
  krb5_asn.c
  txt_db.c
  pqueue.c
  uplink.c
  Generating Code...
     Creating library C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\libeay32.lib and object C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\libeay32.exp
     Creating library C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\libeay32.lib and object C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\libeay32.exp
  libeay32.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\libeay32.dll
------ Build started: Project: Download mpg123, Configuration: Debug Win32 ------
  Downloading Flite.
  Downloading: http://files.freeswitch.org/downloads/libs/mpg123.tar.gz
  Extracting: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\mpg123.tar.gz
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\mpg123.tar.gz
  
  Extracting  mpg123.tar
  
  Everything is Ok
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\mpg123.tar
  
  Extracting  mpg123
  Extracting  mpg123\build
  Extracting  mpg123\build\compile
  Extracting  mpg123\build\config.guess
  Extracting  mpg123\build\config.sub
  Extracting  mpg123\build\depcomp
  Extracting  mpg123\build\install-sh
  Extracting  mpg123\build\ltmain.sh
  Extracting  mpg123\build\missing
  Extracting  mpg123\man1
  Extracting  mpg123\man1\mpg123.1
  Extracting  mpg123\ports
  Extracting  mpg123\ports\MSVC++
  Extracting  mpg123\ports\MSVC++\INCLUDE
  Extracting  mpg123\ports\MSVC++\INCLUDE\CORE
  Extracting  mpg123\ports\MSVC++\INCLUDE\CORE\CORE_FileIn.H
  Extracting  mpg123\ports\MSVC++\INCLUDE\CORE\SourceFilter_MP3.H
  Extracting  mpg123\ports\MSVC++\INCLUDE\IIEP_FileIn.H
  Extracting  mpg123\ports\MSVC++\INCLUDE\IIEP_Def.H
  Extracting  mpg123\ports\MSVC++\SOURCE
  Extracting  mpg123\ports\MSVC++\SOURCE\CORE_Log.CPP
  Extracting  mpg123\ports\MSVC++\SOURCE\CORE_FileIn.CPP
  Extracting  mpg123\ports\MSVC++\SOURCE\SourceFilter_MP3Stream.CPP
  Extracting  mpg123\ports\MSVC++\SOURCE\CORE_Mutex.CPP
  Extracting  mpg123\ports\MSVC++\libMPG123
  Extracting  mpg123\ports\MSVC++\libMPG123\libMPG123.vcproj
  Extracting  mpg123\ports\MSVC++\libMPG123\PLACE_LIBMPG123_SOURCES_HERE
  Extracting  mpg123\ports\MSVC++\README
  Extracting  mpg123\ports\Sony_PSP
  Extracting  mpg123\ports\Sony_PSP\config.h
  Extracting  mpg123\ports\Sony_PSP\README
  Extracting  mpg123\ports\Sony_PSP\Makefile.psp
  Extracting  mpg123\ports\Sony_PSP\readers.c.patch
  Extracting  mpg123\ports\README
  Extracting  mpg123\ports\mpg123_.pas
  Extracting  mpg123\src
  Extracting  mpg123\src\libmpg123
  Extracting  mpg123\src\libmpg123\compat.c
  Extracting  mpg123\src\libmpg123\compat.h
  Extracting  mpg123\src\libmpg123\Makefile.am
  Extracting  mpg123\src\libmpg123\Makefile.in
  Extracting  mpg123\src\libmpg123\mpg123.h.in
  Extracting  mpg123\src\libmpg123\parse.c
  Extracting  mpg123\src\libmpg123\parse.h
  Extracting  mpg123\src\libmpg123\frame.c
  Extracting  mpg123\src\libmpg123\format.c
  Extracting  mpg123\src\libmpg123\frame.h
  Extracting  mpg123\src\libmpg123\reader.h
  Extracting  mpg123\src\libmpg123\debug.h
  Extracting  mpg123\src\libmpg123\decode.h
  Extracting  mpg123\src\libmpg123\decode_2to1.c
  Extracting  mpg123\src\libmpg123\decode_4to1.c
  Extracting  mpg123\src\libmpg123\decode_ntom.c
  Extracting  mpg123\src\libmpg123\equalizer.c
  Extracting  mpg123\src\libmpg123\huffman.h
  Extracting  mpg123\src\libmpg123\icy.c
  Extracting  mpg123\src\libmpg123\icy.h
  Extracting  mpg123\src\libmpg123\icy2utf8.c
  Extracting  mpg123\src\libmpg123\icy2utf8.h
  Extracting  mpg123\src\libmpg123\id3.c
  Extracting  mpg123\src\libmpg123\id3.h
  Extracting  mpg123\src\libmpg123\true.h
  Extracting  mpg123\src\libmpg123\l2tables.h
  Extracting  mpg123\src\libmpg123\layer1.c
  Extracting  mpg123\src\libmpg123\layer2.c
  Extracting  mpg123\src\libmpg123\layer3.c
  Extracting  mpg123\src\libmpg123\getbits.h
  Extracting  mpg123\src\libmpg123\optimize.h
  Extracting  mpg123\src\libmpg123\optimize.c
  Extracting  mpg123\src\libmpg123\readers.c
  Extracting  mpg123\src\libmpg123\tabinit.c
  Extracting  mpg123\src\libmpg123\stringbuf.c
  Extracting  mpg123\src\libmpg123\libmpg123.c
  Extracting  mpg123\src\libmpg123\mpg123lib_intern.h
  Extracting  mpg123\src\libmpg123\mangle.h
  Extracting  mpg123\src\libmpg123\getcpuflags.h
  Extracting  mpg123\src\libmpg123\libmpg123.sym
  Extracting  mpg123\src\libmpg123\dct36_3dnowext.S
  Extracting  mpg123\src\libmpg123\dct36_3dnow.S
  Extracting  mpg123\src\libmpg123\dct64_3dnowext.S
  Extracting  mpg123\src\libmpg123\dct64_3dnow.S
  Extracting  mpg123\src\libmpg123\dct64_altivec.c
  Extracting  mpg123\src\libmpg123\dct64.c
  Extracting  mpg123\src\libmpg123\dct64_i386.c
  Extracting  mpg123\src\libmpg123\dct64_i486.c
  Extracting  mpg123\src\libmpg123\dct64_mmx.S
  Extracting  mpg123\src\libmpg123\dct64_sse.S
  Extracting  mpg123\src\libmpg123\decode_3dnowext.S
  Extracting  mpg123\src\libmpg123\decode_3dnow.S
  Extracting  mpg123\src\libmpg123\decode_altivec.c
  Extracting  mpg123\src\libmpg123\decode.c
  Extracting  mpg123\src\libmpg123\decode_i386.c
  Extracting  mpg123\src\libmpg123\decode_i486.c
  Extracting  mpg123\src\libmpg123\decode_i586_dither.S
  Extracting  mpg123\src\libmpg123\decode_i586.S
  Extracting  mpg123\src\libmpg123\decode_mmx.S
  Extracting  mpg123\src\libmpg123\decode_sse3d.h
  Extracting  mpg123\src\libmpg123\decode_sse.S
  Extracting  mpg123\src\libmpg123\equalizer_3dnow.S
  Extracting  mpg123\src\libmpg123\tabinit_mmx.S
  Extracting  mpg123\src\libmpg123\getcpuflags.S
  Extracting  mpg123\src\libmpg123\testcpu.c
  Extracting  mpg123\src\libmpg123\dnoise.sh
  Extracting  mpg123\src\libmpg123\dnoise.dat
  Extracting  mpg123\src\Makefile.am
  Extracting  mpg123\src\Makefile.in
  Extracting  mpg123\src\config.h.in
  Extracting  mpg123\src\audio.c
  Extracting  mpg123\src\audio.h
  Extracting  mpg123\src\buffer.c
  Extracting  mpg123\src\buffer.h
  Extracting  mpg123\src\common.c
  Extracting  mpg123\src\common.h
  Extracting  mpg123\src\control_generic.c
  Extracting  mpg123\src\getlopt.c
  Extracting  mpg123\src\getlopt.h
  Extracting  mpg123\src\httpget.c
  Extracting  mpg123\src\httpget.h
  Extracting  mpg123\src\resolver.c
  Extracting  mpg123\src\resolver.h
  Extracting  mpg123\src\genre.h
  Extracting  mpg123\src\genre.c
  Extracting  mpg123\src\module.h
  Extracting  mpg123\src\mpg123.c
  Extracting  mpg123\src\mpg123app.h
  Extracting  mpg123\src\metaprint.c
  Extracting  mpg123\src\metaprint.h
  Extracting  mpg123\src\playlist.c
  Extracting  mpg123\src\playlist.h
  Extracting  mpg123\src\sfifo.c
  Extracting  mpg123\src\sfifo.h
  Extracting  mpg123\src\term.c
  Extracting  mpg123\src\term.h
  Extracting  mpg123\src\wav.c
  Extracting  mpg123\src\xfermem.c
  Extracting  mpg123\src\xfermem.h
  Extracting  mpg123\src\Makefile.legacy
  Extracting  mpg123\src\config.h.legacy
  Extracting  mpg123\src\legacy_module.c
  Extracting  mpg123\src\module.c
  Extracting  mpg123\src\output
  Extracting  mpg123\src\output\Makefile.am
  Extracting  mpg123\src\output\Makefile.in
  Extracting  mpg123\src\output\aix.c
  Extracting  mpg123\src\output\alib.c
  Extracting  mpg123\src\output\alsa.c
  Extracting  mpg123\src\output\arts.c
  Extracting  mpg123\src\output\coreaudio.c
  Extracting  mpg123\src\output\dummy.c
  Extracting  mpg123\src\output\esd.c
  Extracting  mpg123\src\output\hp.c
  Extracting  mpg123\src\output\jack.c
  Extracting  mpg123\src\output\mint.c
  Extracting  mpg123\src\output\nas.c
  Extracting  mpg123\src\output\os2.c
  Extracting  mpg123\src\output\oss.c
  Extracting  mpg123\src\output\portaudio.c
  Extracting  mpg123\src\output\pulse.c
  Extracting  mpg123\src\output\sdl.c
  Extracting  mpg123\src\output\sgi.c
  Extracting  mpg123\src\output\sun.c
  Extracting  mpg123\src\output\win32.c
  Extracting  mpg123\test
  Extracting  mpg123\test\forkfaint.c
  Extracting  mpg123\test\rms16.c
  Extracting  mpg123\xmms2-plugin
  Extracting  mpg123\xmms2-plugin\mpg123
  Extracting  mpg123\xmms2-plugin\mpg123\mpg123.c
  Extracting  mpg123\xmms2-plugin\mpg123\wscript
  Extracting  mpg123\xmms2-plugin\README
  Extracting  mpg123\README
  Extracting  mpg123\configure.ac
  Extracting  mpg123\aclocal.m4
  Extracting  mpg123\Makefile.am
  Extracting  mpg123\Makefile.in
  Extracting  mpg123\libmpg123.pc.in
  Extracting  mpg123\configure
  Extracting  mpg123\AUTHORS
  Extracting  mpg123\COPYING
  Extracting  mpg123\ChangeLog
  Extracting  mpg123\INSTALL
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  Extracting  mpg123\MakeLegacy.sh
  Extracting  mpg123\mpg123.spec.in
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  Extracting  mpg123\libltdl
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  Extracting  mpg123\libltdl\COPYING.LIB
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  Extracting  mpg123\libltdl\config.sub
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  Extracting  mpg123\libltdl\missing
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  Extracting  mpg123\doc\examples\mpg123_to_wav.c
  Extracting  mpg123\doc\examples\scan.c
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  Extracting  mpg123\doc\examples\id3dump.c
  Extracting  mpg123\doc\examples\Makefile
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  Extracting  mpg123\doc\BENCHMARKING
  Extracting  mpg123\doc\BUGS
  Extracting  mpg123\doc\CONTACT
  Extracting  mpg123\doc\PATENTS
  Extracting  mpg123\doc\README.3DNOW
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  Extracting  mpg123\doc\libmpg123_speed.txt
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  Extracting  mpg123\doc\doxy_examples.c
  Extracting  mpg123\doc\doxygen.conf
  
  Everything is Ok
------ Build started: Project: pthread, Configuration: Debug DLL Win32 ------
  pthread.c
     Creating library C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\debug\pthread.lib and object C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\debug\pthread.exp
  pthread.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\debug\pthread.dll
------ Build started: Project: FreeSwitchCoreLib, Configuration: Debug Win32 ------
  Generating switch_version.h
  Downloading: http://files.freeswitch.org/downloads/win32/fs_svnversion.exe
  Downloading: http://files.freeswitch.org/downloads/win32/libdb44.dll
  Downloading: http://files.freeswitch.org/downloads/win32/libsvn_diff-1.dll
  Downloading: http://files.freeswitch.org/downloads/win32/libsvn_subr-1.dll
  Downloading: http://files.freeswitch.org/downloads/win32/libsvn_wc-1.dll
  Downloading: http://files.freeswitch.org/downloads/win32/intl3_svn.dll
  Downloading: http://files.freeswitch.org/downloads/win32/libapr-1.dll
  Downloading: http://files.freeswitch.org/downloads/win32/libaprutil-1.dll
  Downloading: http://files.freeswitch.org/downloads/win32/libapriconv-1.dll
  Downloading: http://files.freeswitch.org/downloads/win32/libsvn_delta-1.dll
  switch_buffer.c
  stfu.c
  inet_pton.c
  g711.c
  Generating Code...
  switch_xml_config.c
  switch_xml.c
  switch_utils.c
  switch_time.c
  switch_stun.c
  switch_scheduler.c
  switch_rtp.c
  switch_resample.c
  switch_regex.c
  switch_profile.c
  switch_pcm.c
  switch_odbc.c
  switch_mprintf.c
  switch_log.c
  switch_loadable_module.c
  switch_limit.c
  switch_json.c
  switch_ivr_say.c
  switch_ivr_play_say.c
  switch_ivr_menu.c
  Generating Code...
  Compiling...
  switch_ivr_bridge.c
  switch_ivr_async.c
  switch_ivr.c
  switch_event.c
  switch_dso.c
  switch_core_timer.c
  switch_core_state_machine.c
  switch_core_sqldb.c
  switch_core_speech.c
  switch_core_rwlock.c
  switch_core_port_allocator.c
  switch_core_memory.c
  switch_core_media_bug.c
  switch_core_io.c
  switch_core_hash.c
  switch_core_file.c
  switch_core_event_hook.c
  switch_core_directory.c
  switch_core_db.c
  switch_core_codec.c
  Generating Code...
  Compiling...
  switch_core_asr.c
  switch_core.c
  switch_console.c
  switch_config.c
  switch_channel.c
  switch_caller.c
  switch_apr.c
  Generating Code...
  switch_core_session.c
  switch_cpp.cpp
cl : Command line warning D9025: overriding '/analyze-' with '/analyze:stacksize17000'
  switch_ivr_originate.c
  switch_nat.c
  upnpreplyparse.c
  upnperrors.c
  upnpcommands.c
  minixml.c
  miniwget.c
  minissdpc.c
  minisoap.c
  igd_desc_parse.c
  Generating Code...
  miniupnpc.c
  natpmp.c
  getgateway.c
  Generating Code...
LINK : fatal error LNK1104: cannot open file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\pcre\Win32\Debug\libpcre.lib'
------ Build started: Project: Download sphinxmodel, Configuration: Debug Win32 ------
  Downloading sphinxmodel.
  Downloading: http://files.freeswitch.org/downloads/libs/communicator_semi_6000_20080321.tar.gz
  Extracting: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\communicator_semi_6000_20080321.tar.gz
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\communicator_semi_6000_20080321.tar.gz
  
  Extracting  communicator_semi_6000_20080321.tar
  
  Everything is Ok
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\communicator_semi_6000_20080321.tar
  
  Extracting  Communicator_semi_40.cd_semi_6000
  Extracting  Communicator_semi_40.cd_semi_6000\transition_matrices
  Extracting  Communicator_semi_40.cd_semi_6000\mdef
  Extracting  Communicator_semi_40.cd_semi_6000\feat.params
  Extracting  Communicator_semi_40.cd_semi_6000\means
  Extracting  Communicator_semi_40.cd_semi_6000\noisedict
  Extracting  Communicator_semi_40.cd_semi_6000\variances
  Extracting  Communicator_semi_40.cd_semi_6000\sendump
  Extracting  Communicator_semi_40.cd_semi_6000\COPYING
  
  Everything is Ok
------ Build started: Project: Download sphinxbase, Configuration: Debug Win32 ------
  Downloading sphinxbase.
  Downloading: http://files.freeswitch.org/downloads/libs/sphinxbase-0.4.99-20091212.tar.gz
  Extracting: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sphinxbase-0.4.99-20091212.tar.gz
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sphinxbase-0.4.99-20091212.tar.gz
  
  Extracting  sphinxbase-0.4.99-20091212.tar
  
  Everything is Ok
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sphinxbase-0.4.99-20091212.tar
  
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  Everything is Ok
------ Build started: Project: libogg, Configuration: Debug Win32 ------
  framing.c
  bitwise.c
  Generating Code...
  libogg.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\libogg\Win32\Debug\libogg.lib
------ Build started: Project: xmlrpc, Configuration: Debug Win32 ------
  xmlrpc_wininet_transport.c
  xmlrpc_struct.c
  xmlrpc_string.c
  xmlrpc_server_info.c
  xmlrpc_server_abyss.c
  xmlrpc_serialize.c
  xmlrpc_parse.c
  xmlrpc_expat.c
  xmlrpc_decompose.c
  xmlrpc_datetime.c
  xmlrpc_data.c
  xmlrpc_client_global.c
  xmlrpc_client.c
  xmlrpc_build.c
  xmlrpc_base64.c
  xmlrpc_authcookie.c
  xmlrpc_array.c
  version.c
  utf8.c
  trace.c
  Generating Code...
  Compiling...
  time.c
  system_method.c
  sleep.c
  select.c
  resource.c
  registry.c
  pthreadx_win32.c
  parse_value.c
  method.c
  memblock.c
  make_printable.c
  error.c
  double.c
  asprintf.c
  Generating Code...
  xmlrpc.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\xmlrpc-c\Windows\Win32\Debug\xmlrpc.lib
------ Build started: Project: libbroadvoice, Configuration: Debug Win32 ------
  gettimeofday.c
  bv32tables.c
  bv32ptquan.c
  bv32ptdec.c
  bv32plc.c
  bv32lspquan.c
floating\bv32\bv32lspquan.c(156): warning C4244: '=' : conversion from 'int' to 'int16_t', possible loss of data
floating\bv32\bv32lspquan.c(214): warning C4244: '=' : conversion from 'int' to 'int16_t', possible loss of data
floating\bv32\bv32lspquan.c(259): warning C4244: '=' : conversion from 'int' to 'int16_t', possible loss of data
  bv32lspdec.c
  bv32levelest.c
  bv32gainquan.c
  bv32gaindec.c
  bv32fine_pitch.c
  bv32excquan.c
floating\bv32\bv32excquan.c(189): warning C4244: '=' : conversion from 'int' to 'int16_t', possible loss of data
floating\bv32\bv32excquan.c(191): warning C4244: '=' : conversion from 'int' to 'int16_t', possible loss of data
  bv32excdec.c
  bv32coarse_pitch.c
  bv32encoder.c
floating\bv32\bv32encoder.c(165): warning C4244: '=' : conversion from 'int' to 'int16_t', possible loss of data
floating\bv32\bv32encoder.c(168): warning C4244: '=' : conversion from 'int' to 'int16_t', possible loss of data
floating\bv32\bv32encoder.c(194): warning C4244: '=' : conversion from 'int' to 'int16_t', possible loss of data
  bv32decoder.c
  bitpack32.c
bitpack32.c(85): warning C4244: '=' : conversion from 'uint32_t' to 'int16_t', possible loss of data
bitpack32.c(86): warning C4244: '=' : conversion from 'uint32_t' to 'int16_t', possible loss of data
bitpack32.c(87): warning C4244: '=' : conversion from 'uint32_t' to 'int16_t', possible loss of data
bitpack32.c(89): warning C4244: '=' : conversion from 'uint32_t' to 'int16_t', possible loss of data
bitpack32.c(90): warning C4244: '=' : conversion from 'uint32_t' to 'int16_t', possible loss of data
bitpack32.c(91): warning C4244: '=' : conversion from 'uint32_t' to 'int16_t', possible loss of data
bitpack32.c(92): warning C4244: '=' : conversion from 'uint32_t' to 'int16_t', possible loss of data
bitpack32.c(95): warning C4244: '=' : conversion from 'uint32_t' to 'int16_t', possible loss of data
  bv16tables.c
  bv16ptquan.c
  bv16postfilter.c
  Generating Code...
c:\users\dell 1\documents\freeswitch\freeswitch\libs\broadvoice\src\floating\bv32\bv32excdec.c(87): warning C4701: potentially uninitialized local variable 'E' used
  Compiling...
  bv16ptdec.c
  bv16plc.c
  bv16lspquan.c
floating\bv16\bv16lspquan.c(169): warning C4244: '=' : conversion from 'int' to 'int16_t', possible loss of data
floating\bv16\bv16lspquan.c(224): warning C4244: '=' : conversion from 'int' to 'int16_t', possible loss of data
floating\bv16\bv16lspquan.c(251): warning C4244: '=' : conversion from 'int' to 'int16_t', possible loss of data
floating\bv16\bv16lspquan.c(268): warning C4244: '=' : conversion from 'int' to 'int16_t', possible loss of data
  bv16lspdec.c
  bv16levelest.c
  bv16gainquan.c
  bv16gaindec.c
  bv16fine_pitch.c
  bv16excquan.c
floating\bv16\bv16excquan.c(244): warning C4244: '=' : conversion from 'int' to 'int16_t', possible loss of data
floating\bv16\bv16excquan.c(246): warning C4244: '=' : conversion from 'int' to 'int16_t', possible loss of data
  bv16excdec.c
  bv16coarse_pitch.c
  bv16encoder.c
floating\bv16\bv16encoder.c(173): warning C4244: '=' : conversion from 'int' to 'int16_t', possible loss of data
floating\bv16\bv16encoder.c(176): warning C4244: '=' : conversion from 'int' to 'int16_t', possible loss of data
floating\bv16\bv16encoder.c(188): warning C4244: '=' : conversion from 'int' to 'int16_t', possible loss of data
  bv16decoder.c
  bitpack16.c
bitpack16.c(83): warning C4244: '=' : conversion from 'uint32_t' to 'int16_t', possible loss of data
bitpack16.c(84): warning C4244: '=' : conversion from 'uint32_t' to 'int16_t', possible loss of data
bitpack16.c(85): warning C4244: '=' : conversion from 'uint32_t' to 'int16_t', possible loss of data
bitpack16.c(86): warning C4244: '=' : conversion from 'uint32_t' to 'int16_t', possible loss of data
bitpack16.c(87): warning C4244: '=' : conversion from 'uint32_t' to 'int16_t', possible loss of data
bitpack16.c(90): warning C4244: '=' : conversion from 'uint32_t' to 'int16_t', possible loss of data
  utility.c
  stblzlsp.c
  stblchck.c
  lsp2a.c
  levdur.c
  cmtables.c
  Generating Code...
  Compiling...
  autocor.c
  allzero.c
  allpole.c
  a2lsp.c
  bitstream.c
  Generating Code...
     Creating library C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\libbroadvoice.lib and object C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\libbroadvoice.exp
     Creating library C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\libbroadvoice.lib and object C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\libbroadvoice.exp
  libbroadvoice.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\libbroadvoice.dll
------ Build started: Project: xmltok, Configuration: Debug Win32 ------
  xmltok.c
  xmlrole.c
  Generating Code...
  xmltok.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\xmlrpc-c\Windows\Win32\Debug\xmltok.lib
------ Build started: Project: abyss, Configuration: Debug Win32 ------
  trace.c
  token.c
  thread_windows.c
  socket_win.c
  socket.c
  session.c
  server.c
  response.c
  init.c
  http.c
  handler.c
  file.c
  date.c
  data.c
  conn.c
  conf.c
  chanswitch.c
  channel.c
  Generating Code...
  abyss.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\xmlrpc-c\Windows\Win32\Debug\abyss.lib
------ Build started: Project: portaudio, Configuration: Debug Win32 ------
  pa_x86_plain_converters.c
  pa_win_wdmks_utils.c
  pa_win_waveformat.c
  pa_win_util.c
  pa_win_hostapis.c
  pa_win_wmme.c
  pa_trace.c
  pa_stream.c
  pa_skeleton.c
  pa_process.c
  pa_front.c
  pa_dither.c
  pa_debugprint.c
  pa_cpuload.c
  pa_converters.c
  pa_allocation.c
  Generating Code...
  portaudio.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\portaudio\build\msvc\Win32\Debug\portaudio.lib
------ Build started: Project: Download pocketsphinx, Configuration: Debug Win32 ------
  Downloading pocketsphinx.
  Downloading: http://files.freeswitch.org/downloads/libs/pocketsphinx-0.5.99-20091212.tar.gz
  Extracting: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\pocketsphinx-0.5.99-20091212.tar.gz
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\pocketsphinx-0.5.99-20091212.tar.gz
  
  Extracting  pocketsphinx-0.5.99-20091212.tar
  
  Everything is Ok
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\pocketsphinx-0.5.99-20091212.tar
  
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  Extracting  pocketsphinx-0.5.99\win32
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  Everything is Ok
------ Build started: Project: sphinxbase, Configuration: Debug Win32 ------
  yin.c
  unlimit.c
  strfuncs.c
  slapack_lite.c
  slamch.c
  sbthread.c
  rec_win32.c
  profile.c
  play_win32.c
  pio.c
  ngram_model_set.c
  ngram_model_dmp32.c
  ngram_model_dmp.c
  ngram_model_arpa.c
  ngram_model.c
  mmio.c
  matrix.c
  logmath.c
  lm3g_model.c
  listelem_alloc.c
  Generating Code...
  Compiling...
  lda.c
  jsgf_scanner.c
  jsgf_parser.c
  jsgf.c
  info.c
  heap.c
  hash_table.c
  glist.c
  genrand.c
  fsg_model.c
  fixlog.c
  filename.c
  feat.c
  fe_warp_piecewise_linear.c
  fe_warp_inverse_linear.c
  fe_warp_affine.c
  fe_warp.c
  fe_sigproc.c
  fe_interface.c
  f2c_lite.c
  Generating Code...
  Compiling...
  err.c
  dtoa.c
  cont_ad_base.c
  cmn_prior.c
  cmn.c
  cmd_ln.c
  ckd_alloc.c
  case.c
  blas_lite.c
  bitvec.c
  bio.c
  agc.c
  Generating Code...
     Creating library C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\sphinxbase.lib and object C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\sphinxbase.exp
     Creating library C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\sphinxbase.lib and object C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\sphinxbase.exp
  sphinxbase.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\sphinxbase.dll
------ Skipped Build: Project: Download 16khz music, Configuration: Debug Win32 ------
Project not selected to build for this solution configuration 
------ Build started: Project: Download 8khz music, Configuration: Debug Win32 ------
  Downloading 8khzsound.
  Downloading: http://files.freeswitch.org/freeswitch-sounds-music-8000-1.0.8.tar.gz
  Extracting: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\freeswitch-sounds-music-8000-1.0.8.tar.gz
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\freeswitch-sounds-music-8000-1.0.8.tar.gz
  
  Extracting  freeswitch-sounds-music-8000-1.0.8.tar
  
  Everything is Ok
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\freeswitch-sounds-music-8000-1.0.8.tar
  
  Extracting  music\8000
  Extracting  music\8000\partita-no-3-in-e-major-bwv-1006-1-preludio.wav
  Extracting  music\8000\ponce-preludio-in-e-major.wav
  Extracting  music\8000\suite-espanola-op-47-leyenda.wav
  Extracting  music\8000\danza-espanola-op-37-h-142-xii-arabesca.wav
  
  Everything is Ok
------ Skipped Build: Project: Download 32khz music, Configuration: Debug Win32 ------
Project not selected to build for this solution configuration 
------ Build started: Project: Download 8khzsound, Configuration: Debug Win32 ------
  Downloading 8khzsound.
  Sound name: en-us-callie Version 1.0.13
  URL: http://files.freeswitch.org/freeswitch-sounds-en-us-callie-8000-1.0.13.tar.gz
  Downloading: http://files.freeswitch.org/freeswitch-sounds-en-us-callie-8000-1.0.13.tar.gz
  Extracting: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\freeswitch-sounds-en-us-callie-8000-1.0.13.tar.gz
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\freeswitch-sounds-en-us-callie-8000-1.0.13.tar.gz
  
  Extracting  freeswitch-sounds-en-us-callie-8000-1.0.13.tar
  
  Everything is Ok
  
  7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
  
  Processing archive: C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\freeswitch-sounds-en-us-callie-8000-1.0.13.tar
  
  Extracting  en
  Extracting  en\us
  Extracting  en\us\callie
  Extracting  en\us\callie\base256
  Extracting  en\us\callie\base256\8000
  Extracting  en\us\callie\base256\8000\travesty.wav
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  Extracting  en\us\callie\base256\8000\scallion.wav
  Extracting  en\us\callie\base256\8000\flatfoot.wav
  Extracting  en\us\callie\base256\8000\misnomer.wav
  Extracting  en\us\callie\base256\8000\ancient.wav
  Extracting  en\us\callie\base256\8000\regain.wav
  Extracting  en\us\callie\base256\8000\breakup.wav
  Extracting  en\us\callie\base256\8000\equation.wav
  Extracting  en\us\callie\base256\8000\dropper.wav
  Extracting  en\us\callie\base256\8000\inferno.wav
  Extracting  en\us\callie\base256\8000\cobra.wav
  Extracting  en\us\callie\base256\8000\fascinate.wav
  Extracting  en\us\callie\base256\8000\sardonic.wav
  Extracting  en\us\callie\base256\8000\concurrent.wav
  Extracting  en\us\callie\base256\8000\Pegasus.wav
  Extracting  en\us\callie\base256\8000\adviser.wav
  Extracting  en\us\callie\base256\8000\universe.wav
  Extracting  en\us\callie\base256\8000\Dupont.wav
  Extracting  en\us\callie\base256\8000\determine.wav
  Extracting  en\us\callie\base256\8000\showgirl.wav
  Extracting  en\us\callie\base256\8000\checkup.wav
  Extracting  en\us\callie\base256\8000\revival.wav
  Extracting  en\us\callie\base256\8000\bookseller.wav
  Extracting  en\us\callie\base256\8000\Ohio.wav
  Extracting  en\us\callie\base256\8000\informant.wav
  Extracting  en\us\callie\base256\8000\crossover.wav
  Extracting  en\us\callie\base256\8000\replica.wav
  Extracting  en\us\callie\base256\8000\wallet.wav
  Extracting  en\us\callie\base256\8000\millionaire.wav
  Extracting  en\us\callie\base256\8000\typewriter.wav
  Extracting  en\us\callie\base256\8000\responsive.wav
  Extracting  en\us\callie\base256\8000\assume.wav
  Extracting  en\us\callie\base256\8000\Wilmington.wav
  Extracting  en\us\callie\base256\8000\rocker.wav
  Extracting  en\us\callie\base256\8000\trauma.wav
  Extracting  en\us\callie\base256\8000\jawbone.wav
  Extracting  en\us\callie\base256\8000\kickoff.wav
  Extracting  en\us\callie\base256\8000\consensus.wav
  Extracting  en\us\callie\base256\8000\sailboat.wav
  Extracting  en\us\callie\base256\8000\sociable.wav
  Extracting  en\us\callie\base256\8000\upcoming.wav
  Extracting  en\us\callie\base256\8000\backfield.wav
  Extracting  en\us\callie\base256\8000\snapline.wav
  Extracting  en\us\callie\base256\8000\decimal.wav
  Extracting  en\us\callie\base256\8000\maritime.wav
  Extracting  en\us\callie\base256\8000\midsummer.wav
  Extracting  en\us\callie\base256\8000\commence.wav
  Extracting  en\us\callie\base256\8000\drifter.wav
  Extracting  en\us\callie\base256\8000\Virginia.wav
  Extracting  en\us\callie\base256\8000\decadence.wav
  Extracting  en\us\callie\base256\8000\aggregate.wav
  Extracting  en\us\callie\base256\8000\surrender.wav
  Extracting  en\us\callie\base256\8000\breadline.wav
  Extracting  en\us\callie\base256\8000\existence.wav
  Extracting  en\us\callie\base256\8000\Wyoming.wav
  Extracting  en\us\callie\phonetic-ascii
  Extracting  en\us\callie\phonetic-ascii\8000
  Extracting  en\us\callie\phonetic-ascii\8000\108.wav
  Extracting  en\us\callie\phonetic-ascii\8000\119.wav
  Extracting  en\us\callie\phonetic-ascii\8000\118.wav
  Extracting  en\us\callie\phonetic-ascii\8000\98.wav
  Extracting  en\us\callie\phonetic-ascii\8000\112.wav
  Extracting  en\us\callie\phonetic-ascii\8000\42.wav
  Extracting  en\us\callie\phonetic-ascii\8000\122.wav
  Extracting  en\us\callie\phonetic-ascii\8000\102.wav
  Extracting  en\us\callie\phonetic-ascii\8000\105.wav
  Extracting  en\us\callie\phonetic-ascii\8000\99.wav
  Extracting  en\us\callie\phonetic-ascii\8000\116.wav
  Extracting  en\us\callie\phonetic-ascii\8000\110.wav
  Extracting  en\us\callie\phonetic-ascii\8000\121.wav
  Extracting  en\us\callie\phonetic-ascii\8000\35.wav
  Extracting  en\us\callie\phonetic-ascii\8000\120.wav
  Extracting  en\us\callie\phonetic-ascii\8000\107.wav
  Extracting  en\us\callie\phonetic-ascii\8000\97.wav
  Extracting  en\us\callie\phonetic-ascii\8000\104.wav
  Extracting  en\us\callie\phonetic-ascii\8000\113.wav
  Extracting  en\us\callie\phonetic-ascii\8000\103.wav
  Extracting  en\us\callie\phonetic-ascii\8000\32.wav
  Extracting  en\us\callie\phonetic-ascii\8000\106.wav
  Extracting  en\us\callie\phonetic-ascii\8000\100.wav
  Extracting  en\us\callie\phonetic-ascii\8000\117.wav
  Extracting  en\us\callie\phonetic-ascii\8000\114.wav
  Extracting  en\us\callie\phonetic-ascii\8000\115.wav
  Extracting  en\us\callie\phonetic-ascii\8000\111.wav
  Extracting  en\us\callie\phonetic-ascii\8000\109.wav
  Extracting  en\us\callie\phonetic-ascii\8000\46.wav
  Extracting  en\us\callie\phonetic-ascii\8000\101.wav
  Extracting  en\us\callie\currency
  Extracting  en\us\callie\currency\8000
  Extracting  en\us\callie\currency\8000\minus.wav
  Extracting  en\us\callie\currency\8000\negative.wav
  Extracting  en\us\callie\currency\8000\cent.wav
  Extracting  en\us\callie\currency\8000\dollars.wav
  Extracting  en\us\callie\currency\8000\cents.wav
  Extracting  en\us\callie\currency\8000\and.wav
  Extracting  en\us\callie\currency\8000\cents-per-minute.wav
  Extracting  en\us\callie\currency\8000\central.wav
  Extracting  en\us\callie\currency\8000\dollar.wav
  Extracting  en\us\callie\misc
  Extracting  en\us\callie\misc\8000
  Extracting  en\us\callie\misc\8000\we_are_trying_to_reach.wav
  Extracting  en\us\callie\misc\8000\sorry.wav
  Extracting  en\us\callie\misc\8000\call_secured.wav
  Extracting  en\us\callie\misc\8000\provide_reference_number.wav
  Extracting  en\us\callie\misc\8000\misc-your_call_will_be_terminated_in.wav
  Extracting  en\us\callie\misc\8000\es.wav
  Extracting  en\us\callie\misc\8000\misc-your_call_has_been_terminated.wav
  Extracting  en\us\callie\misc\8000\en.wav
  Extracting  en\us\callie\misc\8000\followed.wav
  Extracting  en\us\callie\misc\8000\if_you_are_this_person.wav
  Extracting  en\us\callie\misc\8000\phone_not_auth.wav
  Extracting  en\us\callie\misc\8000\error.wav
  Extracting  en\us\callie\misc\8000\if_you_would_like_to.wav
  Extracting  en\us\callie\misc\8000\invalid_extension.wav
  Extracting  en\us\callie\misc\8000\call_monitoring_blurb.wav
  Extracting  en\us\callie\misc\8000\transfer1.wav
  Extracting  en\us\callie\misc\8000\transfer2.wav
  Extracting  en\us\callie\ivr
  Extracting  en\us\callie\ivr\8000
  Extracting  en\us\callie\ivr\8000\ivr-sample_submenu.wav
  Extracting  en\us\callie\ivr\8000\ivr-recording_saved.wav
  Extracting  en\us\callie\ivr\8000\ivr-you_are_now_logged_in.wav
  Extracting  en\us\callie\ivr\8000\ivr-last_name_first.wav
  Extracting  en\us\callie\ivr\8000\ivr-generic_greeting.wav
  Extracting  en\us\callie\ivr\8000\ivr-estimated_hold_time.wav
  Extracting  en\us\callie\ivr\8000\ivr-press_one_q_or_z.wav
  Extracting  en\us\callie\ivr\8000\ivr-begin_recording.wav
  Extracting  en\us\callie\ivr\8000\ivr-to_do_a_freeswitch_echo_test.wav
  Extracting  en\us\callie\ivr\8000\ivr-im_sorry.wav
  Extracting  en\us\callie\ivr\8000\ivr-to_call_the_freeswitch_conference.wav
  Extracting  en\us\callie\ivr\8000\ivr-to_speak_with_an_operator.wav
  Extracting  en\us\callie\ivr\8000\ivr-enter_ext_pound.wav
  Extracting  en\us\callie\ivr\8000\ivr-to_listen_to_moh.wav
  Extracting  en\us\callie\ivr\8000\ivr-operator.wav
  Extracting  en\us\callie\ivr\8000\ivr-you_are_number.wav
  Extracting  en\us\callie\ivr\8000\ivr-you_may.wav
  Extracting  en\us\callie\ivr\8000\ivr-pin_or_extension_is-invalid.wav
  Extracting  en\us\callie\ivr\8000\ivr-please_state_your_name_and_reason_for_calling.wav
  Extracting  en\us\callie\ivr\8000\ivr-no_menu_items.wav
  Extracting  en\us\callie\ivr\8000\ivr-call_from.wav
  Extracting  en\us\callie\ivr\8000\ivr-accept_reject_voicemail.wav
  Extracting  en\us\callie\ivr\8000\ivr-hold_connect_call.wav
  Extracting  en\us\callie\ivr\8000\ivr-provision_phone_permanently_to_extension.wav
  Extracting  en\us\callie\ivr\8000\ivr-more_than.wav
  Extracting  en\us\callie\ivr\8000\ivr-please_enter_the_phone_number.wav
  Extracting  en\us\callie\ivr\8000\ivr-account_balance_is.wav
  Extracting  en\us\callie\ivr\8000\ivr-use_telephone_keypad.wav
  Extracting  en\us\callie\ivr\8000\ivr-followed_by_pound.wav
  Extracting  en\us\callie\ivr\8000\ivr-you_may_exit_by_hanging_up.wav
  Extracting  en\us\callie\ivr\8000\ivr-this_ivr_will_let_you_test_features.wav
  Extracting  en\us\callie\ivr\8000\ivr-to_repeat_these_options.wav
  Extracting  en\us\callie\ivr\8000\ivr-finished_pound_hash_key.wav
  Extracting  en\us\callie\ivr\8000\ivr-if_correct_press.wav
  Extracting  en\us\callie\ivr\8000\ivr-files.wav
  Extracting  en\us\callie\ivr\8000\ivr-to_log_in.wav
  Extracting  en\us\callie\ivr\8000\ivr-number.wav
  Extracting  en\us\callie\ivr\8000\ivr-call_forwarding_has_been_cancelled.wav
  Extracting  en\us\callie\ivr\8000\ivr-sales.wav
  Extracting  en\us\callie\ivr\8000\ivr-please_try_again.wav
  Extracting  en\us\callie\ivr\8000\ivr-welcome_to_freeswitch.wav
  Extracting  en\us\callie\ivr\8000\ivr-say_name.wav
  Extracting  en\us\callie\ivr\8000\ivr-thank_you.wav
  Extracting  en\us\callie\ivr\8000\ivr-any_other_digit.wav
  Extracting  en\us\callie\ivr\8000\ivr-extension_to_provision_this_phone.wav
  Extracting  en\us\callie\ivr\8000\ivr-stay_on_line_call_answered_momentarily.wav
  Extracting  en\us\callie\ivr\8000\ivr-please.wav
  Extracting  en\us\callie\ivr\8000\ivr-you_are_about_to_provision_this_phone.wav
  Extracting  en\us\callie\ivr\8000\ivr-to_hear_screaming_monkeys.wav
  Extracting  en\us\callie\ivr\8000\ivr-connect_to_caller.wav
  Extracting  en\us\callie\ivr\8000\ivr-file.wav
  Extracting  en\us\callie\ivr\8000\ivr-this_phone_will_now_reboot.wav
  Extracting  en\us\callie\ivr\8000\ivr-please_enter_pin_followed_by_pound.wav
  Extracting  en\us\callie\ivr\8000\ivr-please_return_our_call_at.wav
  Extracting  en\us\callie\ivr\8000\ivr-unable_save.wav
  Extracting  en\us\callie\ivr\8000\ivr-enter_queue_number.wav
  Extracting  en\us\callie\ivr\8000\ivr-account_number.wav
  Extracting  en\us\callie\ivr\8000\ivr-not.wav
  Extracting  en\us\callie\ivr\8000\ivr-save_review_record.wav
  Extracting  en\us\callie\ivr\8000\ivr-send_to_voicemail.wav
  Extracting  en\us\callie\ivr\8000\ivr-less_than.wav
  Extracting  en\us\callie\ivr\8000\ivr-speak_to_a_customer_service_representative.wav
  Extracting  en\us\callie\ivr\8000\ivr-thank_you_for_holding.wav
  Extracting  en\us\callie\ivr\8000\ivr-for_directory_press.wav
  Extracting  en\us\callie\ivr\8000\ivr-you_are_now_logged_out.wav
  Extracting  en\us\callie\ivr\8000\ivr-thank_you_for_calling.wav
  Extracting  en\us\callie\ivr\8000\ivr-enter_ext.wav
  Extracting  en\us\callie\ivr\8000\ivr-customer_service.wav
  Extracting  en\us\callie\ivr\8000\ivr-using_telephone_keypad.wav
  Extracting  en\us\callie\ivr\8000\ivr-to_return_to_previous_menu.wav
  Extracting  en\us\callie\ivr\8000\ivr-dnd_cancelled.wav
  Extracting  en\us\callie\ivr\8000\ivr-incoming_call.wav
  Extracting  en\us\callie\ivr\8000\ivr-enjoy_music_while_transfer.wav
  Extracting  en\us\callie\ivr\8000\ivr-to_hear_sample_submenu.wav
  Extracting  en\us\callie\ivr\8000\ivr-please_enter_extension_followed_by_pound.wav
  Extracting  en\us\callie\ivr\8000\ivr-technical_support.wav
  Extracting  en\us\callie\ivr\8000\ivr-for_this_person.wav
  Extracting  en\us\callie\ivr\8000\ivr-call.wav
  Extracting  en\us\callie\ivr\8000\ivr-call_being_transferred.wav
  Extracting  en\us\callie\ivr\8000\ivr-take_a_message.wav
  Extracting  en\us\callie\ivr\8000\ivr-register_for_cluecon.wav
  Extracting  en\us\callie\ivr\8000\ivr-in_line.wav
  Extracting  en\us\callie\ivr\8000\ivr-regarding_reference_number.wav
  Extracting  en\us\callie\ivr\8000\ivr-hello.wav
  Extracting  en\us\callie\ivr\8000\ivr-please_enter_the.wav
  Extracting  en\us\callie\ivr\8000\ivr-one_yes_two_no.wav
  Extracting  en\us\callie\ivr\8000\ivr-to_do_a_fwd_echo_test.wav
  Extracting  en\us\callie\ivr\8000\ivr-call_forwarding_has_been_set.wav
  Extracting  en\us\callie\ivr\8000\ivr-dnd_activated.wav
  Extracting  en\us\callie\ivr\8000\ivr-this_is_a_call_from.wav
  Extracting  en\us\callie\ivr\8000\ivr-that_was_an_invalid_entry.wav
  Extracting  en\us\callie\ivr\8000\ivr-or.wav
  Extracting  en\us\callie\ivr\8000\ivr-please_reenter_your_pin.wav
  Extracting  en\us\callie\ivr\8000\ivr-spell_name.wav
  Extracting  en\us\callie\ivr\8000\ivr-if_not_press.wav
  Extracting  en\us\callie\ivr\8000\ivr-you_have_dialed_an_invalid_extension.wav
  Extracting  en\us\callie\ivr\8000\ivr-first_name_first.wav
  Extracting  en\us\callie\ivr\8000\ivr-to_log_out.wav
  Extracting  en\us\callie\conference
  Extracting  en\us\callie\conference\8000
  Extracting  en\us\callie\conference\8000\conf-alone.wav
  Extracting  en\us\callie\conference\8000\conf-welcome.wav
  Extracting  en\us\callie\conference\8000\conf-is-locked.wav
  Extracting  en\us\callie\conference\8000\conf-bad-pin.wav
  Extracting  en\us\callie\conference\8000\conf-muted.wav
  Extracting  en\us\callie\conference\8000\conf-is-unlocked.wav
  Extracting  en\us\callie\conference\8000\conf-goodbye.wav
  Extracting  en\us\callie\conference\8000\conf-locked.wav
  Extracting  en\us\callie\conference\8000\conf-unmuted.wav
  Extracting  en\us\callie\conference\8000\conf-pin.wav
  Extracting  en\us\callie\conference\8000\conf-kicked.wav
  Extracting  en\us\callie\time
  Extracting  en\us\callie\time\8000
  Extracting  en\us\callie\time\8000\oh.wav
  Extracting  en\us\callie\time\8000\second.wav
  Extracting  en\us\callie\time\8000\a-m.wav
  Extracting  en\us\callie\time\8000\today.wav
  Extracting  en\us\callie\time\8000\mon-4.wav
  Extracting  en\us\callie\time\8000\day-3.wav
  Extracting  en\us\callie\time\8000\p-m.wav
  Extracting  en\us\callie\time\8000\hours.wav
  Extracting  en\us\callie\time\8000\minutes.wav
  Extracting  en\us\callie\time\8000\mon-2.wav
  Extracting  en\us\callie\time\8000\oclock.wav
  Extracting  en\us\callie\time\8000\hour.wav
  Extracting  en\us\callie\time\8000\tomorrow.wav
  Extracting  en\us\callie\time\8000\mon-10.wav
  Extracting  en\us\callie\time\8000\seconds.wav
  Extracting  en\us\callie\time\8000\mon-3.wav
  Extracting  en\us\callie\time\8000\day-6.wav
  Extracting  en\us\callie\time\8000\day-1.wav
  Extracting  en\us\callie\time\8000\day-5.wav
  Extracting  en\us\callie\time\8000\mon-7.wav
  Extracting  en\us\callie\time\8000\yesterday.wav
  Extracting  en\us\callie\time\8000\mon-0.wav
  Extracting  en\us\callie\time\8000\mon-6.wav
  Extracting  en\us\callie\time\8000\minute.wav
  Extracting  en\us\callie\time\8000\mon-1.wav
  Extracting  en\us\callie\time\8000\mon-11.wav
  Extracting  en\us\callie\time\8000\mon-5.wav
  Extracting  en\us\callie\time\8000\day-2.wav
  Extracting  en\us\callie\time\8000\mon-9.wav
  Extracting  en\us\callie\time\8000\day-4.wav
  Extracting  en\us\callie\time\8000\at.wav
  Extracting  en\us\callie\time\8000\mon-8.wav
  Extracting  en\us\callie\time\8000\day-0.wav
  Extracting  en\us\callie\directory
  Extracting  en\us\callie\directory\8000
  Extracting  en\us\callie\directory\8000\dir-enter_person.wav
  Extracting  en\us\callie\directory\8000\dir-no_matching_results.wav
  Extracting  en\us\callie\directory\8000\dir-please_try_again.wav
  Extracting  en\us\callie\directory\8000\dir-result_match.wav
  Extracting  en\us\callie\directory\8000\dir-to_search_by.wav
  Extracting  en\us\callie\directory\8000\dir-at_extension.wav
  Extracting  en\us\callie\directory\8000\dir-no_more_results.wav
  Extracting  en\us\callie\directory\8000\dir-first_name.wav
  Extracting  en\us\callie\directory\8000\dir-for_next.wav
  Extracting  en\us\callie\directory\8000\dir-start_new_search.wav
  Extracting  en\us\callie\directory\8000\dir-result_number.wav
  Extracting  en\us\callie\directory\8000\dir-last_name.wav
  Extracting  en\us\callie\directory\8000\dir-too_many_result.wav
  Extracting  en\us\callie\directory\8000\dir-to_select_entry.wav
  Extracting  en\us\callie\directory\8000\dir-specify_mininum.wav
  Extracting  en\us\callie\directory\8000\dir-letters_of_person_name.wav
  Extracting  en\us\callie\ascii
  Extracting  en\us\callie\ascii\8000
  Extracting  en\us\callie\ascii\8000\108.wav
  Extracting  en\us\callie\ascii\8000\119.wav
  Extracting  en\us\callie\ascii\8000\118.wav
  Extracting  en\us\callie\ascii\8000\98.wav
  Extracting  en\us\callie\ascii\8000\112.wav
  Extracting  en\us\callie\ascii\8000\42.wav
  Extracting  en\us\callie\ascii\8000\122.wav
  Extracting  en\us\callie\ascii\8000\102.wav
  Extracting  en\us\callie\ascii\8000\105.wav
  Extracting  en\us\callie\ascii\8000\99.wav
  Extracting  en\us\callie\ascii\8000\116.wav
  Extracting  en\us\callie\ascii\8000\110.wav
  Extracting  en\us\callie\ascii\8000\121.wav
  Extracting  en\us\callie\ascii\8000\35.wav
  Extracting  en\us\callie\ascii\8000\120.wav
  Extracting  en\us\callie\ascii\8000\107.wav
  Extracting  en\us\callie\ascii\8000\97.wav
  Extracting  en\us\callie\ascii\8000\104.wav
  Extracting  en\us\callie\ascii\8000\113.wav
  Extracting  en\us\callie\ascii\8000\103.wav
  Extracting  en\us\callie\ascii\8000\32.wav
  Extracting  en\us\callie\ascii\8000\106.wav
  Extracting  en\us\callie\ascii\8000\100.wav
  Extracting  en\us\callie\ascii\8000\117.wav
  Extracting  en\us\callie\ascii\8000\114.wav
  Extracting  en\us\callie\ascii\8000\115.wav
  Extracting  en\us\callie\ascii\8000\111.wav
  Extracting  en\us\callie\ascii\8000\109.wav
  Extracting  en\us\callie\ascii\8000\46.wav
  Extracting  en\us\callie\ascii\8000\101.wav
  Extracting  en\us\callie\digits
  Extracting  en\us\callie\digits\8000
  Extracting  en\us\callie\digits\8000\13.wav
  Extracting  en\us\callie\digits\8000\8.wav
  Extracting  en\us\callie\digits\8000\h-2.wav
  Extracting  en\us\callie\digits\8000\60.wav
  Extracting  en\us\callie\digits\8000\17.wav
  Extracting  en\us\callie\digits\8000\14.wav
  Extracting  en\us\callie\digits\8000\h-1.wav
  Extracting  en\us\callie\digits\8000\1.wav
  Extracting  en\us\callie\digits\8000\10.wav
  Extracting  en\us\callie\digits\8000\50.wav
  Extracting  en\us\callie\digits\8000\h-10.wav
  Extracting  en\us\callie\digits\8000\h-9.wav
  Extracting  en\us\callie\digits\8000\h-17.wav
  Extracting  en\us\callie\digits\8000\15.wav
  Extracting  en\us\callie\digits\8000\9.wav
  Extracting  en\us\callie\digits\8000\11.wav
  Extracting  en\us\callie\digits\8000\h-19.wav
  Extracting  en\us\callie\digits\8000\0.wav
  Extracting  en\us\callie\digits\8000\h-5.wav
  Extracting  en\us\callie\digits\8000\6.wav
  Extracting  en\us\callie\digits\8000\h-8.wav
  Extracting  en\us\callie\digits\8000\period.wav
  Extracting  en\us\callie\digits\8000\million.wav
  Extracting  en\us\callie\digits\8000\h-13.wav
  Extracting  en\us\callie\digits\8000\h-3.wav
  Extracting  en\us\callie\digits\8000\18.wav
  Extracting  en\us\callie\digits\8000\40.wav
  Extracting  en\us\callie\digits\8000\pound.wav
  Extracting  en\us\callie\digits\8000\4.wav
  Extracting  en\us\callie\digits\8000\2.wav
  Extracting  en\us\callie\digits\8000\h-20.wav
  Extracting  en\us\callie\digits\8000\h-11.wav
  Extracting  en\us\callie\digits\8000\point.wav
  Extracting  en\us\callie\digits\8000\7.wav
  Extracting  en\us\callie\digits\8000\thousand.wav
  Extracting  en\us\callie\digits\8000\star.wav
  Extracting  en\us\callie\digits\8000\h-6.wav
  Extracting  en\us\callie\digits\8000\dot.wav
  Extracting  en\us\callie\digits\8000\h-15.wav
  Extracting  en\us\callie\digits\8000\h-14.wav
  Extracting  en\us\callie\digits\8000\h-12.wav
  Extracting  en\us\callie\digits\8000\5.wav
  Extracting  en\us\callie\digits\8000\h-7.wav
  Extracting  en\us\callie\digits\8000\hundred.wav
  Extracting  en\us\callie\digits\8000\19.wav
  Extracting  en\us\callie\digits\8000\20.wav
  Extracting  en\us\callie\digits\8000\3.wav
  Extracting  en\us\callie\digits\8000\30.wav
  Extracting  en\us\callie\digits\8000\12.wav
  Extracting  en\us\callie\digits\8000\70.wav
  Extracting  en\us\callie\digits\8000\h-18.wav
  Extracting  en\us\callie\digits\8000\h-16.wav
  Extracting  en\us\callie\digits\8000\h-4.wav
  Extracting  en\us\callie\digits\8000\80.wav
  Extracting  en\us\callie\digits\8000\90.wav
  Extracting  en\us\callie\digits\8000\16.wav
  Extracting  en\us\callie\digits\8000\h-30.wav
  Extracting  en\us\callie\voicemail
  Extracting  en\us\callie\voicemail\8000
  Extracting  en\us\callie\voicemail\8000\vm-new.wav
  Extracting  en\us\callie\voicemail\8000\vm-delete_message.wav
  Extracting  en\us\callie\voicemail\8000\vm-play_previous_message.wav
  Extracting  en\us\callie\voicemail\8000\vm-save_message.wav
  Extracting  en\us\callie\voicemail\8000\vm-to_forward.wav
  Extracting  en\us\callie\voicemail\8000\vm-that_was_an_invalid_ext.wav
  Extracting  en\us\callie\voicemail\8000\vm-you_have.wav
  Extracting  en\us\callie\voicemail\8000\vm-followed_by.wav
  Extracting  en\us\callie\voicemail\8000\vm-play_greeting.wav
  Extracting  en\us\callie\voicemail\8000\vm-choose_greeting_fail.wav
  Extracting  en\us\callie\voicemail\8000\vm-choose_greeting.wav
  Extracting  en\us\callie\voicemail\8000\vm-record_name1.wav
  Extracting  en\us\callie\voicemail\8000\vm-listen_new.wav
  Extracting  en\us\callie\voicemail\8000\vm-to_record_greeting.wav
  Extracting  en\us\callie\voicemail\8000\vm-forward_to_email.wav
  Extracting  en\us\callie\voicemail\8000\vm-send_message_now.wav
  Extracting  en\us\callie\voicemail\8000\vm-to_exit_alt.wav
  Extracting  en\us\callie\voicemail\8000\vm-mark_message_new.wav
  Extracting  en\us\callie\voicemail\8000\vm-change_password.wav
  Extracting  en\us\callie\voicemail\8000\vm-next.wav
  Extracting  en\us\callie\voicemail\8000\vm-abort.wav
  Extracting  en\us\callie\voicemail\8000\vm-has_been_changed_to.wav
  Extracting  en\us\callie\voicemail\8000\vm-in_folder.wav
  Extracting  en\us\callie\voicemail\8000\vm-not_available.wav
  Extracting  en\us\callie\voicemail\8000\vm-marked-urgent.wav
  Extracting  en\us\callie\voicemail\8000\vm-choose_greeting_choose.wav
  Extracting  en\us\callie\voicemail\8000\vm-saved.wav
  Extracting  en\us\callie\voicemail\8000\vm-listen_to_recording.wav
  Extracting  en\us\callie\voicemail\8000\vm-mark-urgent.wav
  Extracting  en\us\callie\voicemail\8000\vm-save_recording.wav
  Extracting  en\us\callie\voicemail\8000\vm-greeting.wav
  Extracting  en\us\callie\voicemail\8000\vm-message.wav
  Extracting  en\us\callie\voicemail\8000\vm-message_envelope.wav
  Extracting  en\us\callie\voicemail\8000\vm-forward_add_intro.wav
  Extracting  en\us\callie\voicemail\8000\vm-message_alt.wav
  Extracting  en\us\callie\voicemail\8000\vm-main_menu.wav
  Extracting  en\us\callie\voicemail\8000\vm-mailbox_full.wav
  Extracting  en\us\callie\voicemail\8000\vm-goodbye.wav
  Extracting  en\us\callie\voicemail\8000\vm-last.wav
  Extracting  en\us\callie\voicemail\8000\vm-undelete_message.wav
  Extracting  en\us\callie\voicemail\8000\vm-followed_by_pound.wav
  Extracting  en\us\callie\voicemail\8000\vm-hello.wav
  Extracting  en\us\callie\voicemail\8000\vm-repeat_message.wav
  Extracting  en\us\callie\voicemail\8000\vm-no_more_messages.wav
  Extracting  en\us\callie\voicemail\8000\vm-main_menu_alt.wav
  Extracting  en\us\callie\voicemail\8000\vm-message_number.wav
  Extracting  en\us\callie\voicemail\8000\vm-selected.wav
  Extracting  en\us\callie\voicemail\8000\vm-emailed.wav
  Extracting  en\us\callie\voicemail\8000\vm-return_call.wav
  Extracting  en\us\callie\voicemail\8000\vm-play_next_message.wav
  Extracting  en\us\callie\voicemail\8000\vm-fail_auth.wav
  Extracting  en\us\callie\voicemail\8000\vm-continue.wav
  Extracting  en\us\callie\voicemail\8000\vm-listen_to_recording_again.wav
  Extracting  en\us\callie\voicemail\8000\vm-rerecord.wav
  Extracting  en\us\callie\voicemail\8000\vm-tutorial_record_name.wav
  Extracting  en\us\callie\voicemail\8000\vm-enter_pass.wav
  Extracting  en\us\callie\voicemail\8000\vm-delete_recording.wav
  Extracting  en\us\callie\voicemail\8000\vm-tutorial_yes_no.wav
  Extracting  en\us\callie\voicemail\8000\vm-from.wav
  Extracting  en\us\callie\voicemail\8000\vm-enter_id.wav
  Extracting  en\us\callie\voicemail\8000\vm-marked_new.wav
  Extracting  en\us\callie\voicemail\8000\vm-urgent-new.wav
  Extracting  en\us\callie\voicemail\8000\vm-undeleted.wav
  Extracting  en\us\callie\voicemail\8000\vm-to_exit.wav
  Extracting  en\us\callie\voicemail\8000\vm-messages_alt.wav
  Extracting  en\us\callie\voicemail\8000\vm-urgent.wav
  Extracting  en\us\callie\voicemail\8000\vm-listen_saved.wav
  Extracting  en\us\callie\voicemail\8000\vm-record_message.wav
  Extracting  en\us\callie\voicemail\8000\vm-forward_enter_ext.wav
  Extracting  en\us\callie\voicemail\8000\vm-record_name2.wav
  Extracting  en\us\callie\voicemail\8000\vm-person.wav
  Extracting  en\us\callie\voicemail\8000\vm-advanced.wav
  Extracting  en\us\callie\voicemail\8000\vm-messages.wav
  Extracting  en\us\callie\voicemail\8000\vm-advanced_alt.wav
  Extracting  en\us\callie\voicemail\8000\vm-record_greeting.wav
  Extracting  en\us\callie\voicemail\8000\vm-press.wav
  Extracting  en\us\callie\voicemail\8000\vm-received.wav
  Extracting  en\us\callie\voicemail\8000\vm-too-small.wav
  Extracting  en\us\callie\voicemail\8000\vm-urgent-saved.wav
  Extracting  en\us\callie\voicemail\8000\vm-tutorial_change_pin.wav
  Extracting  en\us\callie\voicemail\8000\vm-deleted.wav
  Extracting  en\us\callie\zrtp
  Extracting  en\us\callie\zrtp\8000
  Extracting  en\us\callie\zrtp\8000\zrtp-status_secure.wav
  Extracting  en\us\callie\zrtp\8000\zrtp-enroll_not_sip_registered.wav
  Extracting  en\us\callie\zrtp\8000\zrtp-enroll_already_enrolled.wav
  Extracting  en\us\callie\zrtp\8000\zrtp-status_notsecure.wav
  Extracting  en\us\callie\zrtp\8000\zrtp-status_error.wav
  Extracting  en\us\callie\zrtp\8000\zrtp-is_secure.wav
  Extracting  en\us\callie\zrtp\8000\zrtp-somethings_wrong.wav
  Extracting  en\us\callie\zrtp\8000\zrtp-is_unverified.wav
  Extracting  en\us\callie\zrtp\8000\zrtp-thankyou_goodbye.wav
  Extracting  en\us\callie\zrtp\8000\zrtp-status_securing.wav
  Extracting  en\us\callie\zrtp\8000\zrtp-check_sas.wav
  Extracting  en\us\callie\zrtp\8000\zrtp-enroll_notzrtp.wav
  Extracting  en\us\callie\zrtp\8000\zrtp-enroll_confirmed.wav
  Extracting  en\us\callie\zrtp\8000\zrtp-enroll_welcome.wav
  Extracting  en\us\callie\zrtp\8000\zrtp-is_verified.wav
  
  Everything is Ok
------ Skipped Build: Project: Download 16khzsound, Configuration: Debug Win32 ------
Project not selected to build for this solution configuration 
------ Build started: Project: libudns, Configuration: Debug Win32 ------
  udns_rr_txt.c
  udns_rr_srv.c
  udns_rr_ptr.c
  udns_rr_naptr.c
  udns_rr_mx.c
  udns_rr_a.c
  udns_resolver.c
  udns_parse.c
  udns_misc.c
  udns_dntosp.c
  udns_dn.c
  udns_codes.c
  udns_bl.c
  inet_pton.c
..\..\udns\inet_pton.c(47): warning C4005: 'EAFNOSUPPORT' : macro redefinition
          C:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\errno.h(94) : see previous definition of 'EAFNOSUPPORT'
  Generating Code...
  libudns.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\udns\Win32\Debug\libudns.lib
------ Build started: Project: libsofia_sip_ua_static, Configuration: Debug Win32 ------
  Downloading: http://files.freeswitch.org/downloads/win32/gawk.exe
  multipart mismatch with Recursive multipart ()
  Suppress-Body-If-Match header is experimental
  Suppress-Notify-If-Match header is experimental
  Suppress-Body-If-Match header is experimental
  Suppress-Notify-If-Match header is experimental
  Suppress-Body-If-Match header is experimental
  Suppress-Notify-If-Match header is experimental
  
  NOTE:
  NOTE: Remember to install pthreadVC2.dll to your path, too!
  NOTE:
  soa_tag_ref.c
  soa_tag.c
  soa_static.c
  soa.c
  sdp_tag_ref.c
  sdp_tag.c
  sdp_print.c
  sdp_parse.c
  sdp.c
  tport_type_udp.c
  tport_type_tls.c
  tport_type_tcp.c
  tport_type_connect.c
  tport_tls.c
  tport_tag_ref.c
  tport_tag.c
  tport_stub_stun.c
  tport_stub_sigcomp.c
  tport_logging.c
  tport.c
  Generating Code...
  Compiling...
  sl_utils_print.c
  sl_utils_log.c
  sl_read_payload.c
  nta_tag_ref.c
  nta_tag.c
  nta_check.c
  nta.c
  outbound.c
  nua_tag_ref.c
  nua_tag.c
  nua_subnotref.c
  nua_stack.c
  nua_session.c
  nua_server.c
  nua_registrar.c
  nua_register.c
  nua_publish.c
  nua_params.c
  nua_options.c
  nua_notifier.c
  Generating Code...
  Compiling...
  nua_message.c
  nua_extension.c
  nua_event_server.c
  nua_dialog.c
  nua_common.c
  nua_client.c
  nua.c
  stun_tag_ref.c
  stun_tag.c
  stun_mini.c
  stun_dns.c
  stun_common.c
  stun.c
  iptsec_debug.c
  auth_tag_ref.c
  auth_tag.c
  auth_plugin_delayed.c
  auth_plugin.c
  auth_module_sip.c
  auth_module_http.c
  Generating Code...
  Compiling...
  auth_module.c
  auth_digest.c
  auth_common.c
  auth_client.c
  nea_tag_ref.c
  nea_tag.c
  nea_server.c
  nea_event.c
  nea_debug.c
  nea.c
  sresolv.c
  sres_sip.c
  sres_cache.c
  sres_blocking.c
  sres.c
  nth_tag_ref.c
  nth_tag.c
  nth_server.c
  nth_client.c
  http_tag_ref.c
  Generating Code...
  Compiling...
  http_tag_class.c
  http_tag.c
  http_status.c
  http_parser_table.c
  http_parser.c
  http_header.c
  http_extra.c
  http_basic.c
  sip_util.c
  sip_time.c
  sip_tag_ref.c
  sip_tag_class.c
  sip_tag.c
  sip_status.c
  sip_session.c
  sip_security.c
  sip_refer.c
  sip_reason.c
  sip_pref_util.c
  sip_prack.c
  Generating Code...
  Compiling...
  sip_parser_table.c
  sip_parser.c
  sip_mime.c
  sip_header.c
  sip_feature.c
  sip_extra.c
  sip_event.c
  sip_caller_prefs.c
  sip_basic.c
  strtoull.c
  strcasestr.c
  memspn.c
  memmem.c
  memcspn.c
  msg_tag.c
  msg_parser_util.c
  msg_parser.c
  msg_mime_table.c
  msg_mime.c
  msg_mclass.c
  Generating Code...
  Compiling...
  msg_header_make.c
  msg_header_copy.c
  msg_generic.c
  msg_date.c
  msg_basic.c
  msg_auth.c
  msg.c
  bnf.c
  features.c
  url_tag_ref.c
  url_tag.c
  url.c
  token64.c
  rc4.c
  base64.c
  su_win32_port.c
  su_wait.c
  su_vector.c
  su_uniqueid.c
  su_timer.c
  Generating Code...
  Compiling...
  su_time0.c
  su_time.c
  su_taglist.c
  su_tag_io.c
  su_tag.c
  su_strlst.c
  su_string.c
  su_strdup.c
  su_sprintf.c
  su_socket_port.c
  su_root.c
  su_pthread_port.c
  su_port.c
  su_os_nw.c
  su_md5.c
  su_log.c
  su_localinfo.c
  su_global_log.c
  su_errno.c
  su_default_log.c
  Generating Code...
  Compiling...
  su_bm.c
  su_base_port.c
  su_alloc_lock.c
  su_alloc.c
  su_addrinfo.c
  su.c
  string0.c
  smoothsort.c
  inet_pton.c
  inet_ntop.c
  Generating Code...
  libsofia_sip_ua_static.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\sofia\Win32\Debug\libsofia_sip_ua_static.lib
------ Skipped Build: Project: Download 32khzsound, Configuration: Debug Win32 ------
Project not selected to build for this solution configuration 
------ Build started: Project: flite, Configuration: Debug Win32 ------
  cmu_us_rms_cg_phonestate.c
  cmu_us_rms_cg_params.c
  cmu_us_rms_cg_mcep_trees.c
  cmu_us_rms_cg_f0_trees.c
  cmu_us_rms_cg_durmodel.c
  cmu_us_rms_cg.c
  cmu_us_rms.c
  cmu_us_awb_cg_phonestate.c
  cmu_us_awb_cg_params.c
  cmu_us_awb_cg_mcep_trees.c
  cmu_us_awb_cg_f0_trees.c
  cmu_us_awb_cg_durmodel.c
  cmu_us_awb_cg.c
  cmu_us_awb.c
  cmu_us_slt_cg_phonestate.c
  cmu_us_slt_cg_params.c
  cmu_us_slt_cg_mcep_trees.c
  cmu_us_slt_cg_f0_trees.c
  cmu_us_slt_cg_durmodel.c
  cmu_us_slt_cg.c
  Generating Code...
  Compiling...
  cmu_us_slt.c
  usenglish.c
  us_text.c
  us_phrasing_cart.c
  us_phoneset.c
  us_nums_cart.c
  us_int_tone_cart.c
  us_int_accent_cart.c
  us_gpos.c
  us_ffeatures.c
  us_f0lr.c
  us_f0_model.c
  us_expand.c
  us_durz_cart.c
  us_dur_stats.c
  us_aswd.c
  regsub.c
  regexp.c
  rateconv.c
  flite.c
  Generating Code...
  Compiling...
  cst_wave_utils.c
  cst_wave_io.c
  cst_wave.c
  cst_voice.c
  cst_viterbi.c
  cst_vc.c
  cst_val_user.c
  cst_val_const.c
  cst_val.c
  cst_utterance.c
  cst_utt_utils.c
  cst_units.c
  cst_track_io.c
  cst_track.c
  cst_tokenstream.c
  cst_synth.c
  cst_sts.c
  cst_string.c
  cst_ss.c
  cst_socket.c
  Generating Code...
  Compiling...
  cst_sigpr.c
  cst_relation.c
  cst_rel_io.c
  cst_regex.c
  cst_reflpc.c
  cst_phoneset.c
  cst_mmap_win32.c
  cst_mlsa.c
  cst_mlpg.c
  cst_lts_rewrites.c
  cst_lts.c
  cst_lpcres.c
  cst_lexicon.c
  cst_item.c
  cst_file_stdio.c
  cst_ffeatures.c
  cst_ffeature.c
  cst_features.c
  cst_error.c
  cst_endian.c
  Generating Code...
  Compiling...
  cst_diphone.c
  cst_clunits.c
  cst_cg.c
  cst_cart.c
  cst_args.c
  cst_alloc.c
  cmu_us_kal_residx.c
  cmu_us_kal_res.c
  cmu_us_kal_lpc.c
  cmu_us_kal_diphone.c
  cmu_us_kal.c
  cmu_time_awb_mcep.c
  cmu_time_awb_lpc.c
  cmu_time_awb_lex_entry.c
  cmu_time_awb_clunits.c
  cmu_time_awb_cart.c
  cmu_postlex.c
  cmu_lts_rules.c
  cmu_lts_model.c
  cmu_lex_entries.c
  Generating Code...
  Compiling...
  cmu_lex_data.c
  cmu_lex.c
..\..\flite-1.3.99\lang\cmulex\cmu_lex.c(356): warning C4090: '=' : different 'const' qualifiers
  auserver.c
  audio.c
  au_wince.c
  au_streaming.c
  au_none.c
  au_command.c
  Generating Code...
  cmu_time_awb.c
..\..\flite-1.3.99\lang\cmu_time_awb\cmu_time_awb.c(81): warning C4090: '=' : different 'const' qualifiers
..\..\flite-1.3.99\lang\cmu_time_awb\cmu_time_awb.c(82): warning C4090: '=' : different 'const' qualifiers
  flite.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\flite\Win32\Debug\flite.lib
------ Build started: Project: lua51, Configuration: Debug Win32 ------
  lapi.c
  lauxlib.c
  lbaselib.c
  lcode.c
  ldblib.c
  ldebug.c
  ldo.c
  ldump.c
  lfunc.c
  lgc.c
  linit.c
  liolib.c
  llex.c
  lmathlib.c
  lmem.c
  loadlib.c
  lobject.c
  lopcodes.c
  loslib.c
  lparser.c
  Generating Code...
  Compiling...
  lstate.c
  lstring.c
  lstrlib.c
  ltable.c
  ltablib.c
  ltm.c
  lundump.c
  lvm.c
  lzio.c
  Generating Code...
     Creating library C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\lua51.lib and object C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\lua51.exp
  lua.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\lua51.dll
------ Build started: Project: libilbc, Configuration: Debug Win32 ------
  syntFilter.c
  StateSearchW.c
  StateConstructW.c
  packing.c
  lsf.c
  LPCencode.c
  LPCdecode.c
  iLBC_encode.c
  iLBC_decode.c
  iCBSearch.c
  iCBConstruct.c
  hpOutput.c
  hpInput.c
  helpfun.c
  getCBvec.c
  gainquant.c
  FrameClassify.c
  filter.c
  enhancer.c
  doCPLC.c
  Generating Code...
  Compiling...
  createCB.c
  constants.c
  anaFilter.c
  Generating Code...
  libilbc.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\ilbc\Win32\Debug\libilbc.lib
------ Build started: Project: libsndfile, Configuration: Debug Win32 ------
  g72x.c
  g723_40.c
  g723_24.c
  g723_16.c
  g721.c
  table.c
  short_term.c
  rpe.c
  preprocess.c
  lpc.c
  long_term.c
  gsm_option.c
  gsm_encode.c
  gsm_destroy.c
  gsm_decode.c
  gsm_create.c
  decode.c
  code.c
  add.c
  xi.c
  Generating Code...
  Compiling...
  wve.c
  wav_w64.c
  wav.c
  w64.c
  vox_adpcm.c
  voc.c
  ulaw.c
  txw.c
  svx.c
  strings.c
  sndfile.c
  sds.c
  sd2.c
  rx2.c
  rf64.c
  raw.c
  pvf.c
  pcm.c
  paf.c
  ogg.c
  Generating Code...
  Compiling...
  nist.c
  ms_adpcm.c
  mpc2k.c
  mat5.c
  mat4.c
  ircam.c
  ima_oki_adpcm.c
  ima_adpcm.c
  htk.c
  gsm610.c
  float32.c
  flac.c
  file_io.c
  dwvw.c
  dwd.c
  double64.c
  dither.c
  common.c
  command.c
  chunk.c
  Generating Code...
  Compiling...
  caf.c
  broadcast.c
  avr.c
  audio_detect.c
  au.c
  alaw.c
  aiff.c
  Generating Code...
  g72x.c
  libsndfile.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\libsndfile\Win32\Debug\libsndfile.lib
------ Build started: Project: libspeex, Configuration: Debug Win32 ------
  window.c
  vq.c
  vbr.c
  stereo.c
  speex_header.c
  speex_callbacks.c
  speex.c
  sb_celp.c
  quant_lsp.c
  nb_celp.c
  modes_wb.c
  modes.c
  ltp.c
  lsp_tables_nb.c
  lsp.c
  lpc.c
  high_lsp_tables.c
  hexc_table.c
  hexc_10_32_table.c
  gain_table_lbr.c
  Generating Code...
  Compiling...
  gain_table.c
  filters.c
  exc_8_128_table.c
  exc_5_64_table.c
  exc_5_256_table.c
  exc_20_32_table.c
  exc_10_32_table.c
  exc_10_16_table.c
  cb_search.c
  bits.c
  Generating Code...
  libspeex.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\speex\win32\VS2008\libspeex\Win32\Debug\libspeex.lib
------ Build started: Project: libcelt, Configuration: Debug Win32 ------
  vq.c
..\..\celt-0.7.1\libcelt\vq.c(63): warning C4244: '=' : conversion from 'double' to 'celt_word16', possible loss of data
..\..\celt-0.7.1\libcelt\vq.c(64): warning C4244: '=' : conversion from 'double' to 'celt_word16', possible loss of data
..\..\celt-0.7.1\libcelt\vq.c(196): warning C4244: '=' : conversion from 'double' to 'int', possible loss of data
..\..\celt-0.7.1\libcelt\vq.c(198): warning C4244: '=' : conversion from 'int' to 'celt_norm', possible loss of data
..\..\celt-0.7.1\libcelt\vq.c(224): warning C4244: '=' : conversion from 'int' to 'celt_word16', possible loss of data
..\..\celt-0.7.1\libcelt\vq.c(258): warning C4244: '=' : conversion from 'int' to 'celt_word16', possible loss of data
  rate.c
  rangeenc.c
..\..\celt-0.7.1\libcelt\rangeenc.c(100): warning C4554: '<<' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\rangeenc.c(103): warning C4554: '<<' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\rangeenc.c(113): warning C4554: '<<' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\rangeenc.c(143): warning C4018: '>=' : signed/unsigned mismatch
..\..\celt-0.7.1\libcelt\rangeenc.c(167): warning C4554: '>>' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\rangeenc.c(185): warning C4554: '<<' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\rangeenc.c(185): warning C4554: '>>' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\rangeenc.c(194): warning C4554: '<<' : check operator precedence for possible error; use parentheses to clarify precedence
  rangedec.c
..\..\celt-0.7.1\libcelt\rangedec.c(120): warning C4554: '<<' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\rangedec.c(128): warning C4554: '>>' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\rangedec.c(129): warning C4554: '<<' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\rangedec.c(129): warning C4554: '<<' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\rangedec.c(133): warning C4554: '<<' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\rangedec.c(141): warning C4554: '>>' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\rangedec.c(161): warning C4018: '<' : signed/unsigned mismatch
..\..\celt-0.7.1\libcelt\rangedec.c(168): warning C4018: '>=' : signed/unsigned mismatch
..\..\celt-0.7.1\libcelt\rangedec.c(201): warning C4554: '>>' : check operator precedence for possible error; use parentheses to clarify precedence
  quant_bands.c
c:\users\dell 1\documents\freeswitch\freeswitch\libs\celt-0.7.1\libcelt\quant_bands.h(44): warning C4244: 'return' : conversion from 'double' to 'celt_word16', possible loss of data
c:\users\dell 1\documents\freeswitch\freeswitch\libs\celt-0.7.1\libcelt\quant_bands.h(49): warning C4244: 'return' : conversion from 'double' to 'celt_word32', possible loss of data
..\..\celt-0.7.1\libcelt\quant_bands.c(131): warning C4018: '>' : signed/unsigned mismatch
..\..\celt-0.7.1\libcelt\quant_bands.c(139): warning C4244: '=' : conversion from 'int' to 'celt_word16', possible loss of data
..\..\celt-0.7.1\libcelt\quant_bands.c(253): warning C4244: '=' : conversion from 'int' to 'celt_word16', possible loss of data
  pitch.c
..\..\celt-0.7.1\libcelt\pitch.c(78): warning C4244: '=' : conversion from 'double' to 'float', possible loss of data
  modes.c
c:\users\dell 1\documents\freeswitch\freeswitch\libs\celt-0.7.1\libcelt\quant_bands.h(44): warning C4244: 'return' : conversion from 'double' to 'celt_word16', possible loss of data
c:\users\dell 1\documents\freeswitch\freeswitch\libs\celt-0.7.1\libcelt\quant_bands.h(49): warning C4244: 'return' : conversion from 'double' to 'celt_word32', possible loss of data
..\..\celt-0.7.1\libcelt\modes.c(359): warning C4244: '=' : conversion from 'double' to 'celt_word16', possible loss of data
  mdct.c
..\..\celt-0.7.1\libcelt\mdct.c(86): warning C4244: '=' : conversion from 'double' to 'float', possible loss of data
  laplace.c
..\..\celt-0.7.1\libcelt\laplace.c(70): warning C4018: '<=' : signed/unsigned mismatch
..\..\celt-0.7.1\libcelt\laplace.c(112): warning C4018: '<=' : signed/unsigned mismatch
  kiss_fft.c
..\..\celt-0.7.1\libcelt\kiss_fft.c(220): warning C4244: '=' : conversion from 'double' to 'float', possible loss of data
..\..\celt-0.7.1\libcelt\kiss_fft.c(221): warning C4244: '=' : conversion from 'double' to 'float', possible loss of data
..\..\celt-0.7.1\libcelt\kiss_fft.c(262): warning C4244: '=' : conversion from 'double' to 'float', possible loss of data
..\..\celt-0.7.1\libcelt\kiss_fft.c(263): warning C4244: '=' : conversion from 'double' to 'float', possible loss of data
..\..\celt-0.7.1\libcelt\kiss_fft.c(632): warning C4244: '=' : conversion from 'double' to 'float', possible loss of data
  header.c
  entenc.c
  entdec.c
  entcode.c
  cwrs.c
..\..\celt-0.7.1\libcelt\cwrs.c(56): warning C4554: '>>' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\cwrs.c(56): warning C4554: '<<' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\cwrs.c(56): warning C4554: '<<' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\cwrs.c(56): warning C4554: '>>' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\cwrs.c(58): warning C4554: '<<' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\cwrs.c(65): warning C4554: '>>' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\cwrs.c(66): warning C4554: '>>' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\cwrs.c(73): warning C4554: '<<' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\cwrs.c(143): warning C4554: '>>' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\cwrs.c(148): warning C4554: '>>' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\cwrs.c(162): warning C4554: '>>' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\cwrs.c(166): warning C4554: '<<' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\cwrs.c(442): warning C4554: '>>' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\cwrs.c(471): warning C4554: '>>' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\cwrs.c(492): warning C4554: '>>' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\cwrs.c(519): warning C4554: '>>' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\cwrs.c(550): warning C4554: '>>' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\cwrs.c(741): warning C4554: '>>' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\cwrs.c(764): warning C4554: '>>' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\cwrs.c(773): warning C4554: '>>' : check operator precedence for possible error; use parentheses to clarify precedence
..\..\celt-0.7.1\libcelt\cwrs.c(803): warning C4554: '>>' : check operator precedence for possible error; use parentheses to clarify precedence
  celt.c
c:\users\dell 1\documents\freeswitch\freeswitch\libs\celt-0.7.1\libcelt\quant_bands.h(44): warning C4244: 'return' : conversion from 'double' to 'celt_word16', possible loss of data
c:\users\dell 1\documents\freeswitch\freeswitch\libs\celt-0.7.1\libcelt\quant_bands.h(49): warning C4244: 'return' : conversion from 'double' to 'celt_word32', possible loss of data
c:\users\dell 1\documents\freeswitch\freeswitch\libs\celt-0.7.1\libcelt\plc.c(34): warning C4244: '=' : conversion from 'double' to 'float', possible loss of data
c:\users\dell 1\documents\freeswitch\freeswitch\libs\celt-0.7.1\libcelt\plc.c(124): warning C4244: '*=' : conversion from 'double' to 'float', possible loss of data
c:\users\dell 1\documents\freeswitch\freeswitch\libs\celt-0.7.1\libcelt\plc.c(125): warning C4244: '*=' : conversion from 'double' to 'float', possible loss of data
..\..\celt-0.7.1\libcelt\celt.c(69): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\celt-0.7.1\libcelt\celt.c(69): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\celt-0.7.1\libcelt\celt.c(69): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\celt-0.7.1\libcelt\celt.c(69): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\celt-0.7.1\libcelt\celt.c(70): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\celt-0.7.1\libcelt\celt.c(70): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\celt-0.7.1\libcelt\celt.c(70): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\celt-0.7.1\libcelt\celt.c(70): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\celt-0.7.1\libcelt\celt.c(71): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\celt-0.7.1\libcelt\celt.c(71): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\celt-0.7.1\libcelt\celt.c(71): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\celt-0.7.1\libcelt\celt.c(71): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\celt-0.7.1\libcelt\celt.c(72): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\celt-0.7.1\libcelt\celt.c(72): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\celt-0.7.1\libcelt\celt.c(72): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\celt-0.7.1\libcelt\celt.c(72): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\celt-0.7.1\libcelt\celt.c(251): warning C4244: '=' : conversion from 'double' to 'float', possible loss of data
..\..\celt-0.7.1\libcelt\celt.c(506): warning C4244: '=' : conversion from 'double' to 'celt_word16', possible loss of data
..\..\celt-0.7.1\libcelt\celt.c(592): warning C4244: 'initializing' : conversion from 'double' to 'celt_sig', possible loss of data
..\..\celt-0.7.1\libcelt\celt.c(594): warning C4244: '=' : conversion from 'double' to 'celt_word16', possible loss of data
..\..\celt-0.7.1\libcelt\celt.c(625): warning C4244: '=' : conversion from 'double' to 'float', possible loss of data
..\..\celt-0.7.1\libcelt\celt.c(819): warning C4244: '+=' : conversion from 'celt_word16' to 'celt_int32', possible loss of data
..\..\celt-0.7.1\libcelt\celt.c(949): warning C4244: '=' : conversion from 'double' to 'celt_sig', possible loss of data
..\..\celt-0.7.1\libcelt\celt.c(1304): warning C4305: '*=' : truncation from 'double' to 'float'
..\..\celt-0.7.1\libcelt\celt.c(1309): warning C4244: '-=' : conversion from 'double' to 'float', possible loss of data
..\..\celt-0.7.1\libcelt\celt.c(1329): warning C4244: '=' : conversion from 'double' to 'float', possible loss of data
..\..\celt-0.7.1\libcelt\celt.c(1343): warning C4244: '+=' : conversion from 'double' to 'float', possible loss of data
..\..\celt-0.7.1\libcelt\celt.c(1351): warning C4244: '+=' : conversion from 'double' to 'float', possible loss of data
..\..\celt-0.7.1\libcelt\celt.c(1359): warning C4244: 'initializing' : conversion from 'double' to 'float', possible loss of data
..\..\celt-0.7.1\libcelt\celt.c(1244): warning C4101: 'freq' : unreferenced local variable
  bands.c
..\..\celt-0.7.1\libcelt\bands.c(125): warning C4305: 'initializing' : truncation from 'double' to 'celt_word32'
..\..\celt-0.7.1\libcelt\bands.c(128): warning C4244: '=' : conversion from 'double' to 'celt_ener', possible loss of data
..\..\celt-0.7.1\libcelt\bands.c(170): warning C4244: 'initializing' : conversion from 'double' to 'celt_word16', possible loss of data
..\..\celt-0.7.1\libcelt\bands.c(278): warning C4244: 'initializing' : conversion from 'double' to 'float', possible loss of data
..\..\celt-0.7.1\libcelt\bands.c(281): warning C4244: '=' : conversion from 'double' to 'celt_word16', possible loss of data
..\..\celt-0.7.1\libcelt\bands.c(285): warning C4244: '=' : conversion from 'double' to 'int', possible loss of data
..\..\celt-0.7.1\libcelt\bands.c(290): warning C4244: '=' : conversion from 'double' to 'celt_word16', possible loss of data
..\..\celt-0.7.1\libcelt\bands.c(291): warning C4244: '=' : conversion from 'double' to 'celt_word16', possible loss of data
..\..\celt-0.7.1\libcelt\bands.c(318): warning C4244: '=' : conversion from 'double' to 'celt_word16', possible loss of data
..\..\celt-0.7.1\libcelt\bands.c(408): warning C4244: '=' : conversion from 'double' to 'celt_word32', possible loss of data
..\..\celt-0.7.1\libcelt\bands.c(423): warning C4244: '=' : conversion from 'double' to 'celt_word16', possible loss of data
..\..\celt-0.7.1\libcelt\bands.c(575): warning C4244: '=' : conversion from 'double' to 'int', possible loss of data
..\..\celt-0.7.1\libcelt\bands.c(724): warning C4244: '=' : conversion from 'double' to 'celt_word16', possible loss of data
..\..\celt-0.7.1\libcelt\bands.c(725): warning C4244: '=' : conversion from 'double' to 'celt_word16', possible loss of data
..\..\celt-0.7.1\libcelt\bands.c(932): warning C4244: '=' : conversion from 'double' to 'celt_word16', possible loss of data
..\..\celt-0.7.1\libcelt\bands.c(933): warning C4244: '=' : conversion from 'double' to 'celt_word16', possible loss of data
  Generating Code...
  libcelt.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\celt\Win32\Debug\libcelt.lib
------ Build started: Project: libjson, Configuration: Debug Win32 ------
  printbuf.c
..\..\json-c-0.9\printbuf.c(98): warning C4996: 'vsprintf': This function or variable may be unsafe. Consider using vsprintf_s instead. To disable deprecation, use _CRT_SECURE_NO_WARNINGS. See online help for details.
          C:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\stdio.h(371) : see declaration of 'vsprintf'
..\..\json-c-0.9\printbuf.c(118): warning C4996: '_vsnprintf': This function or variable may be unsafe. Consider using _vsnprintf_s instead. To disable deprecation, use _CRT_SECURE_NO_WARNINGS. See online help for details.
          C:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\stdio.h(363) : see declaration of '_vsnprintf'
  linkhash.c
  json_util.c
..\..\json-c-0.9\json_util.c(63): warning C4996: '_open': This function or variable may be unsafe. Consider using _sopen_s instead. To disable deprecation, use _CRT_SECURE_NO_WARNINGS. See online help for details.
          C:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\io.h(235) : see declaration of '_open'
..\..\json-c-0.9\json_util.c(65): warning C4996: 'strerror': This function or variable may be unsafe. Consider using strerror_s instead. To disable deprecation, use _CRT_SECURE_NO_WARNINGS. See online help for details.
          C:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\string.h(157) : see declaration of 'strerror'
..\..\json-c-0.9\json_util.c(72): warning C4996: 'read': The POSIX name for this item is deprecated. Instead, use the ISO C++ conformant name: _read. See online help for details.
          C:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\io.h(313) : see declaration of 'read'
..\..\json-c-0.9\json_util.c(75): warning C4996: 'close': The POSIX name for this item is deprecated. Instead, use the ISO C++ conformant name: _close. See online help for details.
          C:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\io.h(302) : see declaration of 'close'
..\..\json-c-0.9\json_util.c(78): warning C4996: 'strerror': This function or variable may be unsafe. Consider using strerror_s instead. To disable deprecation, use _CRT_SECURE_NO_WARNINGS. See online help for details.
          C:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\string.h(157) : see declaration of 'strerror'
..\..\json-c-0.9\json_util.c(98): warning C4996: '_open': This function or variable may be unsafe. Consider using _sopen_s instead. To disable deprecation, use _CRT_SECURE_NO_WARNINGS. See online help for details.
          C:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\io.h(235) : see declaration of '_open'
..\..\json-c-0.9\json_util.c(100): warning C4996: 'strerror': This function or variable may be unsafe. Consider using strerror_s instead. To disable deprecation, use _CRT_SECURE_NO_WARNINGS. See online help for details.
          C:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\string.h(157) : see declaration of 'strerror'
..\..\json-c-0.9\json_util.c(110): warning C4996: 'write': The POSIX name for this item is deprecated. Instead, use the ISO C++ conformant name: _write. See online help for details.
          C:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\io.h(318) : see declaration of 'write'
..\..\json-c-0.9\json_util.c(111): warning C4996: 'close': The POSIX name for this item is deprecated. Instead, use the ISO C++ conformant name: _close. See online help for details.
          C:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\io.h(302) : see declaration of 'close'
..\..\json-c-0.9\json_util.c(113): warning C4996: 'strerror': This function or variable may be unsafe. Consider using strerror_s instead. To disable deprecation, use _CRT_SECURE_NO_WARNINGS. See online help for details.
          C:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\string.h(157) : see declaration of 'strerror'
..\..\json-c-0.9\json_util.c(121): warning C4996: 'close': The POSIX name for this item is deprecated. Instead, use the ISO C++ conformant name: _close. See online help for details.
          C:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\io.h(302) : see declaration of 'close'
  json_tokener.c
..\..\json-c-0.9\json_tokener.c(279): warning C4018: '<' : signed/unsigned mismatch
..\..\json-c-0.9\json_tokener.c(442): warning C4018: '<' : signed/unsigned mismatch
..\..\json-c-0.9\json_tokener.c(450): warning C4018: '<' : signed/unsigned mismatch
..\..\json-c-0.9\json_tokener.c(483): warning C4996: 'sscanf': This function or variable may be unsafe. Consider using sscanf_s instead. To disable deprecation, use _CRT_SECURE_NO_WARNINGS. See online help for details.
          C:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\stdio.h(325) : see declaration of 'sscanf'
..\..\json-c-0.9\json_tokener.c(485): warning C4996: 'sscanf': This function or variable may be unsafe. Consider using sscanf_s instead. To disable deprecation, use _CRT_SECURE_NO_WARNINGS. See online help for details.
          C:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\stdio.h(325) : see declaration of 'sscanf'
..\..\json-c-0.9\json_tokener.c(553): warning C4996: 'strdup': The POSIX name for this item is deprecated. Instead, use the ISO C++ conformant name: _strdup. See online help for details.
          C:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\string.h(238) : see declaration of 'strdup'
  json_object.c
..\..\json-c-0.9\json_object.c(267): warning C4996: 'strdup': The POSIX name for this item is deprecated. Instead, use the ISO C++ conformant name: _strdup. See online help for details.
          C:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\string.h(238) : see declaration of 'strdup'
..\..\json-c-0.9\json_object.c(347): warning C4996: 'sscanf': This function or variable may be unsafe. Consider using sscanf_s instead. To disable deprecation, use _CRT_SECURE_NO_WARNINGS. See online help for details.
          C:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\stdio.h(325) : see declaration of 'sscanf'
..\..\json-c-0.9\json_object.c(384): warning C4996: 'sscanf': This function or variable may be unsafe. Consider using sscanf_s instead. To disable deprecation, use _CRT_SECURE_NO_WARNINGS. See online help for details.
          C:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\stdio.h(325) : see declaration of 'sscanf'
..\..\json-c-0.9\json_object.c(414): warning C4996: 'strdup': The POSIX name for this item is deprecated. Instead, use the ISO C++ conformant name: _strdup. See online help for details.
          C:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\string.h(238) : see declaration of 'strdup'
  debug.c
  arraylist.c
  Generating Code...
  libjson.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\json\Win32\Debug\libjson.lib
------ Build started: Project: Silk_FIX, Configuration: Debug Win32 ------
  SKP_Silk_VQ_nearest_neighbor_FIX.c
  SKP_Silk_VAD.c
  SKP_Silk_tables_type_offset.c
  SKP_Silk_tables_sign.c
  SKP_Silk_tables_pulses_per_block.c
  SKP_Silk_tables_pitch_lag.c
  SKP_Silk_tables_other.c
  SKP_Silk_tables_NLSF_CB1_16.c
  SKP_Silk_tables_NLSF_CB1_10.c
  SKP_Silk_tables_NLSF_CB0_16.c
  SKP_Silk_tables_NLSF_CB0_10.c
  SKP_Silk_tables_LTP.c
  SKP_Silk_tables_gain.c
  SKP_Silk_sum_sqr_shift.c
  SKP_Silk_sort.c
  SKP_Silk_solve_LS_FIX.c
  SKP_Silk_sigm_Q15.c
  SKP_Silk_shell_coder.c
  SKP_Silk_schur64.c
  SKP_Silk_schur.c
  Generating Code...
  Compiling...
  SKP_Silk_scale_vector.c
  SKP_Silk_scale_copy_vector16.c
  SKP_Silk_residual_energy_FIX.c
  SKP_Silk_residual_energy16_FIX.c
  SKP_Silk_resample_4_3.c
  SKP_Silk_resample_3_4.c
  SKP_Silk_resample_3_2_rom.c
  SKP_Silk_resample_3_2.c
  SKP_Silk_resample_3_1.c
  SKP_Silk_resample_2_3_rom.c
  SKP_Silk_resample_2_3_coarsest.c
  SKP_Silk_resample_2_3_coarse.c
  SKP_Silk_resample_2_3.c
  SKP_Silk_resample_2_1_coarse.c
  SKP_Silk_resample_1_3.c
  SKP_Silk_resample_1_2_coarsest.c
  SKP_Silk_resample_1_2_coarse.c
  SKP_Silk_resample_1_2.c
  SKP_Silk_regularize_correlations_FIX.c
  SKP_Silk_range_coder.c
  Generating Code...
  Compiling...
  SKP_Silk_quant_LTP_gains_FIX.c
  SKP_Silk_pulses_to_bytes.c
  SKP_Silk_process_NLSFs_FIX.c
  SKP_Silk_process_gains_FIX.c
  SKP_Silk_prefilter_FIX.c
  SKP_Silk_PLC.c
  SKP_Silk_pitch_est_tables.c
  SKP_Silk_pitch_analysis_core.c
  SKP_Silk_NSQ_del_dec.c
  SKP_Silk_NSQ.c
  SKP_Silk_noise_shape_analysis_FIX.c
  SKP_Silk_NLSF_VQ_weights_laroia.c
  SKP_Silk_NLSF_VQ_sum_error_FIX.c
  SKP_Silk_NLSF_VQ_rate_distortion_FIX.c
  SKP_Silk_NLSF_stabilize.c
  SKP_Silk_NLSF_MSVQ_encode_FIX.c
  SKP_Silk_NLSF_MSVQ_decode.c
  SKP_Silk_NLSF2A_stable.c
  SKP_Silk_NLSF2A.c
  SKP_Silk_MA.c
  Generating Code...
  Compiling...
  SKP_Silk_LTP_scale_ctrl_FIX.c
  SKP_Silk_LTP_analysis_filter_FIX.c
  SKP_Silk_LSF_cos_table.c
  SKP_Silk_LPC_synthesis_order16.c
  SKP_Silk_LPC_synthesis_filter.c
  SKP_Silk_LPC_stabilize.c
  SKP_Silk_LPC_inv_pred_gain.c
  SKP_Silk_LP_variable_cutoff.c
  SKP_Silk_lowpass_short.c
  SKP_Silk_lowpass_int.c
  SKP_Silk_log2lin.c
  SKP_Silk_lin2log.c
  SKP_Silk_LBRR_reset.c
  SKP_Silk_k2a_Q16.c
  SKP_Silk_k2a.c
  SKP_Silk_interpolate.c
  SKP_Silk_inner_prod_aligned.c
  SKP_Silk_init_encoder_FIX.c
  SKP_Silk_HP_variable_cutoff_FIX.c
  SKP_Silk_gain_quant.c
  Generating Code...
  Compiling...
  SKP_Silk_find_pred_coefs_FIX.c
  SKP_Silk_find_pitch_lags_FIX.c
  SKP_Silk_find_LTP_FIX.c
  SKP_Silk_find_LPC_FIX.c
  SKP_Silk_encode_pulses.c
  SKP_Silk_encode_parameters_v4.c
  SKP_Silk_encode_parameters.c
  SKP_Silk_encode_frame_FIX.c
  SKP_Silk_enc_API.c
  SKP_Silk_detect_SWB_input.c
  SKP_Silk_decoder_set_fs.c
  SKP_Silk_decode_pulses.c
  SKP_Silk_decode_parameters_v4.c
  SKP_Silk_decode_parameters.c
  SKP_Silk_decode_indices_v4.c
  SKP_Silk_decode_frame.c
  SKP_Silk_decode_core.c
  SKP_Silk_dec_API.c
  SKP_Silk_create_init_destroy.c
  SKP_Silk_corrMatrix_FIX.c
  Generating Code...
  Compiling...
  SKP_Silk_control_codec_FIX.c
  SKP_Silk_code_signs.c
  SKP_Silk_CNG.c
  SKP_Silk_bwexpander_32.c
  SKP_Silk_bwexpander.c
  SKP_Silk_burg_modified.c
  SKP_Silk_biquad_alt.c
  SKP_Silk_biquad.c
  SKP_Silk_autocorr.c
  SKP_Silk_array_maxabs.c
  SKP_Silk_apply_sine_window.c
  SKP_Silk_ana_filt_bank_1.c
  SKP_Silk_allpass_int.c
  SKP_Silk_A2NLSF.c
  Generating Code...
  Silk_FIX.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\silk\src\Win32\Debug\Silk_FIX.lib
------ Build started: Project: libdingaling, Configuration: Debug Win32 ------
  sha1.c
  libdingaling.c
  Generating Code...
  libdingaling.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\libdingaling\Win32\Debug\libdingaling.lib
------ Build started: Project: esl, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  esl_buffer.c
  esl_threadmutex.c
  esl_json.c
  esl_event.c
  esl_config.c
  esl.c
  Generating Code...
  esl.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\esl\src\Win32\Debug\esl.lib
------ Build started: Project: mod_spidermonkey, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_spidermonkey.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: libg722_1, Configuration: Debug Win32 ------
  utilities.c
  tables.c
..\..\libg722_1\src\tables.c(66): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(67): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(68): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(69): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(70): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(71): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(72): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(73): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(74): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(75): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(76): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(77): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(78): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(79): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(80): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(81): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(82): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(83): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(84): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(85): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(86): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(87): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(88): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(89): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(91): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(92): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(93): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(94): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(95): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(96): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(97): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(98): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(99): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(100): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(101): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(102): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(103): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(104): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(105): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(106): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(107): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(108): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(109): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(110): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(111): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(112): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(113): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(114): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(115): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(116): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(117): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(118): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(119): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(120): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(121): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(122): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(123): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(124): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(125): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(126): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(127): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(128): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(130): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(147): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(148): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(149): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(150): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(151): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(152): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(153): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(154): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(155): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(156): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(157): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(158): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(159): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(160): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(161): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(162): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(163): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(164): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(165): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(166): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(167): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(168): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(169): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(170): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(172): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(173): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(174): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(175): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(176): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(177): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(178): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(179): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(180): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(181): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(182): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(183): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(184): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(185): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(186): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(187): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(188): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(189): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(190): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(191): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(192): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(193): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(194): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(195): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(196): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(197): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(198): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(199): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(200): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(201): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(202): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(203): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(204): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(205): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(206): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(207): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(208): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(209): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(211): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(242): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(244): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(246): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(248): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(250): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(280): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(281): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(282): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(283): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(284): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(285): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(286): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(287): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(288): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(289): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(290): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(291): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(334): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(335): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(336): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(337): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(338): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(339): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(340): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(341): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(342): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(344): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(348): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(349): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(350): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(351): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(352): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(353): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(354): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(355): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(356): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(357): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(358): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(359): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(360): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(361): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(362): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(363): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(364): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(365): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(366): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(367): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(368): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(369): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(370): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(371): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(372): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(373): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(374): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(375): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(376): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(377): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(378): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(379): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(380): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(381): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(382): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(383): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(384): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(385): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(386): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(387): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(388): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(389): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(390): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(391): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(392): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(402): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(403): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(404): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(405): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(406): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(407): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(408): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(409): warning C4305: 'initializing' : truncation from 'double' to 'const float'
..\..\libg722_1\src\tables.c(411): warning C4305: 'initializing' : truncation from 'double' to 'const float'
  sam2coef.c
  huff_tab.c
  encoderf.c
  encoder.c
  decoderf.c
..\..\libg722_1\src\decoderf.c(453): warning C4244: '=' : conversion from 'double' to 'float', possible loss of data
  decoder.c
  dct4_s.c
  dct4_a.c
  dct4.c
  commonf.c
  common.c
  coef2sam.c
  bitstream.c
  basop32.c
  Generating Code...
  libg722_1.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\libg722_1\Win32\Debug\libg722_1.lib
------ Build started: Project: libmp3lame, Configuration: Debug Win32 ------
  bitstream.c
  encoder.c
  fft.c
  gain_analysis.c
  id3tag.c
  lame.c
  mpglib_interface.c
  newmdct.c
  presets.c
  psymodel.c
  quantize.c
  quantize_pvt.c
  reservoir.c
  set_get.c
  takehiro.c
  util.c
  vbrquantize.c
  VbrTag.c
  version.c
  Generating Code...
  tables.c
  libmp3lame.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\libmp3lame\Win32\Debug\libmp3lame.lib
------ Build started: Project: libshout, Configuration: Debug Win32 ------
  util.c
  timing.c
  thread.c
  sock.c
  shout.c
  resolver.c
  ogg.c
  mp3.c
  httpp.c
  avl.c
  Generating Code...
  libshout.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\libshout\Win32\Debug\libshout.lib
------ Build started: Project: libmpg123, Configuration: Debug Win32 ------
  tabinit.c
  stringbuf.c
  readers.c
  parse.c
  optimize.c
  libmpg123.c
  layer3.c
  layer2.c
  layer1.c
  id3.c
  icy2utf8.c
  icy.c
  frame.c
  format.c
  equalizer.c
  decode_ntom.c
  decode_4to1.c
  decode_2to1.c
  decode.c
  dct64.c
  Generating Code...
  Compiling...
  compat.c
  Generating Code...
  libmpg123.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\mpg123\Win32\Debug\libmpg123.lib
------ Build started: Project: ssleay32, Configuration: Debug Win32 ------
          1 file(s) copied.
  bio_ssl.c
  d1_both.c
  d1_clnt.c
  d1_enc.c
  d1_lib.c
  d1_meth.c
  d1_pkt.c
  d1_srvr.c
  kssl.c
  s23_clnt.c
  s23_lib.c
  s23_meth.c
  s23_pkt.c
  s23_srvr.c
  s2_clnt.c
  s2_enc.c
  s2_lib.c
  s2_meth.c
  s2_pkt.c
  s2_srvr.c
  Generating Code...
  Compiling...
  s3_both.c
  s3_clnt.c
  s3_enc.c
  s3_lib.c
  s3_meth.c
  s3_pkt.c
  s3_srvr.c
  ssl_algs.c
  ssl_asn1.c
  ssl_cert.c
  ssl_ciph.c
  ssl_err.c
  ssl_err2.c
  ssl_lib.c
  ssl_rsa.c
  ssl_sess.c
  ssl_stat.c
  ssl_txt.c
  t1_clnt.c
  t1_enc.c
  Generating Code...
  Compiling...
  t1_lib.c
  t1_meth.c
  t1_reneg.c
  t1_srvr.c
  Generating Code...
     Creating library C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\ssleay32.lib and object C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\ssleay32.exp
     Creating library C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\ssleay32.lib and object C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\ssleay32.exp
  ssleay32.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\ssleay32.dll
------ Build started: Project: xmlparse, Configuration: Debug Win32 ------
  xmlparse.c
  xmlparse.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\xmlrpc-c\Windows\Win32\Debug\xmlparse.lib
------ Build started: Project: mod_say_zh, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_say_zh.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_snom, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_snom.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_managed, Configuration: Debug_CLR Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  freeswitch_managed.cpp
freeswitch_managed.cpp : fatal error C1192: #using failed on 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\mod\FreeSWITCH.Managed.dll'
          'The system cannot find the file specified.'
  freeswitch_wrap.2010.cxx
freeswitch_wrap.2010.cxx : fatal error C1192: #using failed on 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\mod\FreeSWITCH.Managed.dll'
          'The system cannot find the file specified.'
  mod_managed.cpp
mod_managed.cpp : fatal error C1192: #using failed on 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\mod\FreeSWITCH.Managed.dll'
          'The system cannot find the file specified.'
  Generating Code...
------ Build started: Project: mod_shout, Configuration: Debug Win32 ------
  mod_shout.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_loopback, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_loopback.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_vmd, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_vmd.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_siren, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_siren.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: fs_cli, Configuration: Debug Win32 ------
  getopt_long.c
  fs_cli.c
  Generating Code...
  fs_cli.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\fs_cli.exe
------ Build started: Project: mod_easyroute, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_easyroute.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_lcr, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_lcr.c
mod_lcr.c(637): warning C4244: '=' : conversion from 'float' to 'int', possible loss of data
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Skipped Build: Project: mod_flite, Configuration: Debug Win32 ------
Project not selected to build for this solution configuration 
------ Skipped Build: Project: mod_opal, Configuration: Debug Win32 ------
Project not selected to build for this solution configuration 
------ Skipped Build: Project: 32khz, Configuration: Debug Win32 ------
Project not selected to build for this solution configuration 
------ Skipped Build: Project: mod_skinny, Configuration: Debug Win32 ------
Project not selected to build for this solution configuration 
------ Skipped Build: Project: 16khz, Configuration: Debug Win32 ------
Project not selected to build for this solution configuration 
------ Build started: Project: 8khz, Configuration: Debug Win32 ------
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-abort.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-advanced.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-advanced_alt.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-change_password.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-choose_greeting.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-choose_greeting_choose.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-choose_greeting_fail.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-continue.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-deleted.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-delete_message.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-delete_recording.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-emailed.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-enter_id.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-enter_pass.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-fail_auth.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-followed_by.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-followed_by_pound.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-forward_add_intro.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-forward_enter_ext.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-forward_to_email.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-from.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-goodbye.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-greeting.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-has_been_changed_to.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-hello.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-in_folder.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-last.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-listen_new.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-listen_saved.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-listen_to_recording.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-listen_to_recording_again.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-mailbox_full.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-main_menu.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-main_menu_alt.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-mark-urgent.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-marked-urgent.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-marked_new.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-mark_message_new.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-message.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-messages.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-messages_alt.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-message_alt.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-message_envelope.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-message_number.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-new.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-next.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-not_available.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-no_more_messages.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-person.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-play_greeting.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-play_next_message.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-play_previous_message.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-press.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-received.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-record_greeting.wav
  C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\sounds\en\us\callie\voicemail\8000\vm-record_message.wav
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------ Skipped Build: Project: mod_skel, Configuration: Debug Win32 ------
Project not selected to build for this solution configuration 
------ Skipped Build: Project: mod_skypopen, Configuration: Debug Win32 ------
Project not selected to build for this solution configuration 
------ Build started: Project: pocketsphinx, Configuration: Debug Win32 ------
  vithist.c
  vector.c
  tst_search.c
  tmat.c
  s3dict.c
  s2_semi_mgau.c
  ps_mllr.c
  ps_lattice.c
  pocketsphinx.c
  phone_loop_search.c
  ngram_search_fwdtree.c
  ngram_search_fwdflat.c
  ngram_search.c
  ms_senone.c
  ms_mgau.c
  ms_gauden.c
  mdef.c
  lextree.c
  kdtree.c
  hmm.c
  Generating Code...
  Compiling...
  fsg_search.c
  fsg_lextree.c
  fsg_history.c
  fillpen.c
  dict2pid.c
  cmu6_lts_rules.c
  blkarray_list.c
  bin_mdef.c
  acmod.c
  Generating Code...
     Creating library C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\pocketsphinx.lib and object C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\pocketsphinx.exp
     Creating library C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\pocketsphinx.lib and object C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\pocketsphinx.exp
  pocketsphinx.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\pocketsphinx.dll
------ Build started: Project: 8khz music, Configuration: Debug Win32 ------
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------ Skipped Build: Project: 16khz music, Configuration: Debug Win32 ------
Project not selected to build for this solution configuration 
------ Skipped Build: Project: 32khz music, Configuration: Debug Win32 ------
Project not selected to build for this solution configuration 
------ Build started: Project: mod_file_string, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_file_string.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_nibblebill, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_nibblebill.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_say_ru, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_say_ru.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_valet_parking, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_valet_parking.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_bv, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_bv.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_lua, Configuration: Debug Win32 ------
  freeswitch_lua.cpp
  mod_lua.cpp
  Generating Code...
  mod_lua_wrap.cpp
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_spidermonkey_curl, Configuration: Debug Win32 ------
C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\w32\modules.props(4,5): warning MSB4011: "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\w32\winlibs.props" cannot be imported again. It was already imported at "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\src\mod\languages\mod_spidermonkey\mod_spidermonkey_curl.2010.vcxproj (55,5)". This is most likely a build authoring error. This subsequent import will be ignored. 
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_spidermonkey_curl.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_fsv, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_fsv.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_tone_stream, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_tone_stream.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_cdr_csv, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_cdr_csv.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_logfile, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_logfile.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_dialplan_asterisk, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_dialplan_asterisk.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_expr, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  expreval.c
  exprfunc.c
  exprinit.c
  exprmem.c
  exprobj.c
  exprpars.c
  exprutil.c
  exprval.c
  mod_expr.c
  Generating Code...
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_db, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_db.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_fifo, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_fifo.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_say_nl, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_say_nl.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_say_it, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_say_it.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_say_fr, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_say_fr.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_say_es, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_say_es.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_say_de, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_say_de.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_voicemail, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_voicemail.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_spidermonkey_socket, Configuration: Debug Win32 ------
C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\w32\modules.props(4,5): warning MSB4011: "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\w32\winlibs.props" cannot be imported again. It was already imported at "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\src\mod\languages\mod_spidermonkey\mod_spidermonkey_socket.2010.vcxproj (57,5)". This is most likely a build authoring error. This subsequent import will be ignored. 
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_spidermonkey_socket.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_local_stream, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_local_stream.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_esf, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_esf.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_h26x, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_h26x.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_amr, Configuration: Debug Passthrough Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_amr.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_xml_cdr, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_xml_cdr.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: aprtoolkit, Configuration: Debug Win32 ------
C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\apr.props(4,5): warning MSB4011: "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\unibase.props" cannot be imported again. It was already imported at "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\unidebug.props (4,5)". This is most likely a build authoring error. This subsequent import will be ignored. [C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\libs\apr-toolkit\aprtoolkit.2010.vcxproj]
  apt_text_stream.c
  apt_test_suite.c
  apt_task_msg.c
  apt_task.c
  apt_string_table.c
  apt_pool.c
  apt_pollset.c
  apt_pair.c
  apt_obj_list.c
  apt_nlsml_doc.c
  apt_net_server_task.c
  apt_net_client_task.c
  apt_net.c
  apt_log.c
  apt_dir_layout.c
  apt_cyclic_queue.c
  apt_consumer_task.c
  Generating Code...
  aprtoolkit.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\libs\apr-toolkit\Win32\Debug\aprtoolkit.lib
------ Build started: Project: mod_say_en, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_say_en.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_xml_curl, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_xml_curl.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_spidermonkey_odbc, Configuration: Debug Win32 ------
C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\w32\modules.props(4,5): warning MSB4011: "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\w32\winlibs.props" cannot be imported again. It was already imported at "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\src\mod\languages\mod_spidermonkey\mod_spidermonkey_odbc.2010.vcxproj (55,5)". This is most likely a build authoring error. This subsequent import will be ignored. 
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_spidermonkey_odbc.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_enum, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_enum.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mpf, Configuration: Debug Win32 ------
C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\apr.props(4,5): warning MSB4011: "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\unibase.props" cannot be imported again. It was already imported at "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\unidebug.props (4,5)". This is most likely a build authoring error. This subsequent import will be ignored. [C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\libs\mpf\mpf.2010.vcxproj]
  mpf_timer_manager.c
  mpf_termination_factory.c
  mpf_termination.c
  mpf_stream.c
  mpf_scheduler.c
  mpf_rtp_termination_factory.c
  mpf_rtp_stream.c
  mpf_rtp_attribs.c
  mpf_resampler.c
  mpf_named_event.c
  mpf_multiplier.c
  mpf_mixer.c
  mpf_jitter_buffer.c
  mpf_frame_buffer.c
  mpf_file_termination_factory.c
  mpf_engine.c
  mpf_encoder.c
  mpf_dtmf_generator.c
  mpf_dtmf_detector.c
  mpf_decoder.c
  Generating Code...
  Compiling...
  mpf_context.c
  mpf_codec_manager.c
  mpf_codec_linear.c
  mpf_codec_g711.c
  mpf_codec_descriptor.c
  mpf_buffer.c
  mpf_bridge.c
  mpf_audio_file_stream.c
  mpf_activity_detector.c
  g711.c
  Generating Code...
  mpf.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\libs\mpf\Win32\Debug\mpf.lib
------ Build started: Project: mod_spidermonkey_teletone, Configuration: Debug Win32 ------
C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\w32\modules.props(4,5): warning MSB4011: "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\w32\winlibs.props" cannot be imported again. It was already imported at "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\src\mod\languages\mod_spidermonkey\mod_spidermonkey_teletone.2010.vcxproj (55,5)". This is most likely a build authoring error. This subsequent import will be ignored. 
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_spidermonkey_teletone.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_spidermonkey_core_db, Configuration: Debug Win32 ------
C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\w32\modules.props(4,5): warning MSB4011: "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\w32\winlibs.props" cannot be imported again. It was already imported at "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\src\mod\languages\mod_spidermonkey\mod_spidermonkey_core_db.2010.vcxproj (55,5)". This is most likely a build authoring error. This subsequent import will be ignored. 
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_spidermonkey_core_db.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_native_file, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_native_file.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_g723_1, Configuration: Debug Passthrough Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_g723_1.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mrcp, Configuration: Debug Win32 ------
C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\apr.props(4,5): warning MSB4011: "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\unibase.props" cannot be imported again. It was already imported at "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\unidebug.props (4,5)". This is most likely a build authoring error. This subsequent import will be ignored. [C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\libs\mrcp\mrcp.2010.vcxproj]
  mrcp_synth_resource.c
  mrcp_synth_header.c
  mrcp_recorder_resource.c
  mrcp_recorder_header.c
  mrcp_recog_resource.c
  mrcp_recog_header.c
  mrcp_stream.c
  mrcp_resource_loader.c
  mrcp_resource_factory.c
  mrcp_start_line.c
  mrcp_message.c
  mrcp_header_accessor.c
  mrcp_generic_header.c
  Generating Code...
  mrcp.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\libs\mrcp\Win32\Debug\mrcp.lib
------ Build started: Project: mrcpclient, Configuration: Debug Win32 ------
C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\apr.props(4,5): warning MSB4011: "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\unibase.props" cannot be imported again. It was already imported at "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\unidebug.props (4,5)". This is most likely a build authoring error. This subsequent import will be ignored. [C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\libs\mrcp-client\mrcpclient.2010.vcxproj]
C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\mrcp.props(4,5): warning MSB4011: "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\apt.props" cannot be imported again. It was already imported at "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\mpf.props (4,5)". This is most likely a build authoring error. This subsequent import will be ignored. [C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\libs\mrcp-client\mrcpclient.2010.vcxproj]
C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\mrcpv2transport.props(4,5): warning MSB4011: "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\mrcp.props" cannot be imported again. It was already imported at "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\mrcpsignaling.props (5,5)". This is most likely a build authoring error. This subsequent import will be ignored. [C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\libs\mrcp-client\mrcpclient.2010.vcxproj]
  mrcp_client_session.c
  mrcp_client.c
  Generating Code...
  mrcpclient.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\libs\mrcp-client\Win32\Debug\mrcpclient.lib
------ Build started: Project: mrcpsignaling, Configuration: Debug Win32 ------
C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\apr.props(4,5): warning MSB4011: "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\unibase.props" cannot be imported again. It was already imported at "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\unidebug.props (4,5)". This is most likely a build authoring error. This subsequent import will be ignored. [C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\libs\mrcp-signaling\mrcpsignaling.2010.vcxproj]
C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\mrcp.props(4,5): warning MSB4011: "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\apt.props" cannot be imported again. It was already imported at "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\mpf.props (4,5)". This is most likely a build authoring error. This subsequent import will be ignored. [C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\libs\mrcp-signaling\mrcpsignaling.2010.vcxproj]
  mrcp_sig_agent.c
  mrcp_session_descriptor.c
  Generating Code...
  mrcpsignaling.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\libs\mrcp-signaling\Win32\Debug\mrcpsignaling.lib
------ Build started: Project: mod_sofia, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_sofia.c
  sip-dig.c
  sofia.c
sofia.c(2372): warning C4133: 'function' : incompatible types - from 'switch_rtp_bug_flag_t *' to 'uint32_t *'
sofia.c(2374): warning C4133: 'function' : incompatible types - from 'switch_rtp_bug_flag_t *' to 'uint32_t *'
sofia.c(3107): warning C4133: 'function' : incompatible types - from 'switch_rtp_bug_flag_t *' to 'uint32_t *'
sofia.c(3109): warning C4133: 'function' : incompatible types - from 'switch_rtp_bug_flag_t *' to 'uint32_t *'
  sofia_glue.c
sofia_glue.c(541): warning C4244: 'function' : conversion from 'uint32_t' to 'switch_port_t', possible loss of data
sofia_glue.c(555): warning C4244: 'function' : conversion from 'uint32_t' to 'switch_port_t', possible loss of data
sofia_glue.c(3117): warning C4133: 'function' : incompatible types - from 'switch_rtp_bug_flag_t *' to 'uint32_t *'
sofia_glue.c(4370): warning C4244: 'function' : conversion from 'unsigned int' to 'switch_payload_t', possible loss of data
  sofia_presence.c
  sofia_reg.c
  sofia_sla.c
  Generating Code...
c:\users\dell 1\documents\freeswitch\freeswitch\src\mod\endpoints\mod_sofia\sofia_glue.c(2697): warning C4702: unreachable code
c:\users\dell 1\documents\freeswitch\freeswitch\src\mod\endpoints\mod_sofia\sofia_glue.c(3944): warning C4702: unreachable code
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mrcpv2transport, Configuration: Debug Win32 ------
C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\apr.props(4,5): warning MSB4011: "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\unibase.props" cannot be imported again. It was already imported at "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\unidebug.props (4,5)". This is most likely a build authoring error. This subsequent import will be ignored. [C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\libs\mrcpv2-transport\mrcpv2transport.2010.vcxproj]
  mrcp_server_connection.c
  mrcp_control_descriptor.c
  mrcp_connection.c
  mrcp_client_connection.c
  Generating Code...
  mrcpv2transport.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\libs\mrcpv2-transport\Win32\Debug\mrcpv2transport.lib
------ Build started: Project: xml, Configuration: Debug Win32 ------
  Creating config.h from winconfig.h
  Creating expat.h from expat.h.in
  xmlparse.c
  xmlrole.c
  xmltok.c
  Generating Code...
  xml.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\win32\apr-util\Win32\Debug\xml.lib
------ Build started: Project: unirtsp, Configuration: Debug Win32 ------
C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\apr.props(4,5): warning MSB4011: "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\unibase.props" cannot be imported again. It was already imported at "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\unidebug.props (4,5)". This is most likely a build authoring error. This subsequent import will be ignored. [C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\libs\uni-rtsp\unirtsp.2010.vcxproj]
  rtsp_stream.c
  rtsp_start_line.c
  rtsp_server.c
  rtsp_message.c
  rtsp_header.c
  rtsp_client.c
  Generating Code...
  unirtsp.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\libs\uni-rtsp\Win32\Debug\unirtsp.lib
------ Build started: Project: mrcpsofiasip, Configuration: Debug Win32 ------
C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\sofiasip.props(4,5): warning MSB4011: "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\unibase.props" cannot be imported again. It was already imported at "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\unidebug.props (4,5)". This is most likely a build authoring error. This subsequent import will be ignored. [C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\modules\mrcp-sofiasip\mrcpsofiasip.2010.vcxproj]
C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\apr.props(4,5): warning MSB4011: "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\unibase.props" cannot be imported again. It was already imported at "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\unidebug.props (4,5)". This is most likely a build authoring error. This subsequent import will be ignored. [C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\modules\mrcp-sofiasip\mrcpsofiasip.2010.vcxproj]
C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\mpf.props(4,5): warning MSB4011: "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\apt.props" cannot be imported again. It was already imported at "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\mrcp.props (4,5)". This is most likely a build authoring error. This subsequent import will be ignored. [C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\modules\mrcp-sofiasip\mrcpsofiasip.2010.vcxproj]
C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\mrcpsignaling.props(5,5): warning MSB4011: "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\mrcp.props" cannot be imported again. It was already imported at "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\mrcpv2transport.props (4,5)". This is most likely a build authoring error. This subsequent import will be ignored. [C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\modules\mrcp-sofiasip\mrcpsofiasip.2010.vcxproj]
  mrcp_sofiasip_server_agent.c
  mrcp_sofiasip_client_agent.c
  mrcp_sdp.c
  Generating Code...
  mrcpsofiasip.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\modules\mrcp-sofiasip\Win32\Debug\mrcpsofiasip.lib
------ Build started: Project: mrcpunirtsp, Configuration: Debug Win32 ------
C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\apr.props(4,5): warning MSB4011: "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\unibase.props" cannot be imported again. It was already imported at "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\unidebug.props (4,5)". This is most likely a build authoring error. This subsequent import will be ignored. [C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\modules\mrcp-unirtsp\mrcpunirtsp.2010.vcxproj]
C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\mrcp.props(4,5): warning MSB4011: "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\apt.props" cannot be imported again. It was already imported at "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\mpf.props (4,5)". This is most likely a build authoring error. This subsequent import will be ignored. [C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\modules\mrcp-unirtsp\mrcpunirtsp.2010.vcxproj]
C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\unirtsp.props(4,5): warning MSB4011: "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\apt.props" cannot be imported again. It was already imported at "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\mpf.props (4,5)". This is most likely a build authoring error. This subsequent import will be ignored. [C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\modules\mrcp-unirtsp\mrcpunirtsp.2010.vcxproj]
C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\sofiasip.props(4,5): warning MSB4011: "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\unibase.props" cannot be imported again. It was already imported at "C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\build\vsprops\unidebug.props (4,5)". This is most likely a build authoring error. This subsequent import will be ignored. [C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\modules\mrcp-unirtsp\mrcpunirtsp.2010.vcxproj]
  mrcp_unirtsp_server_agent.c
  mrcp_unirtsp_sdp.c
  mrcp_unirtsp_client_agent.c
  Generating Code...
  mrcpunirtsp.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\libs\unimrcp\modules\mrcp-unirtsp\Win32\Debug\mrcpunirtsp.lib
------ Build started: Project: mod_celt, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_celt.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Skipped Build: Project: FSComm, Configuration: Debug Win32 ------
Project not selected to build for this solution configuration 
------ Build started: Project: mod_curl, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_curl.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_silk, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_silk.c
mod_silk.c(283): warning C4244: 'initializing' : conversion from 'int' to 'short', possible loss of data
mod_silk.c(286): warning C4244: 'initializing' : conversion from 'uint32_t' to 'short', possible loss of data
mod_silk.c(294): warning C4244: '=' : conversion from 'int' to 'short', possible loss of data
mod_silk.c(330): warning C4244: '=' : conversion from 'int' to 'short', possible loss of data
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_avmd, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  amplitude.c
  buffer.c
  desa2.c
  fast_acosf.c
  goertzel.c
  mod_avmd.c
  Generating Code...
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_spandsp, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_spandsp.c
  mod_spandsp_codecs.c
  mod_spandsp_dsp.c
  mod_spandsp_fax.c
mod_spandsp_fax.c(1616): warning C4047: '=' : 'zap_socket_t' differs in levels of indirection from 'int'
mod_spandsp_fax.c(1617): warning C4047: '=' : 'zap_socket_t' differs in levels of indirection from 'int'
mod_spandsp_fax.c(1618): warning C4047: 'function' : 'int' differs in levels of indirection from 'zap_socket_t'
mod_spandsp_fax.c(1618): warning C4024: 'close' : different types for formal and actual parameter 1
mod_spandsp_fax.c(1639): warning C4047: 'function' : 'int' differs in levels of indirection from 'zap_socket_t'
mod_spandsp_fax.c(1639): warning C4024: 'write' : different types for formal and actual parameter 1
mod_spandsp_fax.c(1663): warning C4047: 'function' : 'int' differs in levels of indirection from 'zap_socket_t'
mod_spandsp_fax.c(1663): warning C4024: 'write' : different types for formal and actual parameter 1
mod_spandsp_fax.c(1685): warning C4047: 'function' : 'int' differs in levels of indirection from 'zap_socket_t'
mod_spandsp_fax.c(1685): warning C4024: 'close' : different types for formal and actual parameter 1
mod_spandsp_fax.c(1690): warning C4047: 'function' : 'int' differs in levels of indirection from 'zap_socket_t'
mod_spandsp_fax.c(1690): warning C4024: 'close' : different types for formal and actual parameter 1
  udptl.c
udptl.c(557): warning C4100: 's' : unreferenced formal parameter
  Generating Code...
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_event_socket, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_event_socket.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_dptools, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_dptools.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_conference, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_conference.c
mod_conference.c(2037): warning C4018: '>' : signed/unsigned mismatch
mod_conference.c(2055): warning C4018: '<' : signed/unsigned mismatch
mod_conference.c(2292): warning C4018: '<' : signed/unsigned mismatch
mod_conference.c(2295): warning C4244: '=' : conversion from 'int32_t' to 'int16_t', possible loss of data
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_rss, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_rss.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_xml_rpc, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_xml_rpc.c
mod_xml_rpc.c(306): warning C4090: 'function' : different 'const' qualifiers
mod_xml_rpc.c(306): warning C4244: 'function' : conversion from 'int' to 'const xmlrpc_uint16_t', possible loss of data
mod_xml_rpc.c(244): warning C4189: 'status' : local variable is initialized but not referenced
mod_xml_rpc.c(385): warning C4090: 'function' : different 'const' qualifiers
mod_xml_rpc.c(465): warning C4090: 'function' : different 'const' qualifiers
mod_xml_rpc.c(466): warning C4090: 'function' : different 'const' qualifiers
mod_xml_rpc.c(467): warning C4090: 'function' : different 'const' qualifiers
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_console, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_console.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_commands, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_commands.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_dingaling, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_dingaling.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_ilbc, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_ilbc.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Skipped Build: Project: mod_cepstral, Configuration: Debug Win32 ------
Project not selected to build for this solution configuration 
------ Build started: Project: mod_hash, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_hash.c
mod_hash.c(764): warning C4244: 'function' : conversion from 'int' to 'esl_port_t', possible loss of data
mod_hash.c(818): warning C4244: '=' : conversion from 'switch_time_t' to 'uint32_t', possible loss of data
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: openssl, Configuration: Debug Win32 ------
  x509.c
  winrand.c
  version.c
  verify.c
  ts.c
  spkac.c
  speed.c
  smime.c
  sess_id.c
  s_time.c
  s_socket.c
  s_server.c
  s_client.c
  s_cb.c
  rsautl.c
  rsa.c
  req.c
  rand.c
  prime.c
  pkeyutl.c
  Generating Code...
  Compiling...
  pkeyparam.c
  pkey.c
  pkcs8.c
  pkcs7.c
  pkcs12.c
  passwd.c
  openssl.c
  ocsp.c
  nseq.c
  genrsa.c
  genpkey.c
  gendsa.c
  gendh.c
  errstr.c
  engine.c
  enc.c
  ecparam.c
  ec.c
  dsaparam.c
  dsa.c
  Generating Code...
  Compiling...
  dhparam.c
  dh.c
  dgst.c
  crl2p7.c
  crl.c
  cms.c
  ciphers.c
  ca.c
  asn1pars.c
  apps.c
  app_rand.c
  Generating Code...
  openssl.2010.vcxproj -> C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\openssl.exe
------ Build started: Project: mod_event_multicast, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_event_multicast.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_dialplan_directory, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_dialplan_directory.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_ldap, Configuration: Debug MS-LDAP Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_ldap.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_dialplan_xml, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_dialplan_xml.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Skipped Build: Project: docs, Configuration: Debug Win32 ------
Project not selected to build for this solution configuration 
------ Build started: Project: mod_speex, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_speex.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_PortAudio, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_PortAudio.c
  pa_ringbuffer.c
  pablio.c
  Generating Code...
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_sndfile, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_sndfile.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Build started: Project: mod_g729, Configuration: Debug Passthrough Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  mod_g729.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
------ Skipped Build: Project: mod_directory, Configuration: Debug Win32 ------
Project not selected to build for this solution configuration 
------ Build started: Project: FreeSwitchConsole, Configuration: Debug Win32 ------
cl : Command line warning D9040: ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
  switch.c
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll 1\Documents\FreeSWITCH\freeswitch\Win32\Debug\FreeSwitchCore.lib'
========== Build: 69 succeeded, 72 failed, 0 up-to-date, 17 skipped ==========

From infos at madovsky.org  Sat Jan  1 02:53:12 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 31 Dec 2010 18:53:12 -0500
Subject: [Freeswitch-users] tone_detect and dinging
References: <C5A8430009F1461592CDA253B4F64DC9@e1705>
	<23BB6F88-A7A8-4A84-A325-D4974B10BD41@gmail.com>
Message-ID: <EE2EAA0A62754F2186DC0BC81EC34DFB@e1705>

yes it is.
  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Friday, December 31, 2010 5:48 PM
  Subject: Re: [Freeswitch-users] tone_detect and dinging


  Is tone_detect answering the call?


  Enable debug level log output and you'll be able to see what's going on.

  Steve on iPhone

  On 31 Dec 2010, at 20:51, "Madovsky" <infos at madovsky.org> wrote:


    I use this in my dialplan before a bridge

    ...
    <action application="set" data="instant_ringback=true"/>
    <action application="ring_ready"/>
    <action application="tone_detect" data="fax 1100 r +5000 transfer 'receivefax XML features' 1" />
    ....

    but no ring is back to the caller. if I remove tone_detect the ringback is working again.
    is anyone knows why ?

    Thanks
    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org



------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From anthony.minessale at gmail.com  Sat Jan  1 08:46:44 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 31 Dec 2010 23:46:44 -0600
Subject: [Freeswitch-users] tone_detect and dinging
In-Reply-To: <AANLkTik7HWHgUZ9B3GogE-59twhyiAVMAUdviBY-QMvC@mail.gmail.com>
References: <C5A8430009F1461592CDA253B4F64DC9@e1705>
	<23BB6F88-A7A8-4A84-A325-D4974B10BD41@gmail.com>
	<EE2EAA0A62754F2186DC0BC81EC34DFB@e1705>
	<AANLkTik7HWHgUZ9B3GogE-59twhyiAVMAUdviBY-QMvC@mail.gmail.com>
Message-ID: <AANLkTi=gh6YHnaa_XcY=-AjPpLiZ4nzXHtW+3bmaJ_Ft@mail.gmail.com>

It pre answers early media because it requires media to work. Activate it in
execute_on_media instead
On Dec 31, 2010 5:53 PM, "Madovsky" <infos at madovsky.org> wrote:
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From infos at madovsky.org  Sat Jan  1 09:36:51 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 1 Jan 2011 01:36:51 -0500
Subject: [Freeswitch-users] tone_detect and dinging
References: <C5A8430009F1461592CDA253B4F64DC9@e1705><23BB6F88-A7A8-4A84-A325-D4974B10BD41@gmail.com><EE2EAA0A62754F2186DC0BC81EC34DFB@e1705><AANLkTik7HWHgUZ9B3GogE-59twhyiAVMAUdviBY-QMvC@mail.gmail.com>
	<AANLkTi=gh6YHnaa_XcY=-AjPpLiZ4nzXHtW+3bmaJ_Ft@mail.gmail.com>
Message-ID: <6F843F7F37D144EDB027067B323A3DA8@e1705>

ok thanks and Happy New Year ;)
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: FreeSWITCH Users Help 
  Sent: Saturday, January 01, 2011 12:46 AM
  Subject: Re: [Freeswitch-users] tone_detect and dinging


  It pre answers early media because it requires media to work. Activate it in execute_on_media instead

  On Dec 31, 2010 5:53 PM, "Madovsky" <infos at madovsky.org> wrote:



------------------------------------------------------------------------------


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From babak.freeswitch at gmail.com  Sat Jan  1 16:33:20 2011
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sat, 1 Jan 2011 17:03:20 +0330
Subject: [Freeswitch-users] Can't build on Windows 7 64bit C++ Express!?
In-Reply-To: <AANLkTincSQSt-Pg00T4qVG+GXfVw2XpE6FU+2Ffe253y@mail.gmail.com>
References: <AANLkTincSQSt-Pg00T4qVG+GXfVw2XpE6FU+2Ffe253y@mail.gmail.com>
Message-ID: <AANLkTinPiO-fMHGc7jVj=if5mo1yykVWW8qNgn_NcFNv@mail.gmail.com>

if u need mod_managed u should build it on vc# express cause vc++ can not
build it
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From babak.freeswitch at gmail.com  Sat Jan  1 19:13:22 2011
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sat, 1 Jan 2011 19:43:22 +0330
Subject: [Freeswitch-users] session_in_hangup_hook
Message-ID: <AANLkTi=j5oxuqO=5GE+3ghL2jbZDwV7i-Mz3Buzjk9K4@mail.gmail.com>

Happy new year
I'm trying to get spandsp fax related channel variables after hangup. I'm
using session_in_hangup_hook=true in mod_managed. some times it is working
(session variables are accessible) and some times not. I've read on the ML
that it's just applicable in lua javas.... but why it is sometimes
working,should I do any special configuration so it work always!?
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From infos at madovsky.org  Sat Jan  1 19:28:08 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 1 Jan 2011 11:28:08 -0500
Subject: [Freeswitch-users] session_in_hangup_hook
References: <AANLkTi=j5oxuqO=5GE+3ghL2jbZDwV7i-Mz3Buzjk9K4@mail.gmail.com>
Message-ID: <94AB4D8CC2074310B120A74E682F7F14@e1705>

you can use also Perl and bash script.
for me it works at any time, using last git
  ----- Original Message ----- 
  From: babak yakhchali 
  To: FreeSWITCH Users Help 
  Sent: Saturday, January 01, 2011 11:13 AM
  Subject: [Freeswitch-users] session_in_hangup_hook


  Happy new year

  I'm trying to get spandsp fax related channel variables after hangup. I'm using session_in_hangup_hook=true in mod_managed. some times it is working (session variables are accessible) and some times not. I've read on the ML that it's just applicable in lua javas.... but why it is sometimes working,should I do any special configuration so it work always!?


------------------------------------------------------------------------------


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From infos at madovsky.org  Sat Jan  1 22:12:03 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 1 Jan 2011 14:12:03 -0500
Subject: [Freeswitch-users] tone_detect and dinging
References: <C5A8430009F1461592CDA253B4F64DC9@e1705><23BB6F88-A7A8-4A84-A325-D4974B10BD41@gmail.com><EE2EAA0A62754F2186DC0BC81EC34DFB@e1705><AANLkTik7HWHgUZ9B3GogE-59twhyiAVMAUdviBY-QMvC@mail.gmail.com>
	<AANLkTi=gh6YHnaa_XcY=-AjPpLiZ4nzXHtW+3bmaJ_Ft@mail.gmail.com>
Message-ID: <4063AE799A58452F8CAFF7D38FA07C03@e1705>

it works well now, thanks Tony.
I will try to add some wiki lines for execute_on_media and execute_on_preanswer,
there are not on channel variables page
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: FreeSWITCH Users Help 
  Sent: Saturday, January 01, 2011 12:46 AM
  Subject: Re: [Freeswitch-users] tone_detect and dinging


  It pre answers early media because it requires media to work. Activate it in execute_on_media instead

  On Dec 31, 2010 5:53 PM, "Madovsky" <infos at madovsky.org> wrote:



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From infos at madovsky.org  Sat Jan  1 23:13:39 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 1 Jan 2011 15:13:39 -0500
Subject: [Freeswitch-users] voicmeail operator transfer
Message-ID: <073D091BA0994867A629B371BEEFCA68@e1705>

in voicemail.conf.xml I have this

      <param name="operator-key" value="9"/>
      <param name="operator-extension" value="conference XML features"/>

it works well but are all channel variables transferred also ?

Thanks
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From Avi at aMarcus.com  Sat Jan  1 23:20:32 2011
From: Avi at aMarcus.com (Avi Marcus)
Date: Sat, 1 Jan 2011 22:20:32 +0200
Subject: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please Help
	setting up
Message-ID: <AANLkTi=6ZwmiMHLcBM7EndP8apmXf7CO9XFied1UOvRh@mail.gmail.com>

Hi - I'm setting up 2 parallel computers for a HA setup. (low volume so far,
no need to load balance.)
I already have heartbeat able to grab the public IP if the first box goes
down, but since I'm basically completely unfamiar with pacemaker, I'd like
some help.
I'd imagine I'm not the only one doing this, there should be no reason for
me to recreate the wheel.

Info should be on: http://wiki.freeswitch.org/wiki/Freeswitch_HA

Specifically:
1) It seems freeswitch takes 8 seconds to restart. So if it crashes
(unlikely) calls should immediately get shunted over to the other box. How
do I set that up?
2) How do I automatically trigger a sofia recover? I know via commanline
it's "fs_cli -x "sofia recover" but where do I tell pacemaker to do that?
And I suppose a reload_xml before (maybe after?) is a good idea, too.

Also, can I put <param name="track-calls" value="true"/> in a global file,
or does it need to be in the actual profile files?
And anything else I may be overlooking.

Thanks guys!
-Avi Marcus
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From infos at madovsky.org  Sat Jan  1 23:31:05 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 1 Jan 2011 15:31:05 -0500
Subject: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please
	Helpsetting up
References: <AANLkTi=6ZwmiMHLcBM7EndP8apmXf7CO9XFied1UOvRh@mail.gmail.com>
Message-ID: <EDF753B814C44E44BD108AE6DCEF73E8@e1705>

1)  and 2), use startup script in /etc/init.d to do whatever you want
HB/Pacemaker is reacting only from these scripts (LSB, OCF compliant)
use also cron tasks
  ----- Original Message ----- 
  From: Avi Marcus 
  To: FreeSWITCH Users Help 
  Sent: Saturday, January 01, 2011 3:20 PM
  Subject: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please Helpsetting up


  Hi - I'm setting up 2 parallel computers for a HA setup. (low volume so far, no need to load balance.)
  I already have heartbeat able to grab the public IP if the first box goes down, but since I'm basically completely unfamiar with pacemaker, I'd like some help.
  I'd imagine I'm not the only one doing this, there should be no reason for me to recreate the wheel.


  Info should be on: http://wiki.freeswitch.org/wiki/Freeswitch_HA


  Specifically:
  1) It seems freeswitch takes 8 seconds to restart. So if it crashes (unlikely) calls should immediately get shunted over to the other box. How do I set that up?
  2) How do I automatically trigger a sofia recover? I know via commanline it's "fs_cli -x "sofia recover" but where do I tell pacemaker to do that? And I suppose a reload_xml before (maybe after?) is a good idea, too.


  Also, can I put <param name="track-calls" value="true"/> in a global file, or does it need to be in the actual profile files?
  And anything else I may be overlooking.


  Thanks guys!
  -Avi Marcus




------------------------------------------------------------------------------


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From Avi at aMarcus.com  Sat Jan  1 23:46:12 2011
From: Avi at aMarcus.com (Avi Marcus)
Date: Sat, 1 Jan 2011 22:46:12 +0200
Subject: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please
 Helpsetting up
In-Reply-To: <EDF753B814C44E44BD108AE6DCEF73E8@e1705>
References: <AANLkTi=6ZwmiMHLcBM7EndP8apmXf7CO9XFied1UOvRh@mail.gmail.com>
	<EDF753B814C44E44BD108AE6DCEF73E8@e1705>
Message-ID: <AANLkTi=Vhia_GWskAB4mX_nSMNdb-wALyGW8__2sy94o@mail.gmail.com>

I had planned on leaving freeswitch running on both machines because it
seemed to take 8 seconds to start up and I'd like instant recovery.
So.. I'm not dealing with init scripts?

I've not seen much documentation on ha/pacemaker that made sense to me, can
you please explain what you mean?
-Avi

On Sat, Jan 1, 2011 at 10:31 PM, Madovsky <infos at madovsky.org> wrote:

>  1)  and 2), use startup script in /etc/init.d to do whatever you want
> HB/Pacemaker is reacting only from these scripts (LSB, OCF compliant)
> use also cron tasks
>
> ----- Original Message -----
> *From:* Avi Marcus <Avi at aMarcus.com>
> *To:* FreeSWITCH Users Help <FreeSWITCH-users at lists.freeswitch.org>
> *Sent:* Saturday, January 01, 2011 3:20 PM
> *Subject:* [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please
> Helpsetting up
>
> Hi - I'm setting up 2 parallel computers for a HA setup. (low volume so
> far, no need to load balance.)
> I already have heartbeat able to grab the public IP if the first box goes
> down, but since I'm basically completely unfamiar with pacemaker, I'd like
> some help.
> I'd imagine I'm not the only one doing this, there should be no reason for
> me to recreate the wheel.
>
> Info should be on: http://wiki.freeswitch.org/wiki/Freeswitch_HA
>
> Specifically:
> 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
> (unlikely) calls should immediately get shunted over to the other box. How
> do I set that up?
> 2) How do I automatically trigger a sofia recover? I know via commanline
> it's "fs_cli -x "sofia recover" but where do I tell pacemaker to do that?
> And I suppose a reload_xml before (maybe after?) is a good idea, too.
>
> Also, can I put <param name="track-calls" value="true"/> in a global file,
> or does it need to be in the actual profile files?
> And anything else I may be overlooking.
>
> Thanks guys!
> -Avi Marcus
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
>
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From infos at madovsky.org  Sat Jan  1 23:57:56 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 1 Jan 2011 15:57:56 -0500
Subject: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please
	Helpsetting up
References: <AANLkTi=6ZwmiMHLcBM7EndP8apmXf7CO9XFied1UOvRh@mail.gmail.com><EDF753B814C44E44BD108AE6DCEF73E8@e1705>
	<AANLkTi=Vhia_GWskAB4mX_nSMNdb-wALyGW8__2sy94o@mail.gmail.com>
Message-ID: <8611CEF54D7047ADB28396926C1568C7@e1705>

I mean that if you want specific procedure witn freeswitch service so you have to create
bash script inside your init script because HB/Pacemaker don't care every specification of every service
  ----- Original Message ----- 
  From: Avi Marcus 
  To: FreeSWITCH Users Help 
  Sent: Saturday, January 01, 2011 3:46 PM
  Subject: Re: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please Helpsetting up


  I had planned on leaving freeswitch running on both machines because it seemed to take 8 seconds to start up and I'd like instant recovery.
  So.. I'm not dealing with init scripts?


  I've not seen much documentation on ha/pacemaker that made sense to me, can you please explain what you mean?
  -Avi


  On Sat, Jan 1, 2011 at 10:31 PM, Madovsky <infos at madovsky.org> wrote:

    1)  and 2), use startup script in /etc/init.d to do whatever you want
    HB/Pacemaker is reacting only from these scripts (LSB, OCF compliant)
    use also cron tasks
      ----- Original Message ----- 
      From: Avi Marcus 
      To: FreeSWITCH Users Help 
      Sent: Saturday, January 01, 2011 3:20 PM
      Subject: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please Helpsetting up


      Hi - I'm setting up 2 parallel computers for a HA setup. (low volume so far, no need to load balance.) 
      I already have heartbeat able to grab the public IP if the first box goes down, but since I'm basically completely unfamiar with pacemaker, I'd like some help.
      I'd imagine I'm not the only one doing this, there should be no reason for me to recreate the wheel.


      Info should be on: http://wiki.freeswitch.org/wiki/Freeswitch_HA


      Specifically:
      1) It seems freeswitch takes 8 seconds to restart. So if it crashes (unlikely) calls should immediately get shunted over to the other box. How do I set that up?
      2) How do I automatically trigger a sofia recover? I know via commanline it's "fs_cli -x "sofia recover" but where do I tell pacemaker to do that? And I suppose a reload_xml before (maybe after?) is a good idea, too.


      Also, can I put <param name="track-calls" value="true"/> in a global file, or does it need to be in the actual profile files?
      And anything else I may be overlooking.


      Thanks guys!
      -Avi Marcus




--------------------------------------------------------------------------


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From infos at madovsky.org  Sun Jan  2 00:08:51 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 1 Jan 2011 16:08:51 -0500
Subject: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please
	Helpsetting up
Message-ID: <10E419E930D94077B5A29001ED71734D@e1705>

also be prepared to learn intensively for at least 6 months.
cluster administration is not an easy task...
  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Saturday, January 01, 2011 3:57 PM
  Subject: Re: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please Helpsetting up


  I mean that if you want specific procedure witn freeswitch service so you have to create
  bash script inside your init script because HB/Pacemaker don't care every specification of every service
    ----- Original Message ----- 
    From: Avi Marcus 
    To: FreeSWITCH Users Help 
    Sent: Saturday, January 01, 2011 3:46 PM
    Subject: Re: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please Helpsetting up


    I had planned on leaving freeswitch running on both machines because it seemed to take 8 seconds to start up and I'd like instant recovery. 
    So.. I'm not dealing with init scripts?


    I've not seen much documentation on ha/pacemaker that made sense to me, can you please explain what you mean?
    -Avi


    On Sat, Jan 1, 2011 at 10:31 PM, Madovsky <infos at madovsky.org> wrote:

      1)  and 2), use startup script in /etc/init.d to do whatever you want
      HB/Pacemaker is reacting only from these scripts (LSB, OCF compliant)
      use also cron tasks
        ----- Original Message ----- 
        From: Avi Marcus 
        To: FreeSWITCH Users Help 
        Sent: Saturday, January 01, 2011 3:20 PM
        Subject: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please Helpsetting up


        Hi - I'm setting up 2 parallel computers for a HA setup. (low volume so far, no need to load balance.) 
        I already have heartbeat able to grab the public IP if the first box goes down, but since I'm basically completely unfamiar with pacemaker, I'd like some help.
        I'd imagine I'm not the only one doing this, there should be no reason for me to recreate the wheel.


        Info should be on: http://wiki.freeswitch.org/wiki/Freeswitch_HA


        Specifically:
        1) It seems freeswitch takes 8 seconds to restart. So if it crashes (unlikely) calls should immediately get shunted over to the other box. How do I set that up?
        2) How do I automatically trigger a sofia recover? I know via commanline it's "fs_cli -x "sofia recover" but where do I tell pacemaker to do that? And I suppose a reload_xml before (maybe after?) is a good idea, too.


        Also, can I put <param name="track-calls" value="true"/> in a global file, or does it need to be in the actual profile files?
        And anything else I may be overlooking.


        Thanks guys!
        -Avi Marcus




------------------------------------------------------------------------


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        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Sun Jan  2 00:33:45 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 1 Jan 2011 16:33:45 -0500
Subject: [Freeswitch-users] user variables access
Message-ID: <97830643AF844DFD81290B671FF97033@e1705>


is it possible to access to legB vars (of course if legB is a FS user)
in XML dialplan ? if yes how ?
I need to check the vars I set in  directory/default/legB_user.xml

thanks
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From infos at madovsky.org  Sun Jan  2 00:39:21 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 1 Jan 2011 16:39:21 -0500
Subject: [Freeswitch-users] user variables access
References: <97830643AF844DFD81290B671FF97033@e1705>
Message-ID: <F5461F85BE7B4DA8B76DCE0FED482045@e1705>

ok found the answer

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, January 01, 2011 4:33 PM
  Subject: [Freeswitch-users] user variables access



  is it possible to access to legB vars (of course if legB is a FS user)
  in XML dialplan ? if yes how ?
  I need to check the vars I set in  directory/default/legB_user.xml

  thanks


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From Avi at aMarcus.com  Sun Jan  2 00:39:58 2011
From: Avi at aMarcus.com (Avi Marcus)
Date: Sat, 1 Jan 2011 23:39:58 +0200
Subject: [Freeswitch-users] user variables access
In-Reply-To: <97830643AF844DFD81290B671FF97033@e1705>
References: <97830643AF844DFD81290B671FF97033@e1705>
Message-ID: <AANLkTin1i_J_Hs2qAQZeuxzBrbYKg72YDeSKLKhxo0OU@mail.gmail.com>

check "import". Or export it from leg b.
-Avi

On Sat, Jan 1, 2011 at 11:33 PM, Madovsky <infos at madovsky.org> wrote:

>
> is it possible to access to legB vars (of course if legB is a FS user)
> in XML dialplan ? if yes how ?
> I need to check the vars I set in  directory/default/legB_user.xml
>
> thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From frank at telonium.com  Sun Jan  2 00:46:25 2011
From: frank at telonium.com (Frank Park)
Date: Sat, 1 Jan 2011 16:46:25 -0500
Subject: [Freeswitch-users] Logging "dropped calls"
Message-ID: <AANLkTi=OP-eVAr2QpxEghVN1v-ASFHUgC9n_3MyPRsm1@mail.gmail.com>

Happy New Years!

This is merely a wish list for me, but I wanted to know if there's a
way for me to log all calls that disconnects without the proper SIP
termination..
I would like to build some level of debug tool where I can track
dropped calls due to freeswitch, or due to any other external issues
(which I assume most are).
In the same statistics, I am also interested in logging who initially
sent the BYE message (the caller, the callee, or the switch).

A direction to do this might be sufficient at this point.

Thank you!
Frank



----=======================----
Frank Park
Telonium Communications, LLC
frank at telonium.com
http://www.telonium.com
Follow Us on Twitter: @GetTelonium
404-566-8888 x1001 Office
404-939-4242 Cell
----=======================----


From steveayre at gmail.com  Sun Jan  2 02:58:55 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 1 Jan 2011 23:58:55 +0000
Subject: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please
	Help setting up
In-Reply-To: <AANLkTi=6ZwmiMHLcBM7EndP8apmXf7CO9XFied1UOvRh@mail.gmail.com>
References: <AANLkTi=6ZwmiMHLcBM7EndP8apmXf7CO9XFied1UOvRh@mail.gmail.com>
Message-ID: <65B881CD-D420-444B-91FD-04360AB97644@gmail.com>

> 1) It seems freeswitch takes 8 seconds to restart. So if it crashes (unlikely) calls should immediately get shunted over to the other box. How do I set that up?

Just a tip, I start fs with -nonat and find that makes it start faster.

Steve on iPhone

On 1 Jan 2011, at 20:20, Avi Marcus <Avi at aMarcus.com> wrote:

> Hi - I'm setting up 2 parallel computers for a HA setup. (low volume so far, no need to load balance.)
> I already have heartbeat able to grab the public IP if the first box goes down, but since I'm basically completely unfamiar with pacemaker, I'd like some help.
> I'd imagine I'm not the only one doing this, there should be no reason for me to recreate the wheel.
> 
> Info should be on: http://wiki.freeswitch.org/wiki/Freeswitch_HA
> 
> Specifically:
> 1) It seems freeswitch takes 8 seconds to restart. So if it crashes (unlikely) calls should immediately get shunted over to the other box. How do I set that up?
> 2) How do I automatically trigger a sofia recover? I know via commanline it's "fs_cli -x "sofia recover" but where do I tell pacemaker to do that? And I suppose a reload_xml before (maybe after?) is a good idea, too.
> 
> Also, can I put <param name="track-calls" value="true"/> in a global file, or does it need to be in the actual profile files?
> And anything else I may be overlooking.
> 
> Thanks guys!
> -Avi Marcus
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From george.niculae79 at gmail.com  Sun Jan  2 03:12:45 2011
From: george.niculae79 at gmail.com (George Niculae)
Date: Sun, 2 Jan 2011 02:12:45 +0200
Subject: [Freeswitch-users] call dropped while trying to transfer
In-Reply-To: <AANLkTikxbsAKhsybVp2SKG7_t3CmB+Hv_HZ_fxSAgbEb@mail.gmail.com>
References: <AANLkTiku8ZRDxFxQECys14BW94X2J28UpxKZu+p_BOXq@mail.gmail.com>
	<AANLkTin0Lmz64n+chbcQ6UqYY704bcim+yOf5AOcV6sv@mail.gmail.com>
	<AANLkTintrgfrhB33pRydJzniksont2BPtu0eVpj+=fUG@mail.gmail.com>
	<AANLkTinEiSKfA3Aw4LjXQ6cEYdEqiGN87thUUHQEh7QQ@mail.gmail.com>
	<AANLkTi==EKk_igs7kNnzgiS8XZs9S5L_Dp1=NcXBma0+@mail.gmail.com>
	<AANLkTikxbsAKhsybVp2SKG7_t3CmB+Hv_HZ_fxSAgbEb@mail.gmail.com>
Message-ID: <AANLkTiknRJradKkiGDZE3PqmbsNeWdWmQDdDg_hE3E60@mail.gmail.com>

I figured that this is due to a bug in my code (bad event handling) by
creating a lua script that performs exactly the same steps as the java app.

Thanks for your help,
George

2010/12/28 George Niculae <george.niculae79 at gmail.com>

> Hi Michael,
>
> is the provided trace OK or should I collect new logs?
>
> Thanks,
> George
>
> 2010/12/23 George Niculae <george.niculae79 at gmail.com>
>
> Here it is: http://pastebin.freeswitch.org/14873
>>
>> <http://pastebin.freeswitch.org/14873>Thanks,
>> George
>>
>> 2010/12/23 Michael Collins <msc at freeswitch.org>
>>
>> Try turning on the siptrace as well so we can see the sip traffic:
>>>
>>> sofia profile internal siptrace on
>>>
>>> Then do another test & pastebin the debug output.
>>> -MC
>>>
>>>
>>> On Thu, Dec 23, 2010 at 4:36 AM, George Niculae <george at ezuce.com>wrote:
>>>
>>>> Michael,
>>>>
>>>> the commands are written on socket using PrintWriter.printf() and in
>>>> this case is something like:
>>>> api uuid_deflect f5539b24-0e8e-11e0-9a0e-c37fe40448c1
>>>> sip:101 at dizzy.dizzysip.ro <sip%3A101 at dizzy.dizzysip.ro>
>>>> Please see here all commands sent (prefixed with FSES::cmd):
>>>> http://pastebin.freeswitch.org/14868 , uuid deflect at line 27
>>>> New console output (for correlating uuid's if needed):
>>>> http://pastebin.freeswitch.org/14867
>>>>
>>>> Thanks,
>>>> George
>>>>
>>>> On Thu, Dec 23, 2010 at 2:07 AM, Michael Collins <msc at freeswitch.org>
>>>> wrote:
>>>> > Please pastebin the code that performs the uuid_deflect so that we can
>>>> see
>>>> > what you are doing to produce this symptom.
>>>> > -MC
>>>> >
>>>> > On Wed, Dec 22, 2010 at 8:35 AM, George Niculae <george at ezuce.com>
>>>> wrote:
>>>> >>
>>>> >> Hi All,
>>>> >>
>>>> >> I am working on an IVR application based on FS (running FreeSWITCH
>>>> >> Version 1.0.head (git-43393f2 2010-12-15 20-59-42 -0600) where the
>>>> >> following scenario fails:
>>>> >> user 201 calls to 100 (autoattendant), hears menu then press # to
>>>> >> transfer to voicemail (101), but the call is dropped (transfer is
>>>> made
>>>> >> using uuid_deflect api command)
>>>> >> Dialplan extension configured like:
>>>> >>
>>>> >>  <extension name="IVR">
>>>> >>    <condition field="destination_number" expression="^IVR${dollar}">
>>>> >>      <action application="socket" data="192.168.0.114:8084 async
>>>> full"/>
>>>> >>    </condition>
>>>> >>  </extension>
>>>> >>
>>>> >> Actions taken are:
>>>> >> - when call arrives to extension 100 call is bridged
>>>> >> (hangup_after_bridge=true)
>>>> >> - answer the call, autoattendant menu is played and DTMF collected
>>>> >> - when # pressed, call is transfered to 101 using uuid_deflect
>>>> >> - call arrives to voicemail extension and is again bridged
>>>> >> - call is answered - at this point in time the initial bridge hangs
>>>> up
>>>> >> and the whole call is dropped
>>>> >> Please see console output http://pastebin.freeswitch.org/14855
>>>> >>
>>>> >> When debugging the application, If I keep the first channel connected
>>>> >> transfer works just fine without dropping the call.
>>>> >> Pretty sure I'm missing something here, any suggestion highly
>>>> appreciated
>>>> >>
>>>> >> Thanks,
>>>> >> George
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
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From dome at tel.co.th  Sun Jan  2 04:38:05 2011
From: dome at tel.co.th (dome at tel.co.th)
Date: Sun, 2 Jan 2011 08:38:05 +0700
Subject: [Freeswitch-users] increase Max session
Message-ID: <AANLkTik=Fa3ZAX5XP_ubZ7ekrx8M8A7D5J0L=fqepEUg@mail.gmail.com>

Dear All,
    My system running with FS 1.0.6 on debian over openvz x86_64. i'm
seting up max session to 2000 after live traffic coming max session
reduce to 917.
FS run in none proxy mode and CPU use about 5%
     before start FS i use ulimit -s 240  and other

So i want to know how to increase max session to 2000.

BG

Dome C.


From lloyd.aloysius at gmail.com  Sun Jan  2 04:51:46 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sat, 1 Jan 2011 20:51:46 -0500
Subject: [Freeswitch-users] voicemail operator-extension -
	Multi-tenantEnvironment
In-Reply-To: <AANLkTin5nFvEUwUmNj9kf1cHx7wTgcNKsLWKF8D64iw3@mail.gmail.com>
References: <AANLkTimm99xdzt2cW6zTpFZY+FtiRCaHCQSm=Rv__o4G@mail.gmail.com>
	<614459D4C10248C8B913E4422CFC7C35@e1705>
	<D330BE13-B0A1-4D76-9DFA-B80B7F3534D8@gmail.com>
	<AANLkTinATfPEbDkzfZSyXHgVYM0mjtsgvzk28xYBotXt@mail.gmail.com>
	<AANLkTik=afD7Zjsb_jPw1aNSXsxLCtHeshiVbK-aHTbA@mail.gmail.com>
	<AANLkTikfETbZCi080u1uuLQj-VkezQf_2+=eWBUv5mZ+@mail.gmail.com>
	<AANLkTimbpDJ3QrkfzvVygtSA1s-KmpV_f8TKYU4RLgf7@mail.gmail.com>
	<AANLkTin5nFvEUwUmNj9kf1cHx7wTgcNKsLWKF8D64iw3@mail.gmail.com>
Message-ID: <AANLkTikAsVVG=TBEcvOFVVGvhDTrxxW6CRz+oLZ6vHcm@mail.gmail.com>

Dialplan sending calls to voicemail

<action application="bridge" data="loopback/app=voicemail:default
${domain_name} ${dialed_extension}"/>

voicemail.conf.xml

 <param name="operator-extension" value="operator XML features"/>
 <param name="operator-key" value="9"/>

features.xml

    <extension name="operator">
      <condition field="destination_number" expression="^(operator|0)$">
        <action application="set" data="transfer_ringback=$${hold_music}"/>
<action application="transfer" data="1000 XML features"/>
        <action application="transfer" data="operator XML ${domain_name}"/>
      </condition>
    </extension>


Domain Name - compaya.com

pressing 9 .... voice mail operator extension transfer to features context.
But  ${domain_name} lost the value compaya.com , but now the ${ domain_name}
have the IP address.

How to get the ${domain_name} value in features context?



Thanks and regards,
Lloyd

On Tue, Dec 28, 2010 at 5:44 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> When I press 9 the call get transfered to the default context. Then I try
> get the ${domain_name} that is giving the default domain_name. I could not
> find a way to get the correct voicemal domain_name from the from default
> context.
>
> xml_curl .... right now I am using xml_curl. All users defined in mysql,
>  then I use a php script for the user informations.
>
> How to use xml_curl for voicemail ? please let me know if there any
> help/docs on this.
>
> Thanks
> LLoyd
>
>
> On Tue, Dec 28, 2010 at 4:29 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Tue, Dec 28, 2010 at 9:57 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com
>> > wrote:
>>
>>> Michael,
>>>
>>> Thank you for the suggestion but this is not working . I think because of
>>> the the  <param name="callback-context" value="default"/>
>>>
>>
>> Please define "not working" - either you press 9 and the call is x-fer'd
>> or it is not. Once it is x-fer'd to the dialplan you should be able to do
>> whatever you want with the call. Or do what bkw says and use xml_curl.
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From lloyd.aloysius at gmail.com  Sun Jan  2 04:56:39 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sat, 1 Jan 2011 20:56:39 -0500
Subject: [Freeswitch-users] voicemail operator-extension -
	Multi-tenantEnvironment
In-Reply-To: <AANLkTikAsVVG=TBEcvOFVVGvhDTrxxW6CRz+oLZ6vHcm@mail.gmail.com>
References: <AANLkTimm99xdzt2cW6zTpFZY+FtiRCaHCQSm=Rv__o4G@mail.gmail.com>
	<614459D4C10248C8B913E4422CFC7C35@e1705>
	<D330BE13-B0A1-4D76-9DFA-B80B7F3534D8@gmail.com>
	<AANLkTinATfPEbDkzfZSyXHgVYM0mjtsgvzk28xYBotXt@mail.gmail.com>
	<AANLkTik=afD7Zjsb_jPw1aNSXsxLCtHeshiVbK-aHTbA@mail.gmail.com>
	<AANLkTikfETbZCi080u1uuLQj-VkezQf_2+=eWBUv5mZ+@mail.gmail.com>
	<AANLkTimbpDJ3QrkfzvVygtSA1s-KmpV_f8TKYU4RLgf7@mail.gmail.com>
	<AANLkTin5nFvEUwUmNj9kf1cHx7wTgcNKsLWKF8D64iw3@mail.gmail.com>
	<AANLkTikAsVVG=TBEcvOFVVGvhDTrxxW6CRz+oLZ6vHcm@mail.gmail.com>
Message-ID: <AANLkTinWG0GfDSn-YZ0bHrf_=sd5uuqEehy+k=Xz+8ZN@mail.gmail.com>

There is a mistake in the last email . Please ignore.

Dialplan sending calls to voicemail

<action application="bridge" data="loopback/app=voicemail:default
${domain_name} ${dialed_extension}"/>

voicemail.conf.xml

 <param name="operator-extension" value="operator XML features"/>
 <param name="operator-key" value="9"/>

features.xml

    <extension name="operator">
      <condition field="destination_number" expression="^(operator|0)$">
        <action application="set" data="transfer_ringback=$${hold_music}"/>
 <action application="transfer" data="operator XML ${domain_name}"/>
      </condition>
    </extension>


Domain Name - compaya.com

pressing 9 .... voice mail operator extension transfer to features context.
But  ${domain_name} lost the value compaya.com , but now the ${ domain_name}
have the IP address.

How to get the ${domain_name} value in features context?



Thanks and regards,
Lloyd


On Sat, Jan 1, 2011 at 8:51 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Dialplan sending calls to voicemail
>
> <action application="bridge" data="loopback/app=voicemail:default
> ${domain_name} ${dialed_extension}"/>
>
> voicemail.conf.xml
>
>  <param name="operator-extension" value="operator XML features"/>
>  <param name="operator-key" value="9"/>
>
> features.xml
>
>     <extension name="operator">
>       <condition field="destination_number" expression="^(operator|0)$">
>         <action application="set" data="transfer_ringback=$${hold_music}"/>
> <action application="transfer" data="1000 XML features"/>
>          <action application="transfer" data="operator XML
> ${domain_name}"/>
>       </condition>
>     </extension>
>
>
>  Domain Name - compaya.com
>
> pressing 9 .... voice mail operator extension transfer to features context.
> But  ${domain_name} lost the value compaya.com , but now the ${
> domain_name} have the IP address.
>
> How to get the ${domain_name} value in features context?
>
>
>
> Thanks and regards,
> Lloyd
>
>
> On Tue, Dec 28, 2010 at 5:44 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> When I press 9 the call get transfered to the default context. Then I try
>> get the ${domain_name} that is giving the default domain_name. I could not
>> find a way to get the correct voicemal domain_name from the from default
>> context.
>>
>> xml_curl .... right now I am using xml_curl. All users defined in mysql,
>>  then I use a php script for the user informations.
>>
>> How to use xml_curl for voicemail ? please let me know if there any
>> help/docs on this.
>>
>> Thanks
>> LLoyd
>>
>>
>> On Tue, Dec 28, 2010 at 4:29 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>>
>>>
>>> On Tue, Dec 28, 2010 at 9:57 AM, Aloysius Lloyd <
>>> lloyd.aloysius at gmail.com> wrote:
>>>
>>>> Michael,
>>>>
>>>> Thank you for the suggestion but this is not working . I
>>>> think because of the the  <param name="callback-context"
>>>> value="default"/>
>>>>
>>>
>>> Please define "not working" - either you press 9 and the call is x-fer'd
>>> or it is not. Once it is x-fer'd to the dialplan you should be able to do
>>> whatever you want with the call. Or do what bkw says and use xml_curl.
>>> -MC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
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From lloyd.aloysius at gmail.com  Sun Jan  2 05:04:03 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sat, 1 Jan 2011 21:04:03 -0500
Subject: [Freeswitch-users] increase Max session
In-Reply-To: <AANLkTik=Fa3ZAX5XP_ubZ7ekrx8M8A7D5J0L=fqepEUg@mail.gmail.com>
References: <AANLkTik=Fa3ZAX5XP_ubZ7ekrx8M8A7D5J0L=fqepEUg@mail.gmail.com>
Message-ID: <AANLkTimZxTYeGGNnSKF2x07wS6mBhU5kyB6xbeTKcAmc@mail.gmail.com>

autoload_configs/switch.conf.xml

<param name="max-sessions" value="1000"/>


Thanks
Lloyd

On Sat, Jan 1, 2011 at 8:38 PM, dome at tel.co.th <dome at tel.co.th> wrote:

> Dear All,
>    My system running with FS 1.0.6 on debian over openvz x86_64. i'm
> seting up max session to 2000 after live traffic coming max session
> reduce to 917.
> FS run in none proxy mode and CPU use about 5%
>     before start FS i use ulimit -s 240  and other
>
> So i want to know how to increase max session to 2000.
>
> BG
>
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dujinfang at gmail.com  Sun Jan  2 05:06:28 2011
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 2 Jan 2011 10:06:28 +0800
Subject: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please
 Help setting up
In-Reply-To: <65B881CD-D420-444B-91FD-04360AB97644@gmail.com>
References: <AANLkTi=6ZwmiMHLcBM7EndP8apmXf7CO9XFied1UOvRh@mail.gmail.com>
	<65B881CD-D420-444B-91FD-04360AB97644@gmail.com>
Message-ID: <AANLkTinoFOPtPW+=uyoogi33u6LvAYz_ta3bDc7XMhs0@mail.gmail.com>

the wiki page already said you can start the spare FS ahead to speed
switch over up.

it means if you use ip_nonlocal_bind and start the spare FS ahead, you
only need to run sofia recover in you init script without waiting for
the actually FS start process.

On Sun, Jan 2, 2011 at 7:58 AM, Steven Ayre <steveayre at gmail.com> wrote:
> 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
> (unlikely) calls should?immediately?get shunted over to the other box. How
> do I set that up?
>
> Just a tip, I start fs with -nonat and find that makes it start faster.
>
> Steve on iPhone
> On 1 Jan 2011, at 20:20, Avi Marcus <Avi at aMarcus.com> wrote:
>
> Hi - I'm setting up 2 parallel computers for a HA setup. (low volume so far,
> no need to load balance.)
> I already have heartbeat able to grab the public IP if the first box goes
> down, but since I'm basically completely unfamiar with pacemaker, I'd like
> some help.
> I'd imagine I'm not the only one doing this, there should be no reason for
> me to recreate the wheel.
> Info should be on:?http://wiki.freeswitch.org/wiki/Freeswitch_HA
> Specifically:
> 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
> (unlikely) calls should?immediately?get shunted over to the other box. How
> do I set that up?
> 2) How do I automatically trigger a sofia recover? I know via commanline
> it's "fs_cli -x "sofia recover" but where do I tell pacemaker to do that?
> And I suppose a reload_xml before (maybe after?) is a good idea, too.
> Also, can I put?<param name="track-calls" value="true"/> in a global file,
> or does it need to be in the actual profile files?
> And anything else I may be overlooking.
> Thanks guys!
> -Avi Marcus
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From dome at tel.co.th  Sun Jan  2 05:21:30 2011
From: dome at tel.co.th (dome at tel.co.th)
Date: Sun, 2 Jan 2011 09:21:30 +0700
Subject: [Freeswitch-users] increase Max session
In-Reply-To: <AANLkTimZxTYeGGNnSKF2x07wS6mBhU5kyB6xbeTKcAmc@mail.gmail.com>
References: <AANLkTik=Fa3ZAX5XP_ubZ7ekrx8M8A7D5J0L=fqepEUg@mail.gmail.com>
	<AANLkTimZxTYeGGNnSKF2x07wS6mBhU5kyB6xbeTKcAmc@mail.gmail.com>
Message-ID: <AANLkTin+uvhAM58ODfz=6jcJsuwDqcNzysEpLO7yXVfc@mail.gmail.com>

Now my config
<param name="max-sessions" value="5000"/>
and from CLI
22 minutes after start
freeswitch at internal> status
UP 0 years, 0 days, 0 hours, 22 minutes, 56 seconds, 321 milliseconds,
876 microseconds
11677 session(s) since startup
830 session(s) 17/100
5000 session(s) max
min idle cpu 0.00/94.00

37 minutes after start

freeswitch at internal> status
UP 0 years, 0 days, 0 hours, 37 minutes, 43 seconds, 697 milliseconds,
625 microseconds
20293 session(s) since startup
899 session(s) 24/100
921 session(s) max
min idle cpu 0.00/89.00

2011/1/2 Aloysius Lloyd <lloyd.aloysius at gmail.com>:
> autoload_configs/switch.conf.xml
> <param name="max-sessions" value="1000"/>
>
> Thanks
> Lloyd
> On Sat, Jan 1, 2011 at 8:38 PM, dome at tel.co.th <dome at tel.co.th> wrote:
>>
>> Dear All,
>> ? ?My system running with FS 1.0.6 on debian over openvz x86_64. i'm
>> seting up max session to 2000 after live traffic coming max session
>> reduce to 917.
>> FS run in none proxy mode and CPU use about 5%
>> ? ? before start FS i use ulimit -s 240 ?and other
>>
>> So i want to know how to increase max session to 2000.
>>
>> BG
>>
>> Dome C.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From infos at madovsky.org  Sun Jan  2 05:30:16 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 1 Jan 2011 21:30:16 -0500
Subject: [Freeswitch-users] increase Max session
References: <AANLkTik=Fa3ZAX5XP_ubZ7ekrx8M8A7D5J0L=fqepEUg@mail.gmail.com><AANLkTimZxTYeGGNnSKF2x07wS6mBhU5kyB6xbeTKcAmc@mail.gmail.com>
	<AANLkTin+uvhAM58ODfz=6jcJsuwDqcNzysEpLO7yXVfc@mail.gmail.com>
Message-ID: <8825DCE39E834012ACD4A4978DFA9AB7@e1705>

I think FS estimates the max possible on your server

----- Original Message ----- 
From: <dome at tel.co.th>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, January 01, 2011 9:21 PM
Subject: Re: [Freeswitch-users] increase Max session


> Now my config
> <param name="max-sessions" value="5000"/>
> and from CLI
> 22 minutes after start
> freeswitch at internal> status
> UP 0 years, 0 days, 0 hours, 22 minutes, 56 seconds, 321 milliseconds,
> 876 microseconds
> 11677 session(s) since startup
> 830 session(s) 17/100
> 5000 session(s) max
> min idle cpu 0.00/94.00
>
> 37 minutes after start
>
> freeswitch at internal> status
> UP 0 years, 0 days, 0 hours, 37 minutes, 43 seconds, 697 milliseconds,
> 625 microseconds
> 20293 session(s) since startup
> 899 session(s) 24/100
> 921 session(s) max
> min idle cpu 0.00/89.00
>
> 2011/1/2 Aloysius Lloyd <lloyd.aloysius at gmail.com>:
>> autoload_configs/switch.conf.xml
>> <param name="max-sessions" value="1000"/>
>>
>> Thanks
>> Lloyd
>> On Sat, Jan 1, 2011 at 8:38 PM, dome at tel.co.th <dome at tel.co.th> wrote:
>>>
>>> Dear All,
>>> My system running with FS 1.0.6 on debian over openvz x86_64. i'm
>>> seting up max session to 2000 after live traffic coming max session
>>> reduce to 917.
>>> FS run in none proxy mode and CPU use about 5%
>>> before start FS i use ulimit -s 240 and other
>>>
>>> So i want to know how to increase max session to 2000.
>>>
>>> BG
>>>
>>> Dome C.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From dome at tel.co.th  Sun Jan  2 05:49:44 2011
From: dome at tel.co.th (dome at tel.co.th)
Date: Sun, 2 Jan 2011 09:49:44 +0700
Subject: [Freeswitch-users] increase Max session
In-Reply-To: <8825DCE39E834012ACD4A4978DFA9AB7@e1705>
References: <AANLkTik=Fa3ZAX5XP_ubZ7ekrx8M8A7D5J0L=fqepEUg@mail.gmail.com>
	<AANLkTimZxTYeGGNnSKF2x07wS6mBhU5kyB6xbeTKcAmc@mail.gmail.com>
	<AANLkTin+uvhAM58ODfz=6jcJsuwDqcNzysEpLO7yXVfc@mail.gmail.com>
	<8825DCE39E834012ACD4A4978DFA9AB7@e1705>
Message-ID: <AANLkTimTQ2mjJ+mGQ+kozTbgfE0qtgR-j32sq_oxYNtd@mail.gmail.com>

92.00 % CPU Idle
7.3 GB MB free

I don't know why FS limit 921 max session

Dome C.

2011/1/2 Madovsky <infos at madovsky.org>:
> I think FS estimates the max possible on your server
>
> ----- Original Message -----
> From: <dome at tel.co.th>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Saturday, January 01, 2011 9:21 PM
> Subject: Re: [Freeswitch-users] increase Max session
>
>
>> Now my config
>> <param name="max-sessions" value="5000"/>
>> and from CLI
>> 22 minutes after start
>> freeswitch at internal> status
>> UP 0 years, 0 days, 0 hours, 22 minutes, 56 seconds, 321 milliseconds,
>> 876 microseconds
>> 11677 session(s) since startup
>> 830 session(s) 17/100
>> 5000 session(s) max
>> min idle cpu 0.00/94.00
>>
>> 37 minutes after start
>>
>> freeswitch at internal> status
>> UP 0 years, 0 days, 0 hours, 37 minutes, 43 seconds, 697 milliseconds,
>> 625 microseconds
>> 20293 session(s) since startup
>> 899 session(s) 24/100
>> 921 session(s) max
>> min idle cpu 0.00/89.00
>>
>> 2011/1/2 Aloysius Lloyd <lloyd.aloysius at gmail.com>:
>>> autoload_configs/switch.conf.xml
>>> <param name="max-sessions" value="1000"/>
>>>
>>> Thanks
>>> Lloyd
>>> On Sat, Jan 1, 2011 at 8:38 PM, dome at tel.co.th <dome at tel.co.th> wrote:
>>>>
>>>> Dear All,
>>>> My system running with FS 1.0.6 on debian over openvz x86_64. i'm
>>>> seting up max session to 2000 after live traffic coming max session
>>>> reduce to 917.
>>>> FS run in none proxy mode and CPU use about 5%
>>>> before start FS i use ulimit -s 240 and other
>>>>
>>>> So i want to know how to increase max session to 2000.
>>>>
>>>> BG
>>>>
>>>> Dome C.
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From lloyd.aloysius at gmail.com  Sun Jan  2 05:58:35 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sat, 1 Jan 2011 21:58:35 -0500
Subject: [Freeswitch-users] increase Max session
In-Reply-To: <AANLkTimTQ2mjJ+mGQ+kozTbgfE0qtgR-j32sq_oxYNtd@mail.gmail.com>
References: <AANLkTik=Fa3ZAX5XP_ubZ7ekrx8M8A7D5J0L=fqepEUg@mail.gmail.com>
	<AANLkTimZxTYeGGNnSKF2x07wS6mBhU5kyB6xbeTKcAmc@mail.gmail.com>
	<AANLkTin+uvhAM58ODfz=6jcJsuwDqcNzysEpLO7yXVfc@mail.gmail.com>
	<8825DCE39E834012ACD4A4978DFA9AB7@e1705>
	<AANLkTimTQ2mjJ+mGQ+kozTbgfE0qtgR-j32sq_oxYNtd@mail.gmail.com>
Message-ID: <AANLkTi=YR=LxqLy+PzjO_NTiRf+-cAVQ0D=yguZkUw_P@mail.gmail.com>

make sure the switch.conf.xml have the right value. Did you rotate the log
file between 22 min and 37 min.

try the cli command fsctl max_session 5000 see what happen in couple of
hours.

Thanks
Lloyd


On Sat, Jan 1, 2011 at 9:49 PM, dome at tel.co.th <dome at tel.co.th> wrote:

> 92.00 % CPU Idle
> 7.3 GB MB free
>
> I don't know why FS limit 921 max session
>
> Dome C.
>
> 2011/1/2 Madovsky <infos at madovsky.org>:
> > I think FS estimates the max possible on your server
> >
> > ----- Original Message -----
> > From: <dome at tel.co.th>
> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > Sent: Saturday, January 01, 2011 9:21 PM
> > Subject: Re: [Freeswitch-users] increase Max session
> >
> >
> >> Now my config
> >> <param name="max-sessions" value="5000"/>
> >> and from CLI
> >> 22 minutes after start
> >> freeswitch at internal> status
> >> UP 0 years, 0 days, 0 hours, 22 minutes, 56 seconds, 321 milliseconds,
> >> 876 microseconds
> >> 11677 session(s) since startup
> >> 830 session(s) 17/100
> >> 5000 session(s) max
> >> min idle cpu 0.00/94.00
> >>
> >> 37 minutes after start
> >>
> >> freeswitch at internal> status
> >> UP 0 years, 0 days, 0 hours, 37 minutes, 43 seconds, 697 milliseconds,
> >> 625 microseconds
> >> 20293 session(s) since startup
> >> 899 session(s) 24/100
> >> 921 session(s) max
> >> min idle cpu 0.00/89.00
> >>
> >> 2011/1/2 Aloysius Lloyd <lloyd.aloysius at gmail.com>:
> >>> autoload_configs/switch.conf.xml
> >>> <param name="max-sessions" value="1000"/>
> >>>
> >>> Thanks
> >>> Lloyd
> >>> On Sat, Jan 1, 2011 at 8:38 PM, dome at tel.co.th <dome at tel.co.th> wrote:
> >>>>
> >>>> Dear All,
> >>>> My system running with FS 1.0.6 on debian over openvz x86_64. i'm
> >>>> seting up max session to 2000 after live traffic coming max session
> >>>> reduce to 917.
> >>>> FS run in none proxy mode and CPU use about 5%
> >>>> before start FS i use ulimit -s 240 and other
> >>>>
> >>>> So i want to know how to increase max session to 2000.
> >>>>
> >>>> BG
> >>>>
> >>>> Dome C.
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dome at tel.co.th  Sun Jan  2 06:12:36 2011
From: dome at tel.co.th (dome at tel.co.th)
Date: Sun, 2 Jan 2011 10:12:36 +0700
Subject: [Freeswitch-users] increase Max session
In-Reply-To: <AANLkTi=YR=LxqLy+PzjO_NTiRf+-cAVQ0D=yguZkUw_P@mail.gmail.com>
References: <AANLkTik=Fa3ZAX5XP_ubZ7ekrx8M8A7D5J0L=fqepEUg@mail.gmail.com>
	<AANLkTimZxTYeGGNnSKF2x07wS6mBhU5kyB6xbeTKcAmc@mail.gmail.com>
	<AANLkTin+uvhAM58ODfz=6jcJsuwDqcNzysEpLO7yXVfc@mail.gmail.com>
	<8825DCE39E834012ACD4A4978DFA9AB7@e1705>
	<AANLkTimTQ2mjJ+mGQ+kozTbgfE0qtgR-j32sq_oxYNtd@mail.gmail.com>
	<AANLkTi=YR=LxqLy+PzjO_NTiRf+-cAVQ0D=yguZkUw_P@mail.gmail.com>
Message-ID: <AANLkTint0ypRO68sRdeASqy5s4OC38xK8SPeMXigm0_9@mail.gmail.com>

2011/1/2 Aloysius Lloyd <lloyd.aloysius at gmail.com>:
> make sure the switch.conf.xml have the right value. Did you rotate the log
> file between 22 min and 37 min.
Yes
> try the cli command?fsctl max_session 5000 see what happen in couple of
> hours.
i try fsctl max_session 5000
after that FS switch back to 921 again

> Thanks
> Lloyd
>
>
> On Sat, Jan 1, 2011 at 9:49 PM, dome at tel.co.th <dome at tel.co.th> wrote:
>>
>> 92.00 % CPU Idle
>> 7.3 GB MB free
>>
>> I don't know why FS limit 921 max session
>>
>> Dome C.
>>
>> 2011/1/2 Madovsky <infos at madovsky.org>:
>> > I think FS estimates the max possible on your server
>> >
>> > ----- Original Message -----
>> > From: <dome at tel.co.th>
>> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> > Sent: Saturday, January 01, 2011 9:21 PM
>> > Subject: Re: [Freeswitch-users] increase Max session
>> >
>> >
>> >> Now my config
>> >> <param name="max-sessions" value="5000"/>
>> >> and from CLI
>> >> 22 minutes after start
>> >> freeswitch at internal> status
>> >> UP 0 years, 0 days, 0 hours, 22 minutes, 56 seconds, 321 milliseconds,
>> >> 876 microseconds
>> >> 11677 session(s) since startup
>> >> 830 session(s) 17/100
>> >> 5000 session(s) max
>> >> min idle cpu 0.00/94.00
>> >>
>> >> 37 minutes after start
>> >>
>> >> freeswitch at internal> status
>> >> UP 0 years, 0 days, 0 hours, 37 minutes, 43 seconds, 697 milliseconds,
>> >> 625 microseconds
>> >> 20293 session(s) since startup
>> >> 899 session(s) 24/100
>> >> 921 session(s) max
>> >> min idle cpu 0.00/89.00
>> >>
>> >> 2011/1/2 Aloysius Lloyd <lloyd.aloysius at gmail.com>:
>> >>> autoload_configs/switch.conf.xml
>> >>> <param name="max-sessions" value="1000"/>
>> >>>
>> >>> Thanks
>> >>> Lloyd
>> >>> On Sat, Jan 1, 2011 at 8:38 PM, dome at tel.co.th <dome at tel.co.th> wrote:
>> >>>>
>> >>>> Dear All,
>> >>>> My system running with FS 1.0.6 on debian over openvz x86_64. i'm
>> >>>> seting up max session to 2000 after live traffic coming max session
>> >>>> reduce to 917.
>> >>>> FS run in none proxy mode and CPU use about 5%
>> >>>> before start FS i use ulimit -s 240 and other
>> >>>>
>> >>>> So i want to know how to increase max session to 2000.
>> >>>>
>> >>>> BG
>> >>>>
>> >>>> Dome C.
>> >>>>
>> >>>> _______________________________________________
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>
>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From lloyd.aloysius at sunteltech.ca  Sun Jan  2 06:24:24 2011
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Sat, 1 Jan 2011 22:24:24 -0500
Subject: [Freeswitch-users] increase Max session
In-Reply-To: <AANLkTint0ypRO68sRdeASqy5s4OC38xK8SPeMXigm0_9@mail.gmail.com>
References: <AANLkTik=Fa3ZAX5XP_ubZ7ekrx8M8A7D5J0L=fqepEUg@mail.gmail.com>
	<AANLkTimZxTYeGGNnSKF2x07wS6mBhU5kyB6xbeTKcAmc@mail.gmail.com>
	<AANLkTin+uvhAM58ODfz=6jcJsuwDqcNzysEpLO7yXVfc@mail.gmail.com>
	<8825DCE39E834012ACD4A4978DFA9AB7@e1705>
	<AANLkTimTQ2mjJ+mGQ+kozTbgfE0qtgR-j32sq_oxYNtd@mail.gmail.com>
	<AANLkTi=YR=LxqLy+PzjO_NTiRf+-cAVQ0D=yguZkUw_P@mail.gmail.com>
	<AANLkTint0ypRO68sRdeASqy5s4OC38xK8SPeMXigm0_9@mail.gmail.com>
Message-ID: <AANLkTi=2C6vOJiHWK8r2EdfdjBhca1_tYUXAZ1_dCpK0@mail.gmail.com>

32 bit or 64 bit ?


On Sat, Jan 1, 2011 at 10:12 PM, dome at tel.co.th <dome at tel.co.th> wrote:

> 2011/1/2 Aloysius Lloyd <lloyd.aloysius at gmail.com>:
> > make sure the switch.conf.xml have the right value. Did you rotate the
> log
> > file between 22 min and 37 min.
> Yes
> > try the cli command fsctl max_session 5000 see what happen in couple of
> > hours.
> i try fsctl max_session 5000
> after that FS switch back to 921 again
>
> > Thanks
> > Lloyd
> >
> >
> > On Sat, Jan 1, 2011 at 9:49 PM, dome at tel.co.th <dome at tel.co.th> wrote:
> >>
> >> 92.00 % CPU Idle
> >> 7.3 GB MB free
> >>
> >> I don't know why FS limit 921 max session
> >>
> >> Dome C.
> >>
> >> 2011/1/2 Madovsky <infos at madovsky.org>:
> >> > I think FS estimates the max possible on your server
> >> >
> >> > ----- Original Message -----
> >> > From: <dome at tel.co.th>
> >> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> >> > Sent: Saturday, January 01, 2011 9:21 PM
> >> > Subject: Re: [Freeswitch-users] increase Max session
> >> >
> >> >
> >> >> Now my config
> >> >> <param name="max-sessions" value="5000"/>
> >> >> and from CLI
> >> >> 22 minutes after start
> >> >> freeswitch at internal> status
> >> >> UP 0 years, 0 days, 0 hours, 22 minutes, 56 seconds, 321
> milliseconds,
> >> >> 876 microseconds
> >> >> 11677 session(s) since startup
> >> >> 830 session(s) 17/100
> >> >> 5000 session(s) max
> >> >> min idle cpu 0.00/94.00
> >> >>
> >> >> 37 minutes after start
> >> >>
> >> >> freeswitch at internal> status
> >> >> UP 0 years, 0 days, 0 hours, 37 minutes, 43 seconds, 697
> milliseconds,
> >> >> 625 microseconds
> >> >> 20293 session(s) since startup
> >> >> 899 session(s) 24/100
> >> >> 921 session(s) max
> >> >> min idle cpu 0.00/89.00
> >> >>
> >> >> 2011/1/2 Aloysius Lloyd <lloyd.aloysius at gmail.com>:
> >> >>> autoload_configs/switch.conf.xml
> >> >>> <param name="max-sessions" value="1000"/>
> >> >>>
> >> >>> Thanks
> >> >>> Lloyd
> >> >>> On Sat, Jan 1, 2011 at 8:38 PM, dome at tel.co.th <dome at tel.co.th>
> wrote:
> >> >>>>
> >> >>>> Dear All,
> >> >>>> My system running with FS 1.0.6 on debian over openvz x86_64. i'm
> >> >>>> seting up max session to 2000 after live traffic coming max session
> >> >>>> reduce to 917.
> >> >>>> FS run in none proxy mode and CPU use about 5%
> >> >>>> before start FS i use ulimit -s 240 and other
> >> >>>>
> >> >>>> So i want to know how to increase max session to 2000.
> >> >>>>
> >> >>>> BG
> >> >>>>
> >> >>>> Dome C.
> >> >>>>
> >> >>>> _______________________________________________
> >> >>>> FreeSWITCH-users mailing list
> >> >>>> FreeSWITCH-users at lists.freeswitch.org
> >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>>
> >> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>>> http://www.freeswitch.org
> >> >>>
> >> >>>
> >> >>> _______________________________________________
> >> >>> FreeSWITCH-users mailing list
> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>
> >> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> http://www.freeswitch.org
> >> >>>
> >> >>>
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dome at tel.co.th  Sun Jan  2 06:39:39 2011
From: dome at tel.co.th (dome at tel.co.th)
Date: Sun, 2 Jan 2011 10:39:39 +0700
Subject: [Freeswitch-users] increase Max session
In-Reply-To: <AANLkTi=2C6vOJiHWK8r2EdfdjBhca1_tYUXAZ1_dCpK0@mail.gmail.com>
References: <AANLkTik=Fa3ZAX5XP_ubZ7ekrx8M8A7D5J0L=fqepEUg@mail.gmail.com>
	<AANLkTimZxTYeGGNnSKF2x07wS6mBhU5kyB6xbeTKcAmc@mail.gmail.com>
	<AANLkTin+uvhAM58ODfz=6jcJsuwDqcNzysEpLO7yXVfc@mail.gmail.com>
	<8825DCE39E834012ACD4A4978DFA9AB7@e1705>
	<AANLkTimTQ2mjJ+mGQ+kozTbgfE0qtgR-j32sq_oxYNtd@mail.gmail.com>
	<AANLkTi=YR=LxqLy+PzjO_NTiRf+-cAVQ0D=yguZkUw_P@mail.gmail.com>
	<AANLkTint0ypRO68sRdeASqy5s4OC38xK8SPeMXigm0_9@mail.gmail.com>
	<AANLkTi=2C6vOJiHWK8r2EdfdjBhca1_tYUXAZ1_dCpK0@mail.gmail.com>
Message-ID: <AANLkTik9AiZ0GpmtKENmO_Y15rvxyDqm+Ga7Ryyt8vML@mail.gmail.com>

64 bit over openvz

2011/1/2 Aloysius Lloyd <lloyd.aloysius at sunteltech.ca>:
> 32 bit or 64 bit ?
>
>
> On Sat, Jan 1, 2011 at 10:12 PM, dome at tel.co.th <dome at tel.co.th> wrote:
>>
>> 2011/1/2 Aloysius Lloyd <lloyd.aloysius at gmail.com>:
>> > make sure the switch.conf.xml have the right value. Did you rotate the
>> > log
>> > file between 22 min and 37 min.
>> Yes
>> > try the cli command?fsctl max_session 5000 see what happen in couple of
>> > hours.
>> i try fsctl max_session 5000
>> after that FS switch back to 921 again
>>
>> > Thanks
>> > Lloyd
>> >
>> >
>> > On Sat, Jan 1, 2011 at 9:49 PM, dome at tel.co.th <dome at tel.co.th> wrote:
>> >>
>> >> 92.00 % CPU Idle
>> >> 7.3 GB MB free
>> >>
>> >> I don't know why FS limit 921 max session
>> >>
>> >> Dome C.
>> >>
>> >> 2011/1/2 Madovsky <infos at madovsky.org>:
>> >> > I think FS estimates the max possible on your server
>> >> >
>> >> > ----- Original Message -----
>> >> > From: <dome at tel.co.th>
>> >> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> >> > Sent: Saturday, January 01, 2011 9:21 PM
>> >> > Subject: Re: [Freeswitch-users] increase Max session
>> >> >
>> >> >
>> >> >> Now my config
>> >> >> <param name="max-sessions" value="5000"/>
>> >> >> and from CLI
>> >> >> 22 minutes after start
>> >> >> freeswitch at internal> status
>> >> >> UP 0 years, 0 days, 0 hours, 22 minutes, 56 seconds, 321
>> >> >> milliseconds,
>> >> >> 876 microseconds
>> >> >> 11677 session(s) since startup
>> >> >> 830 session(s) 17/100
>> >> >> 5000 session(s) max
>> >> >> min idle cpu 0.00/94.00
>> >> >>
>> >> >> 37 minutes after start
>> >> >>
>> >> >> freeswitch at internal> status
>> >> >> UP 0 years, 0 days, 0 hours, 37 minutes, 43 seconds, 697
>> >> >> milliseconds,
>> >> >> 625 microseconds
>> >> >> 20293 session(s) since startup
>> >> >> 899 session(s) 24/100
>> >> >> 921 session(s) max
>> >> >> min idle cpu 0.00/89.00
>> >> >>
>> >> >> 2011/1/2 Aloysius Lloyd <lloyd.aloysius at gmail.com>:
>> >> >>> autoload_configs/switch.conf.xml
>> >> >>> <param name="max-sessions" value="1000"/>
>> >> >>>
>> >> >>> Thanks
>> >> >>> Lloyd
>> >> >>> On Sat, Jan 1, 2011 at 8:38 PM, dome at tel.co.th <dome at tel.co.th>
>> >> >>> wrote:
>> >> >>>>
>> >> >>>> Dear All,
>> >> >>>> My system running with FS 1.0.6 on debian over openvz x86_64. i'm
>> >> >>>> seting up max session to 2000 after live traffic coming max
>> >> >>>> session
>> >> >>>> reduce to 917.
>> >> >>>> FS run in none proxy mode and CPU use about 5%
>> >> >>>> before start FS i use ulimit -s 240 and other
>> >> >>>>
>> >> >>>> So i want to know how to increase max session to 2000.
>> >> >>>>
>> >> >>>> BG
>> >> >>>>
>> >> >>>> Dome C.
>> >> >>>>
>> >> >>>> _______________________________________________
>> >> >>>> FreeSWITCH-users mailing list
>> >> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>>
>> >> >>>>
>> >> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>>> http://www.freeswitch.org
>> >> >>>
>> >> >>>
>> >> >>> _______________________________________________
>> >> >>> FreeSWITCH-users mailing list
>> >> >>> FreeSWITCH-users at lists.freeswitch.org
>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>
>> >> >>>
>> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> http://www.freeswitch.org
>> >> >>>
>> >> >>>
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >>
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From joaocarlosleme at gmail.com  Sun Jan  2 07:08:37 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Sat, 1 Jan 2011 20:08:37 -0800
Subject: [Freeswitch-users] Can't build on Windows 7 64bit C++ Express!?
In-Reply-To: <AANLkTinPiO-fMHGc7jVj=if5mo1yykVWW8qNgn_NcFNv@mail.gmail.com>
References: <AANLkTincSQSt-Pg00T4qVG+GXfVw2XpE6FU+2Ffe253y@mail.gmail.com>
	<AANLkTinPiO-fMHGc7jVj=if5mo1yykVWW8qNgn_NcFNv@mail.gmail.com>
Message-ID: <AANLkTimK-9_fiAoCRwBYRhmTCg-me2jrDJ-r5LKKJYSm@mail.gmail.com>

I don't need mod_managed at this time. So should I build on vc# express?
Meaning the instructions on
http://wiki.freeswitch.org/wiki/Installation_for_Windows are wrong?
I don't know if
Thanks,
John

On Sat, Jan 1, 2011 at 5:33 AM, babak yakhchali
<babak.freeswitch at gmail.com>wrote:

> if u need mod_managed u should build it on vc# express cause vc++ can not
> build it
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From babak.freeswitch at gmail.com  Sun Jan  2 08:51:24 2011
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sun, 2 Jan 2011 09:21:24 +0330
Subject: [Freeswitch-users] Can't build on Windows 7 64bit C++ Express!?
In-Reply-To: <AANLkTimK-9_fiAoCRwBYRhmTCg-me2jrDJ-r5LKKJYSm@mail.gmail.com>
References: <AANLkTincSQSt-Pg00T4qVG+GXfVw2XpE6FU+2Ffe253y@mail.gmail.com>
	<AANLkTinPiO-fMHGc7jVj=if5mo1yykVWW8qNgn_NcFNv@mail.gmail.com>
	<AANLkTimK-9_fiAoCRwBYRhmTCg-me2jrDJ-r5LKKJYSm@mail.gmail.com>
Message-ID: <AANLkTikoZCpjat2eWJw3_c3T-x2WJx=MrvP5RBH-951_@mail.gmail.com>

you just need to build the project in
D:\gitRepos\2010exp-freeswitch\freeswitch\src\mod\languages\mod_managed\managed
using vc# express, others should just work using vc++
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From joaocarlosleme at gmail.com  Sun Jan  2 10:28:21 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Sat, 1 Jan 2011 23:28:21 -0800
Subject: [Freeswitch-users] Can't build on Windows 7 64bit C++ Express!?
In-Reply-To: <AANLkTikoZCpjat2eWJw3_c3T-x2WJx=MrvP5RBH-951_@mail.gmail.com>
References: <AANLkTincSQSt-Pg00T4qVG+GXfVw2XpE6FU+2Ffe253y@mail.gmail.com>
	<AANLkTinPiO-fMHGc7jVj=if5mo1yykVWW8qNgn_NcFNv@mail.gmail.com>
	<AANLkTimK-9_fiAoCRwBYRhmTCg-me2jrDJ-r5LKKJYSm@mail.gmail.com>
	<AANLkTikoZCpjat2eWJw3_c3T-x2WJx=MrvP5RBH-951_@mail.gmail.com>
Message-ID: <AANLkTinnrdqJn-SLSaaiFDd04S89GH1wmdLtvKreabbB@mail.gmail.com>

Ok Thanks. Any idea why all the errors and why i can't build? I've built
before on Windows Vista 64bit and VS2008Pro with no problems but can't get
it to work on Express edition.

On Sat, Jan 1, 2011 at 9:51 PM, babak yakhchali
<babak.freeswitch at gmail.com>wrote:

> you just need to build the project in
>
> D:\gitRepos\2010exp-freeswitch\freeswitch\src\mod\languages\mod_managed\managed
> using vc# express, others should just work using vc++
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Avi at aMarcus.com  Sun Jan  2 12:43:56 2011
From: Avi at aMarcus.com (Avi Marcus)
Date: Sun, 2 Jan 2011 11:43:56 +0200
Subject: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please
 Help setting up
In-Reply-To: <AANLkTinoFOPtPW+=uyoogi33u6LvAYz_ta3bDc7XMhs0@mail.gmail.com>
References: <AANLkTi=6ZwmiMHLcBM7EndP8apmXf7CO9XFied1UOvRh@mail.gmail.com>
	<65B881CD-D420-444B-91FD-04360AB97644@gmail.com>
	<AANLkTinoFOPtPW+=uyoogi33u6LvAYz_ta3bDc7XMhs0@mail.gmail.com>
Message-ID: <AANLkTikBrP6ezm9uPrU7JdG29HP7BDZ7U5nReavz--6=@mail.gmail.com>

Thanks, I noticed the -nonat already.
What do you mean by running sofia recover from my init script? I
don't understand how that would be triggered by heartbeat/pacemaker. Any
explanatory docs on the issue would be nice, but everything I've seen was
explaining exact parts and nothing ever seemed to explain the general syntax
of everything.

Does anyone have a working example that they can wikify and explain?
Thanks,
Avi

On Sun, Jan 2, 2011 at 4:06 AM, Seven Du <dujinfang at gmail.com> wrote:

> the wiki page already said you can start the spare FS ahead to speed
> switch over up.
>
> it means if you use ip_nonlocal_bind and start the spare FS ahead, you
> only need to run sofia recover in you init script without waiting for
> the actually FS start process.
>
> On Sun, Jan 2, 2011 at 7:58 AM, Steven Ayre <steveayre at gmail.com> wrote:
> > 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
> > (unlikely) calls should immediately get shunted over to the other box.
> How
> > do I set that up?
> >
> > Just a tip, I start fs with -nonat and find that makes it start faster.
> >
> > Steve on iPhone
> > On 1 Jan 2011, at 20:20, Avi Marcus <Avi at aMarcus.com> wrote:
> >
> > Hi - I'm setting up 2 parallel computers for a HA setup. (low volume so
> far,
> > no need to load balance.)
> > I already have heartbeat able to grab the public IP if the first box goes
> > down, but since I'm basically completely unfamiar with pacemaker, I'd
> like
> > some help.
> > I'd imagine I'm not the only one doing this, there should be no reason
> for
> > me to recreate the wheel.
> > Info should be on: http://wiki.freeswitch.org/wiki/Freeswitch_HA
> > Specifically:
> > 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
> > (unlikely) calls should immediately get shunted over to the other box.
> How
> > do I set that up?
> > 2) How do I automatically trigger a sofia recover? I know via commanline
> > it's "fs_cli -x "sofia recover" but where do I tell pacemaker to do that?
> > And I suppose a reload_xml before (maybe after?) is a good idea, too.
> > Also, can I put <param name="track-calls" value="true"/> in a global
> file,
> > or does it need to be in the actual profile files?
> > And anything else I may be overlooking.
> > Thanks guys!
> > -Avi Marcus
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Sun Jan  2 13:31:44 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 2 Jan 2011 10:31:44 +0000
Subject: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please
 Help setting up
In-Reply-To: <AANLkTikBrP6ezm9uPrU7JdG29HP7BDZ7U5nReavz--6=@mail.gmail.com>
References: <AANLkTi=6ZwmiMHLcBM7EndP8apmXf7CO9XFied1UOvRh@mail.gmail.com>
	<65B881CD-D420-444B-91FD-04360AB97644@gmail.com>
	<AANLkTinoFOPtPW+=uyoogi33u6LvAYz_ta3bDc7XMhs0@mail.gmail.com>
	<AANLkTikBrP6ezm9uPrU7JdG29HP7BDZ7U5nReavz--6=@mail.gmail.com>
Message-ID: <AANLkTiksQD5QA8vUBHe2yBsh6ahA+2xcZ+LFGNst0dGT@mail.gmail.com>

It's an API command that you can run via ESL. You can do that from a
script using fs_cli and its -x/--execute option.

Something like this:

fs_cli --host=otherhost --password=secretpasswd --execute="sofia recover"

-Steve


On 2 January 2011 09:43, Avi Marcus <Avi at amarcus.com> wrote:
> Thanks, I noticed the -nonat already.
> What do you mean by running sofia recover from my init script? I
> don't?understand?how that would be triggered by heartbeat/pacemaker. Any
> explanatory docs on the issue would be nice, but everything I've seen was
> explaining exact parts and nothing ever seemed to explain the general syntax
> of everything.
> Does anyone have a working example that they can wikify and explain?
> Thanks,
> Avi
>
> On Sun, Jan 2, 2011 at 4:06 AM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> the wiki page already said you can start the spare FS ahead to speed
>> switch over up.
>>
>> it means if you use ip_nonlocal_bind and start the spare FS ahead, you
>> only need to run sofia recover in you init script without waiting for
>> the actually FS start process.
>>
>> On Sun, Jan 2, 2011 at 7:58 AM, Steven Ayre <steveayre at gmail.com> wrote:
>> > 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
>> > (unlikely) calls should?immediately?get shunted over to the other box.
>> > How
>> > do I set that up?
>> >
>> > Just a tip, I start fs with -nonat and find that makes it start faster.
>> >
>> > Steve on iPhone
>> > On 1 Jan 2011, at 20:20, Avi Marcus <Avi at aMarcus.com> wrote:
>> >
>> > Hi - I'm setting up 2 parallel computers for a HA setup. (low volume so
>> > far,
>> > no need to load balance.)
>> > I already have heartbeat able to grab the public IP if the first box
>> > goes
>> > down, but since I'm basically completely unfamiar with pacemaker, I'd
>> > like
>> > some help.
>> > I'd imagine I'm not the only one doing this, there should be no reason
>> > for
>> > me to recreate the wheel.
>> > Info should be on:?http://wiki.freeswitch.org/wiki/Freeswitch_HA
>> > Specifically:
>> > 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
>> > (unlikely) calls should?immediately?get shunted over to the other box.
>> > How
>> > do I set that up?
>> > 2) How do I automatically trigger a sofia recover? I know via commanline
>> > it's "fs_cli -x "sofia recover" but where do I tell pacemaker to do
>> > that?
>> > And I suppose a reload_xml before (maybe after?) is a good idea, too.
>> > Also, can I put?<param name="track-calls" value="true"/> in a global
>> > file,
>> > or does it need to be in the actual profile files?
>> > And anything else I may be overlooking.
>> > Thanks guys!
>> > -Avi Marcus
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> About: http://about.me/dujinfang
>> Blog: http://www.dujinfang.com
>> Proj:? http://www.freeswitch.org.cn
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From daniel.neubert at solomo.de  Sun Jan  2 15:29:09 2011
From: daniel.neubert at solomo.de (Daniel Neubert)
Date: Sun, 2 Jan 2011 13:29:09 +0100
Subject: [Freeswitch-users] Logging "dropped calls"
In-Reply-To: <AANLkTi=OP-eVAr2QpxEghVN1v-ASFHUgC9n_3MyPRsm1@mail.gmail.com>
References: <AANLkTi=OP-eVAr2QpxEghVN1v-ASFHUgC9n_3MyPRsm1@mail.gmail.com>
Message-ID: <4D206F95.20602@solomo.de>

Hi,

you can use ESL (http://wiki.freeswitch.org/wiki/Event_Socket_Library) 
to receive certain events http://wiki.freeswitch.org/wiki/Event_list

or you can use information stored in XML CDR data to do your analysis: 
http://wiki.freeswitch.org/wiki/Mod_xml_cdr

Best regards / Mit freundlichen Gr??en,
Daniel Neubert

Am 01.01.2011 22:46, schrieb Frank Park:
> Happy New Years!
>
> This is merely a wish list for me, but I wanted to know if there's a
> way for me to log all calls that disconnects without the proper SIP
> termination..
> I would like to build some level of debug tool where I can track
> dropped calls due to freeswitch, or due to any other external issues
> (which I assume most are).
> In the same statistics, I am also interested in logging who initially
> sent the BYE message (the caller, the callee, or the switch).
>
> A direction to do this might be sufficient at this point.
>
> Thank you!
> Frank
>
>
>
> ----=======================----
> Frank Park
> Telonium Communications, LLC
> frank at telonium.com
> http://www.telonium.com
> Follow Us on Twitter: @GetTelonium
> 404-566-8888 x1001 Office
> 404-939-4242 Cell
> ----=======================----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mustafa.pk at gmail.com  Sun Jan  2 18:36:32 2011
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Sun, 2 Jan 2011 20:36:32 +0500
Subject: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please
 Help setting up
In-Reply-To: <AANLkTiksQD5QA8vUBHe2yBsh6ahA+2xcZ+LFGNst0dGT@mail.gmail.com>
References: <AANLkTi=6ZwmiMHLcBM7EndP8apmXf7CO9XFied1UOvRh@mail.gmail.com>
	<65B881CD-D420-444B-91FD-04360AB97644@gmail.com>
	<AANLkTinoFOPtPW+=uyoogi33u6LvAYz_ta3bDc7XMhs0@mail.gmail.com>
	<AANLkTikBrP6ezm9uPrU7JdG29HP7BDZ7U5nReavz--6=@mail.gmail.com>
	<AANLkTiksQD5QA8vUBHe2yBsh6ahA+2xcZ+LFGNst0dGT@mail.gmail.com>
Message-ID: <AANLkTimZH-ONShU-e98f-HjDg_UkvcHfp9gAWt5Ki=+h@mail.gmail.com>

Hi, for testing purposes i modified IPaddr ocf script on both nodes
and added following line at the end of start_ipaddr.
/opt/switch/bin/fs_cli -x "sofia recover"

On 1/2/11, Steven Ayre <steveayre at gmail.com> wrote:
> It's an API command that you can run via ESL. You can do that from a
> script using fs_cli and its -x/--execute option.
>
> Something like this:
>
> fs_cli --host=otherhost --password=secretpasswd --execute="sofia recover"
>
> -Steve
>
>
> On 2 January 2011 09:43, Avi Marcus <Avi at amarcus.com> wrote:
>> Thanks, I noticed the -nonat already.
>> What do you mean by running sofia recover from my init script? I
>> don't?understand?how that would be triggered by heartbeat/pacemaker. Any
>> explanatory docs on the issue would be nice, but everything I've seen was
>> explaining exact parts and nothing ever seemed to explain the general
>> syntax
>> of everything.
>> Does anyone have a working example that they can wikify and explain?
>> Thanks,
>> Avi
>>
>> On Sun, Jan 2, 2011 at 4:06 AM, Seven Du <dujinfang at gmail.com> wrote:
>>>
>>> the wiki page already said you can start the spare FS ahead to speed
>>> switch over up.
>>>
>>> it means if you use ip_nonlocal_bind and start the spare FS ahead, you
>>> only need to run sofia recover in you init script without waiting for
>>> the actually FS start process.
>>>
>>> On Sun, Jan 2, 2011 at 7:58 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>> > 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
>>> > (unlikely) calls should?immediately?get shunted over to the other box.
>>> > How
>>> > do I set that up?
>>> >
>>> > Just a tip, I start fs with -nonat and find that makes it start faster.
>>> >
>>> > Steve on iPhone
>>> > On 1 Jan 2011, at 20:20, Avi Marcus <Avi at aMarcus.com> wrote:
>>> >
>>> > Hi - I'm setting up 2 parallel computers for a HA setup. (low volume so
>>> > far,
>>> > no need to load balance.)
>>> > I already have heartbeat able to grab the public IP if the first box
>>> > goes
>>> > down, but since I'm basically completely unfamiar with pacemaker, I'd
>>> > like
>>> > some help.
>>> > I'd imagine I'm not the only one doing this, there should be no reason
>>> > for
>>> > me to recreate the wheel.
>>> > Info should be on:?http://wiki.freeswitch.org/wiki/Freeswitch_HA
>>> > Specifically:
>>> > 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
>>> > (unlikely) calls should?immediately?get shunted over to the other box.
>>> > How
>>> > do I set that up?
>>> > 2) How do I automatically trigger a sofia recover? I know via
>>> > commanline
>>> > it's "fs_cli -x "sofia recover" but where do I tell pacemaker to do
>>> > that?
>>> > And I suppose a reload_xml before (maybe after?) is a good idea, too.
>>> > Also, can I put?<param name="track-calls" value="true"/> in a global
>>> > file,
>>> > or does it need to be in the actual profile files?
>>> > And anything else I may be overlooking.
>>> > Thanks guys!
>>> > -Avi Marcus
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> About: http://about.me/dujinfang
>>> Blog: http://www.dujinfang.com
>>> Proj:? http://www.freeswitch.org.cn
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com


From Avi at aMarcus.com  Sun Jan  2 18:38:41 2011
From: Avi at aMarcus.com (Avi Marcus)
Date: Sun, 2 Jan 2011 17:38:41 +0200
Subject: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please
 Help setting up
In-Reply-To: <AANLkTiksQD5QA8vUBHe2yBsh6ahA+2xcZ+LFGNst0dGT@mail.gmail.com>
References: <AANLkTi=6ZwmiMHLcBM7EndP8apmXf7CO9XFied1UOvRh@mail.gmail.com>
	<65B881CD-D420-444B-91FD-04360AB97644@gmail.com>
	<AANLkTinoFOPtPW+=uyoogi33u6LvAYz_ta3bDc7XMhs0@mail.gmail.com>
	<AANLkTikBrP6ezm9uPrU7JdG29HP7BDZ7U5nReavz--6=@mail.gmail.com>
	<AANLkTiksQD5QA8vUBHe2yBsh6ahA+2xcZ+LFGNst0dGT@mail.gmail.com>
Message-ID: <AANLkTi=Mkz2muxKRYkD=LOCTVGeR1hirD51EQmk+tZA5@mail.gmail.com>

Yes, yes: fs_cli -x "sofia recover" I understand the freeswitch parts.
Now how does that get set up with heartbeat/pacemaker?

You seem to assume I have a working pacemaker understanding, which I do not.
I wouldn't mind learning about it *if I saw any docs that made sense.*
-Avi

On Sun, Jan 2, 2011 at 12:31 PM, Steven Ayre <steveayre at gmail.com> wrote:

> It's an API command that you can run via ESL. You can do that from a
> script using fs_cli and its -x/--execute option.
>
> Something like this:
>
> fs_cli --host=otherhost --password=secretpasswd --execute="sofia recover"
>
> -Steve
>
>
> On 2 January 2011 09:43, Avi Marcus <Avi at amarcus.com> wrote:
> > Thanks, I noticed the -nonat already.
> > What do you mean by running sofia recover from my init script? I
> > don't understand how that would be triggered by heartbeat/pacemaker. Any
> > explanatory docs on the issue would be nice, but everything I've seen was
> > explaining exact parts and nothing ever seemed to explain the general
> syntax
> > of everything.
> > Does anyone have a working example that they can wikify and explain?
> > Thanks,
> > Avi
> >
> > On Sun, Jan 2, 2011 at 4:06 AM, Seven Du <dujinfang at gmail.com> wrote:
> >>
> >> the wiki page already said you can start the spare FS ahead to speed
> >> switch over up.
> >>
> >> it means if you use ip_nonlocal_bind and start the spare FS ahead, you
> >> only need to run sofia recover in you init script without waiting for
> >> the actually FS start process.
> >>
> >> On Sun, Jan 2, 2011 at 7:58 AM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >> > 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
> >> > (unlikely) calls should immediately get shunted over to the other box.
> >> > How
> >> > do I set that up?
> >> >
> >> > Just a tip, I start fs with -nonat and find that makes it start
> faster.
> >> >
> >> > Steve on iPhone
> >> > On 1 Jan 2011, at 20:20, Avi Marcus <Avi at aMarcus.com> wrote:
> >> >
> >> > Hi - I'm setting up 2 parallel computers for a HA setup. (low volume
> so
> >> > far,
> >> > no need to load balance.)
> >> > I already have heartbeat able to grab the public IP if the first box
> >> > goes
> >> > down, but since I'm basically completely unfamiar with pacemaker, I'd
> >> > like
> >> > some help.
> >> > I'd imagine I'm not the only one doing this, there should be no reason
> >> > for
> >> > me to recreate the wheel.
> >> > Info should be on: http://wiki.freeswitch.org/wiki/Freeswitch_HA
> >> > Specifically:
> >> > 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
> >> > (unlikely) calls should immediately get shunted over to the other box.
> >> > How
> >> > do I set that up?
> >> > 2) How do I automatically trigger a sofia recover? I know via
> commanline
> >> > it's "fs_cli -x "sofia recover" but where do I tell pacemaker to do
> >> > that?
> >> > And I suppose a reload_xml before (maybe after?) is a good idea, too.
> >> > Also, can I put <param name="track-calls" value="true"/> in a global
> >> > file,
> >> > or does it need to be in the actual profile files?
> >> > And anything else I may be overlooking.
> >> > Thanks guys!
> >> > -Avi Marcus
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> About: http://about.me/dujinfang
> >> Blog: http://www.dujinfang.com
> >> Proj:  http://www.freeswitch.org.cn
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Avi at aMarcus.com  Sun Jan  2 19:16:15 2011
From: Avi at aMarcus.com (Avi Marcus)
Date: Sun, 2 Jan 2011 18:16:15 +0200
Subject: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please
 Help setting up
In-Reply-To: <AANLkTimZH-ONShU-e98f-HjDg_UkvcHfp9gAWt5Ki=+h@mail.gmail.com>
References: <AANLkTi=6ZwmiMHLcBM7EndP8apmXf7CO9XFied1UOvRh@mail.gmail.com>
	<65B881CD-D420-444B-91FD-04360AB97644@gmail.com>
	<AANLkTinoFOPtPW+=uyoogi33u6LvAYz_ta3bDc7XMhs0@mail.gmail.com>
	<AANLkTikBrP6ezm9uPrU7JdG29HP7BDZ7U5nReavz--6=@mail.gmail.com>
	<AANLkTiksQD5QA8vUBHe2yBsh6ahA+2xcZ+LFGNst0dGT@mail.gmail.com>
	<AANLkTimZH-ONShU-e98f-HjDg_UkvcHfp9gAWt5Ki=+h@mail.gmail.com>
Message-ID: <AANLkTineUendBAcYYd2fNWnPvyoXRoqqkaRS6VJg_-M3@mail.gmail.com>

Where would that go?
/me is confused.

On Sun, Jan 2, 2011 at 5:36 PM, Ghulam Mustafa <mustafa.pk at gmail.com> wrote:

> Hi, for testing purposes i modified IPaddr ocf script on both nodes
> and added following line at the end of start_ipaddr.
> /opt/switch/bin/fs_cli -x "sofia recover"
>
> On 1/2/11, Steven Ayre <steveayre at gmail.com> wrote:
> > It's an API command that you can run via ESL. You can do that from a
> > script using fs_cli and its -x/--execute option.
> >
> > Something like this:
> >
> > fs_cli --host=otherhost --password=secretpasswd --execute="sofia recover"
> >
> > -Steve
> >
> >
> > On 2 January 2011 09:43, Avi Marcus <Avi at amarcus.com> wrote:
> >> Thanks, I noticed the -nonat already.
> >> What do you mean by running sofia recover from my init script? I
> >> don't understand how that would be triggered by heartbeat/pacemaker. Any
> >> explanatory docs on the issue would be nice, but everything I've seen
> was
> >> explaining exact parts and nothing ever seemed to explain the general
> >> syntax
> >> of everything.
> >> Does anyone have a working example that they can wikify and explain?
> >> Thanks,
> >> Avi
> >>
> >> On Sun, Jan 2, 2011 at 4:06 AM, Seven Du <dujinfang at gmail.com> wrote:
> >>>
> >>> the wiki page already said you can start the spare FS ahead to speed
> >>> switch over up.
> >>>
> >>> it means if you use ip_nonlocal_bind and start the spare FS ahead, you
> >>> only need to run sofia recover in you init script without waiting for
> >>> the actually FS start process.
> >>>
> >>> On Sun, Jan 2, 2011 at 7:58 AM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >>> > 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
> >>> > (unlikely) calls should immediately get shunted over to the other
> box.
> >>> > How
> >>> > do I set that up?
> >>> >
> >>> > Just a tip, I start fs with -nonat and find that makes it start
> faster.
> >>> >
> >>> > Steve on iPhone
> >>> > On 1 Jan 2011, at 20:20, Avi Marcus <Avi at aMarcus.com> wrote:
> >>> >
> >>> > Hi - I'm setting up 2 parallel computers for a HA setup. (low volume
> so
> >>> > far,
> >>> > no need to load balance.)
> >>> > I already have heartbeat able to grab the public IP if the first box
> >>> > goes
> >>> > down, but since I'm basically completely unfamiar with pacemaker, I'd
> >>> > like
> >>> > some help.
> >>> > I'd imagine I'm not the only one doing this, there should be no
> reason
> >>> > for
> >>> > me to recreate the wheel.
> >>> > Info should be on: http://wiki.freeswitch.org/wiki/Freeswitch_HA
> >>> > Specifically:
> >>> > 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
> >>> > (unlikely) calls should immediately get shunted over to the other
> box.
> >>> > How
> >>> > do I set that up?
> >>> > 2) How do I automatically trigger a sofia recover? I know via
> >>> > commanline
> >>> > it's "fs_cli -x "sofia recover" but where do I tell pacemaker to do
> >>> > that?
> >>> > And I suppose a reload_xml before (maybe after?) is a good idea, too.
> >>> > Also, can I put <param name="track-calls" value="true"/> in a global
> >>> > file,
> >>> > or does it need to be in the actual profile files?
> >>> > And anything else I may be overlooking.
> >>> > Thanks guys!
> >>> > -Avi Marcus
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>>
> >>>
> >>> --
> >>> About: http://about.me/dujinfang
> >>> Blog: http://www.dujinfang.com
> >>> Proj:  http://www.freeswitch.org.cn
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> --
> Ghulam Mustafa
> cell: +92 333.611.7681
> sip: cyrenity at ekiga.net
> mail: mustafa.pk at gmail.com
> web: cyrenity.wordpress.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Sun Jan  2 19:27:15 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 2 Jan 2011 16:27:15 +0000
Subject: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please
 Help setting up
In-Reply-To: <AANLkTineUendBAcYYd2fNWnPvyoXRoqqkaRS6VJg_-M3@mail.gmail.com>
References: <AANLkTi=6ZwmiMHLcBM7EndP8apmXf7CO9XFied1UOvRh@mail.gmail.com>
	<65B881CD-D420-444B-91FD-04360AB97644@gmail.com>
	<AANLkTinoFOPtPW+=uyoogi33u6LvAYz_ta3bDc7XMhs0@mail.gmail.com>
	<AANLkTikBrP6ezm9uPrU7JdG29HP7BDZ7U5nReavz--6=@mail.gmail.com>
	<AANLkTiksQD5QA8vUBHe2yBsh6ahA+2xcZ+LFGNst0dGT@mail.gmail.com>
	<AANLkTimZH-ONShU-e98f-HjDg_UkvcHfp9gAWt5Ki=+h@mail.gmail.com>
	<AANLkTineUendBAcYYd2fNWnPvyoXRoqqkaRS6VJg_-M3@mail.gmail.com>
Message-ID: <AANLkTi=cEkm0_fTEteE2+f9wk-bhDAVVmkf_EdfbQoR6@mail.gmail.com>

You would create a new fs resource agent
http://www.linux-ha.org/wiki/Resource_Agents

Essentially, they're programs (can be a shell script) given run with
start stop etc arguments when an action needs to be done. You can run
other commands such as fs_cli from there.

I haven't written one from scratch myself though, I just use IPAddr2.
Perhaps it would be useful for someone to create an example one in
trunk or contrib?

-Steve


On 2 January 2011 16:16, Avi Marcus <Avi at amarcus.com> wrote:
> Where would that go?
> /me is confused.
>
> On Sun, Jan 2, 2011 at 5:36 PM, Ghulam Mustafa <mustafa.pk at gmail.com> wrote:
>>
>> Hi, for testing purposes i modified IPaddr ocf script on both nodes
>> and added following line at the end of start_ipaddr.
>> /opt/switch/bin/fs_cli -x "sofia recover"
>>
>> On 1/2/11, Steven Ayre <steveayre at gmail.com> wrote:
>> > It's an API command that you can run via ESL. You can do that from a
>> > script using fs_cli and its -x/--execute option.
>> >
>> > Something like this:
>> >
>> > fs_cli --host=otherhost --password=secretpasswd --execute="sofia
>> > recover"
>> >
>> > -Steve
>> >
>> >
>> > On 2 January 2011 09:43, Avi Marcus <Avi at amarcus.com> wrote:
>> >> Thanks, I noticed the -nonat already.
>> >> What do you mean by running sofia recover from my init script? I
>> >> don't?understand?how that would be triggered by heartbeat/pacemaker.
>> >> Any
>> >> explanatory docs on the issue would be nice, but everything I've seen
>> >> was
>> >> explaining exact parts and nothing ever seemed to explain the general
>> >> syntax
>> >> of everything.
>> >> Does anyone have a working example that they can wikify and explain?
>> >> Thanks,
>> >> Avi
>> >>
>> >> On Sun, Jan 2, 2011 at 4:06 AM, Seven Du <dujinfang at gmail.com> wrote:
>> >>>
>> >>> the wiki page already said you can start the spare FS ahead to speed
>> >>> switch over up.
>> >>>
>> >>> it means if you use ip_nonlocal_bind and start the spare FS ahead, you
>> >>> only need to run sofia recover in you init script without waiting for
>> >>> the actually FS start process.
>> >>>
>> >>> On Sun, Jan 2, 2011 at 7:58 AM, Steven Ayre <steveayre at gmail.com>
>> >>> wrote:
>> >>> > 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
>> >>> > (unlikely) calls should?immediately?get shunted over to the other
>> >>> > box.
>> >>> > How
>> >>> > do I set that up?
>> >>> >
>> >>> > Just a tip, I start fs with -nonat and find that makes it start
>> >>> > faster.
>> >>> >
>> >>> > Steve on iPhone
>> >>> > On 1 Jan 2011, at 20:20, Avi Marcus <Avi at aMarcus.com> wrote:
>> >>> >
>> >>> > Hi - I'm setting up 2 parallel computers for a HA setup. (low volume
>> >>> > so
>> >>> > far,
>> >>> > no need to load balance.)
>> >>> > I already have heartbeat able to grab the public IP if the first box
>> >>> > goes
>> >>> > down, but since I'm basically completely unfamiar with pacemaker,
>> >>> > I'd
>> >>> > like
>> >>> > some help.
>> >>> > I'd imagine I'm not the only one doing this, there should be no
>> >>> > reason
>> >>> > for
>> >>> > me to recreate the wheel.
>> >>> > Info should be on:?http://wiki.freeswitch.org/wiki/Freeswitch_HA
>> >>> > Specifically:
>> >>> > 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
>> >>> > (unlikely) calls should?immediately?get shunted over to the other
>> >>> > box.
>> >>> > How
>> >>> > do I set that up?
>> >>> > 2) How do I automatically trigger a sofia recover? I know via
>> >>> > commanline
>> >>> > it's "fs_cli -x "sofia recover" but where do I tell pacemaker to do
>> >>> > that?
>> >>> > And I suppose a reload_xml before (maybe after?) is a good idea,
>> >>> > too.
>> >>> > Also, can I put?<param name="track-calls" value="true"/> in a global
>> >>> > file,
>> >>> > or does it need to be in the actual profile files?
>> >>> > And anything else I may be overlooking.
>> >>> > Thanks guys!
>> >>> > -Avi Marcus
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >
>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >
>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>>
>> >>>
>> >>>
>> >>> --
>> >>> About: http://about.me/dujinfang
>> >>> Blog: http://www.dujinfang.com
>> >>> Proj:? http://www.freeswitch.org.cn
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>> --
>> Ghulam Mustafa
>> cell: +92 333.611.7681
>> sip: cyrenity at ekiga.net
>> mail: mustafa.pk at gmail.com
>> web: cyrenity.wordpress.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From Avi at aMarcus.com  Sun Jan  2 19:38:44 2011
From: Avi at aMarcus.com (Avi Marcus)
Date: Sun, 2 Jan 2011 18:38:44 +0200
Subject: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please
 Help setting up
In-Reply-To: <AANLkTi=cEkm0_fTEteE2+f9wk-bhDAVVmkf_EdfbQoR6@mail.gmail.com>
References: <AANLkTi=6ZwmiMHLcBM7EndP8apmXf7CO9XFied1UOvRh@mail.gmail.com>
	<65B881CD-D420-444B-91FD-04360AB97644@gmail.com>
	<AANLkTinoFOPtPW+=uyoogi33u6LvAYz_ta3bDc7XMhs0@mail.gmail.com>
	<AANLkTikBrP6ezm9uPrU7JdG29HP7BDZ7U5nReavz--6=@mail.gmail.com>
	<AANLkTiksQD5QA8vUBHe2yBsh6ahA+2xcZ+LFGNst0dGT@mail.gmail.com>
	<AANLkTimZH-ONShU-e98f-HjDg_UkvcHfp9gAWt5Ki=+h@mail.gmail.com>
	<AANLkTineUendBAcYYd2fNWnPvyoXRoqqkaRS6VJg_-M3@mail.gmail.com>
	<AANLkTi=cEkm0_fTEteE2+f9wk-bhDAVVmkf_EdfbQoR6@mail.gmail.com>
Message-ID: <AANLkTikp4eZx=A3dmbkGL+OccC7i56o+GjB7nssYsO9o@mail.gmail.com>

In the git contrib there's /ledr/ha.d/ but it has a haresources file which
isn't how I've set up anything. Threre's a FSSofia batch file to use ESL to
check if certain profile are running, but I don't get how it all fits
together.
I've got the IPAddr2 but no integration into actual freeswitch.
The "anything" resource says it's supposed to be a bin file that keeps
running, not just a script that exits.

-Avi

On Sun, Jan 2, 2011 at 6:27 PM, Steven Ayre <steveayre at gmail.com> wrote:

> You would create a new fs resource agent
> http://www.linux-ha.org/wiki/Resource_Agents
>
> Essentially, they're programs (can be a shell script) given run with
> start stop etc arguments when an action needs to be done. You can run
> other commands such as fs_cli from there.
>
> I haven't written one from scratch myself though, I just use IPAddr2.
> Perhaps it would be useful for someone to create an example one in
> trunk or contrib?
>
> -Steve
>
>
> On 2 January 2011 16:16, Avi Marcus <Avi at amarcus.com> wrote:
> > Where would that go?
> > /me is confused.
> >
> > On Sun, Jan 2, 2011 at 5:36 PM, Ghulam Mustafa <mustafa.pk at gmail.com>
> wrote:
> >>
> >> Hi, for testing purposes i modified IPaddr ocf script on both nodes
> >> and added following line at the end of start_ipaddr.
> >> /opt/switch/bin/fs_cli -x "sofia recover"
> >>
> >> On 1/2/11, Steven Ayre <steveayre at gmail.com> wrote:
> >> > It's an API command that you can run via ESL. You can do that from a
> >> > script using fs_cli and its -x/--execute option.
> >> >
> >> > Something like this:
> >> >
> >> > fs_cli --host=otherhost --password=secretpasswd --execute="sofia
> >> > recover"
> >> >
> >> > -Steve
> >> >
> >> >
> >> > On 2 January 2011 09:43, Avi Marcus <Avi at amarcus.com> wrote:
> >> >> Thanks, I noticed the -nonat already.
> >> >> What do you mean by running sofia recover from my init script? I
> >> >> don't understand how that would be triggered by heartbeat/pacemaker.
> >> >> Any
> >> >> explanatory docs on the issue would be nice, but everything I've seen
> >> >> was
> >> >> explaining exact parts and nothing ever seemed to explain the general
> >> >> syntax
> >> >> of everything.
> >> >> Does anyone have a working example that they can wikify and explain?
> >> >> Thanks,
> >> >> Avi
> >> >>
> >> >> On Sun, Jan 2, 2011 at 4:06 AM, Seven Du <dujinfang at gmail.com>
> wrote:
> >> >>>
> >> >>> the wiki page already said you can start the spare FS ahead to speed
> >> >>> switch over up.
> >> >>>
> >> >>> it means if you use ip_nonlocal_bind and start the spare FS ahead,
> you
> >> >>> only need to run sofia recover in you init script without waiting
> for
> >> >>> the actually FS start process.
> >> >>>
> >> >>> On Sun, Jan 2, 2011 at 7:58 AM, Steven Ayre <steveayre at gmail.com>
> >> >>> wrote:
> >> >>> > 1) It seems freeswitch takes 8 seconds to restart. So if it
> crashes
> >> >>> > (unlikely) calls should immediately get shunted over to the other
> >> >>> > box.
> >> >>> > How
> >> >>> > do I set that up?
> >> >>> >
> >> >>> > Just a tip, I start fs with -nonat and find that makes it start
> >> >>> > faster.
> >> >>> >
> >> >>> > Steve on iPhone
> >> >>> > On 1 Jan 2011, at 20:20, Avi Marcus <Avi at aMarcus.com> wrote:
> >> >>> >
> >> >>> > Hi - I'm setting up 2 parallel computers for a HA setup. (low
> volume
> >> >>> > so
> >> >>> > far,
> >> >>> > no need to load balance.)
> >> >>> > I already have heartbeat able to grab the public IP if the first
> box
> >> >>> > goes
> >> >>> > down, but since I'm basically completely unfamiar with pacemaker,
> >> >>> > I'd
> >> >>> > like
> >> >>> > some help.
> >> >>> > I'd imagine I'm not the only one doing this, there should be no
> >> >>> > reason
> >> >>> > for
> >> >>> > me to recreate the wheel.
> >> >>> > Info should be on: http://wiki.freeswitch.org/wiki/Freeswitch_HA
> >> >>> > Specifically:
> >> >>> > 1) It seems freeswitch takes 8 seconds to restart. So if it
> crashes
> >> >>> > (unlikely) calls should immediately get shunted over to the other
> >> >>> > box.
> >> >>> > How
> >> >>> > do I set that up?
> >> >>> > 2) How do I automatically trigger a sofia recover? I know via
> >> >>> > commanline
> >> >>> > it's "fs_cli -x "sofia recover" but where do I tell pacemaker to
> do
> >> >>> > that?
> >> >>> > And I suppose a reload_xml before (maybe after?) is a good idea,
> >> >>> > too.
> >> >>> > Also, can I put <param name="track-calls" value="true"/> in a
> global
> >> >>> > file,
> >> >>> > or does it need to be in the actual profile files?
> >> >>> > And anything else I may be overlooking.
> >> >>> > Thanks guys!
> >> >>> > -Avi Marcus
> >> >>> >
> >> >>> > _______________________________________________
> >> >>> > FreeSWITCH-users mailing list
> >> >>> > FreeSWITCH-users at lists.freeswitch.org
> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >
> >> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> > http://www.freeswitch.org
> >> >>> >
> >> >>> > _______________________________________________
> >> >>> > FreeSWITCH-users mailing list
> >> >>> > FreeSWITCH-users at lists.freeswitch.org
> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >
> >> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> > http://www.freeswitch.org
> >> >>> >
> >> >>> >
> >> >>>
> >> >>>
> >> >>>
> >> >>> --
> >> >>> About: http://about.me/dujinfang
> >> >>> Blog: http://www.dujinfang.com
> >> >>> Proj:  http://www.freeswitch.org.cn
> >> >>>
> >> >>> _______________________________________________
> >> >>> FreeSWITCH-users mailing list
> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>
> >> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> http://www.freeswitch.org
> >> >>
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >>
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >>
> >>
> >> --
> >> Ghulam Mustafa
> >> cell: +92 333.611.7681
> >> sip: cyrenity at ekiga.net
> >> mail: mustafa.pk at gmail.com
> >> web: cyrenity.wordpress.com
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sun Jan  2 20:12:20 2011
From: brian at freeswitch.org (Brian West)
Date: Sun, 2 Jan 2011 11:12:20 -0600
Subject: [Freeswitch-users] Logging "dropped calls"
In-Reply-To: <4D206F95.20602@solomo.de>
References: <AANLkTi=OP-eVAr2QpxEghVN1v-ASFHUgC9n_3MyPRsm1@mail.gmail.com>
	<4D206F95.20602@solomo.de>
Message-ID: <F0EC2F69-2947-43DC-8904-E3F0BB0B5A78@freeswitch.org>

You must be using AT&T  :P

/b

On Jan 2, 2011, at 6:29 AM, Daniel Neubert wrote:

> Hi,
> 
> you can use ESL (http://wiki.freeswitch.org/wiki/Event_Socket_Library) to receive certain events  http://wiki.freeswitch.org/wiki/Event_list
> 
> or you can use information stored in XML CDR data to do your analysis: http://wiki.freeswitch.org/wiki/Mod_xml_cdr
> 
> Best regards / Mit freundlichen Gr??en,
> Daniel Neubert

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From rafonline at hotmail.com  Sun Jan  2 21:13:18 2011
From: rafonline at hotmail.com (Rafqat .)
Date: Sun, 2 Jan 2011 18:13:18 +0000
Subject: [Freeswitch-users] freeswitch restart problem
Message-ID: <COL122-W221156F7D205C12E057019B5060@phx.gbl>



Hi

I get the following error when i try to restart freeswitch.  It seems to start ok on boot up, but if i shut it down from the  cli and start it again, i get the following:

freeswitch: symbol lookup error: /usr/local/freeswitch/lib/libfreeswitch.so.1: undefined symbol: uuid_generate

Any help will be appreciated.

Cheers

Raf
 		 	   		  
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From infos at madovsky.org  Sun Jan  2 21:54:26 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 2 Jan 2011 13:54:26 -0500
Subject: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please
	Help setting up
References: <AANLkTi=6ZwmiMHLcBM7EndP8apmXf7CO9XFied1UOvRh@mail.gmail.com><65B881CD-D420-444B-91FD-04360AB97644@gmail.com><AANLkTinoFOPtPW+=uyoogi33u6LvAYz_ta3bDc7XMhs0@mail.gmail.com><AANLkTikBrP6ezm9uPrU7JdG29HP7BDZ7U5nReavz--6=@mail.gmail.com><AANLkTiksQD5QA8vUBHe2yBsh6ahA+2xcZ+LFGNst0dGT@mail.gmail.com><AANLkTimZH-ONShU-e98f-HjDg_UkvcHfp9gAWt5Ki=+h@mail.gmail.com><AANLkTineUendBAcYYd2fNWnPvyoXRoqqkaRS6VJg_-M3@mail.gmail.com>
	<AANLkTi=cEkm0_fTEteE2+f9wk-bhDAVVmkf_EdfbQoR6@mail.gmail.com>
Message-ID: <12A6A7ABC6D04D0888956E09FE200C71@e1705>

it's a solution also,
but you can also use LSB FS startup script in HB/Pacemaker
with some modifications to be fully compliant

----- Original Message ----- 
From: "Steven Ayre" <steveayre at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Sunday, January 02, 2011 11:27 AM
Subject: Re: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please 
Help setting up


You would create a new fs resource agent
http://www.linux-ha.org/wiki/Resource_Agents

Essentially, they're programs (can be a shell script) given run with
start stop etc arguments when an action needs to be done. You can run
other commands such as fs_cli from there.

I haven't written one from scratch myself though, I just use IPAddr2.
Perhaps it would be useful for someone to create an example one in
trunk or contrib?

-Steve


On 2 January 2011 16:16, Avi Marcus <Avi at amarcus.com> wrote:
> Where would that go?
> /me is confused.
>
> On Sun, Jan 2, 2011 at 5:36 PM, Ghulam Mustafa <mustafa.pk at gmail.com> 
> wrote:
>>
>> Hi, for testing purposes i modified IPaddr ocf script on both nodes
>> and added following line at the end of start_ipaddr.
>> /opt/switch/bin/fs_cli -x "sofia recover"
>>
>> On 1/2/11, Steven Ayre <steveayre at gmail.com> wrote:
>> > It's an API command that you can run via ESL. You can do that from a
>> > script using fs_cli and its -x/--execute option.
>> >
>> > Something like this:
>> >
>> > fs_cli --host=otherhost --password=secretpasswd --execute="sofia
>> > recover"
>> >
>> > -Steve
>> >
>> >
>> > On 2 January 2011 09:43, Avi Marcus <Avi at amarcus.com> wrote:
>> >> Thanks, I noticed the -nonat already.
>> >> What do you mean by running sofia recover from my init script? I
>> >> don't understand how that would be triggered by heartbeat/pacemaker.
>> >> Any
>> >> explanatory docs on the issue would be nice, but everything I've seen
>> >> was
>> >> explaining exact parts and nothing ever seemed to explain the general
>> >> syntax
>> >> of everything.
>> >> Does anyone have a working example that they can wikify and explain?
>> >> Thanks,
>> >> Avi
>> >>
>> >> On Sun, Jan 2, 2011 at 4:06 AM, Seven Du <dujinfang at gmail.com> wrote:
>> >>>
>> >>> the wiki page already said you can start the spare FS ahead to speed
>> >>> switch over up.
>> >>>
>> >>> it means if you use ip_nonlocal_bind and start the spare FS ahead, 
>> >>> you
>> >>> only need to run sofia recover in you init script without waiting for
>> >>> the actually FS start process.
>> >>>
>> >>> On Sun, Jan 2, 2011 at 7:58 AM, Steven Ayre <steveayre at gmail.com>
>> >>> wrote:
>> >>> > 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
>> >>> > (unlikely) calls should immediately get shunted over to the other
>> >>> > box.
>> >>> > How
>> >>> > do I set that up?
>> >>> >
>> >>> > Just a tip, I start fs with -nonat and find that makes it start
>> >>> > faster.
>> >>> >
>> >>> > Steve on iPhone
>> >>> > On 1 Jan 2011, at 20:20, Avi Marcus <Avi at aMarcus.com> wrote:
>> >>> >
>> >>> > Hi - I'm setting up 2 parallel computers for a HA setup. (low 
>> >>> > volume
>> >>> > so
>> >>> > far,
>> >>> > no need to load balance.)
>> >>> > I already have heartbeat able to grab the public IP if the first 
>> >>> > box
>> >>> > goes
>> >>> > down, but since I'm basically completely unfamiar with pacemaker,
>> >>> > I'd
>> >>> > like
>> >>> > some help.
>> >>> > I'd imagine I'm not the only one doing this, there should be no
>> >>> > reason
>> >>> > for
>> >>> > me to recreate the wheel.
>> >>> > Info should be on: http://wiki.freeswitch.org/wiki/Freeswitch_HA
>> >>> > Specifically:
>> >>> > 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
>> >>> > (unlikely) calls should immediately get shunted over to the other
>> >>> > box.
>> >>> > How
>> >>> > do I set that up?
>> >>> > 2) How do I automatically trigger a sofia recover? I know via
>> >>> > commanline
>> >>> > it's "fs_cli -x "sofia recover" but where do I tell pacemaker to do
>> >>> > that?
>> >>> > And I suppose a reload_xml before (maybe after?) is a good idea,
>> >>> > too.
>> >>> > Also, can I put <param name="track-calls" value="true"/> in a 
>> >>> > global
>> >>> > file,
>> >>> > or does it need to be in the actual profile files?
>> >>> > And anything else I may be overlooking.
>> >>> > Thanks guys!
>> >>> > -Avi Marcus
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >
>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >
>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>>
>> >>>
>> >>>
>> >>> --
>> >>> About: http://about.me/dujinfang
>> >>> Blog: http://www.dujinfang.com
>> >>> Proj: http://www.freeswitch.org.cn
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>> --
>> Ghulam Mustafa
>> cell: +92 333.611.7681
>> sip: cyrenity at ekiga.net
>> mail: mustafa.pk at gmail.com
>> web: cyrenity.wordpress.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 



From infos at madovsky.org  Sun Jan  2 21:57:21 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 2 Jan 2011 13:57:21 -0500
Subject: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please
	Help setting up
References: <AANLkTi=6ZwmiMHLcBM7EndP8apmXf7CO9XFied1UOvRh@mail.gmail.com><65B881CD-D420-444B-91FD-04360AB97644@gmail.com><AANLkTinoFOPtPW+=uyoogi33u6LvAYz_ta3bDc7XMhs0@mail.gmail.com><AANLkTikBrP6ezm9uPrU7JdG29HP7BDZ7U5nReavz--6=@mail.gmail.com><AANLkTiksQD5QA8vUBHe2yBsh6ahA+2xcZ+LFGNst0dGT@mail.gmail.com>
	<AANLkTi=Mkz2muxKRYkD=LOCTVGeR1hirD51EQmk+tZA5@mail.gmail.com>
Message-ID: <BF798C6B3CC042F69904326134467DAA@e1705>

Avi,

I think this thread is more related to HB/Pacemaker.
they have a lot of doc of how to create a cluster.
Maybe go to check it, because as I said to you you can't
build a cluster with only a thread in Freeswitchy
  ----- Original Message ----- 
  From: Avi Marcus 
  To: FreeSWITCH Users Help 
  Sent: Sunday, January 02, 2011 10:38 AM
  Subject: Re: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please Help setting up


  Yes, yes: fs_cli -x "sofia recover" I understand the freeswitch parts.
  Now how does that get set up with heartbeat/pacemaker?


  You seem to assume I have a working pacemaker understanding, which I do not. I wouldn't mind learning about it *if I saw any docs that made sense.*
  -Avi


  On Sun, Jan 2, 2011 at 12:31 PM, Steven Ayre <steveayre at gmail.com> wrote:

    It's an API command that you can run via ESL. You can do that from a
    script using fs_cli and its -x/--execute option.

    Something like this:

    fs_cli --host=otherhost --password=secretpasswd --execute="sofia recover"

    -Steve



    On 2 January 2011 09:43, Avi Marcus <Avi at amarcus.com> wrote:
    > Thanks, I noticed the -nonat already.
    > What do you mean by running sofia recover from my init script? I
    > don't understand how that would be triggered by heartbeat/pacemaker. Any
    > explanatory docs on the issue would be nice, but everything I've seen was
    > explaining exact parts and nothing ever seemed to explain the general syntax
    > of everything.
    > Does anyone have a working example that they can wikify and explain?
    > Thanks,
    > Avi
    >
    > On Sun, Jan 2, 2011 at 4:06 AM, Seven Du <dujinfang at gmail.com> wrote:
    >>
    >> the wiki page already said you can start the spare FS ahead to speed
    >> switch over up.
    >>
    >> it means if you use ip_nonlocal_bind and start the spare FS ahead, you
    >> only need to run sofia recover in you init script without waiting for
    >> the actually FS start process.
    >>
    >> On Sun, Jan 2, 2011 at 7:58 AM, Steven Ayre <steveayre at gmail.com> wrote:
    >> > 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
    >> > (unlikely) calls should immediately get shunted over to the other box.
    >> > How
    >> > do I set that up?
    >> >
    >> > Just a tip, I start fs with -nonat and find that makes it start faster.
    >> >
    >> > Steve on iPhone
    >> > On 1 Jan 2011, at 20:20, Avi Marcus <Avi at aMarcus.com> wrote:
    >> >
    >> > Hi - I'm setting up 2 parallel computers for a HA setup. (low volume so
    >> > far,
    >> > no need to load balance.)
    >> > I already have heartbeat able to grab the public IP if the first box
    >> > goes
    >> > down, but since I'm basically completely unfamiar with pacemaker, I'd
    >> > like
    >> > some help.
    >> > I'd imagine I'm not the only one doing this, there should be no reason
    >> > for
    >> > me to recreate the wheel.
    >> > Info should be on: http://wiki.freeswitch.org/wiki/Freeswitch_HA
    >> > Specifically:
    >> > 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
    >> > (unlikely) calls should immediately get shunted over to the other box.
    >> > How
    >> > do I set that up?
    >> > 2) How do I automatically trigger a sofia recover? I know via commanline
    >> > it's "fs_cli -x "sofia recover" but where do I tell pacemaker to do
    >> > that?
    >> > And I suppose a reload_xml before (maybe after?) is a good idea, too.
    >> > Also, can I put <param name="track-calls" value="true"/> in a global
    >> > file,
    >> > or does it need to be in the actual profile files?
    >> > And anything else I may be overlooking.
    >> > Thanks guys!
    >> > -Avi Marcus
    >> >
    >> > _______________________________________________
    >> > FreeSWITCH-users mailing list
    >> > FreeSWITCH-users at lists.freeswitch.org
    >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    >> > http://www.freeswitch.org
    >> >
    >> > _______________________________________________
    >> > FreeSWITCH-users mailing list
    >> > FreeSWITCH-users at lists.freeswitch.org
    >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    >> > http://www.freeswitch.org
    >> >
    >> >
    >>
    >>
    >>
    >> --
    >> About: http://about.me/dujinfang
    >> Blog: http://www.dujinfang.com
    >> Proj:  http://www.freeswitch.org.cn
    >>
    >> _______________________________________________
    >> FreeSWITCH-users mailing list
    >> FreeSWITCH-users at lists.freeswitch.org
    >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    >> http://www.freeswitch.org
    >
    >
    > _______________________________________________
    > FreeSWITCH-users mailing list
    > FreeSWITCH-users at lists.freeswitch.org
    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > http://www.freeswitch.org
    >
    >

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





------------------------------------------------------------------------------


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Sun Jan  2 22:08:24 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 2 Jan 2011 14:08:24 -0500
Subject: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please
	Help setting up
References: <AANLkTi=6ZwmiMHLcBM7EndP8apmXf7CO9XFied1UOvRh@mail.gmail.com><65B881CD-D420-444B-91FD-04360AB97644@gmail.com><AANLkTinoFOPtPW+=uyoogi33u6LvAYz_ta3bDc7XMhs0@mail.gmail.com>
	<AANLkTikBrP6ezm9uPrU7JdG29HP7BDZ7U5nReavz--6=@mail.gmail.com>
Message-ID: <E9860A29ADD749559B8A7B063C9F3BAC@e1705>

I use HB/Pacemaker, tried to hack the original FS init script,
but didn't succeed to work it well until now (when a FS fails, the failover fails also in Pacemaker.
so I decided to create a cron task that ping/telnet/start/stop every FS

Fanck
  ----- Original Message ----- 
  From: Avi Marcus 
  To: FreeSWITCH Users Help 
  Sent: Sunday, January 02, 2011 4:43 AM
  Subject: Re: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please Help setting up


  Thanks, I noticed the -nonat already.
  What do you mean by running sofia recover from my init script? I don't understand how that would be triggered by heartbeat/pacemaker. Any explanatory docs on the issue would be nice, but everything I've seen was explaining exact parts and nothing ever seemed to explain the general syntax of everything.


  Does anyone have a working example that they can wikify and explain?
  Thanks,
  Avi


  On Sun, Jan 2, 2011 at 4:06 AM, Seven Du <dujinfang at gmail.com> wrote:

    the wiki page already said you can start the spare FS ahead to speed
    switch over up.

    it means if you use ip_nonlocal_bind and start the spare FS ahead, you
    only need to run sofia recover in you init script without waiting for
    the actually FS start process.


    On Sun, Jan 2, 2011 at 7:58 AM, Steven Ayre <steveayre at gmail.com> wrote:
    > 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
    > (unlikely) calls should immediately get shunted over to the other box. How
    > do I set that up?
    >
    > Just a tip, I start fs with -nonat and find that makes it start faster.
    >
    > Steve on iPhone
    > On 1 Jan 2011, at 20:20, Avi Marcus <Avi at aMarcus.com> wrote:
    >
    > Hi - I'm setting up 2 parallel computers for a HA setup. (low volume so far,
    > no need to load balance.)
    > I already have heartbeat able to grab the public IP if the first box goes
    > down, but since I'm basically completely unfamiar with pacemaker, I'd like
    > some help.
    > I'd imagine I'm not the only one doing this, there should be no reason for
    > me to recreate the wheel.
    > Info should be on: http://wiki.freeswitch.org/wiki/Freeswitch_HA
    > Specifically:
    > 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
    > (unlikely) calls should immediately get shunted over to the other box. How
    > do I set that up?
    > 2) How do I automatically trigger a sofia recover? I know via commanline
    > it's "fs_cli -x "sofia recover" but where do I tell pacemaker to do that?
    > And I suppose a reload_xml before (maybe after?) is a good idea, too.
    > Also, can I put <param name="track-calls" value="true"/> in a global file,
    > or does it need to be in the actual profile files?
    > And anything else I may be overlooking.
    > Thanks guys!
    > -Avi Marcus
    >
    > _______________________________________________
    > FreeSWITCH-users mailing list
    > FreeSWITCH-users at lists.freeswitch.org
    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > http://www.freeswitch.org
    >
    > _______________________________________________
    > FreeSWITCH-users mailing list
    > FreeSWITCH-users at lists.freeswitch.org
    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > http://www.freeswitch.org
    >
    >




    --
    About: http://about.me/dujinfang
    Blog: http://www.dujinfang.com
    Proj:  http://www.freeswitch.org.cn


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





------------------------------------------------------------------------------


  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From Avi at aMarcus.com  Sun Jan  2 22:09:49 2011
From: Avi at aMarcus.com (Avi Marcus)
Date: Sun, 2 Jan 2011 21:09:49 +0200
Subject: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please
 Help setting up
In-Reply-To: <BF798C6B3CC042F69904326134467DAA@e1705>
References: <AANLkTi=6ZwmiMHLcBM7EndP8apmXf7CO9XFied1UOvRh@mail.gmail.com>
	<65B881CD-D420-444B-91FD-04360AB97644@gmail.com>
	<AANLkTinoFOPtPW+=uyoogi33u6LvAYz_ta3bDc7XMhs0@mail.gmail.com>
	<AANLkTikBrP6ezm9uPrU7JdG29HP7BDZ7U5nReavz--6=@mail.gmail.com>
	<AANLkTiksQD5QA8vUBHe2yBsh6ahA+2xcZ+LFGNst0dGT@mail.gmail.com>
	<AANLkTi=Mkz2muxKRYkD=LOCTVGeR1hirD51EQmk+tZA5@mail.gmail.com>
	<BF798C6B3CC042F69904326134467DAA@e1705>
Message-ID: <AANLkTi=Gs9a-z6g9bxZm=RdqN7zbX+O-mgwj3U55S-RE@mail.gmail.com>

a) Sounds great. I've googled several times. Where are the docs that you
suggest I start with?
b) Even so, it's been partially documented on the FreeSWITCH wiki and would
be great to be finished.
-Avi

On Sun, Jan 2, 2011 at 8:57 PM, Madovsky <infos at madovsky.org> wrote:

>  Avi,
>
> I think this thread is more related to HB/Pacemaker.
> they have a lot of doc of how to create a cluster.
> Maybe go to check it, because as I said to you you can't
> build a cluster with only a thread in Freeswitchy
>
> ----- Original Message -----
> *From:* Avi Marcus <Avi at aMarcus.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Sunday, January 02, 2011 10:38 AM
> *Subject:* Re: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker -
> Please Help setting up
>
> Yes, yes: fs_cli -x "sofia recover" I understand the freeswitch parts.
> Now how does that get set up with heartbeat/pacemaker?
>
> You seem to assume I have a working pacemaker understanding, which I do
> not. I wouldn't mind learning about it *if I saw any docs that made sense.*
> -Avi
>
> On Sun, Jan 2, 2011 at 12:31 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> It's an API command that you can run via ESL. You can do that from a
>> script using fs_cli and its -x/--execute option.
>>
>> Something like this:
>>
>> fs_cli --host=otherhost --password=secretpasswd --execute="sofia recover"
>>
>> -Steve
>>
>>
>> On 2 January 2011 09:43, Avi Marcus <Avi at amarcus.com> wrote:
>> > Thanks, I noticed the -nonat already.
>> > What do you mean by running sofia recover from my init script? I
>> > don't understand how that would be triggered by heartbeat/pacemaker. Any
>> > explanatory docs on the issue would be nice, but everything I've seen
>> was
>> > explaining exact parts and nothing ever seemed to explain the general
>> syntax
>> > of everything.
>> > Does anyone have a working example that they can wikify and explain?
>> > Thanks,
>> > Avi
>> >
>> > On Sun, Jan 2, 2011 at 4:06 AM, Seven Du <dujinfang at gmail.com> wrote:
>> >>
>> >> the wiki page already said you can start the spare FS ahead to speed
>> >> switch over up.
>> >>
>> >> it means if you use ip_nonlocal_bind and start the spare FS ahead, you
>> >> only need to run sofia recover in you init script without waiting for
>> >> the actually FS start process.
>> >>
>> >> On Sun, Jan 2, 2011 at 7:58 AM, Steven Ayre <steveayre at gmail.com>
>> wrote:
>> >> > 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
>> >> > (unlikely) calls should immediately get shunted over to the other
>> box.
>> >> > How
>> >> > do I set that up?
>> >> >
>> >> > Just a tip, I start fs with -nonat and find that makes it start
>> faster.
>> >> >
>> >> > Steve on iPhone
>> >> > On 1 Jan 2011, at 20:20, Avi Marcus <Avi at aMarcus.com> wrote:
>> >> >
>> >> > Hi - I'm setting up 2 parallel computers for a HA setup. (low volume
>> so
>> >> > far,
>> >> > no need to load balance.)
>> >> > I already have heartbeat able to grab the public IP if the first box
>> >> > goes
>> >> > down, but since I'm basically completely unfamiar with pacemaker, I'd
>> >> > like
>> >> > some help.
>> >> > I'd imagine I'm not the only one doing this, there should be no
>> reason
>> >> > for
>> >> > me to recreate the wheel.
>> >> > Info should be on: http://wiki.freeswitch.org/wiki/Freeswitch_HA
>> >> > Specifically:
>> >> > 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
>> >> > (unlikely) calls should immediately get shunted over to the other
>> box.
>> >> > How
>> >> > do I set that up?
>> >> > 2) How do I automatically trigger a sofia recover? I know via
>> commanline
>> >> > it's "fs_cli -x "sofia recover" but where do I tell pacemaker to do
>> >> > that?
>> >> > And I suppose a reload_xml before (maybe after?) is a good idea, too.
>> >> > Also, can I put <param name="track-calls" value="true"/> in a global
>> >> > file,
>> >> > or does it need to be in the actual profile files?
>> >> > And anything else I may be overlooking.
>> >> > Thanks guys!
>> >> > -Avi Marcus
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> About: http://about.me/dujinfang
>> >> Blog: http://www.dujinfang.com
>> >> Proj:  http://www.freeswitch.org.cn
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
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>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
>  ------------------------------
>
> _______________________________________________
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From rafonline at hotmail.com  Sun Jan  2 22:13:01 2011
From: rafonline at hotmail.com (Rafqat .)
Date: Sun, 2 Jan 2011 19:13:01 +0000
Subject: [Freeswitch-users] calling card app
Message-ID: <COL122-W17F8F5B93182F2AFF31380B5060@phx.gbl>


Hi

As stated in some of my previous posts, I am writing a calling card system (not too sure of potential number of concurrent users yet).

At the moment I am simply doing everything in a single lua script utilising mod_lcr and mod_nibble.  It seems to work ok, but I have not stress tested it yet.

I was wondering (at a high level) if this will suffice or should I be offloading operations such as pin validation and credit checking to another server (maybe utilising mod_rad_auth?).    

Cheers

Raf
 		 	   		  
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From infos at madovsky.org  Sun Jan  2 22:14:13 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 2 Jan 2011 14:14:13 -0500
Subject: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please
	Help setting up
Message-ID: <FD14C3FFADC546FE9D3FFC68F1CA2036@e1705>

the challenge to integrate FS in HB/Pacemaker as a compliant resources is
to manage well the time of start stop delay, since FS doesn't start instantly it can be
considered as failed resource
  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Sunday, January 02, 2011 2:08 PM
  Subject: Re: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please Help setting up


  I use HB/Pacemaker, tried to hack the original FS init script,
  but didn't succeed to work it well until now (when a FS fails, the failover fails also in Pacemaker.
  so I decided to create a cron task that ping/telnet/start/stop every FS

  Fanck
    ----- Original Message ----- 
    From: Avi Marcus 
    To: FreeSWITCH Users Help 
    Sent: Sunday, January 02, 2011 4:43 AM
    Subject: Re: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please Help setting up


    Thanks, I noticed the -nonat already. 
    What do you mean by running sofia recover from my init script? I don't understand how that would be triggered by heartbeat/pacemaker. Any explanatory docs on the issue would be nice, but everything I've seen was explaining exact parts and nothing ever seemed to explain the general syntax of everything.


    Does anyone have a working example that they can wikify and explain?
    Thanks,
    Avi


    On Sun, Jan 2, 2011 at 4:06 AM, Seven Du <dujinfang at gmail.com> wrote:

      the wiki page already said you can start the spare FS ahead to speed
      switch over up.

      it means if you use ip_nonlocal_bind and start the spare FS ahead, you
      only need to run sofia recover in you init script without waiting for
      the actually FS start process.


      On Sun, Jan 2, 2011 at 7:58 AM, Steven Ayre <steveayre at gmail.com> wrote:
      > 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
      > (unlikely) calls should immediately get shunted over to the other box. How
      > do I set that up?
      >
      > Just a tip, I start fs with -nonat and find that makes it start faster.
      >
      > Steve on iPhone
      > On 1 Jan 2011, at 20:20, Avi Marcus <Avi at aMarcus.com> wrote:
      >
      > Hi - I'm setting up 2 parallel computers for a HA setup. (low volume so far,
      > no need to load balance.)
      > I already have heartbeat able to grab the public IP if the first box goes
      > down, but since I'm basically completely unfamiar with pacemaker, I'd like
      > some help.
      > I'd imagine I'm not the only one doing this, there should be no reason for
      > me to recreate the wheel.
      > Info should be on: http://wiki.freeswitch.org/wiki/Freeswitch_HA
      > Specifically:
      > 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
      > (unlikely) calls should immediately get shunted over to the other box. How
      > do I set that up?
      > 2) How do I automatically trigger a sofia recover? I know via commanline
      > it's "fs_cli -x "sofia recover" but where do I tell pacemaker to do that?
      > And I suppose a reload_xml before (maybe after?) is a good idea, too.
      > Also, can I put <param name="track-calls" value="true"/> in a global file,
      > or does it need to be in the actual profile files?
      > And anything else I may be overlooking.
      > Thanks guys!
      > -Avi Marcus
      >
      > _______________________________________________
      > FreeSWITCH-users mailing list
      > FreeSWITCH-users at lists.freeswitch.org
      > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      > http://www.freeswitch.org
      >
      > _______________________________________________
      > FreeSWITCH-users mailing list
      > FreeSWITCH-users at lists.freeswitch.org
      > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      > http://www.freeswitch.org
      >
      >




      --
      About: http://about.me/dujinfang
      Blog: http://www.dujinfang.com
      Proj:  http://www.freeswitch.org.cn


      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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----------------------------------------------------------------------------


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    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From frank at telonium.com  Sun Jan  2 22:14:08 2011
From: frank at telonium.com (Frank Park)
Date: Sun, 2 Jan 2011 14:14:08 -0500
Subject: [Freeswitch-users] Logging "dropped calls"
In-Reply-To: <F0EC2F69-2947-43DC-8904-E3F0BB0B5A78@freeswitch.org>
References: <AANLkTi=OP-eVAr2QpxEghVN1v-ASFHUgC9n_3MyPRsm1@mail.gmail.com>
	<4D206F95.20602@solomo.de>
	<F0EC2F69-2947-43DC-8904-E3F0BB0B5A78@freeswitch.org>
Message-ID: <AANLkTikYxrZ35TLUL2KTeN1oJ0smqOOF9N1MJdbcePEy@mail.gmail.com>

Haha..
Not me! Our servers are in a major colo, doubt they are having a
network issue. Otherwise, I would know about it.
But there's been few customer complaints on dropped calls (some AT&T
indeed).. I usually automatically assume it's their network issue and
not FS issue, but it would be nice to know the stats on what is
happening on our side.
The CDR gives a good info like clearing conditions, but it seems to
always be NORMAL_CLEARING regardless of dropped calls or not.. I don't
know if NORMAL_CLEARING only happens when SIP successfully tears down
both legs or not (anybody to clarify?).. If it doesn't it's not that
useful in my case.

I might have to look into ESL option unless anybody else has a better idea.

Frank


----=======================----
Frank Park
Telonium Communications, LLC
frank at telonium.com
http://www.telonium.com
Follow Us on Twitter: @GetTelonium
404-566-8888 x1001 Office
404-939-4242 Cell
----=======================----



On Sun, Jan 2, 2011 at 12:12 PM, Brian West <brian at freeswitch.org> wrote:
> You must be using AT&T ?:P
> /b
> On Jan 2, 2011, at 6:29 AM, Daniel Neubert wrote:
>
> Hi,
>
> you can use ESL (http://wiki.freeswitch.org/wiki/Event_Socket_Library)?to
> receive certain events??http://wiki.freeswitch.org/wiki/Event_list
>
> or you can use information stored in XML CDR data to do your
> analysis:?http://wiki.freeswitch.org/wiki/Mod_xml_cdr
>
> Best regards / Mit freundlichen Gr??en,
> Daniel Neubert
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From infos at madovsky.org  Sun Jan  2 22:15:05 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 2 Jan 2011 14:15:05 -0500
Subject: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please
	Help setting up
References: <AANLkTi=6ZwmiMHLcBM7EndP8apmXf7CO9XFied1UOvRh@mail.gmail.com><65B881CD-D420-444B-91FD-04360AB97644@gmail.com><AANLkTinoFOPtPW+=uyoogi33u6LvAYz_ta3bDc7XMhs0@mail.gmail.com><AANLkTikBrP6ezm9uPrU7JdG29HP7BDZ7U5nReavz--6=@mail.gmail.com><AANLkTiksQD5QA8vUBHe2yBsh6ahA+2xcZ+LFGNst0dGT@mail.gmail.com><AANLkTi=Mkz2muxKRYkD=LOCTVGeR1hirD51EQmk+tZA5@mail.gmail.com><BF798C6B3CC042F69904326134467DAA@e1705>
	<AANLkTi=Gs9a-z6g9bxZm=RdqN7zbX+O-mgwj3U55S-RE@mail.gmail.com>
Message-ID: <F40C434EAA934C83BA9D498BC9EAD41A@e1705>

http://www.clusterlabs.org/wiki/Documentation
  ----- Original Message ----- 
  From: Avi Marcus 
  To: FreeSWITCH Users Help 
  Sent: Sunday, January 02, 2011 2:09 PM
  Subject: Re: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please Help setting up


  a) Sounds great. I've googled several times. Where are the docs that you suggest I start with? 
  b) Even so, it's been partially documented on the FreeSWITCH wiki and would be great to be finished.
  -Avi


  On Sun, Jan 2, 2011 at 8:57 PM, Madovsky <infos at madovsky.org> wrote:

    Avi,

    I think this thread is more related to HB/Pacemaker.
    they have a lot of doc of how to create a cluster.
    Maybe go to check it, because as I said to you you can't
    build a cluster with only a thread in Freeswitchy
      ----- Original Message ----- 
      From: Avi Marcus 
      To: FreeSWITCH Users Help 
      Sent: Sunday, January 02, 2011 10:38 AM
      Subject: Re: [Freeswitch-users] FreeSWITCH & Heartbeat / Pacemaker - Please Help setting up


      Yes, yes: fs_cli -x "sofia recover" I understand the freeswitch parts. 
      Now how does that get set up with heartbeat/pacemaker?


      You seem to assume I have a working pacemaker understanding, which I do not. I wouldn't mind learning about it *if I saw any docs that made sense.*
      -Avi


      On Sun, Jan 2, 2011 at 12:31 PM, Steven Ayre <steveayre at gmail.com> wrote:

        It's an API command that you can run via ESL. You can do that from a
        script using fs_cli and its -x/--execute option.

        Something like this:

        fs_cli --host=otherhost --password=secretpasswd --execute="sofia recover"

        -Steve



        On 2 January 2011 09:43, Avi Marcus <Avi at amarcus.com> wrote:
        > Thanks, I noticed the -nonat already.
        > What do you mean by running sofia recover from my init script? I
        > don't understand how that would be triggered by heartbeat/pacemaker. Any
        > explanatory docs on the issue would be nice, but everything I've seen was
        > explaining exact parts and nothing ever seemed to explain the general syntax
        > of everything.
        > Does anyone have a working example that they can wikify and explain?
        > Thanks,
        > Avi
        >
        > On Sun, Jan 2, 2011 at 4:06 AM, Seven Du <dujinfang at gmail.com> wrote:
        >>
        >> the wiki page already said you can start the spare FS ahead to speed
        >> switch over up.
        >>
        >> it means if you use ip_nonlocal_bind and start the spare FS ahead, you
        >> only need to run sofia recover in you init script without waiting for
        >> the actually FS start process.
        >>
        >> On Sun, Jan 2, 2011 at 7:58 AM, Steven Ayre <steveayre at gmail.com> wrote:
        >> > 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
        >> > (unlikely) calls should immediately get shunted over to the other box.
        >> > How
        >> > do I set that up?
        >> >
        >> > Just a tip, I start fs with -nonat and find that makes it start faster.
        >> >
        >> > Steve on iPhone
        >> > On 1 Jan 2011, at 20:20, Avi Marcus <Avi at aMarcus.com> wrote:
        >> >
        >> > Hi - I'm setting up 2 parallel computers for a HA setup. (low volume so
        >> > far,
        >> > no need to load balance.)
        >> > I already have heartbeat able to grab the public IP if the first box
        >> > goes
        >> > down, but since I'm basically completely unfamiar with pacemaker, I'd
        >> > like
        >> > some help.
        >> > I'd imagine I'm not the only one doing this, there should be no reason
        >> > for
        >> > me to recreate the wheel.
        >> > Info should be on: http://wiki.freeswitch.org/wiki/Freeswitch_HA
        >> > Specifically:
        >> > 1) It seems freeswitch takes 8 seconds to restart. So if it crashes
        >> > (unlikely) calls should immediately get shunted over to the other box.
        >> > How
        >> > do I set that up?
        >> > 2) How do I automatically trigger a sofia recover? I know via commanline
        >> > it's "fs_cli -x "sofia recover" but where do I tell pacemaker to do
        >> > that?
        >> > And I suppose a reload_xml before (maybe after?) is a good idea, too.
        >> > Also, can I put <param name="track-calls" value="true"/> in a global
        >> > file,
        >> > or does it need to be in the actual profile files?
        >> > And anything else I may be overlooking.
        >> > Thanks guys!
        >> > -Avi Marcus
        >> >
        >> > _______________________________________________
        >> > FreeSWITCH-users mailing list
        >> > FreeSWITCH-users at lists.freeswitch.org
        >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        >> > http://www.freeswitch.org
        >> >
        >> > _______________________________________________
        >> > FreeSWITCH-users mailing list
        >> > FreeSWITCH-users at lists.freeswitch.org
        >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        >> > http://www.freeswitch.org
        >> >
        >> >
        >>
        >>
        >>
        >> --
        >> About: http://about.me/dujinfang
        >> Blog: http://www.dujinfang.com
        >> Proj:  http://www.freeswitch.org.cn
        >>
        >> _______________________________________________
        >> FreeSWITCH-users mailing list
        >> FreeSWITCH-users at lists.freeswitch.org
        >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        >> http://www.freeswitch.org
        >
        >
        > _______________________________________________
        > FreeSWITCH-users mailing list
        > FreeSWITCH-users at lists.freeswitch.org
        > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        > http://www.freeswitch.org
        >
        >

        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org





--------------------------------------------------------------------------


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      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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    _______________________________________________
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From infos at madovsky.org  Sun Jan  2 22:22:59 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 2 Jan 2011 14:22:59 -0500
Subject: [Freeswitch-users] nibblebill pause
Message-ID: <7C2A55D29A6640D8ADF12B687064F706@e1705>

When I use 
<action application="nibblebill" data="pause"/>
in dialplan all the channel is paused, is it the normal
behavior ?

Thanks 
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From rupa at rupa.com  Sun Jan  2 22:43:09 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Sun, 2 Jan 2011 13:43:09 -0600
Subject: [Freeswitch-users] mod lcr
In-Reply-To: <COL122-W25871C908DFBFB412EE3C9B5030@phx.gbl>
References: <COL122-W41FF54690AE9D0D554E13EB5030@phx.gbl>
	<AANLkTinu1DmGDZ3mqB-tbQp2Q3CSWan3gNgSX14P=q3y@mail.gmail.com>
	<COL122-W25871C908DFBFB412EE3C9B5030@phx.gbl>
Message-ID: <AANLkTi=OviZ7tPfA4MKdcvr2ZVt=JYbKhOKC4KjrYsoS@mail.gmail.com>

Again, which rate do you want to use to calculate your 5min duration?
 Depending on your carrier the rate will be significantly different.

Anyway, look at lcr_rate_N where N is the route entry number.  This is set
as a channel var.  After calling lcr, maybe call the info app to see which
lcr variables are available for use?

On Thu, Dec 30, 2010 at 11:05 AM, Rafqat . <rafonline at hotmail.com> wrote:

>
>
> Hi Rupa
>
> I am after the 'nibble_rate' field which is returned from my custom sql:
>
>
> [lcr_carrier=carrier2,lcr_rate=0.12000,nibble_account=12345,nibble_rate=0.24000,lowbal_amt=0.1,nobal_amt=0]sofia/external/01 at proxy.carrier2.net:5060
> |[lcr_carrier=carrier1,lcr_rate=0.15000,nibble_account=12345,nibble_rate=0.30000,lowbal_amt=0.1,nobal_amt=0]sofia/gateway/carrier1/$1
>
> I would like to calculate the time the user has for the call before doing
> any bridging as they might have positive funds, but not enough for a minimum
> call duration of 5 minutes.
>
> Is there away of getting this field into session scope?
>
> Cheers
>
> Raf
>
>
> ________________________________
> > Date: Thu, 30 Dec 2010 10:51:39 -0600
> > From: rupa at rupa.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] mod lcr
> >
> > I guess my first question would be *which* rate field? Notice they are
> > all different. If you mean after the bridge completes, then the var is
> > set on the b-leg which you can use to announce the rate or whatever.
> >
> > On Thu, Dec 30, 2010 at 10:22 AM, Rafqat .
> > > wrote:
> >
> >
> > Hi,
> >
> > I am using mod lcr with mod nibble, my example LCR invocation returns
> > the following in lcr_auto_route:
> >
> >
> [lcr_carrier=carrier2,lcr_rate=0.12000,nibble_account=12345,nibble_rate=0.24000,lowbal_amt=0.1,nobal_amt=0]sofia/external/01 at proxy.carrier2.net:5060
> |[lcr_carrier=carrier1,lcr_rate=0.15000,nibble_account=12345,nibble_rate=0.30000,lowbal_amt=0.1,nobal_amt=0]sofia/gateway/carrier1/$1
> >
> > I was wondering if there is a quick method of accessing the nibble_rate
> > field, so I can give callers an estimate of how long they before
> > bridging their call? I was hoping for some session variable to be
> > populated by lcr to give easy access to this field.
> >
> > Any help will be appreciated
> >
> > Cheers
> >
> >
> > Raf
> >
> >
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > --
> > -Rupa
> >
> > _______________________________________________ FreeSWITCH-users
> > mailing list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From infos at madovsky.org  Sun Jan  2 23:19:23 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 2 Jan 2011 15:19:23 -0500
Subject: [Freeswitch-users] enable_heartbeat_events
Message-ID: <DCAC3F2E8D4E4C609CBDD890BE841C81@e1705>

Rupa,

is enable_heartbeat_events has the same rule as
global heartbeat param ?

Thanks
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From infos at madovsky.org  Mon Jan  3 03:04:11 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 2 Jan 2011 19:04:11 -0500
Subject: [Freeswitch-users] nibblebill schedule
References: <F07B572C409042239691B1F3DE08575E@e1705><AANLkTi=W9BV+JnD4K=Oqv_fGSt-SpVM2jYV3AyL7xzO2@mail.gmail.com><DA3EFAD250384C26947B906385BE23D1@e1705>
	<AANLkTinwV_vFXqSLqwTnyXJZopLzdYQ_22Lodm9DqKn5@mail.gmail.com>
Message-ID: <EA92AD2EDF9A41F3991AEEAEE718DEF0@e1705>

Hi Rupa,

I want to understand the logical.
if I understand when the user is under the nobal param 
nibblebill takes the precedence of any dilaplan right ?
if yes so it means that the only solution is to find a way in the extension
of nobal action...

Thanks to correct me if I'm wrong
  ----- Original Message ----- 
  From: Rupa Schomaker 
  To: FreeSWITCH Users Help 
  Sent: Monday, December 27, 2010 5:27 PM
  Subject: Re: [Freeswitch-users] nibblebill schedule


  Look at setting up nibblebill in paused mode, then use the schedule api to take nibblebill out of pause mode after 30s.


  http://wiki.freeswitch.org/wiki/Mod_nibblebill#Pause
  http://wiki.freeswitch.org/wiki/Mod_commands (look for sched_api)


  On Sat, Dec 25, 2010 at 10:22 AM, Madovsky <infos at madovsky.org> wrote:

    ha ok,
    the concept I thought is to let user to call 30s as a trial and
    start niblebill after this time.... does it need to modify the mod_nibblebill source code ?

    Thanks

    F
      ----- Original Message ----- 
      From: Rupa Schomaker 
      To: FreeSWITCH Users Help 
      Sent: Saturday, December 25, 2010 8:38 AM
      Subject: Re: [Freeswitch-users] nibblebill schedule


      I don't see how without change. 


      You have no idea how fast the $$ is being nibbled away by other calls.  So, while you can account for a 30s buffer in the current call using the current rate, the more calls that are up for that account the more "off" the 30s estimate will be.  


      On Fri, Dec 24, 2010 at 11:58 AM, Madovsky <infos at madovsky.org> wrote:

        Is it possible to schedule the transfer to nibblebill nobal_action ?
        I'd like to schedule of 30s before the call is cut and go to nobal_action extension

        thanks



      -- 
      -Rupa



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  -- 
  -Rupa



------------------------------------------------------------------------------


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From rupa at rupa.com  Mon Jan  3 05:18:02 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Sun, 2 Jan 2011 20:18:02 -0600
Subject: [Freeswitch-users] enable_heartbeat_events
In-Reply-To: <DCAC3F2E8D4E4C609CBDD890BE841C81@e1705>
References: <DCAC3F2E8D4E4C609CBDD890BE841C81@e1705>
Message-ID: <AANLkTik5x1FZ8Hexb92Sh7FU3uj-uZR+fVqOr6QDk4zQ@mail.gmail.com>

I'd have to dig through the source to get an answer to this -- perhaps
someone else will give an answer that knows off the top of their head...

On Sun, Jan 2, 2011 at 2:19 PM, Madovsky <infos at madovsky.org> wrote:

>  Rupa,
>
> is enable_heartbeat_events has the same rule as
> global heartbeat param ?
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From rupa at rupa.com  Mon Jan  3 05:18:53 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Sun, 2 Jan 2011 20:18:53 -0600
Subject: [Freeswitch-users] nibblebill schedule
In-Reply-To: <EA92AD2EDF9A41F3991AEEAEE718DEF0@e1705>
References: <F07B572C409042239691B1F3DE08575E@e1705>
	<AANLkTi=W9BV+JnD4K=Oqv_fGSt-SpVM2jYV3AyL7xzO2@mail.gmail.com>
	<DA3EFAD250384C26947B906385BE23D1@e1705>
	<AANLkTinwV_vFXqSLqwTnyXJZopLzdYQ_22Lodm9DqKn5@mail.gmail.com>
	<EA92AD2EDF9A41F3991AEEAEE718DEF0@e1705>
Message-ID: <AANLkTimKm1PKxHVsN5avkeifRLt51k-RqqpRGUFqyyCE@mail.gmail.com>

Right, the call is transfered to the nobal extension so there is no more
processing of the "original" dialplan.  Perhaps set a channel var that you
can query in the nobal extension if necessary?

On Sun, Jan 2, 2011 at 6:04 PM, Madovsky <infos at madovsky.org> wrote:

>  Hi Rupa,
>
> I want to understand the logical.
> if I understand when the user is under the nobal param
> nibblebill takes the precedence of any dilaplan right ?
> if yes so it means that the only solution is to find a way in the extension
> of nobal action...
>
> Thanks to correct me if I'm wrong
>
> ----- Original Message -----
> *From:* Rupa Schomaker <rupa at rupa.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, December 27, 2010 5:27 PM
> *Subject:* Re: [Freeswitch-users] nibblebill schedule
>
> Look at setting up nibblebill in paused mode, then use the schedule api to
> take nibblebill out of pause mode after 30s.
>
> http://wiki.freeswitch.org/wiki/Mod_nibblebill#Pause
>  <http://wiki.freeswitch.org/wiki/Mod_nibblebill#Pause>
> http://wiki.freeswitch.org/wiki/Mod_commands (look for sched_api)
>
> On Sat, Dec 25, 2010 at 10:22 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  ha ok,
>> the concept I thought is to let user to call 30s as a trial and
>> start niblebill after this time.... does it need to modify the
>> mod_nibblebill source code ?
>>
>> Thanks
>>
>> F
>>
>>   ----- Original Message -----
>> *From:* Rupa Schomaker <rupa at rupa.com>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Sent:* Saturday, December 25, 2010 8:38 AM
>> *Subject:* Re: [Freeswitch-users] nibblebill schedule
>>
>> I don't see how without change.
>>
>> You have no idea how fast the $$ is being nibbled away by other calls.
>>  So, while you can account for a 30s buffer in the current call using the
>> current rate, the more calls that are up for that account the more "off" the
>> 30s estimate will be.
>>
>> On Fri, Dec 24, 2010 at 11:58 AM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  Is it possible to schedule the transfer to nibblebill nobal_action ?
>>> I'd like to schedule of 30s before the call is cut and go to nobal_action
>>> extension
>>>
>>> thanks
>>>
>>>
>> --
>> -Rupa
>>
>> ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From brian at freeswitch.org  Mon Jan  3 05:54:35 2011
From: brian at freeswitch.org (Brian West)
Date: Sun, 2 Jan 2011 20:54:35 -0600
Subject: [Freeswitch-users] Logging "dropped calls"
In-Reply-To: <AANLkTikYxrZ35TLUL2KTeN1oJ0smqOOF9N1MJdbcePEy@mail.gmail.com>
References: <AANLkTi=OP-eVAr2QpxEghVN1v-ASFHUgC9n_3MyPRsm1@mail.gmail.com>
	<4D206F95.20602@solomo.de>
	<F0EC2F69-2947-43DC-8904-E3F0BB0B5A78@freeswitch.org>
	<AANLkTikYxrZ35TLUL2KTeN1oJ0smqOOF9N1MJdbcePEy@mail.gmail.com>
Message-ID: <15C8DEDA-1092-4F55-BDA2-4C99B80803C3@freeswitch.org>

Call drops can show up as normal clearing depending on what your providers are doing to either hide or strip the real hangup cause... you can do the same thing with FS if you put a hangup app in your diaplan ... so its all about how its configured.

/b

On Jan 2, 2011, at 1:14 PM, Frank Park wrote:

> Haha..
> Not me! Our servers are in a major colo, doubt they are having a
> network issue. Otherwise, I would know about it.
> But there's been few customer complaints on dropped calls (some AT&T
> indeed).. I usually automatically assume it's their network issue and
> not FS issue, but it would be nice to know the stats on what is
> happening on our side.
> The CDR gives a good info like clearing conditions, but it seems to
> always be NORMAL_CLEARING regardless of dropped calls or not.. I don't
> know if NORMAL_CLEARING only happens when SIP successfully tears down
> both legs or not (anybody to clarify?).. If it doesn't it's not that
> useful in my case.
> 
> I might have to look into ESL option unless anybody else has a better idea.
> 
> Frank



From jeff at jefflenk.com  Mon Jan  3 07:09:31 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sun, 2 Jan 2011 22:09:31 -0600
Subject: [Freeswitch-users] Can't build on Windows 7 64bit C++ Express!?
Message-ID: <snt0-eas12908A012C361DEE1DF0DC7B9070@phx.gbl>

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From u2nsam at gmail.com  Mon Jan  3 07:30:17 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 3 Jan 2011 10:00:17 +0530
Subject: [Freeswitch-users] no ringback tone
Message-ID: <AANLkTinbd06YgZFWue=eQGG9A5JO2a550tJ+bo-yG2yv@mail.gmail.com>

Hi All,

happy new you to you !

using a sangoma card and when dialing a mobile number which is having
ringback tune / caller tune ;
but a plain ring is heard to the user dialing that mobile through the trunk.

I am using below syntax to dial out:-
<action application="bridge"
data="{monitor_early_media_ring_total=3}freetdm/wp1/a/${destination_number}"/>

Any suggestions.

Regards
Sam
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From u2nsam at gmail.com  Mon Jan  3 08:25:24 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 3 Jan 2011 10:55:24 +0530
Subject: [Freeswitch-users] no ringback tone
In-Reply-To: <AANLkTinbd06YgZFWue=eQGG9A5JO2a550tJ+bo-yG2yv@mail.gmail.com>
References: <AANLkTinbd06YgZFWue=eQGG9A5JO2a550tJ+bo-yG2yv@mail.gmail.com>
Message-ID: <AANLkTinREcSsWF27ZYq-WfhvROyuPzJ3SmEJ3F_m-3Yb@mail.gmail.com>

These was the output to fscli :-



EXECUTE sofia/internal/7001 at 192.168.2.190bridge({monitor_early_media_ring_total=3}freetdm/wp1/a/9322273640)
2011-01-03 10:15:25.554494 [DEBUG] switch_ivr_originate.c:1954 variable
string 0 = [monitor_early_media_ring_total=3]
2011-01-03 10:15:25.554494 [DEBUG] mod_freetdm.c:378 Set codec PCMA 20ms
2011-01-03 10:15:25.554494 [DEBUG] mod_freetdm.c:1361 Connect outbound
channel FreeTDM/1:1/9322273640
2011-01-03 10:15:25.554494 [NOTICE] switch_channel.c:784 New Channel
FreeTDM/1:1/9322273640 [42a43181-4124-426a-ab14-6cea045e5c22]
2011-01-03 10:15:25.554494 [DEBUG] mod_freetdm.c:1374
(FreeTDM/1:1/9322273640) State Change CS_NEW -> CS_INIT
2011-01-03 10:15:25.554494 [DEBUG] switch_core_session.c:1083 Send signal
FreeTDM/1:1/9322273640 [BREAK]
2011-01-03 10:15:25.554494 [DEBUG] ftmod_sangoma_isdn.c:926 [s1c1][1:1]
Changed state from DOWN to DIALING
2011-01-03 10:15:25.555440 [DEBUG] ftmod_sangoma_isdn.c:634 [s1c1][1:1]
processing state change to DIALING
2011-01-03 10:15:25.555440 [INFO] ftmod_sangoma_isdn_stack_out.c:61
[s1c1][1:1] Outgoing call: Called No:[9322273640] Calling No:[7001]
2011-01-03 10:15:25.555440 [INFO] ftmod_sangoma_isdn_stack_out.c:74
[s1c1][1:1] Sending SETUP (suId:1 suInstId:67 spInstId:0 dchan:1 ces:0)
2011-01-03 10:15:25.565340 [DEBUG] switch_core_state_machine.c:320
(FreeTDM/1:1/9322273640) Running State Change CS_INIT
2011-01-03 10:15:25.565340 [DEBUG] switch_core_state_machine.c:356
(FreeTDM/1:1/9322273640) State INIT
2011-01-03 10:15:25.565340 [DEBUG] mod_freetdm.c:406
(FreeTDM/1:1/9322273640) State Change CS_INIT -> CS_ROUTING
2011-01-03 10:15:25.565340 [DEBUG] switch_core_session.c:1083 Send signal
FreeTDM/1:1/9322273640 [BREAK]
2011-01-03 10:15:25.565340 [DEBUG] switch_core_state_machine.c:356
(FreeTDM/1:1/9322273640) State INIT going to sleep
2011-01-03 10:15:25.565340 [DEBUG] switch_core_state_machine.c:320
(FreeTDM/1:1/9322273640) Running State Change CS_ROUTING
2011-01-03 10:15:25.565340 [DEBUG] switch_channel.c:1615
(FreeTDM/1:1/9322273640) Callstate Change DOWN -> RINGING
2011-01-03 10:15:25.565340 [DEBUG] switch_core_state_machine.c:359
(FreeTDM/1:1/9322273640) State ROUTING
2011-01-03 10:15:25.565340 [DEBUG] mod_freetdm.c:431 FreeTDM/1:1/9322273640
CHANNEL ROUTING
2011-01-03 10:15:25.565340 [DEBUG] switch_ivr_originate.c:66
(FreeTDM/1:1/9322273640) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2011-01-03 10:15:25.565340 [DEBUG] switch_core_session.c:1083 Send signal
FreeTDM/1:1/9322273640 [BREAK]
2011-01-03 10:15:25.565340 [DEBUG] switch_core_state_machine.c:359
(FreeTDM/1:1/9322273640) State ROUTING going to sleep
2011-01-03 10:15:25.565340 [DEBUG] switch_core_state_machine.c:320
(FreeTDM/1:1/9322273640) Running State Change CS_CONSUME_MEDIA
2011-01-03 10:15:25.566306 [DEBUG] switch_core_state_machine.c:378
(FreeTDM/1:1/9322273640) State CONSUME_MEDIA
2011-01-03 10:15:25.566306 [DEBUG] switch_core_state_machine.c:378
(FreeTDM/1:1/9322273640) State CONSUME_MEDIA going to sleep
2011-01-03 10:15:25.609441 [INFO] ftmod_sangoma_isdn_stack_rcv.c:172
[s1c1][1:1] Received PROCEED (suId:1 suInstId:67 spInstId:67 ces:0)
2011-01-03 10:15:25.609441 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:351
[s1c1][1:1] Processing PROCEED (suId:1 suInstId:67 spInstId:67 ces:0)
2011-01-03 10:15:25.609441 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:380
[s1c1][1:1] Changed state from DIALING to PROCEED
2011-01-03 10:15:25.609441 [DEBUG] ftmod_sangoma_isdn.c:634 [s1c1][1:1]
processing state change to PROCEED
2011-01-03 10:15:25.610382 [DEBUG] mod_freetdm.c:2116 got clear channel sig
[PROCEED]
2011-01-03 10:15:26.837602 [INFO] ftmod_sangoma_isdn_stack_rcv.c:172
[s1c1][1:1] Received ALERT (suId:1 suInstId:67 spInstId:67 ces:0)
2011-01-03 10:15:26.838544 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:351
[s1c1][1:1] Processing ALERT (suId:1 suInstId:67 spInstId:67 ces:0)
2011-01-03 10:15:26.838544 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:385
[s1c1][1:1] Changed state from PROCEED to RINGING
2011-01-03 10:15:26.838544 [DEBUG] ftmod_sangoma_isdn.c:634 [s1c1][1:1]
processing state change to RINGING
2011-01-03 10:15:26.838544 [DEBUG] mod_freetdm.c:2116 got clear channel sig
[RINGING]
2011-01-03 10:15:26.838544 [NOTICE] mod_freetdm.c:2184 Ring-Ready
FreeTDM/1:1/9322273640!
2011-01-03 10:15:26.839557 [NOTICE] mod_sofia.c:2156 Ring-Ready
sofia/internal/7001 at 192.168.2.190!
2011-01-03 10:15:26.839557 [DEBUG] switch_core_session.c:676 Send signal
sofia/internal/7001 at 192.168.2.190 [BREAK]
2011-01-03 10:15:26.839557 [NOTICE] switch_ivr_originate.c:472 Ring Ready
sofia/internal/7001 at 192.168.2.190!
2011-01-03 10:15:26.839557 [DEBUG] sofia.c:4606 Channel sofia/internal/
7001 at 192.168.2.190 entering state [early][180]
2011-01-03 10:15:30.204270 [INFO] ftmod_sangoma_isdn_stack_rcv.c:172
[s1c1][1:1] Received PROGRESS (suId:1 suInstId:67 spInstId:67 ces:0)
2011-01-03 10:15:30.204270 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:351
[s1c1][1:1] Processing PROGRESS (suId:1 suInstId:67 spInstId:67 ces:0)


Regds
Sam

On Mon, Jan 3, 2011 at 10:00 AM, Sam <u2nsam at gmail.com> wrote:

> Hi All,
>
> happy new you to you !
>
> using a sangoma card and when dialing a mobile number which is having
> ringback tune / caller tune ;
> but a plain ring is heard to the user dialing that mobile through the
> trunk.
>
> I am using below syntax to dial out:-
> <action application="bridge"
> data="{monitor_early_media_ring_total=3}freetdm/wp1/a/${destination_number}"/>
>
> Any suggestions.
>
> Regards
> Sam
>
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From infos at madovsky.org  Mon Jan  3 08:58:49 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 3 Jan 2011 00:58:49 -0500
Subject: [Freeswitch-users] nibblebill schedule
References: <F07B572C409042239691B1F3DE08575E@e1705><AANLkTi=W9BV+JnD4K=Oqv_fGSt-SpVM2jYV3AyL7xzO2@mail.gmail.com><DA3EFAD250384C26947B906385BE23D1@e1705><AANLkTinwV_vFXqSLqwTnyXJZopLzdYQ_22Lodm9DqKn5@mail.gmail.com><EA92AD2EDF9A41F3991AEEAEE718DEF0@e1705>
	<AANLkTimKm1PKxHVsN5avkeifRLt51k-RqqpRGUFqyyCE@mail.gmail.com>
Message-ID: <9320B12404AB430087107D5D1881D0A7@e1705>

I tried this :

nibblebill.conf
<param name="nobal_action" value="trial XML features"/>

dialplan feature
        <extension name="trial">
                <condition field="destination_number" expression="^trial$">
                        <action application="set" data="trial_mode=true"/>
                        <anti-action application="set" data="trial_mode=false"/>
                </condition>
                <condition>
                        <action application="transfer" data="${sip_to_uri} XML default"/>
                </condition>
        </extension>

but it makes a wonderful loop....

  ----- Original Message ----- 
  From: Rupa Schomaker 
  To: FreeSWITCH Users Help 
  Sent: Sunday, January 02, 2011 9:18 PM
  Subject: Re: [Freeswitch-users] nibblebill schedule


  Right, the call is transfered to the nobal extension so there is no more processing of the "original" dialplan.  Perhaps set a channel var that you can query in the nobal extension if necessary?


  On Sun, Jan 2, 2011 at 6:04 PM, Madovsky <infos at madovsky.org> wrote:

    Hi Rupa,

    I want to understand the logical.
    if I understand when the user is under the nobal param 
    nibblebill takes the precedence of any dilaplan right ?
    if yes so it means that the only solution is to find a way in the extension
    of nobal action...

    Thanks to correct me if I'm wrong
      ----- Original Message ----- 
      From: Rupa Schomaker 
      To: FreeSWITCH Users Help 
      Sent: Monday, December 27, 2010 5:27 PM
      Subject: Re: [Freeswitch-users] nibblebill schedule


      Look at setting up nibblebill in paused mode, then use the schedule api to take nibblebill out of pause mode after 30s. 


      http://wiki.freeswitch.org/wiki/Mod_nibblebill#Pause
      http://wiki.freeswitch.org/wiki/Mod_commands (look for sched_api)


      On Sat, Dec 25, 2010 at 10:22 AM, Madovsky <infos at madovsky.org> wrote:

        ha ok,
        the concept I thought is to let user to call 30s as a trial and
        start niblebill after this time.... does it need to modify the mod_nibblebill source code ?

        Thanks

        F
          ----- Original Message ----- 
          From: Rupa Schomaker 
          To: FreeSWITCH Users Help 
          Sent: Saturday, December 25, 2010 8:38 AM
          Subject: Re: [Freeswitch-users] nibblebill schedule


          I don't see how without change. 


          You have no idea how fast the $$ is being nibbled away by other calls.  So, while you can account for a 30s buffer in the current call using the current rate, the more calls that are up for that account the more "off" the 30s estimate will be.  


          On Fri, Dec 24, 2010 at 11:58 AM, Madovsky <infos at madovsky.org> wrote:

            Is it possible to schedule the transfer to nibblebill nobal_action ?
            I'd like to schedule of 30s before the call is cut and go to nobal_action extension

            thanks



          -- 
          -Rupa



----------------------------------------------------------------------


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      -- 
      -Rupa



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From infos at madovsky.org  Mon Jan  3 09:32:48 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 3 Jan 2011 01:32:48 -0500
Subject: [Freeswitch-users] nibblebill schedule
Message-ID: <AEE685ED71404CDB84785B5B5360D505@e1705>

it seems also that the transfer action in nibblebill conf

<param name="nobal_action" value="trial XML features"/>

is more a execute_extension that a transfer application ?
  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 03, 2011 12:58 AM
  Subject: Re: [Freeswitch-users] nibblebill schedule


  I tried this :

  nibblebill.conf
  <param name="nobal_action" value="trial XML features"/>

  dialplan feature
          <extension name="trial">
                  <condition field="destination_number" expression="^trial$">
                          <action application="set" data="trial_mode=true"/>
                          <anti-action application="set" data="trial_mode=false"/>
                  </condition>
                  <condition>
                          <action application="transfer" data="${sip_to_uri} XML default"/>
                  </condition>
          </extension>

  but it makes a wonderful loop....

    ----- Original Message ----- 
    From: Rupa Schomaker 
    To: FreeSWITCH Users Help 
    Sent: Sunday, January 02, 2011 9:18 PM
    Subject: Re: [Freeswitch-users] nibblebill schedule


    Right, the call is transfered to the nobal extension so there is no more processing of the "original" dialplan.  Perhaps set a channel var that you can query in the nobal extension if necessary?


    On Sun, Jan 2, 2011 at 6:04 PM, Madovsky <infos at madovsky.org> wrote:

      Hi Rupa,

      I want to understand the logical.
      if I understand when the user is under the nobal param 
      nibblebill takes the precedence of any dilaplan right ?
      if yes so it means that the only solution is to find a way in the extension
      of nobal action...

      Thanks to correct me if I'm wrong
        ----- Original Message ----- 
        From: Rupa Schomaker 
        To: FreeSWITCH Users Help 
        Sent: Monday, December 27, 2010 5:27 PM
        Subject: Re: [Freeswitch-users] nibblebill schedule


        Look at setting up nibblebill in paused mode, then use the schedule api to take nibblebill out of pause mode after 30s. 


        http://wiki.freeswitch.org/wiki/Mod_nibblebill#Pause
        http://wiki.freeswitch.org/wiki/Mod_commands (look for sched_api)


        On Sat, Dec 25, 2010 at 10:22 AM, Madovsky <infos at madovsky.org> wrote:

          ha ok,
          the concept I thought is to let user to call 30s as a trial and
          start niblebill after this time.... does it need to modify the mod_nibblebill source code ?

          Thanks

          F
            ----- Original Message ----- 
            From: Rupa Schomaker 
            To: FreeSWITCH Users Help 
            Sent: Saturday, December 25, 2010 8:38 AM
            Subject: Re: [Freeswitch-users] nibblebill schedule


            I don't see how without change. 


            You have no idea how fast the $$ is being nibbled away by other calls.  So, while you can account for a 30s buffer in the current call using the current rate, the more calls that are up for that account the more "off" the 30s estimate will be.  


            On Fri, Dec 24, 2010 at 11:58 AM, Madovsky <infos at madovsky.org> wrote:

              Is it possible to schedule the transfer to nibblebill nobal_action ?
              I'd like to schedule of 30s before the call is cut and go to nobal_action extension

              thanks



            -- 
            -Rupa



--------------------------------------------------------------------


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        -- 
        -Rupa



------------------------------------------------------------------------


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From grsingh750 at gmail.com  Mon Jan  3 12:25:19 2011
From: grsingh750 at gmail.com (guru singh)
Date: Mon, 3 Jan 2011 14:55:19 +0530
Subject: [Freeswitch-users] Sangoma Cards or FXS gateway for high port
	density setup (~ 160 fxs)
Message-ID: <AANLkTinQ5VUf7nt7XY7ZBPELm+Qxg9MQFgRzwCLY1es2@mail.gmail.com>

Hi,

Happy New Year!

What would you recommend for a solution that requires ~160 fxs ports?
Should I get a Sangoma A200, in fact 7 of them and put them on to my
server or look for other products like Audiocodes/Patton fxs gateways
etc?
If I choose Sangoma cards, then I don't think I can set up a HA
scenario since, all fxs ports will be on the FS box itself.
But from what I've read people have horrible things to say about most
gateway devices. They are prone to failure and the support is also
very lax.
So what good would be HA anyway, if one of these gateways fails and
knocks 32 fxs ports off!
Also I see that Sangoma is actively involved with the FreeSwitch
project and digging through the mailing lists, I could see great
support as well.
Is it advisable to have 6-7 cards with 12 fxs modules each on my FS server?

Any suggestions/pointers/heads-up would be really appreciated.

Thanks
guru


From u2nsam at gmail.com  Mon Jan  3 13:37:11 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 3 Jan 2011 16:07:11 +0530
Subject: [Freeswitch-users] multi company
Message-ID: <AANLkTikjjhHMMxKmsEGpDYxXBpozjNycdimV6zZFmhL7@mail.gmail.com>

Hi,

Was using multi company setup,

it gave an error while using below syntax

<domain name="xyz">

</domain>

Cannot Initialize [[error near line 2868]: unexpected closing tag </domain>]

And when i remove , it dont gives an error .
the file is in directory/xyz.xml


same happens on freeswitch.xml file

when i remove the syntax

<section name="directory" description="User Directory">
    <X-PRE-PROCESS cmd="include" data="directory/*.xml"/>
  </section>

it works when removed the section other wise give error as unexpected
closing tag </section>]

Regds
Sam
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From lists at infosecurity.ch  Mon Jan  3 13:59:39 2011
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Mon, 03 Jan 2011 11:59:39 +0100
Subject: [Freeswitch-users] Moving RTP relay to another FS BOX?
	(FS+MediaProxy?)
In-Reply-To: <4D1C6640.3050209@infosecurity.ch>
References: <4D1C46AA.3060702@infosecurity.ch>
	<AANLkTime6GQtbQS3x2GC3yQh_=fGnMafYg12FWwUmgOe@mail.gmail.com>
	<4D1C6640.3050209@infosecurity.ch>
Message-ID: <4D21AC1B.1060808@infosecurity.ch>

Hi all,

to solve my problem of moving deterministically (based on source ip) the
RTP flow to another box i was wondering whether it's possible to connect
FS + MediaProxy .

The typical infrastructure requirement i have (1 SIP server + multiple
RTP relay) is typically done with:
- 1 OpenSIPS/Kamailio + Multiple MediaProxy

Well, i am wondering whether it could be done with:
- 1 FreeSWITCH  + Multiple Mediaproxy

I read there about the mediaproxy-ng
http://mediaproxy-ng.org/wiki/InstallationGuide and it has a protocol
dispatcher.

Does anyone have an idea on how to be able to have FS dispatching calls
for RTP proxy to multiple mediaproxy?

Fabio


On 30/12/10 12.00, Fabio Pietrosanti (naif) wrote:
> Hi Steven,
>
> if i understand correctly in your scenario "C2" is SIP registered to FS2.
>
> While i would like to have C1 and C2 both SIP registered to FS1, but if
> they match certain parameters (that's application logic), i want their
> RTP flow to goes proxed trough FS2.
>
> FS1 is in Europe.
> FS2 is in India.
> C1 and C2 are in India.
>
> C1 and C2 are connected to FS1 in Europe for SIP.
>
> I would like to have the flow as follow:
> SIP Flow: C1 -> FS1 -> C2
> RTP flow: C1 -> FS2 -> C2
>
> Obviously FS1 need in some way to be able to "instruct" C1 and C2 to go
> trough FS2, and FS2 to handle RTP relay.
>
> From my basic feeling i would need to move to a Kamailio+RTPProxy
> solutions, but if FS could have the flexibility to implement such
> solution it would be *much better* as i am already FS based.
>
> Also if some custom development is required, i would be happy to sponsor
> some bounty about it.
>
> Fabio



From david.ponzone at ipeva.fr  Mon Jan  3 14:12:08 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 3 Jan 2011 12:12:08 +0100
Subject: [Freeswitch-users] Moving RTP relay to another FS BOX?
	(FS+MediaProxy?)
In-Reply-To: <4D21AC1B.1060808@infosecurity.ch>
References: <4D1C46AA.3060702@infosecurity.ch>
	<AANLkTime6GQtbQS3x2GC3yQh_=fGnMafYg12FWwUmgOe@mail.gmail.com>
	<4D1C6640.3050209@infosecurity.ch>
	<4D21AC1B.1060808@infosecurity.ch>
Message-ID: <2B001098-F7F6-4A15-AF1E-D7075D0ECBBB@ipeva.fr>

You would need to implement the Mediaproxy module for FreeSWITCH.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 03/01/2011 ? 11:59, Fabio Pietrosanti (naif) a ?crit :

> Hi all,
> 
> to solve my problem of moving deterministically (based on source ip) the
> RTP flow to another box i was wondering whether it's possible to connect
> FS + MediaProxy .
> 
> The typical infrastructure requirement i have (1 SIP server + multiple
> RTP relay) is typically done with:
> - 1 OpenSIPS/Kamailio + Multiple MediaProxy
> 
> Well, i am wondering whether it could be done with:
> - 1 FreeSWITCH  + Multiple Mediaproxy
> 
> I read there about the mediaproxy-ng
> http://mediaproxy-ng.org/wiki/InstallationGuide and it has a protocol
> dispatcher.
> 
> Does anyone have an idea on how to be able to have FS dispatching calls
> for RTP proxy to multiple mediaproxy?
> 
> Fabio
> 
> 
> On 30/12/10 12.00, Fabio Pietrosanti (naif) wrote:
>> Hi Steven,
>> 
>> if i understand correctly in your scenario "C2" is SIP registered to FS2.
>> 
>> While i would like to have C1 and C2 both SIP registered to FS1, but if
>> they match certain parameters (that's application logic), i want their
>> RTP flow to goes proxed trough FS2.
>> 
>> FS1 is in Europe.
>> FS2 is in India.
>> C1 and C2 are in India.
>> 
>> C1 and C2 are connected to FS1 in Europe for SIP.
>> 
>> I would like to have the flow as follow:
>> SIP Flow: C1 -> FS1 -> C2
>> RTP flow: C1 -> FS2 -> C2
>> 
>> Obviously FS1 need in some way to be able to "instruct" C1 and C2 to go
>> trough FS2, and FS2 to handle RTP relay.
>> 
>> From my basic feeling i would need to move to a Kamailio+RTPProxy
>> solutions, but if FS could have the flexibility to implement such
>> solution it would be *much better* as i am already FS based.
>> 
>> Also if some custom development is required, i would be happy to sponsor
>> some bounty about it.
>> 
>> Fabio
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From lists at infosecurity.ch  Mon Jan  3 14:51:01 2011
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Mon, 03 Jan 2011 12:51:01 +0100
Subject: [Freeswitch-users] Moving RTP relay to another FS BOX?
	(FS+MediaProxy?)
In-Reply-To: <2B001098-F7F6-4A15-AF1E-D7075D0ECBBB@ipeva.fr>
References: <4D1C46AA.3060702@infosecurity.ch>	<AANLkTime6GQtbQS3x2GC3yQh_=fGnMafYg12FWwUmgOe@mail.gmail.com>	<4D1C6640.3050209@infosecurity.ch>	<4D21AC1B.1060808@infosecurity.ch>
	<2B001098-F7F6-4A15-AF1E-D7075D0ECBBB@ipeva.fr>
Message-ID: <4D21B825.8010904@infosecurity.ch>

I find no documentation on OpenSIPS to Media-Dispatcher communication
protocol.

All i just see is from logical point of view is that:
- OpenSIPS talk to MediaDispatcher trough a UnixSocket
- then Media-Dispatcher send a request to Media-Relay via TLS
- Media-Relay answer to Media Dispatcher the allocated IP/UDP ports for
RTP relay
- Media Dispatcher provide the IP/UDP port allocated to OpenSIPS
- OpenSIPS mangle SDP attribute to tell the SIP clients where to send RTP

Now, i am not finding details on OpenSIPS<->MediaDispatcher protocol,
but maybe it's something incredibly easy so that few lines of python
could make it working.

Any idea?

Fabio

On 03/01/11 12.12, David Ponzone wrote:
> You would need to implement the Mediaproxy module for FreeSWITCH.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
> ? l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion non autoris?e est interdite. Tout message ?lectronique est
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous
> n'?tes pas destinataire de ce message, merci de le d?truire
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 03/01/2011 ? 11:59, Fabio Pietrosanti (naif) a ?crit :
>
>> Hi all,
>>
>> to solve my problem of moving deterministically (based on source ip) the
>> RTP flow to another box i was wondering whether it's possible to connect
>> FS + MediaProxy .
>>
>> The typical infrastructure requirement i have (1 SIP server + multiple
>> RTP relay) is typically done with:
>> - 1 OpenSIPS/Kamailio + Multiple MediaProxy
>>
>> Well, i am wondering whether it could be done with:
>> - 1 FreeSWITCH  + Multiple Mediaproxy
>>
>> I read there about the mediaproxy-ng
>> http://mediaproxy-ng.org/wiki/InstallationGuide and it has a protocol
>> dispatcher.
>>
>> Does anyone have an idea on how to be able to have FS dispatching calls
>> for RTP proxy to multiple mediaproxy?
>>
>> Fabio
>>
>>
>> On 30/12/10 12.00, Fabio Pietrosanti (naif) wrote:
>>> Hi Steven,
>>>
>>> if i understand correctly in your scenario "C2" is SIP registered to
>>> FS2.
>>>
>>> While i would like to have C1 and C2 both SIP registered to FS1, but if
>>> they match certain parameters (that's application logic), i want their
>>> RTP flow to goes proxed trough FS2.
>>>
>>> FS1 is in Europe.
>>> FS2 is in India.
>>> C1 and C2 are in India.
>>>
>>> C1 and C2 are connected to FS1 in Europe for SIP.
>>>
>>> I would like to have the flow as follow:
>>> SIP Flow: C1 -> FS1 -> C2
>>> RTP flow: C1 -> FS2 -> C2
>>>
>>> Obviously FS1 need in some way to be able to "instruct" C1 and C2 to go
>>> trough FS2, and FS2 to handle RTP relay.
>>>
>>> From my basic feeling i would need to move to a Kamailio+RTPProxy
>>> solutions, but if FS could have the flexibility to implement such
>>> solution it would be *much better* as i am already FS based.
>>>
>>> Also if some custom development is required, i would be happy to sponsor
>>> some bounty about it.
>>>
>>> Fabio
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From erik.dekkers at wvds.nl  Mon Jan  3 14:52:50 2011
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Mon, 3 Jan 2011 12:52:50 +0100
Subject: [Freeswitch-users] Moving RTP relay to another FS
	BOX?	(FS+MediaProxy?)
In-Reply-To: <4D21AC1B.1060808@infosecurity.ch>
References: <4D1C46AA.3060702@infosecurity.ch>
	<AANLkTime6GQtbQS3x2GC3yQh_=fGnMafYg12FWwUmgOe@mail.gmail.com>
	<4D1C6640.3050209@infosecurity.ch> <4D21AC1B.1060808@infosecurity.ch>
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE077E4DA9DA@FS1.wvds.local>

If you read the mediaproxy wiki a little bit better you should see that mediaproxy is made for opensips, not freeswitch.

-----Oorspronkelijk bericht-----
Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Fabio Pietrosanti (naif)
Verzonden: maandag 3 januari 2011 12:00
Aan: freeswitch-users at lists.freeswitch.org
Onderwerp: Re: [Freeswitch-users] Moving RTP relay to another FS BOX? (FS+MediaProxy?)

Hi all,

to solve my problem of moving deterministically (based on source ip) the RTP flow to another box i was wondering whether it's possible to connect FS + MediaProxy .

The typical infrastructure requirement i have (1 SIP server + multiple RTP relay) is typically done with:
- 1 OpenSIPS/Kamailio + Multiple MediaProxy

Well, i am wondering whether it could be done with:
- 1 FreeSWITCH  + Multiple Mediaproxy

I read there about the mediaproxy-ng
http://mediaproxy-ng.org/wiki/InstallationGuide and it has a protocol dispatcher.

Does anyone have an idea on how to be able to have FS dispatching calls for RTP proxy to multiple mediaproxy?

Fabio


On 30/12/10 12.00, Fabio Pietrosanti (naif) wrote:
> Hi Steven,
>
> if i understand correctly in your scenario "C2" is SIP registered to FS2.
>
> While i would like to have C1 and C2 both SIP registered to FS1, but 
> if they match certain parameters (that's application logic), i want 
> their RTP flow to goes proxed trough FS2.
>
> FS1 is in Europe.
> FS2 is in India.
> C1 and C2 are in India.
>
> C1 and C2 are connected to FS1 in Europe for SIP.
>
> I would like to have the flow as follow:
> SIP Flow: C1 -> FS1 -> C2
> RTP flow: C1 -> FS2 -> C2
>
> Obviously FS1 need in some way to be able to "instruct" C1 and C2 to 
> go trough FS2, and FS2 to handle RTP relay.
>
> From my basic feeling i would need to move to a Kamailio+RTPProxy 
> solutions, but if FS could have the flexibility to implement such 
> solution it would be *much better* as i am already FS based.
>
> Also if some custom development is required, i would be happy to 
> sponsor some bounty about it.
>
> Fabio


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From lists at infosecurity.ch  Mon Jan  3 15:24:26 2011
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Mon, 03 Jan 2011 13:24:26 +0100
Subject: [Freeswitch-users] Moving RTP relay to another FS BOX? (What's
 about REDIRECT)?
In-Reply-To: <4D1C6640.3050209@infosecurity.ch>
References: <4D1C46AA.3060702@infosecurity.ch>
	<AANLkTime6GQtbQS3x2GC3yQh_=fGnMafYg12FWwUmgOe@mail.gmail.com>
	<4D1C6640.3050209@infosecurity.ch>
Message-ID: <4D21BFFA.9020704@infosecurity.ch>

Another possible proposal to move RTP relay processing to another
FreeSWITCH could be as follow, please tell me if it sounds reasonable
(it sounds like a very dirty hack).
Consider that i just need to do RTP relay between 2 users (no
transcoding, no advanced features) on boxes differents from the FS
acting as SIP registration server.

FS-UK is in Europe.
FS-IN is in India.
C1 and C2 are in India.

C1 and C2 are connected to FS-UK in Europe for SIP.

I would like to have the flow as follow:
SIP Flow: C1 -> FS-UK -> C2
RTP flow: C1 -> FS-IN -> C2

A possible approach is to use the REDIRECT feature of FS.

FS-UK has bypass_media=true  & proxy_media=false
FS-IN has bypass_media=false & proxy_media=true

STEP1) C1 call C2 and FS-UK send a REDIRECT C2 at FS-IN

C1                             FS-UK

INVITE ------------------------------->
           <------------------------------ 100 Trying
           <------------------------------ 302 Moved Temporary C2 at FS2
ACK    ------------------------------->

<action application="redirect" data="sip:C2 at FS-IN"/>


STEP 2) C1 call C2 on C2 at FS-IN in India because of the redirect

C1                             FS-IN

INVITE C2 at FS-IN--------------------------------------------------------------------------------->

           <------------------------------ 100 Trying

STEP 3) FS-IN make a call back to C2 at FS-UK back to FS-UK

Now FS-IN make a bridge for C2 back on FS-UK (where C2 is SIP registered):

<action application="bridge" data="C2 at FS-UK"/>

FS-IN                             FS-UK

INVITE ------------------------------->
           <------------------------------ 100 Trying

STEP 4) FS-UK (which has bypass_media) let the C2 ring and answer the
call and provide the IP address of the leg on FS-IN

FS-UK                             C2

INVITE ------------------------------->
           <------------------------------ xxx ring, answer, etc

200 OK -------------------------------> For RTP it provide the IP address of FS-IN that has proxy_media=true


STEP 5) Now C1 and C2 are sending their RTP flow to FS-IN in india due to it's proxy_media SDP rewriting


I don't know if it was clear.

To summarize:

>From SIP point of view
C1 -> FS-UK (redirect) -> FS-IN (bridge back to FS-UK) -> FS-UK -> C2

>From RTP point of view
C1 -> FS-IN (proxy_media) -> C2

I just want to move certain users to their near media relay and install
one FS box for each continent so that my users will use the media-relay
near to them.

Does it seems feasible?

Fabio

On 30/12/10 12.00, Fabio Pietrosanti (naif) wrote:
> Hi Steven,
>
> if i understand correctly in your scenario "C2" is SIP registered to FS2.
>
> While i would like to have C1 and C2 both SIP registered to FS1, but if
> they match certain parameters (that's application logic), i want their
> RTP flow to goes proxed trough FS2.
>
> FS1 is in Europe.
> FS2 is in India.
> C1 and C2 are in India.
>
> C1 and C2 are connected to FS1 in Europe for SIP.
>
> I would like to have the flow as follow:
> SIP Flow: C1 -> FS1 -> C2
> RTP flow: C1 -> FS2 -> C2
>
> Obviously FS1 need in some way to be able to "instruct" C1 and C2 to go
> trough FS2, and FS2 to handle RTP relay.
>
> From my basic feeling i would need to move to a Kamailio+RTPProxy
> solutions, but if FS could have the flexibility to implement such
> solution it would be *much better* as i am already FS based.
>
> Also if some custom development is required, i would be happy to sponsor
> some bounty about it.
>
> Fabio

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From Nabble at slickdeals.endjunk.com  Mon Jan  3 17:33:17 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 3 Jan 2011 06:33:17 -0800 (PST)
Subject: [Freeswitch-users] Regarding mod_java application in freeswitch
In-Reply-To: <1294039064455-5880529.post@n2.nabble.com>
References: <1294039064455-5880529.post@n2.nabble.com>
Message-ID: <1294065197457-5880847.post@n2.nabble.com>



kapil.rastogi wrote:
> Can u plz send me a sample code of java with its configuration detail.
Here is FS  http://wiki.freeswitch.org/wiki/Mod_java wiki  w.r.t. mod_java.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Regarding-mod-java-application-in-freeswitch-tp5880529p5880847.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From tim at compnetwork.net  Mon Jan  3 17:39:47 2011
From: tim at compnetwork.net (Tim King)
Date: Mon, 3 Jan 2011 09:39:47 -0500
Subject: [Freeswitch-users] ACL Problem
Message-ID: <AANLkTimi0HhWu0cuQXdsZm1kCCx4t-OXW0vSrsoW3==Q@mail.gmail.com>

*I am getting this error sending calls from my Kamailio box on
192.168.0.250.*

2011-01-03 14:22:49.930896 [WARNING] sofia.c:6369 IP 192.168.0.250 Rejected
by acl "domains"

*Here is the contents of my ACL:*

<configuration name="acl.conf" description="Network Lists">
  <network-lists>

    <list name="lan" default="allow">
      <node type="allow" cidr="192.168.0.250/32"/>
    </list>

    <list name="domains" default="deny">
        <node type="allow" domain="$${domain}"/>
        <node type="allow" domain="192.168.0.250/32"/>
    </list>

  </network-lists>
</configuration>

*I do not understand why calls from this machine are being rejected.
Kamailio and Freeswitch are on the same machine sharing the same IP. Any
ideas what I am missing?*
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From brian at freeswitch.org  Mon Jan  3 18:03:51 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 Jan 2011 09:03:51 -0600
Subject: [Freeswitch-users] ACL Problem
In-Reply-To: <AANLkTimi0HhWu0cuQXdsZm1kCCx4t-OXW0vSrsoW3==Q@mail.gmail.com>
References: <AANLkTimi0HhWu0cuQXdsZm1kCCx4t-OXW0vSrsoW3==Q@mail.gmail.com>
Message-ID: <FDE9627A-C5E6-4279-BE52-2FF00AC507D6@freeswitch.org>

Because you need to use cidr= and NOT domain=

/b

On Jan 3, 2011, at 8:39 AM, Tim King wrote:

> I am getting this error sending calls from my Kamailio box on 192.168.0.250.
> 
> 2011-01-03 14:22:49.930896 [WARNING] sofia.c:6369 IP 192.168.0.250 Rejected by acl "domains"
> 
> Here is the contents of my ACL:
> 
> <configuration name="acl.conf" description="Network Lists">
>   <network-lists>
> 
>     <list name="lan" default="allow">
>       <node type="allow" cidr="192.168.0.250/32"/>
>     </list>
> 
>     <list name="domains" default="deny">
>         <node type="allow" domain="$${domain}"/>
>         <node type="allow" domain="192.168.0.250/32"/>
>     </list>
> 
>   </network-lists>
> </configuration>

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From brian at freeswitch.org  Mon Jan  3 18:05:36 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 Jan 2011 09:05:36 -0600
Subject: [Freeswitch-users] increase Max session
In-Reply-To: <AANLkTik9AiZ0GpmtKENmO_Y15rvxyDqm+Ga7Ryyt8vML@mail.gmail.com>
References: <AANLkTik=Fa3ZAX5XP_ubZ7ekrx8M8A7D5J0L=fqepEUg@mail.gmail.com>
	<AANLkTimZxTYeGGNnSKF2x07wS6mBhU5kyB6xbeTKcAmc@mail.gmail.com>
	<AANLkTin+uvhAM58ODfz=6jcJsuwDqcNzysEpLO7yXVfc@mail.gmail.com>
	<8825DCE39E834012ACD4A4978DFA9AB7@e1705>
	<AANLkTimTQ2mjJ+mGQ+kozTbgfE0qtgR-j32sq_oxYNtd@mail.gmail.com>
	<AANLkTi=YR=LxqLy+PzjO_NTiRf+-cAVQ0D=yguZkUw_P@mail.gmail.com>
	<AANLkTint0ypRO68sRdeASqy5s4OC38xK8SPeMXigm0_9@mail.gmail.com>
	<AANLkTi=2C6vOJiHWK8r2EdfdjBhca1_tYUXAZ1_dCpK0@mail.gmail.com>
	<AANLkTik9AiZ0GpmtKENmO_Y15rvxyDqm+Ga7Ryyt8vML@mail.gmail.com>
Message-ID: <41DEB133-E2DD-4DC5-B6DB-320CD2C6BCBA@freeswitch.org>

Your container doesn't have enough resources look at the UB and see what is failing and increase accordingly.

/b

On Jan 1, 2011, at 9:39 PM, dome at tel.co.th wrote:

> 64 bit over openvz
> 
> 2011/1/2 Aloysius Lloyd <lloyd.aloysius at sunteltech.ca>:
>> 32 bit or 64 bit ?
>> 
>> 
>> On Sat, Jan 1, 2011 at 10:12 PM, dome at tel.co.th <dome at tel.co.th> wrote:
>>> 
>>> 2011/1/2 Aloysius Lloyd <lloyd.aloysius at gmail.com>:
>>>> make sure the switch.conf.xml have the right value. Did you rotate the
>>>> log
>>>> file between 22 min and 37 min.
>>> Yes
>>>> try the cli command fsctl max_session 5000 see what happen in couple of
>>>> hours.
>>> i try fsctl max_session 5000
>>> after that FS switch back to 921 again
>>> 
>>>> Thanks
>>>> Lloyd
>>> 

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From xyangni at gmail.com  Mon Jan  3 18:26:01 2011
From: xyangni at gmail.com (xuyan yang)
Date: Mon, 3 Jan 2011 15:26:01 +0000
Subject: [Freeswitch-users] Attack using 5843 and music account?
Message-ID: <AANLkTinNmW4OtL275UWzRoZof0U_YAq3vhfwNkJo3oOG@mail.gmail.com>

Dear all,

Recently,  my FS server are often slowed down at midnight, and system logged
a lot of these lines below:

2011-01-03 04:13:07.494973 [WARNING] sofia_reg.c:1203 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [5843 at 90.192.85.12] from ip
184.106.178.189

2011-01-03 04:11:41.344034 [WARNING] sofia_reg.c:1203 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [music at 90.192.85.12] from ip
184.106.178.189
2011-01-03 04:11:41.503079 [WARNING] sofia_reg.c:1203 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [music at 90.192.85.12] from ip
184.106.178.189
2011-01-03 04:11:41.671564 [WARNING] sofia_reg.c:1203 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [music at 90.192.85.12] from ip
184.106.178.189
2011-01-03 04:11:41.828182 [WARNING] sofia_reg.c:1203 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [music at 90.192.85.12] from ip
184.106.178.189
2011-01-03 04:11:41.998964 [WARNING] sofia_reg.c:1203 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [music at 90.192.85.12] from ip
184.106.178.189
2011-01-03 04:11:42.145093 [WARNING] sofia_reg.c:1203 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [music at 90.192.85.12] from ip
184.106.178.189
2011-01-03 04:11:42.291273 [WARNING] sofia_reg.c:1203 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [music at 90.192.85.12] from ip
184.106.178.189
2011-01-03 04:11:42.448811 [WARNING] sofia_reg.c:1203 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [music at 90.192.85.12] from ip
184.106.178.189
2011-01-03 04:11:42.605709 [WARNING] sofia_reg.c:1203 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [music at 90.192.85.12] from ip
184.106.178.189

I installed fail2ban, but it does not seem to work. After reading these
lines, I found this to be a successful REGISTER instead of a failure.
But I do not have 5843 or music in my directory, and myself can not login to
music account, it generate the following error log:

2011-01-03 15:19:32.360152 [WARNING] sofia_reg.c:1161 SIP auth failure
(REGISTER) on sofia profile 'internal' for [music at 192.168.0.3] from ip
192.168.0.6

So, how can this hacker successfully registered music account and avoid to
be baned? it is strange.

Thanks.
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From lloyd.aloysius at gmail.com  Mon Jan  3 18:28:16 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 3 Jan 2011 10:28:16 -0500
Subject: [Freeswitch-users] voicemail operator-extension -
	Multi-tenantEnvironment
In-Reply-To: <AANLkTinWG0GfDSn-YZ0bHrf_=sd5uuqEehy+k=Xz+8ZN@mail.gmail.com>
References: <AANLkTimm99xdzt2cW6zTpFZY+FtiRCaHCQSm=Rv__o4G@mail.gmail.com>
	<614459D4C10248C8B913E4422CFC7C35@e1705>
	<D330BE13-B0A1-4D76-9DFA-B80B7F3534D8@gmail.com>
	<AANLkTinATfPEbDkzfZSyXHgVYM0mjtsgvzk28xYBotXt@mail.gmail.com>
	<AANLkTik=afD7Zjsb_jPw1aNSXsxLCtHeshiVbK-aHTbA@mail.gmail.com>
	<AANLkTikfETbZCi080u1uuLQj-VkezQf_2+=eWBUv5mZ+@mail.gmail.com>
	<AANLkTimbpDJ3QrkfzvVygtSA1s-KmpV_f8TKYU4RLgf7@mail.gmail.com>
	<AANLkTin5nFvEUwUmNj9kf1cHx7wTgcNKsLWKF8D64iw3@mail.gmail.com>
	<AANLkTikAsVVG=TBEcvOFVVGvhDTrxxW6CRz+oLZ6vHcm@mail.gmail.com>
	<AANLkTinWG0GfDSn-YZ0bHrf_=sd5uuqEehy+k=Xz+8ZN@mail.gmail.com>
Message-ID: <AANLkTikzK8teY7F3e+200fishjy_RUYWSU5oHix3d7XG@mail.gmail.com>

I solved the problem using the following dial plan

 <action application="voicemail" data="default ${domain_name}
${dialed_extension}"/>

But loopback/app  ... behave differently  and not carry the correct
${domain_name} value and it is default to the default domain name.

Thanks
Lloyd

On Sat, Jan 1, 2011 at 8:56 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> There is a mistake in the last email . Please ignore.
>
>  Dialplan sending calls to voicemail
>
> <action application="bridge" data="loopback/app=voicemail:default
> ${domain_name} ${dialed_extension}"/>
>
> voicemail.conf.xml
>
>  <param name="operator-extension" value="operator XML features"/>
>  <param name="operator-key" value="9"/>
>
> features.xml
>
>     <extension name="operator">
>       <condition field="destination_number" expression="^(operator|0)$">
>         <action application="set" data="transfer_ringback=$${hold_music}"/>
>  <action application="transfer" data="operator XML ${domain_name}"/>
>       </condition>
>     </extension>
>
>
> Domain Name - compaya.com
>
> pressing 9 .... voice mail operator extension transfer to features context.
> But  ${domain_name} lost the value compaya.com , but now the ${
> domain_name} have the IP address.
>
> How to get the ${domain_name} value in features context?
>
>
>
> Thanks and regards,
> Lloyd
>
>
> On Sat, Jan 1, 2011 at 8:51 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> Dialplan sending calls to voicemail
>>
>> <action application="bridge" data="loopback/app=voicemail:default
>> ${domain_name} ${dialed_extension}"/>
>>
>> voicemail.conf.xml
>>
>>  <param name="operator-extension" value="operator XML features"/>
>>  <param name="operator-key" value="9"/>
>>
>> features.xml
>>
>>     <extension name="operator">
>>       <condition field="destination_number" expression="^(operator|0)$">
>>         <action application="set"
>> data="transfer_ringback=$${hold_music}"/>
>> <action application="transfer" data="1000 XML features"/>
>>          <action application="transfer" data="operator XML
>> ${domain_name}"/>
>>       </condition>
>>     </extension>
>>
>>
>>  Domain Name - compaya.com
>>
>> pressing 9 .... voice mail operator extension transfer to features
>> context. But  ${domain_name} lost the value compaya.com , but now the ${
>> domain_name} have the IP address.
>>
>> How to get the ${domain_name} value in features context?
>>
>>
>>
>> Thanks and regards,
>> Lloyd
>>
>>
>> On Tue, Dec 28, 2010 at 5:44 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com
>> > wrote:
>>
>>> When I press 9 the call get transfered to the default context. Then I try
>>> get the ${domain_name} that is giving the default domain_name. I could not
>>> find a way to get the correct voicemal domain_name from the from default
>>> context.
>>>
>>> xml_curl .... right now I am using xml_curl. All users defined in mysql,
>>>  then I use a php script for the user informations.
>>>
>>> How to use xml_curl for voicemail ? please let me know if there any
>>> help/docs on this.
>>>
>>> Thanks
>>> LLoyd
>>>
>>>
>>> On Tue, Dec 28, 2010 at 4:29 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>>
>>>>
>>>> On Tue, Dec 28, 2010 at 9:57 AM, Aloysius Lloyd <
>>>> lloyd.aloysius at gmail.com> wrote:
>>>>
>>>>> Michael,
>>>>>
>>>>> Thank you for the suggestion but this is not working . I
>>>>> think because of the the  <param name="callback-context"
>>>>> value="default"/>
>>>>>
>>>>
>>>> Please define "not working" - either you press 9 and the call is x-fer'd
>>>> or it is not. Once it is x-fer'd to the dialplan you should be able to do
>>>> whatever you want with the call. Or do what bkw says and use xml_curl.
>>>> -MC
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>
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From dome at tel.co.th  Mon Jan  3 18:49:33 2011
From: dome at tel.co.th (dome at tel.co.th)
Date: Mon, 3 Jan 2011 22:49:33 +0700
Subject: [Freeswitch-users] increase Max session
In-Reply-To: <41DEB133-E2DD-4DC5-B6DB-320CD2C6BCBA@freeswitch.org>
References: <AANLkTik=Fa3ZAX5XP_ubZ7ekrx8M8A7D5J0L=fqepEUg@mail.gmail.com>
	<AANLkTimZxTYeGGNnSKF2x07wS6mBhU5kyB6xbeTKcAmc@mail.gmail.com>
	<AANLkTin+uvhAM58ODfz=6jcJsuwDqcNzysEpLO7yXVfc@mail.gmail.com>
	<8825DCE39E834012ACD4A4978DFA9AB7@e1705>
	<AANLkTimTQ2mjJ+mGQ+kozTbgfE0qtgR-j32sq_oxYNtd@mail.gmail.com>
	<AANLkTi=YR=LxqLy+PzjO_NTiRf+-cAVQ0D=yguZkUw_P@mail.gmail.com>
	<AANLkTint0ypRO68sRdeASqy5s4OC38xK8SPeMXigm0_9@mail.gmail.com>
	<AANLkTi=2C6vOJiHWK8r2EdfdjBhca1_tYUXAZ1_dCpK0@mail.gmail.com>
	<AANLkTik9AiZ0GpmtKENmO_Y15rvxyDqm+Ga7Ryyt8vML@mail.gmail.com>
	<41DEB133-E2DD-4DC5-B6DB-320CD2C6BCBA@freeswitch.org>
Message-ID: <AANLkTinyUR2MKrsU9Uc-AH4Go=EvQCu=8+PjgghXasrT@mail.gmail.com>

Thanks brian.
proxmox default config give me 1024 process num :(


Dome C.

2011/1/3 Brian West <brian at freeswitch.org>:
> Your container doesn't have enough resources look at the UB and see what is
> failing and increase accordingly.
> /b
> On Jan 1, 2011, at 9:39 PM, dome at tel.co.th wrote:
>
> 64 bit over openvz
>
> 2011/1/2 Aloysius Lloyd <lloyd.aloysius at sunteltech.ca>:
>
> 32 bit or 64 bit ?
>
>
> On Sat, Jan 1, 2011 at 10:12 PM,?dome at tel.co.th?<dome at tel.co.th> wrote:
>
> 2011/1/2 Aloysius Lloyd <lloyd.aloysius at gmail.com>:
>
> make sure the switch.conf.xml have the right value. Did you rotate the
>
> log
>
> file between 22 min and 37 min.
>
> Yes
>
> try the cli command?fsctl max_session 5000 see what happen in couple of
>
> hours.
>
> i try fsctl max_session 5000
>
> after that FS switch back to 921 again
>
> Thanks
>
> Lloyd
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From Nabble at slickdeals.endjunk.com  Mon Jan  3 19:35:52 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 3 Jan 2011 08:35:52 -0800 (PST)
Subject: [Freeswitch-users] Regarding mod_java application in freeswitch
In-Reply-To: <1294068401368-5880910.post@n2.nabble.com>
References: <1294039064455-5880529.post@n2.nabble.com>
	<1294065197457-5880847.post@n2.nabble.com>
	<1294068401368-5880910.post@n2.nabble.com>
Message-ID: <1294072552545-5880995.post@n2.nabble.com>



kapil.rastogi wrote:
> can you please specify me about these dialplan value if the name of java
> program file is PhoneTest.java and it is located at the
> "/usr/local/freeswitch/scripts/" path.
> 
> <action application="java" data="fully.qualified.class.name arg1 arg2
> arg3"/>
> 
> <action application="java" data="/path/to/file.jar
> fully.qualified.class.name arg1 arg2 arg3"/>Unfortunately, I can't because
> I have never done any Java programming.  Hopefully, experts in this
> mailing list will be able to help. Just give them some times, especially
> after a long holiday.


-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Regarding-mod-java-application-in-freeswitch-tp5880529p5880995.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From xyangni at gmail.com  Mon Jan  3 19:47:16 2011
From: xyangni at gmail.com (xuyan yang)
Date: Mon, 3 Jan 2011 16:47:16 +0000
Subject: [Freeswitch-users] Can't build on Windows 7 64bit C++ Express!?
In-Reply-To: <snt0-eas12908A012C361DEE1DF0DC7B9070@phx.gbl>
References: <snt0-eas12908A012C361DEE1DF0DC7B9070@phx.gbl>
Message-ID: <AANLkTi=6TzBu3BymCiG1j-u5jWfSWv+VT+a+KndywXxN@mail.gmail.com>

Builiding freshly downloaded git version (git clone) on XP with VC2008
Express also generated lots of errors and mod_sophia can not be build.
I just tried to build the solution directly without modify any setting or
choose any project.



On Mon, Jan 3, 2011 at 4:09 AM, Jeff Lenk <jeff at jefflenk.com> wrote:

>  What target are you trying to build? Post a log of the build or pastebin
> reference
>
> Sent from my Windows Phone
> ------------------------------
> From: Joao Leme
> Sent: Sunday, January 02, 2011 1:28 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Can't build on Windows 7 64bit C++
> Express!?
>
>
>
> > Ok Thanks. Any idea why all the errors and why i can't build? I've built
> > before on Windows Vista 64bit and VS2008Pro with no problems but can't
> get
> > it to work on Express edition.
> >
> > On Sat, Jan 1, 2011 at 9:51 PM, babak yakhchali
> ><babak.freeswitch at gmail.com>wrote:
> >
> >> you just need to build the project in
> >>
> >>
> D:\gitRepos\2010exp-freeswitch\freeswitch\src\mod\languages\mod_managed\managed
> >> using vc# express, others should just work using vc++
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Jan  3 19:54:19 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 Jan 2011 10:54:19 -0600
Subject: [Freeswitch-users] Attack using 5843 and music account?
In-Reply-To: <AANLkTinNmW4OtL275UWzRoZof0U_YAq3vhfwNkJo3oOG@mail.gmail.com>
References: <AANLkTinNmW4OtL275UWzRoZof0U_YAq3vhfwNkJo3oOG@mail.gmail.com>
Message-ID: <28AF5B89-AFB3-438E-AB7B-AF598CB18204@freeswitch.org>

Chances are he never received the challenge.. thus never logs an auth failure.

/b

On Jan 3, 2011, at 9:26 AM, xuyan yang wrote:

> 2011-01-03 15:19:32.360152 [WARNING] sofia_reg.c:1161 SIP auth failure (REGISTER) on sofia profile 'internal' for [music at 192.168.0.3] from ip 192.168.0.6
> 
> So, how can this hacker successfully registered music account and avoid to be baned? it is strange.
> 
> Thanks

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From peter.olsson at visionutveckling.se  Mon Jan  3 20:12:10 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 3 Jan 2011 18:12:10 +0100
Subject: [Freeswitch-users] Can't build on Windows 7 64bit C++ Express!?
Message-ID: <BC8E7AC3-4B6A-4759-A179-EE416455D314@visionutveckling.se>

Make sure to disable autocrlf in git, then it will work. Also make sure to read all instructions on the wiki.

/Peter

----- Reply message -----
Fr?n: "xuyan yang" <xyangni at gmail.com>
Datum: m?n, jan 3, 2011 17:54
Rubrik: [Freeswitch-users] Can't build on Windows 7 64bit C++ Express!?
Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

Builiding freshly downloaded git version (git clone) on XP with VC2008 Express also generated lots of errors and mod_sophia can not be build.
I just tried to build the solution directly without modify any setting or choose any project.



On Mon, Jan 3, 2011 at 4:09 AM, Jeff Lenk <jeff at jefflenk.com<mailto:jeff at jefflenk.com>> wrote:
What target are you trying to build? Post a log of the build or pastebin reference

Sent from my Windows Phone
________________________________
From: Joao Leme
Sent: Sunday, January 02, 2011 1:28 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Can't build on Windows 7 64bit C++ Express!?



> Ok Thanks. Any idea why all the errors and why i can't build? I've built
> before on Windows Vista 64bit and VS2008Pro with no problems but can't get
> it to work on Express edition.
>
> On Sat, Jan 1, 2011 at 9:51 PM, babak yakhchali
><babak.freeswitch at gmail.com<mailto:babak.freeswitch at gmail.com>>wrote:
>
>> you just need to build the project in
>>
>> D:\gitRepos\2010exp-freeswitch\freeswitch\src\mod\languages\mod_managed\managed
>> using vc# express, others should just work using vc++
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


!DSPAM:4d21fef732761594845422!


From tim at compnetwork.net  Mon Jan  3 20:40:18 2011
From: tim at compnetwork.net (Tim King)
Date: Mon, 3 Jan 2011 12:40:18 -0500
Subject: [Freeswitch-users] Mail list problem
Message-ID: <AANLkTimx8NcaF4iUz2aq0gPZC8j_Z-B1K_u9mh-aXkdH@mail.gmail.com>

This is a test as I do not seem to be getting any messages from the list.
Can someone please reply and let me know I can can get messages working
again. They are not going to spam.

Thanks

Tim
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From tgraziano at myitdepartment.net  Mon Jan  3 20:47:49 2011
From: tgraziano at myitdepartment.net (Tony Graziano)
Date: Mon, 3 Jan 2011 12:47:49 -0500
Subject: [Freeswitch-users] Mail list problem
In-Reply-To: <AANLkTimx8NcaF4iUz2aq0gPZC8j_Z-B1K_u9mh-aXkdH@mail.gmail.com>
References: <AANLkTimx8NcaF4iUz2aq0gPZC8j_Z-B1K_u9mh-aXkdH@mail.gmail.com>
Message-ID: <AANLkTinahh3iW9pnWx98rJitY2fR=8mj_RBwyKsWAxF0@mail.gmail.com>

Reply...

On Mon, Jan 3, 2011 at 12:40 PM, Tim King <tim at compnetwork.net> wrote:

> This is a test as I do not seem to be getting any messages from the list.
> Can someone please reply and let me know I can can get messages working
> again. They are not going to spam.
>
> Thanks
>
> Tim
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
======================
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgraziano at voice.myitdepartment.net
Fax: 434.326.5325

Email: tgraziano at myitdepartment.net

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpdesk at voice.myitdepartment.net

Helpdesk Contract Customers:
http://support.myitdepartment.net

<http://support.myitdepartment.net>Blog:
http://blog.myitdepartment.net

Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
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From mustafa.pk at gmail.com  Mon Jan  3 20:48:18 2011
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Mon, 3 Jan 2011 22:48:18 +0500
Subject: [Freeswitch-users] Mail list problem
In-Reply-To: <AANLkTimx8NcaF4iUz2aq0gPZC8j_Z-B1K_u9mh-aXkdH@mail.gmail.com>
References: <AANLkTimx8NcaF4iUz2aq0gPZC8j_Z-B1K_u9mh-aXkdH@mail.gmail.com>
Message-ID: <AANLkTi=WfRMQqOgQTo-fph52r-R0wSGWPvLdjAP+g9WK@mail.gmail.com>

bump!

On Mon, Jan 3, 2011 at 10:40 PM, Tim King <tim at compnetwork.net> wrote:

> This is a test as I do not seem to be getting any messages from the list.
> Can someone please reply and let me know I can can get messages working
> again. They are not going to spam.
>
> Thanks
>
> Tim
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com
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From anita.hall at simmortel.com  Mon Jan  3 20:57:07 2011
From: anita.hall at simmortel.com (Anita Hall)
Date: Mon, 3 Jan 2011 23:27:07 +0530
Subject: [Freeswitch-users] libss7 * Sangoma
Message-ID: <AANLkTi=ok4KmurNX8Fha3Ne1yCSexTuDgJPV5f-M72t3@mail.gmail.com>

Hi

This question pertains libss7 with Sangoma A108 card on Asterisk. It does
not concern freeswitch but I suppose Sangoma folks frequent this list more
:)

I am unable to make libss7 work with Sangoma. Here are the details.

Could you please provide me some pointers ?

Thanks,
Anita.

debian:~# uname -a
Linux debian 2.6.26-2-686 #1 SMP Thu Nov 25 01:53:57 UTC 2010 i686 GNU/Linux

debian:~# dahdi_hardware
pci:0000:05:04.0     wanpipe-     1923:0100 Sangoma Technologies Corp. A104d
QUAD T1/E1 AFT card
pci:0000:07:04.0     wanpipe-     1923:0100 Sangoma Technologies Corp. A104d
QUAD T1/E1 AFT card



debian:~# cat /etc/dahdi/system.conf
#autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
#autogenrated on 2011-01-03
#Dahdi Channels Configurations
#For detailed Dahdi options, view /etc/dahdi/system.conf.bak
loadzone=us
defaultzone=us

#Sangoma A108 port 1 [slot:4 bus:5 span:1] <wanpipe1>
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
echocanceller=mg2,1-15,17-31
mtp2=16

#Sangoma A108 port 2 [slot:4 bus:5 span:2] <wanpipe2>
span=2,1,0,ccs,hdb3,crc4
bchan=32-46,48-62
echocanceller=mg2,32-46,48-62
mtp2=47


debian:~# cat /etc/wanpipe/wanpipe1.conf
#=============================
===================
# WANPIPE1 Configuration File
#================================================
#
# Date: Wed Dec  6 20:29:03 UTC 2006
#
# Note: This file was generated automatically
#       by /usr/local/sbin/setup-sangoma program.
#
#       If you want to edit this file, it is
#       recommended that you use wancfg program
#       to do so.
#================================================
# Sangoma Technologies Inc.
#================================================

[devices]
wanpipe1 = WAN_AFT_TE1, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment

[wanpipe1]
CARD_TYPE       = AFT
S514CPU         = A
CommPort        = PRI
AUTO_PCISLOT    = NO
PCISLOT         = 4
PCIBUS          = 5
FE_MEDIA        = E1
FE_LCODE        = HDB3
FE_FRAME        = NCRC4
FE_LINE         = 1
TE_CLOCK        = NORMAL
TE_REF_CLOCK    = 0
TE_SIG_MODE     = CCS
TE_HIGHIMPEDANCE        = NO
LBO             = 120OH
FE_TXTRISTATE   = NO
MTU             = 1500
UDPPORT         = 9000
TTL             = 255
IGNORE_FRONT_END = NO
TDMV_SPAN       = 1
TDMV_DCHAN      = 16
TDMV_HW_DTMF    = NO
TDMV_HW_FAX_DETECT = NO

[w1g1]
ACTIVE_CH       = ALL
TDMV_HWEC       = NO


debian:~# cat /etc/wanpipe/wanpipe2.conf
#================================================
# WANPIPE1 Configuration File
#================================================
#
# Date: Wed Dec  6 20:29:03 UTC 2006
#
# Note: This file was generated automatically
#       by /usr/local/sbin/setup-sangoma program.
#
#       If you want to edit this file, it is
#       recommended that you use wancfg program
#       to do so.
#================================================
# Sangoma Technologies Inc.
#================================================

[devices]
wanpipe2 = WAN_AFT_TE1, Comment

[interfaces]
w2g1 = wanpipe2, , TDM_VOICE, Comment

[wanpipe2]
CARD_TYPE       = AFT
S514CPU         = A
CommPort        = PRI
AUTO_PCISLOT    = NO
PCISLOT         = 4
PCIBUS          = 5
FE_MEDIA        = E1
FE_LCODE        = HDB3
FE_FRAME        = NCRC4
FE_LINE         = 2
TE_CLOCK        = NORMAL
TE_REF_CLOCK    = 0
TE_SIG_MODE     = CCS
TE_HIGHIMPEDANCE        = NO
LBO             = 120OH
FE_TXTRISTATE   = NO
MTU             = 1500
UDPPORT         = 9000
TTL             = 255
IGNORE_FRONT_END = NO
TDMV_SPAN       = 2
TDMV_DCHAN      = 16
TDMV_HW_DTMF    = NO
TDMV_HW_FAX_DETECT = NO

[w2g1]
ACTIVE_CH       = ALL
TDMV_HWEC       = NO

debian:~# cat /etc/asterisk/chan_dahdi.conf
;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
;autogenrated on 2011-01-03
;Dahdi Channels Configurations
;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak

[trunkgroups]

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no

;Sangoma A108 port 1 [slot:4 bus:5 span:1] <wanpipe1>
switchtype=euroisdn
context=tata
group=1
echocancel=yes
signaling=ss7                    ;this is ss7 signaling
ss7type=itu                     ;using the ITU variant
ss7_called_nai=dynamic           ;NAI for outgoing calls
ss7_calling_nai=dynamic          ;NAI for incoming calls
ss7_internationalprefix=00 ;international prefix value for incoming calls
ss7_nationalprefix=0            ;national prefix value for incoming calls
ss7_subscriberprefix=            ;subscriber prefix value for incoming calls
ss7_unknownprefix=              ;unknown prefix value for incoming calls
ss7_explictacm=yes               ;ACM is send as soon as call enters the
dial plan...may not accepted yet though
linkset=1                         ;arbitrary name for this set of channels
pointcode=13323                      ;the point code for this system...aka
SPC
adjpointcode=12650                  ;the point code for the system that we
are signaling to... aka APC
defaultdpc=12650                     ;the point code for the system that the
CICs will be negotiated with...aka DPC
networkindicator=international  ;NI value for MTP3
cicbeginswith=1                  ;the starting value of the CICs
channel =>1-15
cicbeginswith=17                 ;the starting value of the CICs
channel =>17-31         ;the channels that are CICs
sigchan=16                     ;the signaling channel

;Sangoma A108 port 2 [slot:4 bus:5 span:2] <wanpipe2>
switchtype=euroisdn
context=tata
group=1
echocancel=yes
cicbeginswith=32                  ;the starting value of the CICs
channel =>32-46
cicbeginswith=48                  ;the starting value of the CICs
channel =>48-62         ;the channels that are CICs
sigchan=47                       ;the signaling channel


debian:~# ifconfig w1g1
w1g1      Link encap:Point-to-Point Protocol
         UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
         RX packets:772814 errors:0 dropped:0 overruns:0 frame:0
         TX packets:772814 errors:0 dropped:0 overruns:3 carrier:0
         collisions:0 txqueuelen:100
         RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
         Interrupt:16 Memory:f8d00000-f8d01fff

debian:~# ifconfig w1g1
w1g1      Link encap:Point-to-Point Protocol
         UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
         RX packets:774782 errors:0 dropped:0 overruns:0 frame:0
         TX packets:774782 errors:0 dropped:0 overruns:3 carrier:0
         collisions:0 txqueuelen:100
         RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
         Interrupt:16 Memory:f8d00000-f8d01fff

debian:~# ifconfig w1g1
w1g1      Link encap:Point-to-Point Protocol
         UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
         RX packets:775351 errors:0 dropped:0 overruns:0 frame:0
         TX packets:775351 errors:0 dropped:0 overruns:3 carrier:0
         collisions:0 txqueuelen:100
         RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
         Interrupt:16 Memory:f8d00000-f8d01fff

debian:~# ifconfig w1g1
w1g1      Link encap:Point-to-Point Protocol
         UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
         RX packets:775919 errors:0 dropped:0 overruns:0 frame:0
         TX packets:775919 errors:0 dropped:0 overruns:3 carrier:0
         collisions:0 txqueuelen:100
         RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
         Interrupt:16 Memory:f8d00000-f8d01fff

debian:~# ifconfig w1g1
w1g1      Link encap:Point-to-Point Protocol
         UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
         RX packets:776478 errors:0 dropped:0 overruns:0 frame:0
         TX packets:776478 errors:0 dropped:0 overruns:3 carrier:0
         collisions:0 txqueuelen:100
         RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
         Interrupt:16 Memory:f8d00000-f8d01fff

debian:~# ifconfig w1g1
w1g1      Link encap:Point-to-Point Protocol
         UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
         RX packets:777079 errors:0 dropped:0 overruns:0 frame:0
         TX packets:777079 errors:0 dropped:0 overruns:3 carrier:0
         collisions:0 txqueuelen:100
         RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
         Interrupt:16 Memory:f8d00000-f8d01fff

debian:~# ifconfig w1g1
w1g1      Link encap:Point-to-Point Protocol
         UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
         RX packets:778134 errors:0 dropped:0 overruns:0 frame:0
         TX packets:778134 errors:0 dropped:0 overruns:3 carrier:0
         collisions:0 txqueuelen:100
         RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
         Interrupt:16 Memory:f8d00000-f8d01fff


debian:~# ifconfig w2g1
w2g1      Link encap:Point-to-Point Protocol
         UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
         RX packets:792251 errors:0 dropped:0 overruns:0 frame:0
         TX packets:792251 errors:0 dropped:0 overruns:1 carrier:0
         collisions:0 txqueuelen:100
         RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
         Interrupt:16 Memory:f8d00000-f8d01fff

debian:~# ifconfig w2g1
w2g1      Link encap:Point-to-Point Protocol
         UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
         RX packets:792715 errors:0 dropped:0 overruns:0 frame:0
         TX packets:792715 errors:0 dropped:0 overruns:1 carrier:0
         collisions:0 txqueuelen:100
         RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
         Interrupt:16 Memory:f8d00000-f8d01fff

debian:~# ifconfig w2g1
w2g1      Link encap:Point-to-Point Protocol
         UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
         RX packets:793219 errors:0 dropped:0 overruns:0 frame:0
         TX packets:793219 errors:0 dropped:0 overruns:1 carrier:0
         collisions:0 txqueuelen:100
         RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
         Interrupt:16 Memory:f8d00000-f8d01fff


wanpipemon -i w2g1 -c Ta

***** w2g1: E1 Alarms (Framer) *****

ALOS:   OFF     | LOS:  OFF
RED:    OFF     | AIS:  OFF
LOF:    OFF     | RAI:  OFF

***** w2g1: E1 Alarms (LIU) *****

Short Circuit:  OFF
Open Circuit:   OFF
Loss of Signal: OFF


***** w2g1: E1 Performance Monitoring Counters *****

Line Code Violation     : 0
Far End Block Errors    : 0
CRC4 Errors             : 0
FAS Errors              : 0


Rx Level        : > -2.5db


debian:~# wanpipemon -i w1g1 -c Ta

***** w1g1: E1 Alarms (Framer) *****

ALOS:   OFF     | LOS:  OFF
RED:    OFF     | AIS:  OFF
LOF:    OFF     | RAI:  OFF

***** w1g1: E1 Alarms (LIU) *****

Short Circuit:  OFF
Open Circuit:   OFF
Loss of Signal: OFF


***** w1g1: E1 Performance Monitoring Counters *****

Line Code Violation     : 371
Far End Block Errors    : 0
CRC4 Errors             : 0
FAS Errors              : 0


Rx Level        : > -2.5db


Jan  3 20:33:34 debian kernel: [55441.172179] wanpipe1: Shutting Down!
Jan  3 20:33:34 debian kernel: [55441.172179] wanpipe1: E1 Front End
unconfigation!
Jan  3 20:33:34 debian kernel: [55441.172179] wanpipe1: Unregister Wanpipe
device from Zaptel!
Jan  3 20:33:34 debian kernel: [55441.178246] wanpipe1: unregistering 'w1g1'
Jan  3 20:33:34 debian kernel: [55441.248278] wanpipe1: TASKQ Not Running
Jan  3 20:33:34 debian kernel: [55441.248281] wanpipe1: E1 Front End
unconfigation!
Jan  3 20:33:34 debian kernel: [55441.248314] wanpipe1: AFT communications
disabled! (Dev Cnt: 1 Cause: Device Down)
Jan  3 20:33:34 debian kernel: [55441.248335] wanpipe1: E1 Front End
unconfigation!
Jan  3 20:33:34 debian kernel: [55441.248373] wanpipe1: AFT communications
disabled! (Dev Cnt: 1 Cause: Device Down)
Jan  3 20:33:34 debian kernel: [55441.248415] wanpipe1: Global Chip Shutdown
Usage=1
Jan  3 20:33:34 debian kernel: [55441.248419] wanpipe1: Global E1 Front End
unconfigation!
Jan  3 20:33:34 debian kernel: [55441.250485] wanpipe1: Master shutting down
Jan  3 20:33:34 debian kernel: [55441.444571] wanpipe1: Starting WAN Setup
Jan  3 20:33:34 debian kernel: [55441.444575] wanpipe1: Locating:
A101/1D/A102/2D/4/4D/8 card, CPU A, PciSlot=4, PciBus=5
Jan  3 20:33:34 debian kernel: [55441.444579] wanpipe1: Found:
A101/1D/A102/2D/4/4D/8 card, CPU A, PciSlot=4, PciBus=5, Port=0
Jan  3 20:33:34 debian kernel: [55441.444599] wanpipe1: AFT PCI memory at
0xD3200000
Jan  3 20:33:34 debian kernel: [55441.444600] wanpipe1: IRQ 16 allocated to
the AFT PCI card
Jan  3 20:33:34 debian kernel: [55441.444608] wanpipe1: Starting AFT 2/4/8
Hardware Init.
Jan  3 20:33:34 debian kernel: [55441.444614] wanpipe1: Enabling front end
link monitor
Jan  3 20:33:34 debian kernel: [55441.444616] wanpipe1: Global Chip
Configuration: used=1
Jan  3 20:33:34 debian kernel: [55441.448455] wanpipe1: Global E1 Front End
configuration
Jan  3 20:33:34 debian kernel: [55441.448455] wanpipe1: AFT Data Mux Bit
Map: 0x01234567
Jan  3 20:33:34 debian kernel: [55441.448455] wanpipe1: Defaulting E1 Rx
Sens. Gain= 43 db
Jan  3 20:33:34 debian kernel: [55441.448455] wanpipe1: Configuring DS
DS26528 E1 FE
Jan  3 20:33:34 debian kernel: [55441.448455] wanpipe1:    Port
1,HDB3,non-CRC4,120OH
Jan  3 20:33:34 debian kernel: [55441.448455] wanpipe1:    Clk Normal:0,
Channels: FFFFFFFF
Jan  3 20:33:34 debian kernel: [55441.448455] wanpipe1:    Sig Mode CCS
Jan  3 20:33:34 debian kernel: [55441.456490] wanpipe1:    Rx Sensitivity
Gain 43dB (default).
Jan  3 20:33:34 debian kernel: [55441.457242] wanpipe1: Front end successful
Jan  3 20:33:34 debian kernel: [55441.457469] wanpipe1: Front End Interface
Ready 0x40000000
Jan  3 20:33:34 debian kernel: [55441.457473] wanpipe1: WARNING: No Echo
Canceller channels are available!
Jan  3 20:33:34 debian kernel: [55441.457494] wanpipe1: Configuring Device
:wanpipe1  FrmVr=39
Jan  3 20:33:34 debian kernel: [55441.457495] wanpipe1:    Global MTU
    = 1500
Jan  3 20:33:34 debian kernel: [55441.457496] wanpipe1:    Global MRU
    = 1500
Jan  3 20:33:34 debian kernel: [55441.457497] wanpipe1:    Data Mux Map
    = 0x01234567
Jan  3 20:33:34 debian kernel: [55441.457498] wanpipe1:    Rx CRC Bytes
    = 0
Jan  3 20:33:34 debian kernel: [55441.457499] wanpipe1:    Global TDM Int
    = Enabled
Jan  3 20:33:34 debian kernel: [55441.457500] wanpipe1:    Global TDM Ring
   = Enabled
Jan  3 20:33:34 debian kernel: [55441.457501] wanpipe1:    TDMV HW DTMF/FAX
    = Disabled/Disabled(0)
Jan  3 20:33:34 debian kernel: [55441.457502] wanpipe1:    TDMV Span
   = 1 : Enabled
Jan  3 20:33:34 debian kernel: [55441.457503] wanpipe1:    TDMV Dummy
= Disabled
Jan  3 20:33:34 debian kernel: [55441.457505] wanpipe1:    RTP TAP
   = Disabled
Jan  3 20:33:34 debian kernel: [55441.457540] wanpipe1: Configuring
Interface: w1g1
Jan  3 20:33:34 debian kernel: [55441.457543] wanpipe1:w1g1: Running in TDM
Voice Zaptel Mode.
Jan  3 20:33:34 debian kernel: [55441.457547] wanpipe1:    Fifo Level
Map:0x01041040
Jan  3 20:33:34 debian kernel: [55441.457549] wanpipe1:    MRU           :8
Jan  3 20:33:34 debian kernel: [55441.457550] wanpipe1:    MTU           :8
Jan  3 20:33:34 debian kernel: [55441.457551] wanpipe1:    HDLC Eng
 :Off (Transparent) | N/A
Jan  3 20:33:34 debian kernel: [55441.457553] wanpipe1:    Data Mux Ctrl :On
Jan  3 20:33:34 debian kernel: [55441.457554] wanpipe1:    Active Ch Map
:0x00000002
Jan  3 20:33:34 debian kernel: [55441.457555] wanpipe1:    First TSlot   :1
Jan  3 20:33:34 debian kernel: [55441.457568] wanpipe1:    DMA/Len/Chain/EC
:4/1024/Off/Off
Jan  3 20:33:34 debian kernel: [55441.457575] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.457581] wanpipe1:    Active Ch Map
:0x00000004
Jan  3 20:33:34 debian kernel: [55441.457582] wanpipe1:    First TSlot   :2
Jan  3 20:33:34 debian kernel: [55441.457597] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.457602] wanpipe1:    Active Ch Map
:0x00000008
Jan  3 20:33:34 debian kernel: [55441.457603] wanpipe1:    First TSlot   :3
Jan  3 20:33:34 debian kernel: [55441.457618] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.457623] wanpipe1:    Active Ch Map
:0x00000010
Jan  3 20:33:34 debian kernel: [55441.457624] wanpipe1:    First TSlot   :4
Jan  3 20:33:34 debian kernel: [55441.457637] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.457643] wanpipe1:    Active Ch Map
:0x00000020
Jan  3 20:33:34 debian kernel: [55441.457644] wanpipe1:    First TSlot   :5
Jan  3 20:33:34 debian kernel: [55441.457657] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.457663] wanpipe1:    Active Ch Map
:0x00000040
Jan  3 20:33:34 debian kernel: [55441.457664] wanpipe1:    First TSlot   :6
Jan  3 20:33:34 debian kernel: [55441.457678] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.457684] wanpipe1:    Active Ch Map
:0x00000080
Jan  3 20:33:34 debian kernel: [55441.457685] wanpipe1:    First TSlot   :7
Jan  3 20:33:34 debian kernel: [55441.457698] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.457705] wanpipe1:    Active Ch Map
:0x00000100
Jan  3 20:33:34 debian kernel: [55441.457706] wanpipe1:    First TSlot   :8
Jan  3 20:33:34 debian kernel: [55441.457719] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.457725] wanpipe1:    Active Ch Map
:0x00000200
Jan  3 20:33:34 debian kernel: [55441.457726] wanpipe1:    First TSlot   :9
Jan  3 20:33:34 debian kernel: [55441.457740] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.457746] wanpipe1:    Active Ch Map
:0x00000400
Jan  3 20:33:34 debian kernel: [55441.457747] wanpipe1:    First TSlot   :10
Jan  3 20:33:34 debian kernel: [55441.457760] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.457765] wanpipe1:    Active Ch Map
:0x00000800
Jan  3 20:33:34 debian kernel: [55441.457766] wanpipe1:    First TSlot   :11
Jan  3 20:33:34 debian kernel: [55441.457781] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.457786] wanpipe1:    Active Ch Map
:0x00001000
Jan  3 20:33:34 debian kernel: [55441.457787] wanpipe1:    First TSlot   :12
Jan  3 20:33:34 debian kernel: [55441.457801] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.457807] wanpipe1:    Active Ch Map
:0x00002000
Jan  3 20:33:34 debian kernel: [55441.457808] wanpipe1:    First TSlot   :13
Jan  3 20:33:34 debian kernel: [55441.457822] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.457828] wanpipe1:    Active Ch Map
:0x00004000
Jan  3 20:33:34 debian kernel: [55441.457829] wanpipe1:    First TSlot   :14
Jan  3 20:33:34 debian kernel: [55441.457842] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.457848] wanpipe1:    Active Ch Map
:0x00008000
Jan  3 20:33:34 debian kernel: [55441.457849] wanpipe1:    First TSlot   :15
Jan  3 20:33:34 debian kernel: [55441.457862] wanpipe1: Configuring
Interface: w1g1
Jan  3 20:33:34 debian kernel: [55441.457864] wanpipe1:w1g1: Running in TDM
DCHAN Voice Zaptel Mode.
Jan  3 20:33:34 debian kernel: [55441.457866] wanpipe1:    MRU
:1500
Jan  3 20:33:34 debian kernel: [55441.457867] wanpipe1:    MTU
:1500
Jan  3 20:33:34 debian kernel: [55441.457868] wanpipe1:    HDLC Eng      :On
| N/A
Jan  3 20:33:34 debian kernel: [55441.457869] wanpipe1:    Data Mux Ctrl
:Off
Jan  3 20:33:34 debian kernel: [55441.457870] wanpipe1:    Active Ch Map
:0x00010000
Jan  3 20:33:34 debian kernel: [55441.457871] wanpipe1:    First TSlot   :16
Jan  3 20:33:34 debian kernel: [55441.457881] wanpipe1:    DMA/Len/Chain/EC
:65/4096/On/Off
Jan  3 20:33:34 debian kernel: [55441.457905] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.457910] wanpipe1:    Active Ch Map
:0x00020000
Jan  3 20:33:34 debian kernel: [55441.457911] wanpipe1:    First TSlot   :17
Jan  3 20:33:34 debian kernel: [55441.457925] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.457931] wanpipe1:    Active Ch Map
:0x00040000
Jan  3 20:33:34 debian kernel: [55441.457932] wanpipe1:    First TSlot   :18
Jan  3 20:33:34 debian kernel: [55441.457945] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.457951] wanpipe1:    Active Ch Map
:0x00080000
Jan  3 20:33:34 debian kernel: [55441.457952] wanpipe1:    First TSlot   :19
Jan  3 20:33:34 debian kernel: [55441.457967] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.457973] wanpipe1:    Active Ch Map
:0x00100000
Jan  3 20:33:34 debian kernel: [55441.457974] wanpipe1:    First TSlot   :20
Jan  3 20:33:34 debian kernel: [55441.457987] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.457993] wanpipe1:    Active Ch Map
:0x00200000
Jan  3 20:33:34 debian kernel: [55441.457994] wanpipe1:    First TSlot   :21
Jan  3 20:33:34 debian kernel: [55441.458009] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.458015] wanpipe1:    Active Ch Map
:0x00400000
Jan  3 20:33:34 debian kernel: [55441.458016] wanpipe1:    First TSlot   :22
Jan  3 20:33:34 debian kernel: [55441.458030] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.458036] wanpipe1:    Active Ch Map
:0x00800000
Jan  3 20:33:34 debian kernel: [55441.458037] wanpipe1:    First TSlot   :23
Jan  3 20:33:34 debian kernel: [55441.458051] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.458057] wanpipe1:    Active Ch Map
:0x01000000
Jan  3 20:33:34 debian kernel: [55441.458058] wanpipe1:    First TSlot   :24
Jan  3 20:33:34 debian kernel: [55441.458072] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.458078] wanpipe1:    Active Ch Map
:0x02000000
Jan  3 20:33:34 debian kernel: [55441.458079] wanpipe1:    First TSlot   :25
Jan  3 20:33:34 debian kernel: [55441.458094] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.458100] wanpipe1:    Active Ch Map
:0x04000000
Jan  3 20:33:34 debian kernel: [55441.458101] wanpipe1:    First TSlot   :26
Jan  3 20:33:34 debian kernel: [55441.458116] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.458122] wanpipe1:    Active Ch Map
:0x08000000
Jan  3 20:33:34 debian kernel: [55441.458123] wanpipe1:    First TSlot   :27
Jan  3 20:33:34 debian kernel: [55441.458137] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.458143] wanpipe1:    Active Ch Map
:0x10000000
Jan  3 20:33:34 debian kernel: [55441.458144] wanpipe1:    First TSlot   :28
Jan  3 20:33:34 debian kernel: [55441.458160] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.458166] wanpipe1:    Active Ch Map
:0x20000000
Jan  3 20:33:34 debian kernel: [55441.458167] wanpipe1:    First TSlot   :29
Jan  3 20:33:34 debian kernel: [55441.458181] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.458187] wanpipe1:    Active Ch Map
:0x40000000
Jan  3 20:33:34 debian kernel: [55441.458188] wanpipe1:    First TSlot   :30
Jan  3 20:33:34 debian kernel: [55441.458202] wanpipe1: Configuring
Interface: w1g1 (log supress)
Jan  3 20:33:34 debian kernel: [55441.458205] wanpipe1: Configuring TDMV
Master dev w1g1
Jan  3 20:33:34 debian kernel: [55441.458209] wanpipe1:    Active Ch Map
:0x80000000
Jan  3 20:33:34 debian kernel: [55441.458210] wanpipe1:    First TSlot   :31
Jan  3 20:33:34 debian kernel: [55441.458226] wanpipe1: Enable Zaptel HW
DCHAN interface
Jan  3 20:33:34 debian kernel: [55441.458462] wanpipe1: Wanpipe device is
registered to Zaptel span # 1!
Jan  3 20:33:34 debian kernel: [55441.458760] wanpipe1: TDM Free Run Timing
Enabled 1 ms
Jan  3 20:33:34 debian kernel: [55441.464708] wanpipe1: Wanpipe Front End
Interrupt Restart Timeout
Jan  3 20:33:40 debian kernel: [55450.402880] wanpipe1: E1 connected!
Jan  3 20:33:40 debian kernel: [55450.407421] wanpipe1: AFT communications
enabled!
Jan  3 20:33:40 debian kernel: [55450.407421] wanpipe1: AFT Global TDM Intr
Jan  3 20:33:40 debian kernel: [55450.409902] wanpipe1: Global TDM Ring
Resync
Jan  3 20:33:41 debian kernel: [55452.785733] wanpipe1: Enable E1 CCS
Signalling mode!


debian*CLI> core show version
Asterisk 1.6.2.0 built by root @ debian on a i686 running Linux on
2011-01-03 08:57:50 UTC


debian*CLI> ss7 show linkset 1
SS7 linkset 1 status: Down


debian*CLI> ss7 set debug off linkset 1
Enabled debugging on linkset 1
Len = 4 [ ff ff 01 03 ]
FSN: 127 FIB 1
BSN: 127 BIB 1
<[0] LSSU SIOS

Link state change: NOTALIGNED -> NOTALIGNED
Len = 4 [ ff ff 01 00 ]
FSN: 127 FIB 1
BSN: 127 BIB 1
<[0] LSSU SIO

Link state change: NOTALIGNED -> ALIGNED
Len = 4 [ ff ff 01 02 ]
FSN: 127 FIB 1
BSN: 127 BIB 1
>[0] LSSU SIE

Link state change: ALIGNED -> IDLE
Link state change: IDLE -> NOTALIGNED
Len = 4 [ ff ff 01 00 ]
FSN: 127 FIB 1
BSN: 127 BIB 1
>[0] LSSU SIO
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From brian at freeswitch.org  Mon Jan  3 21:06:52 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 Jan 2011 12:06:52 -0600
Subject: [Freeswitch-users] Mail list problem
In-Reply-To: <AANLkTimx8NcaF4iUz2aq0gPZC8j_Z-B1K_u9mh-aXkdH@mail.gmail.com>
References: <AANLkTimx8NcaF4iUz2aq0gPZC8j_Z-B1K_u9mh-aXkdH@mail.gmail.com>
Message-ID: <192A873B-A276-4B03-A08D-E3C52CC2297D@freeswitch.org>

I replied and gave you the answer to your ACL question.

/b

On Jan 3, 2011, at 11:40 AM, Tim King wrote:

> This is a test as I do not seem to be getting any messages from the list. Can someone please reply and let me know I can can get messages working again. They are not going to spam.
> 
> Thanks
> 
> Tim
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Mon Jan  3 21:30:07 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Jan 2011 10:30:07 -0800
Subject: [Freeswitch-users] voicmeail operator transfer
In-Reply-To: <073D091BA0994867A629B371BEEFCA68@e1705>
References: <073D091BA0994867A629B371BEEFCA68@e1705>
Message-ID: <AANLkTi=Kd4Eyr5dcxRj2jv6aXHP3mc+t5Qe-wi4t=_Rm@mail.gmail.com>

In the time that it took to type this email you could have thrown an info
dump in your dialplan and  answered your own question! :)
-MC

On Sat, Jan 1, 2011 at 12:13 PM, Madovsky <infos at madovsky.org> wrote:

>  in voicemail.conf.xml I have this
>
>       <param name="operator-key" value="9"/>
>       <param name="operator-extension" value="conference XML features"/>
> it works well but are all channel variables transferred also ?
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Jan  3 21:32:53 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Jan 2011 10:32:53 -0800
Subject: [Freeswitch-users] no ringback tone
In-Reply-To: <AANLkTinbd06YgZFWue=eQGG9A5JO2a550tJ+bo-yG2yv@mail.gmail.com>
References: <AANLkTinbd06YgZFWue=eQGG9A5JO2a550tJ+bo-yG2yv@mail.gmail.com>
Message-ID: <AANLkTinA46HPKmk_pWaAiMpy-hdihc7STfY9_FuNkthx@mail.gmail.com>

Can you clarify what the actual issue is?
-MC

On Sun, Jan 2, 2011 at 8:30 PM, Sam <u2nsam at gmail.com> wrote:

> Hi All,
>
> happy new you to you !
>
> using a sangoma card and when dialing a mobile number which is having
> ringback tune / caller tune ;
> but a plain ring is heard to the user dialing that mobile through the
> trunk.
>
> I am using below syntax to dial out:-
> <action application="bridge"
> data="{monitor_early_media_ring_total=3}freetdm/wp1/a/${destination_number}"/>
>
> Any suggestions.
>
> Regards
> Sam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Jan  3 21:37:18 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Jan 2011 10:37:18 -0800
Subject: [Freeswitch-users] multi company
In-Reply-To: <AANLkTikjjhHMMxKmsEGpDYxXBpozjNycdimV6zZFmhL7@mail.gmail.com>
References: <AANLkTikjjhHMMxKmsEGpDYxXBpozjNycdimV6zZFmhL7@mail.gmail.com>
Message-ID: <AANLkTikW=LmuBhtLZ3Kxqhaqx95oiz4VTs3EfFU2tNVh@mail.gmail.com>

Just for confirmation, have you looked at these two pages?
http://wiki.freeswitch.org/wiki/Multi-tenant
http://wiki.freeswitch.org/wiki/Multiple_Domains

There is quite a bit of information for you to try.
-MC

On Mon, Jan 3, 2011 at 2:37 AM, Sam <u2nsam at gmail.com> wrote:

> Hi,
>
> Was using multi company setup,
>
> it gave an error while using below syntax
>
> <domain name="xyz">
>
> </domain>
>
> Cannot Initialize [[error near line 2868]: unexpected closing tag
> </domain>]
>
> And when i remove , it dont gives an error .
> the file is in directory/xyz.xml
>
>
> same happens on freeswitch.xml file
>
> when i remove the syntax
>
> <section name="directory" description="User Directory">
>     <X-PRE-PROCESS cmd="include" data="directory/*.xml"/>
>   </section>
>
> it works when removed the section other wise give error as unexpected
> closing tag </section>]
>
> Regds
> Sam
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Mon Jan  3 21:41:42 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 3 Jan 2011 13:41:42 -0500
Subject: [Freeswitch-users] voicmeail operator transfer
References: <073D091BA0994867A629B371BEEFCA68@e1705>
	<AANLkTi=Kd4Eyr5dcxRj2jv6aXHP3mc+t5Qe-wi4t=_Rm@mail.gmail.com>
Message-ID: <2DFC4EDE79F245A7B72A01A174E2C1C4@e1705>

yes ;), ti works now after hundreds different tests and ideas...

  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 03, 2011 1:30 PM
  Subject: Re: [Freeswitch-users] voicmeail operator transfer


  In the time that it took to type this email you could have thrown an info dump in your dialplan and  answered your own question! :)
  -MC


  On Sat, Jan 1, 2011 at 12:13 PM, Madovsky <infos at madovsky.org> wrote:

    in voicemail.conf.xml I have this

          <param name="operator-key" value="9"/>
          <param name="operator-extension" value="conference XML features"/>

    it works well but are all channel variables transferred also ?

    Thanks

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org






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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From msc at freeswitch.org  Mon Jan  3 21:50:51 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Jan 2011 10:50:51 -0800
Subject: [Freeswitch-users] voicmeail operator transfer
In-Reply-To: <2DFC4EDE79F245A7B72A01A174E2C1C4@e1705>
References: <073D091BA0994867A629B371BEEFCA68@e1705>
	<AANLkTi=Kd4Eyr5dcxRj2jv6aXHP3mc+t5Qe-wi4t=_Rm@mail.gmail.com>
	<2DFC4EDE79F245A7B72A01A174E2C1C4@e1705>
Message-ID: <AANLkTik0MiC5GkeN6endvkqokJJ4uvFkZe+stp4JFLfQ@mail.gmail.com>

And I'm sure you updated the wiki to help the next person who has this
question... ;)
-MC

On Mon, Jan 3, 2011 at 10:41 AM, Madovsky <infos at madovsky.org> wrote:

>  yes ;), ti works now after hundreds different tests and ideas...
>
>
> ----- Original Message -----
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, January 03, 2011 1:30 PM
> *Subject:* Re: [Freeswitch-users] voicmeail operator transfer
>
> In the time that it took to type this email you could have thrown an info
> dump in your dialplan and  answered your own question! :)
> -MC
>
> On Sat, Jan 1, 2011 at 12:13 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  in voicemail.conf.xml I have this
>>
>>       <param name="operator-key" value="9"/>
>>       <param name="operator-extension" value="conference XML features"/>
>> it works well but are all channel variables transferred also ?
>>
>> Thanks
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Jan  3 22:03:46 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Jan 2011 11:03:46 -0800
Subject: [Freeswitch-users] increase Max session
In-Reply-To: <AANLkTinyUR2MKrsU9Uc-AH4Go=EvQCu=8+PjgghXasrT@mail.gmail.com>
References: <AANLkTik=Fa3ZAX5XP_ubZ7ekrx8M8A7D5J0L=fqepEUg@mail.gmail.com>
	<AANLkTimZxTYeGGNnSKF2x07wS6mBhU5kyB6xbeTKcAmc@mail.gmail.com>
	<AANLkTin+uvhAM58ODfz=6jcJsuwDqcNzysEpLO7yXVfc@mail.gmail.com>
	<8825DCE39E834012ACD4A4978DFA9AB7@e1705>
	<AANLkTimTQ2mjJ+mGQ+kozTbgfE0qtgR-j32sq_oxYNtd@mail.gmail.com>
	<AANLkTi=YR=LxqLy+PzjO_NTiRf+-cAVQ0D=yguZkUw_P@mail.gmail.com>
	<AANLkTint0ypRO68sRdeASqy5s4OC38xK8SPeMXigm0_9@mail.gmail.com>
	<AANLkTi=2C6vOJiHWK8r2EdfdjBhca1_tYUXAZ1_dCpK0@mail.gmail.com>
	<AANLkTik9AiZ0GpmtKENmO_Y15rvxyDqm+Ga7Ryyt8vML@mail.gmail.com>
	<41DEB133-E2DD-4DC5-B6DB-320CD2C6BCBA@freeswitch.org>
	<AANLkTinyUR2MKrsU9Uc-AH4Go=EvQCu=8+PjgghXasrT@mail.gmail.com>
Message-ID: <AANLkTinYbtjCPbz5whwjYA6jxqjmA4XSUsM4YMVT7RcP@mail.gmail.com>

FYI, I started a new wiki page:
http://wiki.freeswitch.org/wiki/Virtualization

It is quite bare, so if you have experience with doing virt's w/ FS please
add your knowledge and experience.

Thanks,
MC

On Mon, Jan 3, 2011 at 7:49 AM, dome at tel.co.th <dome at tel.co.th> wrote:

> Thanks brian.
> proxmox default config give me 1024 process num :(
>
>
> Dome C.
>
> 2011/1/3 Brian West <brian at freeswitch.org>:
> > Your container doesn't have enough resources look at the UB and see what
> is
> > failing and increase accordingly.
> > /b
> > On Jan 1, 2011, at 9:39 PM, dome at tel.co.th wrote:
> >
> > 64 bit over openvz
> >
> > 2011/1/2 Aloysius Lloyd <lloyd.aloysius at sunteltech.ca>:
> >
> > 32 bit or 64 bit ?
> >
> >
> > On Sat, Jan 1, 2011 at 10:12 PM, dome at tel.co.th <dome at tel.co.th> wrote:
> >
> > 2011/1/2 Aloysius Lloyd <lloyd.aloysius at gmail.com>:
> >
> > make sure the switch.conf.xml have the right value. Did you rotate the
> >
> > log
> >
> > file between 22 min and 37 min.
> >
> > Yes
> >
> > try the cli command fsctl max_session 5000 see what happen in couple of
> >
> > hours.
> >
> > i try fsctl max_session 5000
> >
> > after that FS switch back to 921 again
> >
> > Thanks
> >
> > Lloyd
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Jan  3 22:13:07 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Jan 2011 11:13:07 -0800
Subject: [Freeswitch-users] ACL Problem
In-Reply-To: <AANLkTimi0HhWu0cuQXdsZm1kCCx4t-OXW0vSrsoW3==Q@mail.gmail.com>
References: <AANLkTimi0HhWu0cuQXdsZm1kCCx4t-OXW0vSrsoW3==Q@mail.gmail.com>
Message-ID: <AANLkTi=_ozPpurMQBcOFT1XNVwryFT1055NXchL6vTRP@mail.gmail.com>

Per Brian, change this:

<node type="allow" domain="192.168.0.250/32"/>

to this:
<node type="allow" cidr="192.168.0.250/32"/>

That should fix you up nicely.
-MC

On Mon, Jan 3, 2011 at 6:39 AM, Tim King <tim at compnetwork.net> wrote:

> *I am getting this error sending calls from my Kamailio box on
> 192.168.0.250.*
>
> 2011-01-03 14:22:49.930896 [WARNING] sofia.c:6369 IP 192.168.0.250 Rejected
> by acl "domains"
>
> *Here is the contents of my ACL:*
>
> <configuration name="acl.conf" description="Network Lists">
>   <network-lists>
>
>     <list name="lan" default="allow">
>       <node type="allow" cidr="192.168.0.250/32"/>
>     </list>
>
>     <list name="domains" default="deny">
>         <node type="allow" domain="$${domain}"/>
>         <node type="allow" domain="192.168.0.250/32"/>
>     </list>
>
>   </network-lists>
> </configuration>
>
> *I do not understand why calls from this machine are being rejected.
> Kamailio and Freeswitch are on the same machine sharing the same IP. Any
> ideas what I am missing?*
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Mon Jan  3 22:14:23 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 3 Jan 2011 14:14:23 -0500
Subject: [Freeswitch-users] voicmeail operator transfer
References: <073D091BA0994867A629B371BEEFCA68@e1705><AANLkTi=Kd4Eyr5dcxRj2jv6aXHP3mc+t5Qe-wi4t=_Rm@mail.gmail.com><2DFC4EDE79F245A7B72A01A174E2C1C4@e1705>
	<AANLkTik0MiC5GkeN6endvkqokJJ4uvFkZe+stp4JFLfQ@mail.gmail.com>
Message-ID: <DC296868E2B740A7B9E83A51F16ECD60@e1705>

argh, not sure it will help as
I use operator-extension as a strange way ;)
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 03, 2011 1:50 PM
  Subject: Re: [Freeswitch-users] voicmeail operator transfer


  And I'm sure you updated the wiki to help the next person who has this question... ;)
  -MC


  On Mon, Jan 3, 2011 at 10:41 AM, Madovsky <infos at madovsky.org> wrote:

    yes ;), ti works now after hundreds different tests and ideas...

      ----- Original Message ----- 
      From: Michael Collins 
      To: FreeSWITCH Users Help 
      Sent: Monday, January 03, 2011 1:30 PM
      Subject: Re: [Freeswitch-users] voicmeail operator transfer


      In the time that it took to type this email you could have thrown an info dump in your dialplan and  answered your own question! :) 
      -MC


      On Sat, Jan 1, 2011 at 12:13 PM, Madovsky <infos at madovsky.org> wrote:

        in voicemail.conf.xml I have this

              <param name="operator-key" value="9"/>
              <param name="operator-extension" value="conference XML features"/>

        it works well but are all channel variables transferred also ?

        Thanks

        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org






--------------------------------------------------------------------------


      _______________________________________________
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      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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    _______________________________________________
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    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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------------------------------------------------------------------------------


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Mon Jan  3 22:21:29 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 3 Jan 2011 14:21:29 -0500
Subject: [Freeswitch-users] mode_nibblebill and mod_conference
Message-ID: <B18C387D0F5C4A0798B78CAC5087B14F@e1705>

Is nibblebill works with conference ?
I tried to set nibble_account and nibble_rate without success

Thanks
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From infos at madovsky.org  Mon Jan  3 22:30:42 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 3 Jan 2011 14:30:42 -0500
Subject: [Freeswitch-users] mode_nibblebill and mod_conference
Message-ID: <1B18EBDDB6044354A0ACF963C987DDAF@e1705>

in mod_nibblebill wiki page
  a.. Q: Can you bill based on a multi-call B-Leg, where you have, say a conference call with multiple people on it? 
    a.. Yes, see answer above re: B-Leg and not A-Leg. 
but if I have this in my dialplan

    <extension name="conference">
        <condition field="destination_number" expression="^conference$">
                <action application="export" data="nolocal:nibble_account=${dest_nibble_account}"/>
                <action application="export" data="nolocal:nibble_rate=0.03"/>
                <action application="sleep" data="3000"/>
                <action application="answer"/>
                <!-- <action application="speak" data="cepstral|cepstral|connecting your call to the conference"/> -->
                <action application="conference" data="${dest_nibble_account}-$${domain}@default+${dest_conf_pin}"/>
        </condition>
    </extension>

it doesn't work



  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, January 03, 2011 2:21 PM
  Subject: mode_nibblebill and mod_conference


  Is nibblebill works with conference ?
  I tried to set nibble_account and nibble_rate without success

  Thanks
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From msc at freeswitch.org  Mon Jan  3 22:31:57 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Jan 2011 11:31:57 -0800
Subject: [Freeswitch-users] calling card app
In-Reply-To: <COL122-W17F8F5B93182F2AFF31380B5060@phx.gbl>
References: <COL122-W17F8F5B93182F2AFF31380B5060@phx.gbl>
Message-ID: <AANLkTinrNZiP_czOU1Nn_DDzd+EQ4mW-S4xCZ2B_vez4@mail.gmail.com>

If you have a powerful machine then you can probably scale to several
hundred concurrent calls, depending on transcoding, call recording needs,
etc. We've seen some boxes that can handle literally thousands of concurrent
calls, but the scenarios are never exactly the same. Also, Lua is very
lightweight, so if you're using it just to capture a PIN code or something
then you should be okay. Just be sure to exit the Lua script and let the
dialplan handle the bridge app. (See chapter 7 of the FS book for more
information on Lua scripting tips.)

-MC

On Sun, Jan 2, 2011 at 11:13 AM, Rafqat . <rafonline at hotmail.com> wrote:

>  Hi
>
> As stated in some of my previous posts, I am writing a calling card system
> (not too sure of potential number of concurrent users yet).
>
> At the moment I am simply doing everything in a single lua script utilising
> mod_lcr and mod_nibble.  It seems to work ok, but I have not stress tested
> it yet.
>
> I was wondering (at a high level) if this will suffice or should I be
> offloading operations such as pin validation and credit checking to another
> server (maybe utilising mod_rad_auth?).
>
> Cheers
>
> Raf
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Jan  3 23:06:30 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Jan 2011 12:06:30 -0800
Subject: [Freeswitch-users] tone_detect and dinging
In-Reply-To: <4063AE799A58452F8CAFF7D38FA07C03@e1705>
References: <C5A8430009F1461592CDA253B4F64DC9@e1705>
	<23BB6F88-A7A8-4A84-A325-D4974B10BD41@gmail.com>
	<EE2EAA0A62754F2186DC0BC81EC34DFB@e1705>
	<AANLkTik7HWHgUZ9B3GogE-59twhyiAVMAUdviBY-QMvC@mail.gmail.com>
	<AANLkTi=gh6YHnaa_XcY=-AjPpLiZ4nzXHtW+3bmaJ_Ft@mail.gmail.com>
	<4063AE799A58452F8CAFF7D38FA07C03@e1705>
Message-ID: <AANLkTi=U2vcH8p=+eks0Vyf_REVi9CR5-zgaahC_5Cuv@mail.gmail.com>

FYI, I added these to the wiki, including the main chan vars page (under
section #12):
http://wiki.freeswitch.org/wiki/Variable_execute_on_media
http://wiki.freeswitch.org/wiki/Variable_execute_on_preanswer

-MC

On Sat, Jan 1, 2011 at 11:12 AM, Madovsky <infos at madovsky.org> wrote:

>  it works well now, thanks Tony.
> I will try to add some wiki lines for execute_on_media and
> execute_on_preanswer,
> there are not on channel variables page
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Saturday, January 01, 2011 12:46 AM
> *Subject:* Re: [Freeswitch-users] tone_detect and dinging
>
> It pre answers early media because it requires media to work. Activate it
> in execute_on_media instead
> On Dec 31, 2010 5:53 PM, "Madovsky" <infos at madovsky.org> wrote:
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Jan  3 23:13:50 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Jan 2011 12:13:50 -0800
Subject: [Freeswitch-users] Sangoma Cards or FXS gateway for high port
 density setup (~ 160 fxs)
In-Reply-To: <AANLkTinQ5VUf7nt7XY7ZBPELm+Qxg9MQFgRzwCLY1es2@mail.gmail.com>
References: <AANLkTinQ5VUf7nt7XY7ZBPELm+Qxg9MQFgRzwCLY1es2@mail.gmail.com>
Message-ID: <AANLkTikEZjtmjZ5RGPHVjLEtSMoZRcSrtBaEWe76xcXm@mail.gmail.com>

What is the application for 160 FXS ports? Are you hooking up a hotel or
something? I would think the application would determine what level of HA
you need. In some cases having a warm standby on site would be sufficient.
In the case of a failure of box 1 you could have someone go to the switch
room and fire up box 2. This works in cases where a minute or two of
downtime would be inconvenient but not a show-stopper.

-MC

On Mon, Jan 3, 2011 at 1:25 AM, guru singh <grsingh750 at gmail.com> wrote:

> Hi,
>
> Happy New Year!
>
> What would you recommend for a solution that requires ~160 fxs ports?
> Should I get a Sangoma A200, in fact 7 of them and put them on to my
> server or look for other products like Audiocodes/Patton fxs gateways
> etc?
> If I choose Sangoma cards, then I don't think I can set up a HA
> scenario since, all fxs ports will be on the FS box itself.
> But from what I've read people have horrible things to say about most
> gateway devices. They are prone to failure and the support is also
> very lax.
> So what good would be HA anyway, if one of these gateways fails and
> knocks 32 fxs ports off!
> Also I see that Sangoma is actively involved with the FreeSwitch
> project and digging through the mailing lists, I could see great
> support as well.
> Is it advisable to have 6-7 cards with 12 fxs modules each on my FS server?
>
> Any suggestions/pointers/heads-up would be really appreciated.
>
> Thanks
> guru
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Jan  3 23:22:08 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Jan 2011 12:22:08 -0800
Subject: [Freeswitch-users] mode_nibblebill and mod_conference
In-Reply-To: <1B18EBDDB6044354A0ACF963C987DDAF@e1705>
References: <1B18EBDDB6044354A0ACF963C987DDAF@e1705>
Message-ID: <AANLkTik0azNgbcMX4qD4obfj9-GomR=MCVGOPiWsSdiO@mail.gmail.com>

According to the wiki (and page 251 of the FS book, which has this exact
same example) you need something like this:

<action application="bridge"
data="{enable_heartbeat_events=5,nibble_rate=1,nibble_account=0838833133}sofia/external/$1 at tel.co.th"/>

The link is
http://wiki.freeswitch.org/wiki/Mod_nibblebill#Bill_base_on_B_leg_Only

In your example I don't know that you necessarily have a b-leg. How are you
getting the call to the conference? Do you use originate? Or is it an
inbound call?
-MC

On Mon, Jan 3, 2011 at 11:30 AM, Madovsky <infos at madovsky.org> wrote:

>  in mod_nibblebill wiki page
>
>    - Q: Can you bill based on a multi-call B-Leg, where you have, say a
>    conference call with multiple people on it?
>       - Yes, see answer above re: B-Leg and not A-Leg.
>
> but if I have this in my dialplan
>
>     <extension name="conference">
>         <condition field="destination_number" expression="^conference$">
>                 <action application="export"
> data="nolocal:nibble_account=${dest_nibble_account}"/>
>                 <action application="export"
> data="nolocal:nibble_rate=0.03"/>
>                 <action application="sleep" data="3000"/>
>                 <action application="answer"/>
>                 <!-- <action application="speak"
> data="cepstral|cepstral|connecting your call to the conference"/> -->
>                 <action application="conference" data="
> ${dest_nibble_account}-$${domain}@default+${dest_conf_pin}"/<$%7Bdest_nibble_account%7D-$$%7Bdomain%7D at default+$%7Bdest_conf_pin%7D%22/>
> >
>         </condition>
>     </extension>
>
> it doesn't work
>
>
>  ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Monday, January 03, 2011 2:21 PM
> *Subject:* mode_nibblebill and mod_conference
>
> Is nibblebill works with conference ?
> I tried to set nibble_account and nibble_rate without success
>
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Mon Jan  3 23:23:49 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 3 Jan 2011 15:23:49 -0500
Subject: [Freeswitch-users] tone_detect and dinging
References: <C5A8430009F1461592CDA253B4F64DC9@e1705><23BB6F88-A7A8-4A84-A325-D4974B10BD41@gmail.com><EE2EAA0A62754F2186DC0BC81EC34DFB@e1705><AANLkTik7HWHgUZ9B3GogE-59twhyiAVMAUdviBY-QMvC@mail.gmail.com><AANLkTi=gh6YHnaa_XcY=-AjPpLiZ4nzXHtW+3bmaJ_Ft@mail.gmail.com><4063AE799A58452F8CAFF7D38FA07C03@e1705>
	<AANLkTi=U2vcH8p=+eks0Vyf_REVi9CR5-zgaahC_5Cuv@mail.gmail.com>
Message-ID: <1040DD803A534BB6A121FEDAD7590E36@e1705>

excellent Michael, thanks
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 03, 2011 3:06 PM
  Subject: Re: [Freeswitch-users] tone_detect and dinging


  FYI, I added these to the wiki, including the main chan vars page (under section #12):
  http://wiki.freeswitch.org/wiki/Variable_execute_on_media
  http://wiki.freeswitch.org/wiki/Variable_execute_on_preanswer


  -MC


  On Sat, Jan 1, 2011 at 11:12 AM, Madovsky <infos at madovsky.org> wrote:

    it works well now, thanks Tony.
    I will try to add some wiki lines for execute_on_media and execute_on_preanswer,
    there are not on channel variables page
      ----- Original Message ----- 
      From: Anthony Minessale 
      To: FreeSWITCH Users Help 
      Sent: Saturday, January 01, 2011 12:46 AM
      Subject: Re: [Freeswitch-users] tone_detect and dinging


      It pre answers early media because it requires media to work. Activate it in execute_on_media instead

      On Dec 31, 2010 5:53 PM, "Madovsky" <infos at madovsky.org> wrote:



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From infos at madovsky.org  Mon Jan  3 23:27:14 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 3 Jan 2011 15:27:14 -0500
Subject: [Freeswitch-users] tone_detect and dinging
References: <C5A8430009F1461592CDA253B4F64DC9@e1705><23BB6F88-A7A8-4A84-A325-D4974B10BD41@gmail.com><EE2EAA0A62754F2186DC0BC81EC34DFB@e1705><AANLkTik7HWHgUZ9B3GogE-59twhyiAVMAUdviBY-QMvC@mail.gmail.com><AANLkTi=gh6YHnaa_XcY=-AjPpLiZ4nzXHtW+3bmaJ_Ft@mail.gmail.com><4063AE799A58452F8CAFF7D38FA07C03@e1705>
	<AANLkTi=U2vcH8p=+eks0Vyf_REVi9CR5-zgaahC_5Cuv@mail.gmail.com>
Message-ID: <C0E75FBA1F1B4A2F8B2866BF1C70D7CD@e1705>

just corrected your example in execute_on_media
you wrote execute_on_answer instead
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 03, 2011 3:06 PM
  Subject: Re: [Freeswitch-users] tone_detect and dinging


  FYI, I added these to the wiki, including the main chan vars page (under section #12):
  http://wiki.freeswitch.org/wiki/Variable_execute_on_media
  http://wiki.freeswitch.org/wiki/Variable_execute_on_preanswer


  -MC


  On Sat, Jan 1, 2011 at 11:12 AM, Madovsky <infos at madovsky.org> wrote:

    it works well now, thanks Tony.
    I will try to add some wiki lines for execute_on_media and execute_on_preanswer,
    there are not on channel variables page
      ----- Original Message ----- 
      From: Anthony Minessale 
      To: FreeSWITCH Users Help 
      Sent: Saturday, January 01, 2011 12:46 AM
      Subject: Re: [Freeswitch-users] tone_detect and dinging


      It pre answers early media because it requires media to work. Activate it in execute_on_media instead

      On Dec 31, 2010 5:53 PM, "Madovsky" <infos at madovsky.org> wrote:



--------------------------------------------------------------------------


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From infos at madovsky.org  Mon Jan  3 23:29:22 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 3 Jan 2011 15:29:22 -0500
Subject: [Freeswitch-users] mode_nibblebill and mod_conference
References: <1B18EBDDB6044354A0ACF963C987DDAF@e1705>
	<AANLkTik0azNgbcMX4qD4obfj9-GomR=MCVGOPiWsSdiO@mail.gmail.com>
Message-ID: <033E6A69A4574C79806F2B36D262D80A@e1705>

Yes I know, so I need to change
<action application="conference" data="${dest_nibble_account}-$${domain}@default+${dest_conf_pin}"/>
in a loopback bridge ?
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 03, 2011 3:22 PM
  Subject: Re: [Freeswitch-users] mode_nibblebill and mod_conference


  According to the wiki (and page 251 of the FS book, which has this exact same example) you need something like this:


<action application="bridge" data="{enable_heartbeat_events=5,nibble_rate=1,nibble_account=0838833133}sofia/external/$1 at tel.co.th"/>The link is http://wiki.freeswitch.org/wiki/Mod_nibblebill#Bill_base_on_B_leg_Only


  In your example I don't know that you necessarily have a b-leg. How are you getting the call to the conference? Do you use originate? Or is it an inbound call?
  -MC


  On Mon, Jan 3, 2011 at 11:30 AM, Madovsky <infos at madovsky.org> wrote:

    in mod_nibblebill wiki page
      a.. Q: Can you bill based on a multi-call B-Leg, where you have, say a conference call with multiple people on it? 
        a.. Yes, see answer above re: B-Leg and not A-Leg. 
    but if I have this in my dialplan

        <extension name="conference">
            <condition field="destination_number" expression="^conference$">
                    <action application="export" data="nolocal:nibble_account=${dest_nibble_account}"/>
                    <action application="export" data="nolocal:nibble_rate=0.03"/>
                    <action application="sleep" data="3000"/>
                    <action application="answer"/>
                    <!-- <action application="speak" data="cepstral|cepstral|connecting your call to the conference"/> -->
                    <action application="conference" data="${dest_nibble_account}-$${domain}@default+${dest_conf_pin}"/>
            </condition>
        </extension>

    it doesn't work



      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Monday, January 03, 2011 2:21 PM
      Subject: mode_nibblebill and mod_conference


      Is nibblebill works with conference ?
      I tried to set nibble_account and nibble_rate without success

      Thanks

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Mon Jan  3 23:30:47 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 3 Jan 2011 15:30:47 -0500
Subject: [Freeswitch-users] mode_nibblebill and mod_conference
References: <1B18EBDDB6044354A0ACF963C987DDAF@e1705>
	<AANLkTik0azNgbcMX4qD4obfj9-GomR=MCVGOPiWsSdiO@mail.gmail.com>
Message-ID: <2AC2BC2E833247F5B13841D484A31AE0@e1705>

> In your example I don't know that you necessarily have a b-leg. How are you getting the call to the conference? Do you use originate? Or is it an inbound call?

it can be inbound as outbound call. I originate the call from default.xml and public.xml

thanks
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 03, 2011 3:22 PM
  Subject: Re: [Freeswitch-users] mode_nibblebill and mod_conference


  According to the wiki (and page 251 of the FS book, which has this exact same example) you need something like this:


<action application="bridge" data="{enable_heartbeat_events=5,nibble_rate=1,nibble_account=0838833133}sofia/external/$1 at tel.co.th"/>The link is http://wiki.freeswitch.org/wiki/Mod_nibblebill#Bill_base_on_B_leg_Only


  In your example I don't know that you necessarily have a b-leg. How are you getting the call to the conference? Do you use originate? Or is it an inbound call?
  -MC


  On Mon, Jan 3, 2011 at 11:30 AM, Madovsky <infos at madovsky.org> wrote:

    in mod_nibblebill wiki page
      a.. Q: Can you bill based on a multi-call B-Leg, where you have, say a conference call with multiple people on it? 
        a.. Yes, see answer above re: B-Leg and not A-Leg. 
    but if I have this in my dialplan

        <extension name="conference">
            <condition field="destination_number" expression="^conference$">
                    <action application="export" data="nolocal:nibble_account=${dest_nibble_account}"/>
                    <action application="export" data="nolocal:nibble_rate=0.03"/>
                    <action application="sleep" data="3000"/>
                    <action application="answer"/>
                    <!-- <action application="speak" data="cepstral|cepstral|connecting your call to the conference"/> -->
                    <action application="conference" data="${dest_nibble_account}-$${domain}@default+${dest_conf_pin}"/>
            </condition>
        </extension>

    it doesn't work



      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Monday, January 03, 2011 2:21 PM
      Subject: mode_nibblebill and mod_conference


      Is nibblebill works with conference ?
      I tried to set nibble_account and nibble_rate without success

      Thanks

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From grsingh750 at gmail.com  Mon Jan  3 23:39:38 2011
From: grsingh750 at gmail.com (guru singh)
Date: Tue, 4 Jan 2011 02:09:38 +0530
Subject: [Freeswitch-users] Sangoma Cards or FXS gateway for high port
 density setup (~ 160 fxs)
In-Reply-To: <AANLkTikEZjtmjZ5RGPHVjLEtSMoZRcSrtBaEWe76xcXm@mail.gmail.com>
References: <AANLkTinQ5VUf7nt7XY7ZBPELm+Qxg9MQFgRzwCLY1es2@mail.gmail.com>
	<AANLkTikEZjtmjZ5RGPHVjLEtSMoZRcSrtBaEWe76xcXm@mail.gmail.com>
Message-ID: <AANLkTikDdNd-GDdb-C9wMrJf-vmhBpcYVbT9yrRikPJg@mail.gmail.com>

Yes it is a hotel install. Not that I'm hooking up one, but I met
somebody who's a 'system architect' and has been putting proprietary
systems for clients for a while. He got interested and asked me for a
hardware sizing so to speak, if I was to do it my way.
For HA, having a standby machine seems more than sufficient. For ~160
fxs, I think  gateways (Audiocodes MP124?) would be a better idea.
It'll save me doing the transcoding on the FS box as well. I hear the
MP124 is a beast to configure, but works just fine.

-guru

On Tue, Jan 4, 2011 at 1:43 AM, Michael Collins <msc at freeswitch.org> wrote:
> What is the application for 160 FXS ports? Are you hooking up a hotel or
> something? I would think the application would determine what level of HA
> you need. In some cases having a warm standby on site would be sufficient.
> In the case of a failure of box 1 you could have someone go to the switch
> room and fire up box 2. This works in cases where a minute or two of
> downtime would be inconvenient but not a show-stopper.
> -MC
>
> On Mon, Jan 3, 2011 at 1:25 AM, guru singh <grsingh750 at gmail.com> wrote:
>>
>> Hi,
>>
>> Happy New Year!
>>
>> What would you recommend for a solution that requires ~160 fxs ports?
>> Should I get a Sangoma A200, in fact 7 of them and put them on to my
>> server or look for other products like Audiocodes/Patton fxs gateways
>> etc?
>> If I choose Sangoma cards, then I don't think I can set up a HA
>> scenario since, all fxs ports will be on the FS box itself.
>> But from what I've read people have horrible things to say about most
>> gateway devices. They are prone to failure and the support is also
>> very lax.
>> So what good would be HA anyway, if one of these gateways fails and
>> knocks 32 fxs ports off!
>> Also I see that Sangoma is actively involved with the FreeSwitch
>> project and digging through the mailing lists, I could see great
>> support as well.
>> Is it advisable to have 6-7 cards with 12 fxs modules each on my FS
>> server?
>>
>> Any suggestions/pointers/heads-up would be really appreciated.
>>
>> Thanks
>> guru
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Mon Jan  3 23:45:19 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Jan 2011 12:45:19 -0800
Subject: [Freeswitch-users] Sangoma Cards or FXS gateway for high port
 density setup (~ 160 fxs)
In-Reply-To: <AANLkTikDdNd-GDdb-C9wMrJf-vmhBpcYVbT9yrRikPJg@mail.gmail.com>
References: <AANLkTinQ5VUf7nt7XY7ZBPELm+Qxg9MQFgRzwCLY1es2@mail.gmail.com>
	<AANLkTikEZjtmjZ5RGPHVjLEtSMoZRcSrtBaEWe76xcXm@mail.gmail.com>
	<AANLkTikDdNd-GDdb-C9wMrJf-vmhBpcYVbT9yrRikPJg@mail.gmail.com>
Message-ID: <AANLkTini8umpUEtrd6pFE8kH6Z=V7jChiLon01i=7_Vy@mail.gmail.com>

AudioCodes are indeed beastly to configure, so be sure to save your config
once you get it working. :) My only experience with an audiocodes was that
it was a pain to get it working but once it worked it seemed to be okay. It
wasn't my box so I can't speak to the long-term viability of the AudioCodes.

-MC

On Mon, Jan 3, 2011 at 12:39 PM, guru singh <grsingh750 at gmail.com> wrote:

> Yes it is a hotel install. Not that I'm hooking up one, but I met
> somebody who's a 'system architect' and has been putting proprietary
> systems for clients for a while. He got interested and asked me for a
> hardware sizing so to speak, if I was to do it my way.
> For HA, having a standby machine seems more than sufficient. For ~160
> fxs, I think  gateways (Audiocodes MP124?) would be a better idea.
> It'll save me doing the transcoding on the FS box as well. I hear the
> MP124 is a beast to configure, but works just fine.
>
> -guru
>
> On Tue, Jan 4, 2011 at 1:43 AM, Michael Collins <msc at freeswitch.org>
> wrote:
> > What is the application for 160 FXS ports? Are you hooking up a hotel or
> > something? I would think the application would determine what level of HA
> > you need. In some cases having a warm standby on site would be
> sufficient.
> > In the case of a failure of box 1 you could have someone go to the switch
> > room and fire up box 2. This works in cases where a minute or two of
> > downtime would be inconvenient but not a show-stopper.
> > -MC
> >
> > On Mon, Jan 3, 2011 at 1:25 AM, guru singh <grsingh750 at gmail.com> wrote:
> >>
> >> Hi,
> >>
> >> Happy New Year!
> >>
> >> What would you recommend for a solution that requires ~160 fxs ports?
> >> Should I get a Sangoma A200, in fact 7 of them and put them on to my
> >> server or look for other products like Audiocodes/Patton fxs gateways
> >> etc?
> >> If I choose Sangoma cards, then I don't think I can set up a HA
> >> scenario since, all fxs ports will be on the FS box itself.
> >> But from what I've read people have horrible things to say about most
> >> gateway devices. They are prone to failure and the support is also
> >> very lax.
> >> So what good would be HA anyway, if one of these gateways fails and
> >> knocks 32 fxs ports off!
> >> Also I see that Sangoma is actively involved with the FreeSwitch
> >> project and digging through the mailing lists, I could see great
> >> support as well.
> >> Is it advisable to have 6-7 cards with 12 fxs modules each on my FS
> >> server?
> >>
> >> Any suggestions/pointers/heads-up would be really appreciated.
> >>
> >> Thanks
> >> guru
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
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From msc at freeswitch.org  Mon Jan  3 23:45:46 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Jan 2011 12:45:46 -0800
Subject: [Freeswitch-users] tone_detect and dinging
In-Reply-To: <C0E75FBA1F1B4A2F8B2866BF1C70D7CD@e1705>
References: <C5A8430009F1461592CDA253B4F64DC9@e1705>
	<23BB6F88-A7A8-4A84-A325-D4974B10BD41@gmail.com>
	<EE2EAA0A62754F2186DC0BC81EC34DFB@e1705>
	<AANLkTik7HWHgUZ9B3GogE-59twhyiAVMAUdviBY-QMvC@mail.gmail.com>
	<AANLkTi=gh6YHnaa_XcY=-AjPpLiZ4nzXHtW+3bmaJ_Ft@mail.gmail.com>
	<4063AE799A58452F8CAFF7D38FA07C03@e1705>
	<AANLkTi=U2vcH8p=+eks0Vyf_REVi9CR5-zgaahC_5Cuv@mail.gmail.com>
	<C0E75FBA1F1B4A2F8B2866BF1C70D7CD@e1705>
Message-ID: <AANLkTimpfwYH17OLtzTRKDA-gZaHTnOFzj36fys8A_Uk@mail.gmail.com>

thanks

On Mon, Jan 3, 2011 at 12:27 PM, Madovsky <infos at madovsky.org> wrote:

>  just corrected your example in execute_on_media
> you wrote execute_on_answer instead
>
> ----- Original Message -----
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, January 03, 2011 3:06 PM
> *Subject:* Re: [Freeswitch-users] tone_detect and dinging
>
> FYI, I added these to the wiki, including the main chan vars page (under
> section #12):
> http://wiki.freeswitch.org/wiki/Variable_execute_on_media
> http://wiki.freeswitch.org/wiki/Variable_execute_on_preanswer
>
> -MC
>
> On Sat, Jan 1, 2011 at 11:12 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  it works well now, thanks Tony.
>> I will try to add some wiki lines for execute_on_media and
>> execute_on_preanswer,
>> there are not on channel variables page
>>
>>  ----- Original Message -----
>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>  *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Sent:* Saturday, January 01, 2011 12:46 AM
>>  *Subject:* Re: [Freeswitch-users] tone_detect and dinging
>>
>>   It pre answers early media because it requires media to work. Activate
>> it in execute_on_media instead
>> On Dec 31, 2010 5:53 PM, "Madovsky" <infos at madovsky.org> wrote:
>>
>> ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
>
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From darren at aleph-com.net  Mon Jan  3 23:48:26 2011
From: darren at aleph-com.net (Darren Wiebe)
Date: Mon, 03 Jan 2011 13:48:26 -0700
Subject: [Freeswitch-users] Sangoma Cards or FXS gateway for high port
 density setup (~ 160 fxs)
In-Reply-To: <AANLkTini8umpUEtrd6pFE8kH6Z=V7jChiLon01i=7_Vy@mail.gmail.com>
References: <AANLkTinQ5VUf7nt7XY7ZBPELm+Qxg9MQFgRzwCLY1es2@mail.gmail.com>	<AANLkTikEZjtmjZ5RGPHVjLEtSMoZRcSrtBaEWe76xcXm@mail.gmail.com>	<AANLkTikDdNd-GDdb-C9wMrJf-vmhBpcYVbT9yrRikPJg@mail.gmail.com>
	<AANLkTini8umpUEtrd6pFE8kH6Z=V7jChiLon01i=7_Vy@mail.gmail.com>
Message-ID: <4D22361A.3@aleph-com.net>

That's been our experience with AudioCodes as well.  Configuration is a 
real pain but after that they just work.

Darren Wiebe
darren at aleph-com.net

On 03/01/2011 1:45 PM, Michael Collins wrote:
> AudioCodes are indeed beastly to configure, so be sure to save your 
> config once you get it working. :) My only experience with an 
> audiocodes was that it was a pain to get it working but once it worked 
> it seemed to be okay. It wasn't my box so I can't speak to the 
> long-term viability of the AudioCodes.
>
> -MC
>
> On Mon, Jan 3, 2011 at 12:39 PM, guru singh <grsingh750 at gmail.com 
> <mailto:grsingh750 at gmail.com>> wrote:
>
>     Yes it is a hotel install. Not that I'm hooking up one, but I met
>     somebody who's a 'system architect' and has been putting proprietary
>     systems for clients for a while. He got interested and asked me for a
>     hardware sizing so to speak, if I was to do it my way.
>     For HA, having a standby machine seems more than sufficient. For ~160
>     fxs, I think  gateways (Audiocodes MP124?) would be a better idea.
>     It'll save me doing the transcoding on the FS box as well. I hear the
>     MP124 is a beast to configure, but works just fine.
>
>     -guru
>
>     On Tue, Jan 4, 2011 at 1:43 AM, Michael Collins
>     <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>     > What is the application for 160 FXS ports? Are you hooking up a
>     hotel or
>     > something? I would think the application would determine what
>     level of HA
>     > you need. In some cases having a warm standby on site would be
>     sufficient.
>     > In the case of a failure of box 1 you could have someone go to
>     the switch
>     > room and fire up box 2. This works in cases where a minute or two of
>     > downtime would be inconvenient but not a show-stopper.
>     > -MC
>     >
>     > On Mon, Jan 3, 2011 at 1:25 AM, guru singh <grsingh750 at gmail.com
>     <mailto:grsingh750 at gmail.com>> wrote:
>     >>
>     >> Hi,
>     >>
>     >> Happy New Year!
>     >>
>     >> What would you recommend for a solution that requires ~160 fxs
>     ports?
>     >> Should I get a Sangoma A200, in fact 7 of them and put them on
>     to my
>     >> server or look for other products like Audiocodes/Patton fxs
>     gateways
>     >> etc?
>     >> If I choose Sangoma cards, then I don't think I can set up a HA
>     >> scenario since, all fxs ports will be on the FS box itself.
>     >> But from what I've read people have horrible things to say
>     about most
>     >> gateway devices. They are prone to failure and the support is also
>     >> very lax.
>     >> So what good would be HA anyway, if one of these gateways fails and
>     >> knocks 32 fxs ports off!
>     >> Also I see that Sangoma is actively involved with the FreeSwitch
>     >> project and digging through the mailing lists, I could see great
>     >> support as well.
>     >> Is it advisable to have 6-7 cards with 12 fxs modules each on my FS
>     >> server?
>     >>
>     >> Any suggestions/pointers/heads-up would be really appreciated.
>     >>
>     >> Thanks
>     >> guru
>     >>
>     >> _______________________________________________
>     >> FreeSWITCH-users mailing list
>     >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >> http://www.freeswitch.org
>     >
>     >
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     >
>     >
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From rafonline at hotmail.com  Tue Jan  4 00:07:33 2011
From: rafonline at hotmail.com (Rafqat .)
Date: Mon, 3 Jan 2011 21:07:33 +0000
Subject: [Freeswitch-users] calling card app
In-Reply-To: <AANLkTinrNZiP_czOU1Nn_DDzd+EQ4mW-S4xCZ2B_vez4@mail.gmail.com>
References: <COL122-W17F8F5B93182F2AFF31380B5060@phx.gbl>,
	<AANLkTinrNZiP_czOU1Nn_DDzd+EQ4mW-S4xCZ2B_vez4@mail.gmail.com>
Message-ID: <COL122-W48DBC4195FDE17C98363DCB5070@phx.gbl>



Thanks very much for the advice.

Much Appreciated.

Date: Mon, 3 Jan 2011 11:31:57 -0800
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] calling card app

If you have a powerful machine then you can probably scale to several hundred concurrent calls, depending on transcoding, call recording needs, etc. We've seen some boxes that can handle literally thousands of concurrent calls, but the scenarios are never exactly the same. Also, Lua is very lightweight, so if you're using it just to capture a PIN code or something then you should be okay. Just be sure to exit the Lua script and let the dialplan handle the bridge app. (See chapter 7 of the FS book for more information on Lua scripting tips.)

-MC

On Sun, Jan 2, 2011 at 11:13 AM, Rafqat . <rafonline at hotmail.com> wrote:






Hi

As stated in some of my previous posts, I am writing a calling card system (not too sure of potential number of concurrent users yet).

At the moment I am simply doing everything in a single lua script utilising mod_lcr and mod_nibble.  It seems to work ok, but I have not stress tested it yet.


I was wondering (at a high level) if this will suffice or should I be offloading operations such as pin validation and credit checking to another server (maybe utilising mod_rad_auth?).    

Cheers

Raf

 		 	   		  

_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org





_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From rafonline at hotmail.com  Tue Jan  4 00:17:48 2011
From: rafonline at hotmail.com (Rafqat .)
Date: Mon, 3 Jan 2011 21:17:48 +0000
Subject: [Freeswitch-users] calling card app
In-Reply-To: <COL122-W48DBC4195FDE17C98363DCB5070@phx.gbl>
References: <COL122-W17F8F5B93182F2AFF31380B5060@phx.gbl>,
	<AANLkTinrNZiP_czOU1Nn_DDzd+EQ4mW-S4xCZ2B_vez4@mail.gmail.com>,
	<COL122-W48DBC4195FDE17C98363DCB5070@phx.gbl>
Message-ID: <COL122-W46783EE0C5897D2647A606B5070@phx.gbl>



btw.

My lua script currently does the following:

1. Ask for PIN.
2. Gets funds for PIN (DB lookup using freeswitch.Dbh) and informs the user of the funds.
3. Ask for destination number.
4. Checks if enough funds (again using freeswitch Dbh).
5. Gets auto route using mod_lcr.
6. Populates leg b session variables necessary for mod_nibble.
7. Does the bridging.

As per your advice I will move the bridging stuff out of the script.  Do you think the rest will be ok in lua?

CHeers

Raf


From: rafonline at hotmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: RE: [Freeswitch-users] calling card app
Date: Mon, 3 Jan 2011 21:07:33 +0000









Thanks very much for the advice.

Much Appreciated.

Date: Mon, 3 Jan 2011 11:31:57 -0800
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] calling card app

If you have a powerful machine then you can probably scale to several hundred concurrent calls, depending on transcoding, call recording needs, etc. We've seen some boxes that can handle literally thousands of concurrent calls, but the scenarios are never exactly the same. Also, Lua is very lightweight, so if you're using it just to capture a PIN code or something then you should be okay. Just be sure to exit the Lua script and let the dialplan handle the bridge app. (See chapter 7 of the FS book for more information on Lua scripting tips.)

-MC

On Sun, Jan 2, 2011 at 11:13 AM, Rafqat . <rafonline at hotmail.com> wrote:






Hi

As stated in some of my previous posts, I am writing a calling card system (not too sure of potential number of concurrent users yet).

At the moment I am simply doing everything in a single lua script utilising mod_lcr and mod_nibble.  It seems to work ok, but I have not stress tested it yet.


I was wondering (at a high level) if this will suffice or should I be offloading operations such as pin validation and credit checking to another server (maybe utilising mod_rad_auth?).    

Cheers

Raf

 		 	   		  

_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org





_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From grsingh750 at gmail.com  Tue Jan  4 00:19:33 2011
From: grsingh750 at gmail.com (guru singh)
Date: Tue, 4 Jan 2011 02:49:33 +0530
Subject: [Freeswitch-users] Sangoma Cards or FXS gateway for high port
 density setup (~ 160 fxs)
In-Reply-To: <4D22361A.3@aleph-com.net>
References: <AANLkTinQ5VUf7nt7XY7ZBPELm+Qxg9MQFgRzwCLY1es2@mail.gmail.com>
	<AANLkTikEZjtmjZ5RGPHVjLEtSMoZRcSrtBaEWe76xcXm@mail.gmail.com>
	<AANLkTikDdNd-GDdb-C9wMrJf-vmhBpcYVbT9yrRikPJg@mail.gmail.com>
	<AANLkTini8umpUEtrd6pFE8kH6Z=V7jChiLon01i=7_Vy@mail.gmail.com>
	<4D22361A.3@aleph-com.net>
Message-ID: <AANLkTi=HF4DbvWEcYHTJd3dcg_g8xqpDhdjiBypuy6qC@mail.gmail.com>

Thanks for your inputs.

On Tue, Jan 4, 2011 at 2:18 AM, Darren Wiebe <darren at aleph-com.net> wrote:
> That's been our experience with AudioCodes as well.? Configuration is a real
> pain but after that they just work.
>
> Darren Wiebe
> darren at aleph-com.net
>
> On 03/01/2011 1:45 PM, Michael Collins wrote:
>
> AudioCodes are indeed beastly to configure, so be sure to save your config
> once you get it working. :) My only experience with an audiocodes was that
> it was a pain to get it working but once it worked it seemed to be okay. It
> wasn't my box so I can't speak to the long-term viability of the AudioCodes.
> -MC
>
> On Mon, Jan 3, 2011 at 12:39 PM, guru singh <grsingh750 at gmail.com> wrote:
>>
>> Yes it is a hotel install. Not that I'm hooking up one, but I met
>> somebody who's a 'system architect' and has been putting proprietary
>> systems for clients for a while. He got interested and asked me for a
>> hardware sizing so to speak, if I was to do it my way.
>> For HA, having a standby machine seems more than sufficient. For ~160
>> fxs, I think ?gateways (Audiocodes MP124?) would be a better idea.
>> It'll save me doing the transcoding on the FS box as well. I hear the
>> MP124 is a beast to configure, but works just fine.
>>
>> -guru
>>
>> On Tue, Jan 4, 2011 at 1:43 AM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> > What is the application for 160 FXS ports? Are you hooking up a hotel or
>> > something? I would think the application would determine what level of
>> > HA
>> > you need. In some cases having a warm standby on site would be
>> > sufficient.
>> > In the case of a failure of box 1 you could have someone go to the
>> > switch
>> > room and fire up box 2. This works in cases where a minute or two of
>> > downtime would be inconvenient but not a show-stopper.
>> > -MC
>> >
>> > On Mon, Jan 3, 2011 at 1:25 AM, guru singh <grsingh750 at gmail.com> wrote:
>> >>
>> >> Hi,
>> >>
>> >> Happy New Year!
>> >>
>> >> What would you recommend for a solution that requires ~160 fxs ports?
>> >> Should I get a Sangoma A200, in fact 7 of them and put them on to my
>> >> server or look for other products like Audiocodes/Patton fxs gateways
>> >> etc?
>> >> If I choose Sangoma cards, then I don't think I can set up a HA
>> >> scenario since, all fxs ports will be on the FS box itself.
>> >> But from what I've read people have horrible things to say about most
>> >> gateway devices. They are prone to failure and the support is also
>> >> very lax.
>> >> So what good would be HA anyway, if one of these gateways fails and
>> >> knocks 32 fxs ports off!
>> >> Also I see that Sangoma is actively involved with the FreeSwitch
>> >> project and digging through the mailing lists, I could see great
>> >> support as well.
>> >> Is it advisable to have 6-7 cards with 12 fxs modules each on my FS
>> >> server?
>> >>
>> >> Any suggestions/pointers/heads-up would be really appreciated.
>> >>
>> >> Thanks
>> >> guru
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From joaocarlosleme at gmail.com  Tue Jan  4 00:36:49 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Mon, 3 Jan 2011 13:36:49 -0800
Subject: [Freeswitch-users] Can't build on Windows 7 64bit C++ Express!?
In-Reply-To: <BC8E7AC3-4B6A-4759-A179-EE416455D314@visionutveckling.se>
References: <BC8E7AC3-4B6A-4759-A179-EE416455D314@visionutveckling.se>
Message-ID: <AANLkTi=Da+6DkPoNScoxx20Z7yOKE15XRemm8Exyxh4+@mail.gmail.com>

Hi Peter,
Thanks but I did disable autocrlf in git for sure. Any other ideas?
John

On Mon, Jan 3, 2011 at 9:12 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> Make sure to disable autocrlf in git, then it will work. Also make sure to
> read all instructions on the wiki.
>
> /Peter
>
> ----- Reply message -----
> Fr?n: "xuyan yang" <xyangni at gmail.com>
> Datum: m?n, jan 3, 2011 17:54
> Rubrik: [Freeswitch-users] Can't build on Windows 7 64bit C++ Express!?
> Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>
> Builiding freshly downloaded git version (git clone) on XP with VC2008
> Express also generated lots of errors and mod_sophia can not be build.
> I just tried to build the solution directly without modify any setting or
> choose any project.
>
>
>
> On Mon, Jan 3, 2011 at 4:09 AM, Jeff Lenk <jeff at jefflenk.com<mailto:
> jeff at jefflenk.com>> wrote:
> What target are you trying to build? Post a log of the build or pastebin
> reference
>
> Sent from my Windows Phone
> ________________________________
> From: Joao Leme
> Sent: Sunday, January 02, 2011 1:28 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Can't build on Windows 7 64bit C++
> Express!?
>
>
>
> > Ok Thanks. Any idea why all the errors and why i can't build? I've built
> > before on Windows Vista 64bit and VS2008Pro with no problems but can't
> get
> > it to work on Express edition.
> >
> > On Sat, Jan 1, 2011 at 9:51 PM, babak yakhchali
> ><babak.freeswitch at gmail.com<mailto:babak.freeswitch at gmail.com>>wrote:
> >
> >> you just need to build the project in
> >>
> >>
> D:\gitRepos\2010exp-freeswitch\freeswitch\src\mod\languages\mod_managed\managed
> >> using vc# express, others should just work using vc++
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
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> >>
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From msc at freeswitch.org  Tue Jan  4 00:41:57 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Jan 2011 13:41:57 -0800
Subject: [Freeswitch-users] calling card app
In-Reply-To: <COL122-W46783EE0C5897D2647A606B5070@phx.gbl>
References: <COL122-W17F8F5B93182F2AFF31380B5060@phx.gbl>
	<AANLkTinrNZiP_czOU1Nn_DDzd+EQ4mW-S4xCZ2B_vez4@mail.gmail.com>
	<COL122-W48DBC4195FDE17C98363DCB5070@phx.gbl>
	<COL122-W46783EE0C5897D2647A606B5070@phx.gbl>
Message-ID: <AANLkTikQizofgD4_CdWKDzuzRKWrB=U=F3UMMRNmsF6J@mail.gmail.com>

On Mon, Jan 3, 2011 at 1:17 PM, Rafqat . <rafonline at hotmail.com> wrote:

>
> btw.
>
> My lua script currently does the following:
>
> 1. Ask for PIN.
> 2. Gets funds for PIN (DB lookup using freeswitch.Dbh) and informs the user
> of the funds.
> 3. Ask for destination number.
> 4. Checks if enough funds (again using freeswitch Dbh).
> 5. Gets auto route using mod_lcr.
> 6. Populates leg b session variables necessary for mod_nibble.
> 7. Does the bridging.
>
> As per your advice I will move the bridging stuff out of the script.  Do
> you think the rest will be ok in lua?
>

I'd say that looks good. The Lua script only stays active until you get to
step 7 where you do a transfer instead of bridge. It's only a few seconds of
work.
-MC


>
> CHeers
>
> Raf
>
>
> ------------------------------
> From: rafonline at hotmail.com
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: RE: [Freeswitch-users] calling card app
> Date: Mon, 3 Jan 2011 21:07:33 +0000
>
>
>
> Thanks very much for the advice.
>
> Much Appreciated.
>
> ------------------------------
> Date: Mon, 3 Jan 2011 11:31:57 -0800
> From: msc at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] calling card app
>
> If you have a powerful machine then you can probably scale to several
> hundred concurrent calls, depending on transcoding, call recording needs,
> etc. We've seen some boxes that can handle literally thousands of concurrent
> calls, but the scenarios are never exactly the same. Also, Lua is very
> lightweight, so if you're using it just to capture a PIN code or something
> then you should be okay. Just be sure to exit the Lua script and let the
> dialplan handle the bridge app. (See chapter 7 of the FS book for more
> information on Lua scripting tips.)
>
> -MC
>
> On Sun, Jan 2, 2011 at 11:13 AM, Rafqat . <rafonline at hotmail.com> wrote:
>
>  Hi
>
> As stated in some of my previous posts, I am writing a calling card system
> (not too sure of potential number of concurrent users yet).
>
> At the moment I am simply doing everything in a single lua script utilising
> mod_lcr and mod_nibble.  It seems to work ok, but I have not stress tested
> it yet.
>
> I was wondering (at a high level) if this will suffice or should I be
> offloading operations such as pin validation and credit checking to another
> server (maybe utilising mod_rad_auth?).
>
> Cheers
>
> Raf
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From rafonline at hotmail.com  Tue Jan  4 01:02:11 2011
From: rafonline at hotmail.com (Rafqat .)
Date: Mon, 3 Jan 2011 22:02:11 +0000
Subject: [Freeswitch-users] calling card app
In-Reply-To: <AANLkTikQizofgD4_CdWKDzuzRKWrB=U=F3UMMRNmsF6J@mail.gmail.com>
References: <COL122-W17F8F5B93182F2AFF31380B5060@phx.gbl>,
	<AANLkTinrNZiP_czOU1Nn_DDzd+EQ4mW-S4xCZ2B_vez4@mail.gmail.com>,
	<COL122-W48DBC4195FDE17C98363DCB5070@phx.gbl>,
	<COL122-W46783EE0C5897D2647A606B5070@phx.gbl>,
	<AANLkTikQizofgD4_CdWKDzuzRKWrB=U=F3UMMRNmsF6J@mail.gmail.com>
Message-ID: <COL122-W21AE1E697F30571E3F72F8B5070@phx.gbl>



Thanks again for your help MC.

Date: Mon, 3 Jan 2011 13:41:57 -0800
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] calling card app



On Mon, Jan 3, 2011 at 1:17 PM, Rafqat . <rafonline at hotmail.com> wrote:







btw.

My lua script currently does the following:

1. Ask for PIN.
2. Gets funds for PIN (DB lookup using freeswitch.Dbh) and informs the user of the funds.
3. Ask for destination number.
4. Checks if enough funds (again using freeswitch Dbh).

5. Gets auto route using mod_lcr.
6. Populates leg b session variables necessary for mod_nibble.
7. Does the bridging.

As per your advice I will move the bridging stuff out of the script.  Do you think the rest will be ok in lua?


I'd say that looks good. The Lua script only stays active until you get to step 7 where you do a transfer instead of bridge. It's only a few seconds of work.-MC
 
CHeers

Raf


From: rafonline at hotmail.com

To: freeswitch-users at lists.freeswitch.org
Subject: RE: [Freeswitch-users] calling card app
Date: Mon, 3 Jan 2011 21:07:33 +0000










Thanks very much for the advice.

Much Appreciated.

Date: Mon, 3 Jan 2011 11:31:57 -0800
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] calling card app

If you have a powerful machine then you can probably scale to several hundred concurrent calls, depending on transcoding, call recording needs, etc. We've seen some boxes that can handle literally thousands of concurrent calls, but the scenarios are never exactly the same. Also, Lua is very lightweight, so if you're using it just to capture a PIN code or something then you should be okay. Just be sure to exit the Lua script and let the dialplan handle the bridge app. (See chapter 7 of the FS book for more information on Lua scripting tips.)


-MC

On Sun, Jan 2, 2011 at 11:13 AM, Rafqat . <rafonline at hotmail.com> wrote:







Hi

As stated in some of my previous posts, I am writing a calling card system (not too sure of potential number of concurrent users yet).

At the moment I am simply doing everything in a single lua script utilising mod_lcr and mod_nibble.  It seems to work ok, but I have not stress tested it yet.



I was wondering (at a high level) if this will suffice or should I be offloading operations such as pin validation and credit checking to another server (maybe utilising mod_rad_auth?).    

Cheers

Raf


 		 	   		  

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

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From wasim at convergence.pk  Tue Jan  4 04:12:34 2011
From: wasim at convergence.pk (Wasim Baig)
Date: Tue, 4 Jan 2011 06:12:34 +0500
Subject: [Freeswitch-users] libss7 * Sangoma
In-Reply-To: <AANLkTi=ok4KmurNX8Fha3Ne1yCSexTuDgJPV5f-M72t3@mail.gmail.com>
References: <AANLkTi=ok4KmurNX8Fha3Ne1yCSexTuDgJPV5f-M72t3@mail.gmail.com>
Message-ID: <AANLkTimo6C-Sjm7289t=O3qjNCKfyXFUVS-yZ+Bo2qqH@mail.gmail.com>

anita:

asterisk-ss7 is the correct list for this

change
TDMV_DCHAN      = 16
to
TDMV_DCHAN      = 0
in wanpipeX.conf

fwiw, chan_ss7 is nicer ...

-wasim

On Mon, Jan 3, 2011 at 22:57, Anita Hall <anita.hall at simmortel.com> wrote:

> Hi
>
> This question pertains libss7 with Sangoma A108 card on Asterisk. It does
> not concern freeswitch but I suppose Sangoma folks frequent this list more
> :)
>
> I am unable to make libss7 work with Sangoma. Here are the details.
>
> Could you please provide me some pointers ?
>
> Thanks,
> Anita.
>
> debian:~# uname -a
> Linux debian 2.6.26-2-686 #1 SMP Thu Nov 25 01:53:57 UTC 2010 i686
> GNU/Linux
>
> debian:~# dahdi_hardware
> pci:0000:05:04.0     wanpipe-     1923:0100 Sangoma Technologies Corp.
> A104d QUAD T1/E1 AFT card
> pci:0000:07:04.0     wanpipe-     1923:0100 Sangoma Technologies Corp.
> A104d QUAD T1/E1 AFT card
>
>
>
> debian:~# cat /etc/dahdi/system.conf
> #autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
> #autogenrated on 2011-01-03
> #Dahdi Channels Configurations
> #For detailed Dahdi options, view /etc/dahdi/system.conf.bak
> loadzone=us
> defaultzone=us
>
> #Sangoma A108 port 1 [slot:4 bus:5 span:1] <wanpipe1>
> span=1,1,0,ccs,hdb3,crc4
> bchan=1-15,17-31
> echocanceller=mg2,1-15,17-31
> mtp2=16
>
> #Sangoma A108 port 2 [slot:4 bus:5 span:2] <wanpipe2>
> span=2,1,0,ccs,hdb3,crc4
> bchan=32-46,48-62
> echocanceller=mg2,32-46,48-62
> mtp2=47
>
>
> debian:~# cat /etc/wanpipe/wanpipe1.conf
> #=============================
> ===================
> # WANPIPE1 Configuration File
> #================================================
> #
> # Date: Wed Dec  6 20:29:03 UTC 2006
> #
> # Note: This file was generated automatically
> #       by /usr/local/sbin/setup-sangoma program.
> #
> #       If you want to edit this file, it is
> #       recommended that you use wancfg program
> #       to do so.
> #================================================
> # Sangoma Technologies Inc.
> #================================================
>
> [devices]
> wanpipe1 = WAN_AFT_TE1, Comment
>
> [interfaces]
> w1g1 = wanpipe1, , TDM_VOICE, Comment
>
> [wanpipe1]
> CARD_TYPE       = AFT
> S514CPU         = A
> CommPort        = PRI
> AUTO_PCISLOT    = NO
> PCISLOT         = 4
> PCIBUS          = 5
> FE_MEDIA        = E1
> FE_LCODE        = HDB3
> FE_FRAME        = NCRC4
> FE_LINE         = 1
> TE_CLOCK        = NORMAL
> TE_REF_CLOCK    = 0
> TE_SIG_MODE     = CCS
> TE_HIGHIMPEDANCE        = NO
> LBO             = 120OH
> FE_TXTRISTATE   = NO
> MTU             = 1500
> UDPPORT         = 9000
> TTL             = 255
> IGNORE_FRONT_END = NO
> TDMV_SPAN       = 1
> TDMV_DCHAN      = 16
> TDMV_HW_DTMF    = NO
> TDMV_HW_FAX_DETECT = NO
>
> [w1g1]
> ACTIVE_CH       = ALL
> TDMV_HWEC       = NO
>
>
> debian:~# cat /etc/wanpipe/wanpipe2.conf
> #================================================
> # WANPIPE1 Configuration File
> #================================================
> #
> # Date: Wed Dec  6 20:29:03 UTC 2006
> #
> # Note: This file was generated automatically
> #       by /usr/local/sbin/setup-sangoma program.
> #
> #       If you want to edit this file, it is
> #       recommended that you use wancfg program
> #       to do so.
> #================================================
> # Sangoma Technologies Inc.
> #================================================
>
> [devices]
> wanpipe2 = WAN_AFT_TE1, Comment
>
> [interfaces]
> w2g1 = wanpipe2, , TDM_VOICE, Comment
>
> [wanpipe2]
> CARD_TYPE       = AFT
> S514CPU         = A
> CommPort        = PRI
> AUTO_PCISLOT    = NO
> PCISLOT         = 4
> PCIBUS          = 5
> FE_MEDIA        = E1
> FE_LCODE        = HDB3
> FE_FRAME        = NCRC4
> FE_LINE         = 2
> TE_CLOCK        = NORMAL
> TE_REF_CLOCK    = 0
> TE_SIG_MODE     = CCS
> TE_HIGHIMPEDANCE        = NO
> LBO             = 120OH
> FE_TXTRISTATE   = NO
> MTU             = 1500
> UDPPORT         = 9000
> TTL             = 255
> IGNORE_FRONT_END = NO
> TDMV_SPAN       = 2
> TDMV_DCHAN      = 16
> TDMV_HW_DTMF    = NO
> TDMV_HW_FAX_DETECT = NO
>
> [w2g1]
> ACTIVE_CH       = ALL
> TDMV_HWEC       = NO
>
> debian:~# cat /etc/asterisk/chan_dahdi.conf
> ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
> ;autogenrated on 2011-01-03
> ;Dahdi Channels Configurations
> ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak
>
> [trunkgroups]
>
> [channels]
> context=default
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> relaxdtmf=yes
> rxgain=0.0
> txgain=0.0
> group=1
> callgroup=1
> pickupgroup=1
> immediate=no
>
> ;Sangoma A108 port 1 [slot:4 bus:5 span:1] <wanpipe1>
> switchtype=euroisdn
> context=tata
> group=1
> echocancel=yes
> signaling=ss7                    ;this is ss7 signaling
> ss7type=itu                     ;using the ITU variant
> ss7_called_nai=dynamic           ;NAI for outgoing calls
> ss7_calling_nai=dynamic          ;NAI for incoming calls
> ss7_internationalprefix=00 ;international prefix value for incoming calls
> ss7_nationalprefix=0            ;national prefix value for incoming calls
> ss7_subscriberprefix=            ;subscriber prefix value for incoming
> calls
> ss7_unknownprefix=              ;unknown prefix value for incoming calls
> ss7_explictacm=yes               ;ACM is send as soon as call enters the
> dial plan...may not accepted yet though
> linkset=1                         ;arbitrary name for this set of channels
> pointcode=13323                      ;the point code for this system...aka
> SPC
> adjpointcode=12650                  ;the point code for the system that we
> are signaling to... aka APC
> defaultdpc=12650                     ;the point code for the system that
> the CICs will be negotiated with...aka DPC
> networkindicator=international  ;NI value for MTP3
> cicbeginswith=1                  ;the starting value of the CICs
> channel =>1-15
> cicbeginswith=17                 ;the starting value of the CICs
> channel =>17-31         ;the channels that are CICs
> sigchan=16                     ;the signaling channel
>
> ;Sangoma A108 port 2 [slot:4 bus:5 span:2] <wanpipe2>
> switchtype=euroisdn
> context=tata
> group=1
> echocancel=yes
> cicbeginswith=32                  ;the starting value of the CICs
> channel =>32-46
> cicbeginswith=48                  ;the starting value of the CICs
> channel =>48-62         ;the channels that are CICs
> sigchan=47                       ;the signaling channel
>
>
> debian:~# ifconfig w1g1
> w1g1      Link encap:Point-to-Point Protocol
>          UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
>          RX packets:772814 errors:0 dropped:0 overruns:0 frame:0
>          TX packets:772814 errors:0 dropped:0 overruns:3 carrier:0
>          collisions:0 txqueuelen:100
>          RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
>          Interrupt:16 Memory:f8d00000-f8d01fff
>
> debian:~# ifconfig w1g1
> w1g1      Link encap:Point-to-Point Protocol
>          UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
>          RX packets:774782 errors:0 dropped:0 overruns:0 frame:0
>          TX packets:774782 errors:0 dropped:0 overruns:3 carrier:0
>          collisions:0 txqueuelen:100
>          RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
>          Interrupt:16 Memory:f8d00000-f8d01fff
>
> debian:~# ifconfig w1g1
> w1g1      Link encap:Point-to-Point Protocol
>          UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
>          RX packets:775351 errors:0 dropped:0 overruns:0 frame:0
>          TX packets:775351 errors:0 dropped:0 overruns:3 carrier:0
>          collisions:0 txqueuelen:100
>          RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
>          Interrupt:16 Memory:f8d00000-f8d01fff
>
> debian:~# ifconfig w1g1
> w1g1      Link encap:Point-to-Point Protocol
>          UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
>          RX packets:775919 errors:0 dropped:0 overruns:0 frame:0
>          TX packets:775919 errors:0 dropped:0 overruns:3 carrier:0
>          collisions:0 txqueuelen:100
>          RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
>          Interrupt:16 Memory:f8d00000-f8d01fff
>
> debian:~# ifconfig w1g1
> w1g1      Link encap:Point-to-Point Protocol
>          UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
>          RX packets:776478 errors:0 dropped:0 overruns:0 frame:0
>          TX packets:776478 errors:0 dropped:0 overruns:3 carrier:0
>          collisions:0 txqueuelen:100
>          RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
>          Interrupt:16 Memory:f8d00000-f8d01fff
>
> debian:~# ifconfig w1g1
> w1g1      Link encap:Point-to-Point Protocol
>          UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
>          RX packets:777079 errors:0 dropped:0 overruns:0 frame:0
>          TX packets:777079 errors:0 dropped:0 overruns:3 carrier:0
>          collisions:0 txqueuelen:100
>          RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
>          Interrupt:16 Memory:f8d00000-f8d01fff
>
> debian:~# ifconfig w1g1
> w1g1      Link encap:Point-to-Point Protocol
>          UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
>          RX packets:778134 errors:0 dropped:0 overruns:0 frame:0
>          TX packets:778134 errors:0 dropped:0 overruns:3 carrier:0
>          collisions:0 txqueuelen:100
>          RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
>          Interrupt:16 Memory:f8d00000-f8d01fff
>
>
> debian:~# ifconfig w2g1
> w2g1      Link encap:Point-to-Point Protocol
>          UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
>          RX packets:792251 errors:0 dropped:0 overruns:0 frame:0
>          TX packets:792251 errors:0 dropped:0 overruns:1 carrier:0
>          collisions:0 txqueuelen:100
>          RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
>          Interrupt:16 Memory:f8d00000-f8d01fff
>
> debian:~# ifconfig w2g1
> w2g1      Link encap:Point-to-Point Protocol
>          UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
>          RX packets:792715 errors:0 dropped:0 overruns:0 frame:0
>          TX packets:792715 errors:0 dropped:0 overruns:1 carrier:0
>          collisions:0 txqueuelen:100
>          RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
>          Interrupt:16 Memory:f8d00000-f8d01fff
>
> debian:~# ifconfig w2g1
> w2g1      Link encap:Point-to-Point Protocol
>          UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
>          RX packets:793219 errors:0 dropped:0 overruns:0 frame:0
>          TX packets:793219 errors:0 dropped:0 overruns:1 carrier:0
>          collisions:0 txqueuelen:100
>          RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
>          Interrupt:16 Memory:f8d00000-f8d01fff
>
>
> wanpipemon -i w2g1 -c Ta
>
> ***** w2g1: E1 Alarms (Framer) *****
>
> ALOS:   OFF     | LOS:  OFF
> RED:    OFF     | AIS:  OFF
> LOF:    OFF     | RAI:  OFF
>
> ***** w2g1: E1 Alarms (LIU) *****
>
> Short Circuit:  OFF
> Open Circuit:   OFF
> Loss of Signal: OFF
>
>
> ***** w2g1: E1 Performance Monitoring Counters *****
>
> Line Code Violation     : 0
> Far End Block Errors    : 0
> CRC4 Errors             : 0
> FAS Errors              : 0
>
>
> Rx Level        : > -2.5db
>
>
> debian:~# wanpipemon -i w1g1 -c Ta
>
> ***** w1g1: E1 Alarms (Framer) *****
>
> ALOS:   OFF     | LOS:  OFF
> RED:    OFF     | AIS:  OFF
> LOF:    OFF     | RAI:  OFF
>
> ***** w1g1: E1 Alarms (LIU) *****
>
> Short Circuit:  OFF
> Open Circuit:   OFF
> Loss of Signal: OFF
>
>
> ***** w1g1: E1 Performance Monitoring Counters *****
>
> Line Code Violation     : 371
> Far End Block Errors    : 0
> CRC4 Errors             : 0
> FAS Errors              : 0
>
>
> Rx Level        : > -2.5db
>
>
> Jan  3 20:33:34 debian kernel: [55441.172179] wanpipe1: Shutting Down!
> Jan  3 20:33:34 debian kernel: [55441.172179] wanpipe1: E1 Front End
> unconfigation!
> Jan  3 20:33:34 debian kernel: [55441.172179] wanpipe1: Unregister Wanpipe
> device from Zaptel!
> Jan  3 20:33:34 debian kernel: [55441.178246] wanpipe1: unregistering
> 'w1g1'
> Jan  3 20:33:34 debian kernel: [55441.248278] wanpipe1: TASKQ Not Running
> Jan  3 20:33:34 debian kernel: [55441.248281] wanpipe1: E1 Front End
> unconfigation!
> Jan  3 20:33:34 debian kernel: [55441.248314] wanpipe1: AFT communications
> disabled! (Dev Cnt: 1 Cause: Device Down)
> Jan  3 20:33:34 debian kernel: [55441.248335] wanpipe1: E1 Front End
> unconfigation!
> Jan  3 20:33:34 debian kernel: [55441.248373] wanpipe1: AFT communications
> disabled! (Dev Cnt: 1 Cause: Device Down)
> Jan  3 20:33:34 debian kernel: [55441.248415] wanpipe1: Global Chip
> Shutdown Usage=1
> Jan  3 20:33:34 debian kernel: [55441.248419] wanpipe1: Global E1 Front End
> unconfigation!
> Jan  3 20:33:34 debian kernel: [55441.250485] wanpipe1: Master shutting
> down
> Jan  3 20:33:34 debian kernel: [55441.444571] wanpipe1: Starting WAN Setup
> Jan  3 20:33:34 debian kernel: [55441.444575] wanpipe1: Locating:
> A101/1D/A102/2D/4/4D/8 card, CPU A, PciSlot=4, PciBus=5
> Jan  3 20:33:34 debian kernel: [55441.444579] wanpipe1: Found:
> A101/1D/A102/2D/4/4D/8 card, CPU A, PciSlot=4, PciBus=5, Port=0
> Jan  3 20:33:34 debian kernel: [55441.444599] wanpipe1: AFT PCI memory at
> 0xD3200000
> Jan  3 20:33:34 debian kernel: [55441.444600] wanpipe1: IRQ 16 allocated to
> the AFT PCI card
> Jan  3 20:33:34 debian kernel: [55441.444608] wanpipe1: Starting AFT 2/4/8
> Hardware Init.
> Jan  3 20:33:34 debian kernel: [55441.444614] wanpipe1: Enabling front end
> link monitor
> Jan  3 20:33:34 debian kernel: [55441.444616] wanpipe1: Global Chip
> Configuration: used=1
> Jan  3 20:33:34 debian kernel: [55441.448455] wanpipe1: Global E1 Front End
> configuration
> Jan  3 20:33:34 debian kernel: [55441.448455] wanpipe1: AFT Data Mux Bit
> Map: 0x01234567
> Jan  3 20:33:34 debian kernel: [55441.448455] wanpipe1: Defaulting E1 Rx
> Sens. Gain= 43 db
> Jan  3 20:33:34 debian kernel: [55441.448455] wanpipe1: Configuring DS
> DS26528 E1 FE
> Jan  3 20:33:34 debian kernel: [55441.448455] wanpipe1:    Port
> 1,HDB3,non-CRC4,120OH
> Jan  3 20:33:34 debian kernel: [55441.448455] wanpipe1:    Clk Normal:0,
> Channels: FFFFFFFF
> Jan  3 20:33:34 debian kernel: [55441.448455] wanpipe1:    Sig Mode CCS
> Jan  3 20:33:34 debian kernel: [55441.456490] wanpipe1:    Rx Sensitivity
> Gain 43dB (default).
> Jan  3 20:33:34 debian kernel: [55441.457242] wanpipe1: Front end
> successful
> Jan  3 20:33:34 debian kernel: [55441.457469] wanpipe1: Front End Interface
> Ready 0x40000000
> Jan  3 20:33:34 debian kernel: [55441.457473] wanpipe1: WARNING: No Echo
> Canceller channels are available!
> Jan  3 20:33:34 debian kernel: [55441.457494] wanpipe1: Configuring Device
>   :wanpipe1  FrmVr=39
> Jan  3 20:33:34 debian kernel: [55441.457495] wanpipe1:    Global MTU
>     = 1500
> Jan  3 20:33:34 debian kernel: [55441.457496] wanpipe1:    Global MRU
>     = 1500
> Jan  3 20:33:34 debian kernel: [55441.457497] wanpipe1:    Data Mux Map
>     = 0x01234567
> Jan  3 20:33:34 debian kernel: [55441.457498] wanpipe1:    Rx CRC Bytes
>     = 0
> Jan  3 20:33:34 debian kernel: [55441.457499] wanpipe1:    Global TDM Int
>     = Enabled
> Jan  3 20:33:34 debian kernel: [55441.457500] wanpipe1:    Global TDM Ring
>      = Enabled
> Jan  3 20:33:34 debian kernel: [55441.457501] wanpipe1:    TDMV HW DTMF/FAX
>     = Disabled/Disabled(0)
> Jan  3 20:33:34 debian kernel: [55441.457502] wanpipe1:    TDMV Span
>      = 1 : Enabled
> Jan  3 20:33:34 debian kernel: [55441.457503] wanpipe1:    TDMV Dummy
> = Disabled
> Jan  3 20:33:34 debian kernel: [55441.457505] wanpipe1:    RTP TAP
>      = Disabled
> Jan  3 20:33:34 debian kernel: [55441.457540] wanpipe1: Configuring
> Interface: w1g1
> Jan  3 20:33:34 debian kernel: [55441.457543] wanpipe1:w1g1: Running in TDM
> Voice Zaptel Mode.
> Jan  3 20:33:34 debian kernel: [55441.457547] wanpipe1:    Fifo Level
> Map:0x01041040
> Jan  3 20:33:34 debian kernel: [55441.457549] wanpipe1:    MRU           :8
> Jan  3 20:33:34 debian kernel: [55441.457550] wanpipe1:    MTU           :8
> Jan  3 20:33:34 debian kernel: [55441.457551] wanpipe1:    HDLC Eng
>  :Off (Transparent) | N/A
> Jan  3 20:33:34 debian kernel: [55441.457553] wanpipe1:    Data Mux Ctrl
> :On
> Jan  3 20:33:34 debian kernel: [55441.457554] wanpipe1:    Active Ch Map
> :0x00000002
> Jan  3 20:33:34 debian kernel: [55441.457555] wanpipe1:    First TSlot   :1
> Jan  3 20:33:34 debian kernel: [55441.457568] wanpipe1:    DMA/Len/Chain/EC
> :4/1024/Off/Off
> Jan  3 20:33:34 debian kernel: [55441.457575] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.457581] wanpipe1:    Active Ch Map
> :0x00000004
> Jan  3 20:33:34 debian kernel: [55441.457582] wanpipe1:    First TSlot   :2
> Jan  3 20:33:34 debian kernel: [55441.457597] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.457602] wanpipe1:    Active Ch Map
> :0x00000008
> Jan  3 20:33:34 debian kernel: [55441.457603] wanpipe1:    First TSlot   :3
> Jan  3 20:33:34 debian kernel: [55441.457618] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.457623] wanpipe1:    Active Ch Map
> :0x00000010
> Jan  3 20:33:34 debian kernel: [55441.457624] wanpipe1:    First TSlot   :4
> Jan  3 20:33:34 debian kernel: [55441.457637] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.457643] wanpipe1:    Active Ch Map
> :0x00000020
> Jan  3 20:33:34 debian kernel: [55441.457644] wanpipe1:    First TSlot   :5
> Jan  3 20:33:34 debian kernel: [55441.457657] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.457663] wanpipe1:    Active Ch Map
> :0x00000040
> Jan  3 20:33:34 debian kernel: [55441.457664] wanpipe1:    First TSlot   :6
> Jan  3 20:33:34 debian kernel: [55441.457678] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.457684] wanpipe1:    Active Ch Map
> :0x00000080
> Jan  3 20:33:34 debian kernel: [55441.457685] wanpipe1:    First TSlot   :7
> Jan  3 20:33:34 debian kernel: [55441.457698] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.457705] wanpipe1:    Active Ch Map
> :0x00000100
> Jan  3 20:33:34 debian kernel: [55441.457706] wanpipe1:    First TSlot   :8
> Jan  3 20:33:34 debian kernel: [55441.457719] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.457725] wanpipe1:    Active Ch Map
> :0x00000200
> Jan  3 20:33:34 debian kernel: [55441.457726] wanpipe1:    First TSlot   :9
> Jan  3 20:33:34 debian kernel: [55441.457740] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.457746] wanpipe1:    Active Ch Map
> :0x00000400
> Jan  3 20:33:34 debian kernel: [55441.457747] wanpipe1:    First TSlot
> :10
> Jan  3 20:33:34 debian kernel: [55441.457760] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.457765] wanpipe1:    Active Ch Map
> :0x00000800
> Jan  3 20:33:34 debian kernel: [55441.457766] wanpipe1:    First TSlot
> :11
> Jan  3 20:33:34 debian kernel: [55441.457781] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.457786] wanpipe1:    Active Ch Map
> :0x00001000
> Jan  3 20:33:34 debian kernel: [55441.457787] wanpipe1:    First TSlot
> :12
> Jan  3 20:33:34 debian kernel: [55441.457801] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.457807] wanpipe1:    Active Ch Map
> :0x00002000
> Jan  3 20:33:34 debian kernel: [55441.457808] wanpipe1:    First TSlot
> :13
> Jan  3 20:33:34 debian kernel: [55441.457822] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.457828] wanpipe1:    Active Ch Map
> :0x00004000
> Jan  3 20:33:34 debian kernel: [55441.457829] wanpipe1:    First TSlot
> :14
> Jan  3 20:33:34 debian kernel: [55441.457842] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.457848] wanpipe1:    Active Ch Map
> :0x00008000
> Jan  3 20:33:34 debian kernel: [55441.457849] wanpipe1:    First TSlot
> :15
> Jan  3 20:33:34 debian kernel: [55441.457862] wanpipe1: Configuring
> Interface: w1g1
> Jan  3 20:33:34 debian kernel: [55441.457864] wanpipe1:w1g1: Running in TDM
> DCHAN Voice Zaptel Mode.
> Jan  3 20:33:34 debian kernel: [55441.457866] wanpipe1:    MRU
> :1500
> Jan  3 20:33:34 debian kernel: [55441.457867] wanpipe1:    MTU
> :1500
> Jan  3 20:33:34 debian kernel: [55441.457868] wanpipe1:    HDLC Eng
>  :On | N/A
> Jan  3 20:33:34 debian kernel: [55441.457869] wanpipe1:    Data Mux Ctrl
> :Off
> Jan  3 20:33:34 debian kernel: [55441.457870] wanpipe1:    Active Ch Map
> :0x00010000
> Jan  3 20:33:34 debian kernel: [55441.457871] wanpipe1:    First TSlot
> :16
> Jan  3 20:33:34 debian kernel: [55441.457881] wanpipe1:    DMA/Len/Chain/EC
> :65/4096/On/Off
> Jan  3 20:33:34 debian kernel: [55441.457905] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.457910] wanpipe1:    Active Ch Map
> :0x00020000
> Jan  3 20:33:34 debian kernel: [55441.457911] wanpipe1:    First TSlot
> :17
> Jan  3 20:33:34 debian kernel: [55441.457925] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.457931] wanpipe1:    Active Ch Map
> :0x00040000
> Jan  3 20:33:34 debian kernel: [55441.457932] wanpipe1:    First TSlot
> :18
> Jan  3 20:33:34 debian kernel: [55441.457945] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.457951] wanpipe1:    Active Ch Map
> :0x00080000
> Jan  3 20:33:34 debian kernel: [55441.457952] wanpipe1:    First TSlot
> :19
> Jan  3 20:33:34 debian kernel: [55441.457967] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.457973] wanpipe1:    Active Ch Map
> :0x00100000
> Jan  3 20:33:34 debian kernel: [55441.457974] wanpipe1:    First TSlot
> :20
> Jan  3 20:33:34 debian kernel: [55441.457987] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.457993] wanpipe1:    Active Ch Map
> :0x00200000
> Jan  3 20:33:34 debian kernel: [55441.457994] wanpipe1:    First TSlot
> :21
> Jan  3 20:33:34 debian kernel: [55441.458009] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.458015] wanpipe1:    Active Ch Map
> :0x00400000
> Jan  3 20:33:34 debian kernel: [55441.458016] wanpipe1:    First TSlot
> :22
> Jan  3 20:33:34 debian kernel: [55441.458030] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.458036] wanpipe1:    Active Ch Map
> :0x00800000
> Jan  3 20:33:34 debian kernel: [55441.458037] wanpipe1:    First TSlot
> :23
> Jan  3 20:33:34 debian kernel: [55441.458051] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.458057] wanpipe1:    Active Ch Map
> :0x01000000
> Jan  3 20:33:34 debian kernel: [55441.458058] wanpipe1:    First TSlot
> :24
> Jan  3 20:33:34 debian kernel: [55441.458072] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.458078] wanpipe1:    Active Ch Map
> :0x02000000
> Jan  3 20:33:34 debian kernel: [55441.458079] wanpipe1:    First TSlot
> :25
> Jan  3 20:33:34 debian kernel: [55441.458094] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.458100] wanpipe1:    Active Ch Map
> :0x04000000
> Jan  3 20:33:34 debian kernel: [55441.458101] wanpipe1:    First TSlot
> :26
> Jan  3 20:33:34 debian kernel: [55441.458116] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.458122] wanpipe1:    Active Ch Map
> :0x08000000
> Jan  3 20:33:34 debian kernel: [55441.458123] wanpipe1:    First TSlot
> :27
> Jan  3 20:33:34 debian kernel: [55441.458137] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.458143] wanpipe1:    Active Ch Map
> :0x10000000
> Jan  3 20:33:34 debian kernel: [55441.458144] wanpipe1:    First TSlot
> :28
> Jan  3 20:33:34 debian kernel: [55441.458160] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.458166] wanpipe1:    Active Ch Map
> :0x20000000
> Jan  3 20:33:34 debian kernel: [55441.458167] wanpipe1:    First TSlot
> :29
> Jan  3 20:33:34 debian kernel: [55441.458181] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.458187] wanpipe1:    Active Ch Map
> :0x40000000
> Jan  3 20:33:34 debian kernel: [55441.458188] wanpipe1:    First TSlot
> :30
> Jan  3 20:33:34 debian kernel: [55441.458202] wanpipe1: Configuring
> Interface: w1g1 (log supress)
> Jan  3 20:33:34 debian kernel: [55441.458205] wanpipe1: Configuring TDMV
> Master dev w1g1
> Jan  3 20:33:34 debian kernel: [55441.458209] wanpipe1:    Active Ch Map
> :0x80000000
> Jan  3 20:33:34 debian kernel: [55441.458210] wanpipe1:    First TSlot
> :31
> Jan  3 20:33:34 debian kernel: [55441.458226] wanpipe1: Enable Zaptel HW
> DCHAN interface
> Jan  3 20:33:34 debian kernel: [55441.458462] wanpipe1: Wanpipe device is
> registered to Zaptel span # 1!
> Jan  3 20:33:34 debian kernel: [55441.458760] wanpipe1: TDM Free Run Timing
> Enabled 1 ms
> Jan  3 20:33:34 debian kernel: [55441.464708] wanpipe1: Wanpipe Front End
> Interrupt Restart Timeout
> Jan  3 20:33:40 debian kernel: [55450.402880] wanpipe1: E1 connected!
> Jan  3 20:33:40 debian kernel: [55450.407421] wanpipe1: AFT communications
> enabled!
> Jan  3 20:33:40 debian kernel: [55450.407421] wanpipe1: AFT Global TDM Intr
> Jan  3 20:33:40 debian kernel: [55450.409902] wanpipe1: Global TDM Ring
> Resync
> Jan  3 20:33:41 debian kernel: [55452.785733] wanpipe1: Enable E1 CCS
> Signalling mode!
>
>
> debian*CLI> core show version
> Asterisk 1.6.2.0 built by root @ debian on a i686 running Linux on
> 2011-01-03 08:57:50 UTC
>
>
> debian*CLI> ss7 show linkset 1
> SS7 linkset 1 status: Down
>
>
> debian*CLI> ss7 set debug off linkset 1
> Enabled debugging on linkset 1
> Len = 4 [ ff ff 01 03 ]
> FSN: 127 FIB 1
> BSN: 127 BIB 1
> <[0] LSSU SIOS
>
> Link state change: NOTALIGNED -> NOTALIGNED
> Len = 4 [ ff ff 01 00 ]
> FSN: 127 FIB 1
> BSN: 127 BIB 1
> <[0] LSSU SIO
>
> Link state change: NOTALIGNED -> ALIGNED
> Len = 4 [ ff ff 01 02 ]
> FSN: 127 FIB 1
> BSN: 127 BIB 1
> >[0] LSSU SIE
>
> Link state change: ALIGNED -> IDLE
> Link state change: IDLE -> NOTALIGNED
> Len = 4 [ ff ff 01 00 ]
> FSN: 127 FIB 1
> BSN: 127 BIB 1
> >[0] LSSU SIO
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
peace be upon you ...
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From infos at madovsky.org  Tue Jan  4 05:11:05 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 3 Jan 2011 21:11:05 -0500
Subject: [Freeswitch-users] mod_conference with cluster ODBC
Message-ID: <46549E68636443D3BD6AA2E90AE5A86A@e1705>

If I have a conference created on node A with name abc-domain at default
and another with the same name in node B it seems
that FS considers it as 2 distinct conferences, or maybe
I wrongly set any configuration ?

Thanks
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From Nabble at slickdeals.endjunk.com  Tue Jan  4 05:31:17 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 3 Jan 2011 18:31:17 -0800 (PST)
Subject: [Freeswitch-users] spindermonkey: Host execution of a
 cross-compilation of libs/js/nsprpub/config/nsinstall.c
In-Reply-To: <1293570293681-5872965.post@n2.nabble.com>
References: <1293570293681-5872965.post@n2.nabble.com>
Message-ID: <1294108277218-5887451.post@n2.nabble.com>


Any chance to get help on this issue?

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/spindermonkey-Host-execution-of-a-cross-compilation-of-libs-js-nsprpub-config-nsinstall-c-tp5872965p5887451.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From u2nsam at gmail.com  Tue Jan  4 06:03:39 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 4 Jan 2011 08:33:39 +0530
Subject: [Freeswitch-users] no ringback tone
In-Reply-To: <AANLkTinA46HPKmk_pWaAiMpy-hdihc7STfY9_FuNkthx@mail.gmail.com>
References: <AANLkTinbd06YgZFWue=eQGG9A5JO2a550tJ+bo-yG2yv@mail.gmail.com>
	<AANLkTinA46HPKmk_pWaAiMpy-hdihc7STfY9_FuNkthx@mail.gmail.com>
Message-ID: <AANLkTim6iJ6gFc1=21REAKkvz9z8ZgPK2ShHC7pBUpxH@mail.gmail.com>

Some mobile operators give VAS as ringback tone / caller tune to be set as
mobile ringback,
now when dialing out such mobile users who has ringback enabled i just get
plain ring and no proceeding with 183 for the media to listen,
how to enable such so that some one dialing out will hear ringback generated
by the mobile operators.

call flow is FS --> Sangoma --> mobile (ringback enabled by mobile operator
)

Regards
Sam

On Tue, Jan 4, 2011 at 12:02 AM, Michael Collins <msc at freeswitch.org> wrote:

> Can you clarify what the actual issue is?
> -MC
>
> On Sun, Jan 2, 2011 at 8:30 PM, Sam <u2nsam at gmail.com> wrote:
>
>> Hi All,
>>
>> happy new you to you !
>>
>> using a sangoma card and when dialing a mobile number which is having
>> ringback tune / caller tune ;
>> but a plain ring is heard to the user dialing that mobile through the
>> trunk.
>>
>> I am using below syntax to dial out:-
>> <action application="bridge"
>> data="{monitor_early_media_ring_total=3}freetdm/wp1/a/${destination_number}"/>
>>
>> Any suggestions.
>>
>> Regards
>>  Sam
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Tue Jan  4 06:05:10 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 3 Jan 2011 22:05:10 -0500
Subject: [Freeswitch-users] mode_nibblebill and mod_conference
References: <1B18EBDDB6044354A0ACF963C987DDAF@e1705>
	<AANLkTik0azNgbcMX4qD4obfj9-GomR=MCVGOPiWsSdiO@mail.gmail.com>
Message-ID: <A8A56043E970473B8D63E357E99B267C@e1705>

ok Mike, I understood now the concept.
so I need only to "set" the vars and change nibble vars to the dest user on
legA.
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 03, 2011 3:22 PM
  Subject: Re: [Freeswitch-users] mode_nibblebill and mod_conference


  According to the wiki (and page 251 of the FS book, which has this exact same example) you need something like this:


<action application="bridge" data="{enable_heartbeat_events=5,nibble_rate=1,nibble_account=0838833133}sofia/external/$1 at tel.co.th"/>The link is http://wiki.freeswitch.org/wiki/Mod_nibblebill#Bill_base_on_B_leg_Only


  In your example I don't know that you necessarily have a b-leg. How are you getting the call to the conference? Do you use originate? Or is it an inbound call?
  -MC


  On Mon, Jan 3, 2011 at 11:30 AM, Madovsky <infos at madovsky.org> wrote:

    in mod_nibblebill wiki page
      a.. Q: Can you bill based on a multi-call B-Leg, where you have, say a conference call with multiple people on it? 
        a.. Yes, see answer above re: B-Leg and not A-Leg. 
    but if I have this in my dialplan

        <extension name="conference">
            <condition field="destination_number" expression="^conference$">
                    <action application="export" data="nolocal:nibble_account=${dest_nibble_account}"/>
                    <action application="export" data="nolocal:nibble_rate=0.03"/>
                    <action application="sleep" data="3000"/>
                    <action application="answer"/>
                    <!-- <action application="speak" data="cepstral|cepstral|connecting your call to the conference"/> -->
                    <action application="conference" data="${dest_nibble_account}-$${domain}@default+${dest_conf_pin}"/>
            </condition>
        </extension>

    it doesn't work



      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Monday, January 03, 2011 2:21 PM
      Subject: mode_nibblebill and mod_conference


      Is nibblebill works with conference ?
      I tried to set nibble_account and nibble_rate without success

      Thanks

    _______________________________________________
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    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From u2nsam at gmail.com  Tue Jan  4 06:05:23 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 4 Jan 2011 08:35:23 +0530
Subject: [Freeswitch-users] multi company
In-Reply-To: <AANLkTikW=LmuBhtLZ3Kxqhaqx95oiz4VTs3EfFU2tNVh@mail.gmail.com>
References: <AANLkTikjjhHMMxKmsEGpDYxXBpozjNycdimV6zZFmhL7@mail.gmail.com>
	<AANLkTikW=LmuBhtLZ3Kxqhaqx95oiz4VTs3EfFU2tNVh@mail.gmail.com>
Message-ID: <AANLkTikNNi9J8oJOMzqwfuZNSPwEsLnkMe2sX5ZSxvUG@mail.gmail.com>

Did it exactly what they they have suggested in wiki but give that error.

Regards
Sam

On Tue, Jan 4, 2011 at 12:07 AM, Michael Collins <msc at freeswitch.org> wrote:

> Just for confirmation, have you looked at these two pages?
> http://wiki.freeswitch.org/wiki/Multi-tenant
> http://wiki.freeswitch.org/wiki/Multiple_Domains
>
> There is quite a bit of information for you to try.
> -MC
>
> On Mon, Jan 3, 2011 at 2:37 AM, Sam <u2nsam at gmail.com> wrote:
>
>> Hi,
>>
>> Was using multi company setup,
>>
>> it gave an error while using below syntax
>>
>> <domain name="xyz">
>>
>> </domain>
>>
>> Cannot Initialize [[error near line 2868]: unexpected closing tag
>> </domain>]
>>
>> And when i remove , it dont gives an error .
>> the file is in directory/xyz.xml
>>
>>
>> same happens on freeswitch.xml file
>>
>> when i remove the syntax
>>
>> <section name="directory" description="User Directory">
>>     <X-PRE-PROCESS cmd="include" data="directory/*.xml"/>
>>   </section>
>>
>> it works when removed the section other wise give error as unexpected
>> closing tag </section>]
>>
>> Regds
>> Sam
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From jmesquita at freeswitch.org  Tue Jan  4 07:08:34 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 4 Jan 2011 01:08:34 -0300
Subject: [Freeswitch-users] mod_conference with cluster ODBC
In-Reply-To: <46549E68636443D3BD6AA2E90AE5A86A@e1705>
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705>
Message-ID: <AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com>

They are indeed 2 completely different conferences. There's no
implementation of making these 2 conferences bridge themselves
automatically.

The way I have solved this problem for now is have an ESL daemon "listening"
on the conference creation events and bridging the 2 servers together when
the one with the same name on the same domain is created. This solution
might work for you and it's no too hard to implement.

Regards,
Jo?o Mesquita


On Mon, Jan 3, 2011 at 11:11 PM, Madovsky <infos at madovsky.org> wrote:

>  If I have a conference created on node A with name abc-domain at default
> and another with the same name in node B it seems
> that FS considers it as 2 distinct conferences, or maybe
> I wrongly set any configuration ?
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Tue Jan  4 08:14:48 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 4 Jan 2011 00:14:48 -0500
Subject: [Freeswitch-users] mod_conference with cluster ODBC
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705>
	<AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com>
Message-ID: <623BCB43D9ED45269190F7D08A802469@e1705>

ha ok, 
any example to listening conference with ESL ?
are bridge 2 conferences made the same way as 2 users ?

suggestion: as there is one DB with ODBC for all nodes why not
create a conference table as sip_registration to manage the ip of every first user in a conference
and redirect the other automatically to the right node ?

Thanks
  ----- Original Message ----- 
  From: Jo?o Mesquita 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 03, 2011 11:08 PM
  Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


  They are indeed 2 completely different conferences. There's no implementation of making these 2 conferences bridge themselves automatically.


  The way I have solved this problem for now is have an ESL daemon "listening" on the conference creation events and bridging the 2 servers together when the one with the same name on the same domain is created. This solution might work for you and it's no too hard to implement.


  Regards,
  Jo?o Mesquita



  On Mon, Jan 3, 2011 at 11:11 PM, Madovsky <infos at madovsky.org> wrote:

    If I have a conference created on node A with name abc-domain at default
    and another with the same name in node B it seems
    that FS considers it as 2 distinct conferences, or maybe
    I wrongly set any configuration ?

    Thanks

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From xyangni at gmail.com  Tue Jan  4 08:34:10 2011
From: xyangni at gmail.com (xuyan yang)
Date: Tue, 4 Jan 2011 13:34:10 +0800
Subject: [Freeswitch-users] Attack using 5843 and music account?
In-Reply-To: <28AF5B89-AFB3-438E-AB7B-AF598CB18204@freeswitch.org>
References: <AANLkTinNmW4OtL275UWzRoZof0U_YAq3vhfwNkJo3oOG@mail.gmail.com>
	<28AF5B89-AFB3-438E-AB7B-AF598CB18204@freeswitch.org>
Message-ID: <AANLkTi=vdoj5D_KXD06JpM_QObeuieWVN6XGn_ia74_B@mail.gmail.com>

Got it. But if no failure log. fail2ban will not work. So how can we protect
fs from this kind of attack besides manually setup firewall rules 1 by 1 on
discovery?

On Tue, Jan 4, 2011 at 12:54 AM, Brian West <brian at freeswitch.org> wrote:

> Chances are he never received the challenge.. thus never logs an auth
> failure.
>
> /b
>
> On Jan 3, 2011, at 9:26 AM, xuyan yang wrote:
>
> 2011-01-03 15:19:32.360152 [WARNING] sofia_reg.c:1161 SIP auth failure
> (REGISTER) on sofia profile 'internal' for [music at 192.168.0.3] from ip
> 192.168.0.6
>
> So, how can this hacker successfully registered music account and avoid to
> be baned? it is strange.
>
> Thanks
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From u2nsam at gmail.com  Tue Jan  4 09:37:51 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 4 Jan 2011 12:07:51 +0530
Subject: [Freeswitch-users] mod_conference with cluster ODBC
In-Reply-To: <623BCB43D9ED45269190F7D08A802469@e1705>
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705>
	<AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com>
	<623BCB43D9ED45269190F7D08A802469@e1705>
Message-ID: <AANLkTin+-cou_QLsFXqVJpQmyk8o7Ke6TjDQc6UL3tzh@mail.gmail.com>

Can this be done with the primary database sqlite ?

Regds
Sam

On Tue, Jan 4, 2011 at 10:44 AM, Madovsky <infos at madovsky.org> wrote:

>  ha ok,
> any example to listening conference with ESL ?
> are bridge 2 conferences made the same way as 2 users ?
>
> suggestion: as there is one DB with ODBC for all nodes why not
> create a conference table as sip_registration to manage the ip of every
> first user in a conference
> and redirect the other automatically to the right node ?
>
> Thanks
>
> ----- Original Message -----
> *From:* Jo?o Mesquita <jmesquita at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, January 03, 2011 11:08 PM
> *Subject:* Re: [Freeswitch-users] mod_conference with cluster ODBC
>
> They are indeed 2 completely different conferences. There's no
> implementation of making these 2 conferences bridge themselves
> automatically.
>
> The way I have solved this problem for now is have an ESL daemon
> "listening" on the conference creation events and bridging the 2 servers
> together when the one with the same name on the same domain is created. This
> solution might work for you and it's no too hard to implement.
>
> Regards,
> Jo?o Mesquita
>
>
> On Mon, Jan 3, 2011 at 11:11 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  If I have a conference created on node A with name abc-domain at default
>> and another with the same name in node B it seems
>> that FS considers it as 2 distinct conferences, or maybe
>> I wrongly set any configuration ?
>>
>> Thanks
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From u2nsam at gmail.com  Tue Jan  4 10:55:06 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 4 Jan 2011 13:25:06 +0530
Subject: [Freeswitch-users] conferencing
Message-ID: <AANLkTikgBYfg_UiZnbc==8dgs32G8fDkSvak=-DBG_Ak@mail.gmail.com>

hello,

how to increase the scalability / performance of the conference .

Are there any parameters to be observed that could do a trick ?

The scenario would be multiple conferencing bridges and multiple codecs
involved.

Is is possible to multiple threading of profiles for conferences so that it
gets the scalability ?

Regards
Sam
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From u2nsam at gmail.com  Tue Jan  4 11:03:46 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 4 Jan 2011 13:33:46 +0530
Subject: [Freeswitch-users] mod callcenter
In-Reply-To: <AANLkTin0_cUT1Wb-SanDj_sD_JWnrzJ_QC1YpUMbBDc0@mail.gmail.com>
References: <AANLkTikdRNMo4CgFcuon0X4KkyZSzZYHK6RECaDxrNwa@mail.gmail.com>
	<AANLkTin0_cUT1Wb-SanDj_sD_JWnrzJ_QC1YpUMbBDc0@mail.gmail.com>
Message-ID: <AANLkTikTRte3_848LiNH0=ZjqL7Atmzx_wBaLziHzUnJ@mail.gmail.com>

i have pasted more logs for the error on jira as it showed up again.
FS-2952

Regds
Sam



On Thu, Dec 30, 2010 at 3:54 AM, Michael Collins <msc at freeswitch.org> wrote:

> Open a tick on jira.freeswitch.org and Moc will take a look. Be sure to
> provide as much info as possible.
> -MC
>
> On Wed, Dec 29, 2010 at 6:48 AM, Sam <u2nsam at gmail.com> wrote:
>
>> Hello,
>>
>> Was testing callcenter module and found out that at times it gives error "
>> invalid application callcenter " and after
>> reloading the module it works fine.
>> Some time also happens that if I reload the module it do not reloads the
>> parameters of the callcenter like the agents & tires.
>> It just unload & reloads even if there are changes to the specifications
>> of agents.
>>
>> Regds
>> Sam
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From hwnorman at hotmail.com  Tue Jan  4 11:36:52 2011
From: hwnorman at hotmail.com (Norman Lam)
Date: Tue, 4 Jan 2011 16:36:52 +0800
Subject: [Freeswitch-users] skypopen error when making a call
Message-ID: <SNT109-DS23DD745C224DDC1DCC71ADC1080@phx.gbl>

Hi Everyone 

 

I am trying to call skypopen(skype network) from freeswitch console and I am
getting this error 

 

freeswitch at jfp8> skypopen interface1 hwnorman

 

Using interface: globals.SKYPOPEN_INTERFACES[1].name=|||interface1|||

 

2011-01-04 16:28:01.687500 [ERR] skypopen_protocol.c:259     [git-] [ERRORA

  259  ][interface1     ][IDLE,IDLE] Skype got ERROR: |||ERROR 2 Unknown
command

|||

2011-01-04 16:28:01.687500 [ERR] skypopen_protocol.c:261     [git-] [ERRORA

  261  ][interface1     ][IDLE,FNSHED] skype_call now is DOWN

 

But my interface is O.K , I can make a call from skype to freeswitch
destination to the x-lite is O.K 

 

Please advise 

 

Norman Lam 

 

Here is the skypopen interface 

 

freeswitch at jfp8> sk list

 

sk console is NOT yet assigned

F ID        Name        IB (F/T)    OB (F/T)    State   CallFlw         UUID

= ====    ========      =======     =======     ======  ============
======

  1     [interface1]      0/1        0/0        IDLE    IDLE

  2     [interface2]      0/0        0/0        IDLE    IDLE

  3     [interface3]      0/0        0/0        IDLE    IDLE

  4     [interface4]      0/0        0/0        IDLE    IDLE

 

Total Interfaces: 4  IB Calls(Failed/Total): 0/1  OB Calls(Failed/Total):
0/0

 

 

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From Avi at aMarcus.com  Tue Jan  4 12:26:48 2011
From: Avi at aMarcus.com (Avi Marcus)
Date: Tue, 4 Jan 2011 11:26:48 +0200
Subject: [Freeswitch-users] conferencing
In-Reply-To: <AANLkTikgBYfg_UiZnbc==8dgs32G8fDkSvak=-DBG_Ak@mail.gmail.com>
References: <AANLkTikgBYfg_UiZnbc==8dgs32G8fDkSvak=-DBG_Ak@mail.gmail.com>
Message-ID: <AANLkTino24LWO+V9egF70aPu0Kojj0sLTMiKBDY6E2tr@mail.gmail.com>

It's the transcoding that will kill you. If that's truly needed, you may
want to look into multiple machines handling transcoding and then bridging
into the conference, or the new transcoding hardware from
sangoma<http://www.sangoma.com/products/hardware_products/transcoding.html>
.
-Avi

On Tue, Jan 4, 2011 at 9:55 AM, Sam <u2nsam at gmail.com> wrote:

> hello,
>
> how to increase the scalability / performance of the conference .
>
> Are there any parameters to be observed that could do a trick ?
>
> The scenario would be multiple conferencing bridges and multiple codecs
> involved.
>
> Is is possible to multiple threading of profiles for conferences so that it
> gets the scalability ?
>
> Regards
> Sam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Avi at aMarcus.com  Tue Jan  4 12:28:12 2011
From: Avi at aMarcus.com (Avi Marcus)
Date: Tue, 4 Jan 2011 11:28:12 +0200
Subject: [Freeswitch-users] mod_conference with cluster ODBC
In-Reply-To: <AANLkTin+-cou_QLsFXqVJpQmyk8o7Ke6TjDQc6UL3tzh@mail.gmail.com>
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705>
	<AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com>
	<623BCB43D9ED45269190F7D08A802469@e1705>
	<AANLkTin+-cou_QLsFXqVJpQmyk8o7Ke6TjDQc6UL3tzh@mail.gmail.com>
Message-ID: <AANLkTimSCf1UvCTpEv8QaiG4=f5EsH=qoUbYKw1F_UT+@mail.gmail.com>

The ESL implementation would be the same...

On Tue, Jan 4, 2011 at 8:37 AM, Sam <u2nsam at gmail.com> wrote:

> Can this be done with the primary database sqlite ?
>
> Regds
> Sam
>
>
> On Tue, Jan 4, 2011 at 10:44 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  ha ok,
>> any example to listening conference with ESL ?
>> are bridge 2 conferences made the same way as 2 users ?
>>
>> suggestion: as there is one DB with ODBC for all nodes why not
>> create a conference table as sip_registration to manage the ip of every
>> first user in a conference
>> and redirect the other automatically to the right node ?
>>
>> Thanks
>>
>> ----- Original Message -----
>> *From:* Jo?o Mesquita <jmesquita at freeswitch.org>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Sent:* Monday, January 03, 2011 11:08 PM
>> *Subject:* Re: [Freeswitch-users] mod_conference with cluster ODBC
>>
>> They are indeed 2 completely different conferences. There's no
>> implementation of making these 2 conferences bridge themselves
>> automatically.
>>
>> The way I have solved this problem for now is have an ESL daemon
>> "listening" on the conference creation events and bridging the 2 servers
>> together when the one with the same name on the same domain is created. This
>> solution might work for you and it's no too hard to implement.
>>
>> Regards,
>> Jo?o Mesquita
>>
>>
>> On Mon, Jan 3, 2011 at 11:11 PM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  If I have a conference created on node A with name abc-domain at default
>>> and another with the same name in node B it seems
>>> that FS considers it as 2 distinct conferences, or maybe
>>> I wrongly set any configuration ?
>>>
>>> Thanks
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>  ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
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>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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>
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From u2nsam at gmail.com  Tue Jan  4 13:15:37 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 4 Jan 2011 15:45:37 +0530
Subject: [Freeswitch-users] conferencing
In-Reply-To: <AANLkTino24LWO+V9egF70aPu0Kojj0sLTMiKBDY6E2tr@mail.gmail.com>
References: <AANLkTikgBYfg_UiZnbc==8dgs32G8fDkSvak=-DBG_Ak@mail.gmail.com>
	<AANLkTino24LWO+V9egF70aPu0Kojj0sLTMiKBDY6E2tr@mail.gmail.com>
Message-ID: <AANLkTinLij+oYGQ5Go1_Etg_kZvb_+wtKAQRg4krcwhU@mail.gmail.com>

Ok if i remove transcoding out of the picture and all conf are running on
g711, what rest would matters ?


Regds
Sam


On Tue, Jan 4, 2011 at 2:56 PM, Avi Marcus <Avi at amarcus.com> wrote:

> It's the transcoding that will kill you. If that's truly needed, you may
> want to look into multiple machines handling transcoding and then bridging
> into the conference, or the new transcoding hardware from sangoma<http://www.sangoma.com/products/hardware_products/transcoding.html>
> .
> -Avi
>
> On Tue, Jan 4, 2011 at 9:55 AM, Sam <u2nsam at gmail.com> wrote:
>
>> hello,
>>
>> how to increase the scalability / performance of the conference .
>>
>> Are there any parameters to be observed that could do a trick ?
>>
>> The scenario would be multiple conferencing bridges and multiple codecs
>> involved.
>>
>> Is is possible to multiple threading of profiles for conferences so that
>> it gets the scalability ?
>>
>> Regards
>> Sam
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From steveayre at gmail.com  Tue Jan  4 13:23:32 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 4 Jan 2011 10:23:32 +0000
Subject: [Freeswitch-users] conferencing
In-Reply-To: <AANLkTinLij+oYGQ5Go1_Etg_kZvb_+wtKAQRg4krcwhU@mail.gmail.com>
References: <AANLkTikgBYfg_UiZnbc==8dgs32G8fDkSvak=-DBG_Ak@mail.gmail.com>
	<AANLkTino24LWO+V9egF70aPu0Kojj0sLTMiKBDY6E2tr@mail.gmail.com>
	<AANLkTinLij+oYGQ5Go1_Etg_kZvb_+wtKAQRg4krcwhU@mail.gmail.com>
Message-ID: <AANLkTi=KPRnLDNLL8uUCxCvsFiRHrrJprxNVKjM7Kw=B@mail.gmail.com>

That will still involve transcoding. FS must combine all speaking
channels into a single stream that can be sent to the conference
members. That can't be done in a compressed audio format, so it must
convert G711 to L16 for each speaking member, combine those pieces of
audio, then convert L16 back to G711 to send to the conference
members. Additionally, since the conference members are not
necessarily all using the same codec that last transcoding step will
occur once for each member, not just once for the conference. Your
best bet is to use a codec that uses as little CPU for transcoding as
possible - in software I'm not sure which is best there (G711 is
simple so possibly a good choice), in hardware Sangoma D100/D500 would
almost all the processing off the CPU.

-Steve


On 4 January 2011 10:15, Sam <u2nsam at gmail.com> wrote:
> Ok if i remove transcoding out of the picture and all conf are running on
> g711, what rest would matters ?
>
>
> Regds
> Sam
>
>
> On Tue, Jan 4, 2011 at 2:56 PM, Avi Marcus <Avi at amarcus.com> wrote:
>>
>> It's the transcoding that will kill you. If that's truly needed, you may
>> want to look into multiple machines handling transcoding and then bridging
>> into the conference, or the new?transcoding?hardware from sangoma.
>> -Avi
>> On Tue, Jan 4, 2011 at 9:55 AM, Sam <u2nsam at gmail.com> wrote:
>>>
>>> hello,
>>>
>>> how to increase the scalability / performance of the conference .
>>>
>>> Are there any parameters to be observed that could do a trick ?
>>>
>>> The scenario would be multiple conferencing bridges and multiple codecs
>>> involved.
>>>
>>> Is is possible to multiple threading of profiles for conferences so that
>>> it gets the scalability ?
>>>
>>> Regards
>>> Sam
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From u2nsam at gmail.com  Tue Jan  4 13:45:33 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 4 Jan 2011 16:15:33 +0530
Subject: [Freeswitch-users] conferencing
In-Reply-To: <AANLkTi=KPRnLDNLL8uUCxCvsFiRHrrJprxNVKjM7Kw=B@mail.gmail.com>
References: <AANLkTikgBYfg_UiZnbc==8dgs32G8fDkSvak=-DBG_Ak@mail.gmail.com>
	<AANLkTino24LWO+V9egF70aPu0Kojj0sLTMiKBDY6E2tr@mail.gmail.com>
	<AANLkTinLij+oYGQ5Go1_Etg_kZvb_+wtKAQRg4krcwhU@mail.gmail.com>
	<AANLkTi=KPRnLDNLL8uUCxCvsFiRHrrJprxNVKjM7Kw=B@mail.gmail.com>
Message-ID: <AANLkTikuYNOL55j0rBFgG=FqFQnPWfxT1J13Q1M8ZMf6@mail.gmail.com>

calls would not be TDM it would be ip-ip calls.
Ok G711 uses less CPU than other codecs,how can i utilized multiple
threading/profiles or any other options to scale ? as it would be an
incoming call always on FS.

Regds
Sam

On Tue, Jan 4, 2011 at 3:53 PM, Steven Ayre <steveayre at gmail.com> wrote:

> That will still involve transcoding. FS must combine all speaking
> channels into a single stream that can be sent to the conference
> members. That can't be done in a compressed audio format, so it must
> convert G711 to L16 for each speaking member, combine those pieces of
> audio, then convert L16 back to G711 to send to the conference
> members. Additionally, since the conference members are not
> necessarily all using the same codec that last transcoding step will
> occur once for each member, not just once for the conference. Your
> best bet is to use a codec that uses as little CPU for transcoding as
> possible - in software I'm not sure which is best there (G711 is
> simple so possibly a good choice), in hardware Sangoma D100/D500 would
> almost all the processing off the CPU.
>
> -Steve
>
>
> On 4 January 2011 10:15, Sam <u2nsam at gmail.com> wrote:
> > Ok if i remove transcoding out of the picture and all conf are running on
> > g711, what rest would matters ?
> >
> >
> > Regds
> > Sam
> >
> >
> > On Tue, Jan 4, 2011 at 2:56 PM, Avi Marcus <Avi at amarcus.com> wrote:
> >>
> >> It's the transcoding that will kill you. If that's truly needed, you may
> >> want to look into multiple machines handling transcoding and then
> bridging
> >> into the conference, or the new transcoding hardware from sangoma.
> >> -Avi
> >> On Tue, Jan 4, 2011 at 9:55 AM, Sam <u2nsam at gmail.com> wrote:
> >>>
> >>> hello,
> >>>
> >>> how to increase the scalability / performance of the conference .
> >>>
> >>> Are there any parameters to be observed that could do a trick ?
> >>>
> >>> The scenario would be multiple conferencing bridges and multiple codecs
> >>> involved.
> >>>
> >>> Is is possible to multiple threading of profiles for conferences so
> that
> >>> it gets the scalability ?
> >>>
> >>> Regards
> >>> Sam
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From Avi at aMarcus.com  Tue Jan  4 15:37:34 2011
From: Avi at aMarcus.com (Avi Marcus)
Date: Tue, 4 Jan 2011 14:37:34 +0200
Subject: [Freeswitch-users] conferencing
In-Reply-To: <AANLkTikuYNOL55j0rBFgG=FqFQnPWfxT1J13Q1M8ZMf6@mail.gmail.com>
References: <AANLkTikgBYfg_UiZnbc==8dgs32G8fDkSvak=-DBG_Ak@mail.gmail.com>
	<AANLkTino24LWO+V9egF70aPu0Kojj0sLTMiKBDY6E2tr@mail.gmail.com>
	<AANLkTinLij+oYGQ5Go1_Etg_kZvb_+wtKAQRg4krcwhU@mail.gmail.com>
	<AANLkTi=KPRnLDNLL8uUCxCvsFiRHrrJprxNVKjM7Kw=B@mail.gmail.com>
	<AANLkTikuYNOL55j0rBFgG=FqFQnPWfxT1J13Q1M8ZMf6@mail.gmail.com>
Message-ID: <AANLkTimRVUySdwVL7kNXqzZPqJfyR+XU=pd2dRJKmPhT@mail.gmail.com>

If you mute most of the callers, that would limit the audio that would need
to be processed.
-Avi

On Tue, Jan 4, 2011 at 12:45 PM, Sam <u2nsam at gmail.com> wrote:

> calls would not be TDM it would be ip-ip calls.
> Ok G711 uses less CPU than other codecs,how can i utilized multiple
> threading/profiles or any other options to scale ? as it would be an
> incoming call always on FS.
>
> Regds
> Sam
>
>
> On Tue, Jan 4, 2011 at 3:53 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> That will still involve transcoding. FS must combine all speaking
>> channels into a single stream that can be sent to the conference
>> members. That can't be done in a compressed audio format, so it must
>> convert G711 to L16 for each speaking member, combine those pieces of
>> audio, then convert L16 back to G711 to send to the conference
>> members. Additionally, since the conference members are not
>> necessarily all using the same codec that last transcoding step will
>> occur once for each member, not just once for the conference. Your
>> best bet is to use a codec that uses as little CPU for transcoding as
>> possible - in software I'm not sure which is best there (G711 is
>> simple so possibly a good choice), in hardware Sangoma D100/D500 would
>> almost all the processing off the CPU.
>>
>> -Steve
>>
>>
>> On 4 January 2011 10:15, Sam <u2nsam at gmail.com> wrote:
>> > Ok if i remove transcoding out of the picture and all conf are running
>> on
>> > g711, what rest would matters ?
>> >
>> >
>> > Regds
>> > Sam
>> >
>> >
>> > On Tue, Jan 4, 2011 at 2:56 PM, Avi Marcus <Avi at amarcus.com> wrote:
>> >>
>> >> It's the transcoding that will kill you. If that's truly needed, you
>> may
>> >> want to look into multiple machines handling transcoding and then
>> bridging
>> >> into the conference, or the new transcoding hardware from sangoma.
>> >> -Avi
>> >> On Tue, Jan 4, 2011 at 9:55 AM, Sam <u2nsam at gmail.com> wrote:
>> >>>
>> >>> hello,
>> >>>
>> >>> how to increase the scalability / performance of the conference .
>> >>>
>> >>> Are there any parameters to be observed that could do a trick ?
>> >>>
>> >>> The scenario would be multiple conferencing bridges and multiple
>> codecs
>> >>> involved.
>> >>>
>> >>> Is is possible to multiple threading of profiles for conferences so
>> that
>> >>> it gets the scalability ?
>> >>>
>> >>> Regards
>> >>> Sam
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
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>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
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>
>
> _______________________________________________
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From anita.hall at simmortel.com  Tue Jan  4 15:38:31 2011
From: anita.hall at simmortel.com (Anita Hall)
Date: Tue, 4 Jan 2011 18:08:31 +0530
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 55, Issue 29
In-Reply-To: <mailman.1324.1294103607.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.1324.1294103607.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <AANLkTimT5Xk5JAYUbyT5KhL4T5UR-XJZ0N9PCW=RPYvm@mail.gmail.com>

Thanks Wasim !

That solved the problem. My link is up and am not able to make outbound
calling.
I am taking that discussion to the asterisk-ss7 group.

Thanks,
Anita.

On Tue, Jan 4, 2011 at 6:43 AM, <
freeswitch-users-request at lists.freeswitch.org> wrote:

> Send FreeSWITCH-users mailing list submissions to
>        freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
>        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
>        freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
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>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> Today's Topics:
>
>   1. Re: calling card app (Michael Collins)
>   2. Re: calling card app (Rafqat .)
>   3. Re: libss7 * Sangoma (Wasim Baig)
>
>
> ---------- Forwarded message ----------
> From: Michael Collins <msc at freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Mon, 3 Jan 2011 13:41:57 -0800
> Subject: Re: [Freeswitch-users] calling card app
>
>
> On Mon, Jan 3, 2011 at 1:17 PM, Rafqat . <rafonline at hotmail.com> wrote:
>
>>
>> btw.
>>
>> My lua script currently does the following:
>>
>> 1. Ask for PIN.
>> 2. Gets funds for PIN (DB lookup using freeswitch.Dbh) and informs the
>> user of the funds.
>> 3. Ask for destination number.
>> 4. Checks if enough funds (again using freeswitch Dbh).
>> 5. Gets auto route using mod_lcr.
>> 6. Populates leg b session variables necessary for mod_nibble.
>> 7. Does the bridging.
>>
>> As per your advice I will move the bridging stuff out of the script.  Do
>> you think the rest will be ok in lua?
>>
>
> I'd say that looks good. The Lua script only stays active until you get to
> step 7 where you do a transfer instead of bridge. It's only a few seconds of
> work.
> -MC
>
>
>>
>> CHeers
>>
>> Raf
>>
>>
>> ------------------------------
>> From: rafonline at hotmail.com
>>
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: RE: [Freeswitch-users] calling card app
>> Date: Mon, 3 Jan 2011 21:07:33 +0000
>>
>>
>>
>> Thanks very much for the advice.
>>
>> Much Appreciated.
>>
>> ------------------------------
>> Date: Mon, 3 Jan 2011 11:31:57 -0800
>> From: msc at freeswitch.org
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] calling card app
>>
>> If you have a powerful machine then you can probably scale to several
>> hundred concurrent calls, depending on transcoding, call recording needs,
>> etc. We've seen some boxes that can handle literally thousands of concurrent
>> calls, but the scenarios are never exactly the same. Also, Lua is very
>> lightweight, so if you're using it just to capture a PIN code or something
>> then you should be okay. Just be sure to exit the Lua script and let the
>> dialplan handle the bridge app. (See chapter 7 of the FS book for more
>> information on Lua scripting tips.)
>>
>> -MC
>>
>> On Sun, Jan 2, 2011 at 11:13 AM, Rafqat . <rafonline at hotmail.com> wrote:
>>
>>  Hi
>>
>> As stated in some of my previous posts, I am writing a calling card system
>> (not too sure of potential number of concurrent users yet).
>>
>> At the moment I am simply doing everything in a single lua script
>> utilising mod_lcr and mod_nibble.  It seems to work ok, but I have not
>> stress tested it yet.
>>
>> I was wondering (at a high level) if this will suffice or should I be
>> offloading operations such as pin validation and credit checking to another
>> server (maybe utilising mod_rad_auth?).
>>
>> Cheers
>>
>> Raf
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________ FreeSWITCH-users mailing
>> list FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
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>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>> http://www.freeswitch.org
>>
>>
>
>
> ---------- Forwarded message ----------
> From: "Rafqat ." <rafonline at hotmail.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Date: Mon, 3 Jan 2011 22:02:11 +0000
> Subject: Re: [Freeswitch-users] calling card app
>
> Thanks again for your help MC.
>
> ------------------------------
> Date: Mon, 3 Jan 2011 13:41:57 -0800
> From: msc at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] calling card app
>
>
>
> On Mon, Jan 3, 2011 at 1:17 PM, Rafqat . <rafonline at hotmail.com> wrote:
>
>
> btw.
>
> My lua script currently does the following:
>
> 1. Ask for PIN.
> 2. Gets funds for PIN (DB lookup using freeswitch.Dbh) and informs the user
> of the funds.
> 3. Ask for destination number.
> 4. Checks if enough funds (again using freeswitch Dbh).
> 5. Gets auto route using mod_lcr.
> 6. Populates leg b session variables necessary for mod_nibble.
> 7. Does the bridging.
>
> As per your advice I will move the bridging stuff out of the script.  Do
> you think the rest will be ok in lua?
>
>
> I'd say that looks good. The Lua script only stays active until you get to
> step 7 where you do a transfer instead of bridge. It's only a few seconds of
> work.
> -MC
>
>
>
> CHeers
>
> Raf
>
>
> ------------------------------
> From: rafonline at hotmail.com
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: RE: [Freeswitch-users] calling card app
> Date: Mon, 3 Jan 2011 21:07:33 +0000
>
>
>
> Thanks very much for the advice.
>
> Much Appreciated.
>
> ------------------------------
> Date: Mon, 3 Jan 2011 11:31:57 -0800
> From: msc at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] calling card app
>
> If you have a powerful machine then you can probably scale to several
> hundred concurrent calls, depending on transcoding, call recording needs,
> etc. We've seen some boxes that can handle literally thousands of concurrent
> calls, but the scenarios are never exactly the same. Also, Lua is very
> lightweight, so if you're using it just to capture a PIN code or something
> then you should be okay. Just be sure to exit the Lua script and let the
> dialplan handle the bridge app. (See chapter 7 of the FS book for more
> information on Lua scripting tips.)
>
> -MC
>
> On Sun, Jan 2, 2011 at 11:13 AM, Rafqat . <rafonline at hotmail.com> wrote:
>
>  Hi
>
> As stated in some of my previous posts, I am writing a calling card system
> (not too sure of potential number of concurrent users yet).
>
> At the moment I am simply doing everything in a single lua script utilising
> mod_lcr and mod_nibble.  It seems to work ok, but I have not stress tested
> it yet.
>
> I was wondering (at a high level) if this will suffice or should I be
> offloading operations such as pin validation and credit checking to another
> server (maybe utilising mod_rad_auth?).
>
> Cheers
>
> Raf
>
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>
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> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
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>
>
> ---------- Forwarded message ----------
> From: Wasim Baig <wasim at convergence.pk>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 4 Jan 2011 06:12:34 +0500
> Subject: Re: [Freeswitch-users] libss7 * Sangoma
> anita:
>
> asterisk-ss7 is the correct list for this
>
> change
> TDMV_DCHAN      = 16
> to
> TDMV_DCHAN      = 0
> in wanpipeX.conf
>
> fwiw, chan_ss7 is nicer ...
>
> -wasim
>
> On Mon, Jan 3, 2011 at 22:57, Anita Hall <anita.hall at simmortel.com> wrote:
>
>> Hi
>>
>> This question pertains libss7 with Sangoma A108 card on Asterisk. It does
>> not concern freeswitch but I suppose Sangoma folks frequent this list more
>> :)
>>
>> I am unable to make libss7 work with Sangoma. Here are the details.
>>
>> Could you please provide me some pointers ?
>>
>> Thanks,
>> Anita.
>>
>> debian:~# uname -a
>> Linux debian 2.6.26-2-686 #1 SMP Thu Nov 25 01:53:57 UTC 2010 i686
>> GNU/Linux
>>
>> debian:~# dahdi_hardware
>> pci:0000:05:04.0     wanpipe-     1923:0100 Sangoma Technologies Corp.
>> A104d QUAD T1/E1 AFT card
>> pci:0000:07:04.0     wanpipe-     1923:0100 Sangoma Technologies Corp.
>> A104d QUAD T1/E1 AFT card
>>
>>
>>
>> debian:~# cat /etc/dahdi/system.conf
>> #autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
>> #autogenrated on 2011-01-03
>> #Dahdi Channels Configurations
>> #For detailed Dahdi options, view /etc/dahdi/system.conf.bak
>> loadzone=us
>> defaultzone=us
>>
>> #Sangoma A108 port 1 [slot:4 bus:5 span:1] <wanpipe1>
>> span=1,1,0,ccs,hdb3,crc4
>> bchan=1-15,17-31
>> echocanceller=mg2,1-15,17-31
>> mtp2=16
>>
>> #Sangoma A108 port 2 [slot:4 bus:5 span:2] <wanpipe2>
>> span=2,1,0,ccs,hdb3,crc4
>> bchan=32-46,48-62
>> echocanceller=mg2,32-46,48-62
>> mtp2=47
>>
>>
>> debian:~# cat /etc/wanpipe/wanpipe1.conf
>> #=============================
>> ===================
>> # WANPIPE1 Configuration File
>> #================================================
>> #
>> # Date: Wed Dec  6 20:29:03 UTC 2006
>> #
>> # Note: This file was generated automatically
>> #       by /usr/local/sbin/setup-sangoma program.
>> #
>> #       If you want to edit this file, it is
>> #       recommended that you use wancfg program
>> #       to do so.
>> #================================================
>> # Sangoma Technologies Inc.
>> #================================================
>>
>> [devices]
>> wanpipe1 = WAN_AFT_TE1, Comment
>>
>> [interfaces]
>> w1g1 = wanpipe1, , TDM_VOICE, Comment
>>
>> [wanpipe1]
>> CARD_TYPE       = AFT
>> S514CPU         = A
>> CommPort        = PRI
>> AUTO_PCISLOT    = NO
>> PCISLOT         = 4
>> PCIBUS          = 5
>> FE_MEDIA        = E1
>> FE_LCODE        = HDB3
>> FE_FRAME        = NCRC4
>> FE_LINE         = 1
>> TE_CLOCK        = NORMAL
>> TE_REF_CLOCK    = 0
>> TE_SIG_MODE     = CCS
>> TE_HIGHIMPEDANCE        = NO
>> LBO             = 120OH
>> FE_TXTRISTATE   = NO
>> MTU             = 1500
>> UDPPORT         = 9000
>> TTL             = 255
>> IGNORE_FRONT_END = NO
>> TDMV_SPAN       = 1
>> TDMV_DCHAN      = 16
>> TDMV_HW_DTMF    = NO
>> TDMV_HW_FAX_DETECT = NO
>>
>> [w1g1]
>> ACTIVE_CH       = ALL
>> TDMV_HWEC       = NO
>>
>>
>> debian:~# cat /etc/wanpipe/wanpipe2.conf
>> #================================================
>> # WANPIPE1 Configuration File
>> #================================================
>> #
>> # Date: Wed Dec  6 20:29:03 UTC 2006
>> #
>> # Note: This file was generated automatically
>> #       by /usr/local/sbin/setup-sangoma program.
>> #
>> #       If you want to edit this file, it is
>> #       recommended that you use wancfg program
>> #       to do so.
>> #================================================
>> # Sangoma Technologies Inc.
>> #================================================
>>
>> [devices]
>> wanpipe2 = WAN_AFT_TE1, Comment
>>
>> [interfaces]
>> w2g1 = wanpipe2, , TDM_VOICE, Comment
>>
>> [wanpipe2]
>> CARD_TYPE       = AFT
>> S514CPU         = A
>> CommPort        = PRI
>> AUTO_PCISLOT    = NO
>> PCISLOT         = 4
>> PCIBUS          = 5
>> FE_MEDIA        = E1
>> FE_LCODE        = HDB3
>> FE_FRAME        = NCRC4
>> FE_LINE         = 2
>> TE_CLOCK        = NORMAL
>> TE_REF_CLOCK    = 0
>> TE_SIG_MODE     = CCS
>> TE_HIGHIMPEDANCE        = NO
>> LBO             = 120OH
>> FE_TXTRISTATE   = NO
>> MTU             = 1500
>> UDPPORT         = 9000
>> TTL             = 255
>> IGNORE_FRONT_END = NO
>> TDMV_SPAN       = 2
>> TDMV_DCHAN      = 16
>> TDMV_HW_DTMF    = NO
>> TDMV_HW_FAX_DETECT = NO
>>
>> [w2g1]
>> ACTIVE_CH       = ALL
>> TDMV_HWEC       = NO
>>
>> debian:~# cat /etc/asterisk/chan_dahdi.conf
>> ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
>> ;autogenrated on 2011-01-03
>> ;Dahdi Channels Configurations
>> ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak
>>
>> [trunkgroups]
>>
>> [channels]
>> context=default
>> usecallerid=yes
>> hidecallerid=no
>> callwaiting=yes
>> usecallingpres=yes
>> callwaitingcallerid=yes
>> threewaycalling=yes
>> transfer=yes
>> canpark=yes
>> cancallforward=yes
>> callreturn=yes
>> echocancel=yes
>> echocancelwhenbridged=yes
>> relaxdtmf=yes
>> rxgain=0.0
>> txgain=0.0
>> group=1
>> callgroup=1
>> pickupgroup=1
>> immediate=no
>>
>> ;Sangoma A108 port 1 [slot:4 bus:5 span:1] <wanpipe1>
>> switchtype=euroisdn
>> context=tata
>> group=1
>> echocancel=yes
>> signaling=ss7                    ;this is ss7 signaling
>> ss7type=itu                     ;using the ITU variant
>> ss7_called_nai=dynamic           ;NAI for outgoing calls
>> ss7_calling_nai=dynamic          ;NAI for incoming calls
>> ss7_internationalprefix=00 ;international prefix value for incoming calls
>> ss7_nationalprefix=0            ;national prefix value for incoming calls
>> ss7_subscriberprefix=            ;subscriber prefix value for incoming
>> calls
>> ss7_unknownprefix=              ;unknown prefix value for incoming calls
>> ss7_explictacm=yes               ;ACM is send as soon as call enters the
>> dial plan...may not accepted yet though
>> linkset=1                         ;arbitrary name for this set of channels
>> pointcode=13323                      ;the point code for this system...aka
>> SPC
>> adjpointcode=12650                  ;the point code for the system that we
>> are signaling to... aka APC
>> defaultdpc=12650                     ;the point code for the system that
>> the CICs will be negotiated with...aka DPC
>> networkindicator=international  ;NI value for MTP3
>> cicbeginswith=1                  ;the starting value of the CICs
>> channel =>1-15
>> cicbeginswith=17                 ;the starting value of the CICs
>> channel =>17-31         ;the channels that are CICs
>> sigchan=16                     ;the signaling channel
>>
>> ;Sangoma A108 port 2 [slot:4 bus:5 span:2] <wanpipe2>
>> switchtype=euroisdn
>> context=tata
>> group=1
>> echocancel=yes
>> cicbeginswith=32                  ;the starting value of the CICs
>> channel =>32-46
>> cicbeginswith=48                  ;the starting value of the CICs
>> channel =>48-62         ;the channels that are CICs
>> sigchan=47                       ;the signaling channel
>>
>>
>> debian:~# ifconfig w1g1
>> w1g1      Link encap:Point-to-Point Protocol
>>          UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
>>          RX packets:772814 errors:0 dropped:0 overruns:0 frame:0
>>          TX packets:772814 errors:0 dropped:0 overruns:3 carrier:0
>>          collisions:0 txqueuelen:100
>>          RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
>>          Interrupt:16 Memory:f8d00000-f8d01fff
>>
>> debian:~# ifconfig w1g1
>> w1g1      Link encap:Point-to-Point Protocol
>>          UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
>>          RX packets:774782 errors:0 dropped:0 overruns:0 frame:0
>>          TX packets:774782 errors:0 dropped:0 overruns:3 carrier:0
>>          collisions:0 txqueuelen:100
>>          RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
>>          Interrupt:16 Memory:f8d00000-f8d01fff
>>
>> debian:~# ifconfig w1g1
>> w1g1      Link encap:Point-to-Point Protocol
>>          UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
>>          RX packets:775351 errors:0 dropped:0 overruns:0 frame:0
>>          TX packets:775351 errors:0 dropped:0 overruns:3 carrier:0
>>          collisions:0 txqueuelen:100
>>          RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
>>          Interrupt:16 Memory:f8d00000-f8d01fff
>>
>> debian:~# ifconfig w1g1
>> w1g1      Link encap:Point-to-Point Protocol
>>          UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
>>          RX packets:775919 errors:0 dropped:0 overruns:0 frame:0
>>          TX packets:775919 errors:0 dropped:0 overruns:3 carrier:0
>>          collisions:0 txqueuelen:100
>>          RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
>>          Interrupt:16 Memory:f8d00000-f8d01fff
>>
>> debian:~# ifconfig w1g1
>> w1g1      Link encap:Point-to-Point Protocol
>>          UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
>>          RX packets:776478 errors:0 dropped:0 overruns:0 frame:0
>>          TX packets:776478 errors:0 dropped:0 overruns:3 carrier:0
>>          collisions:0 txqueuelen:100
>>          RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
>>          Interrupt:16 Memory:f8d00000-f8d01fff
>>
>> debian:~# ifconfig w1g1
>> w1g1      Link encap:Point-to-Point Protocol
>>          UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
>>          RX packets:777079 errors:0 dropped:0 overruns:0 frame:0
>>          TX packets:777079 errors:0 dropped:0 overruns:3 carrier:0
>>          collisions:0 txqueuelen:100
>>          RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
>>          Interrupt:16 Memory:f8d00000-f8d01fff
>>
>> debian:~# ifconfig w1g1
>> w1g1      Link encap:Point-to-Point Protocol
>>          UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
>>          RX packets:778134 errors:0 dropped:0 overruns:0 frame:0
>>          TX packets:778134 errors:0 dropped:0 overruns:3 carrier:0
>>          collisions:0 txqueuelen:100
>>          RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
>>          Interrupt:16 Memory:f8d00000-f8d01fff
>>
>>
>> debian:~# ifconfig w2g1
>> w2g1      Link encap:Point-to-Point Protocol
>>          UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
>>          RX packets:792251 errors:0 dropped:0 overruns:0 frame:0
>>          TX packets:792251 errors:0 dropped:0 overruns:1 carrier:0
>>          collisions:0 txqueuelen:100
>>          RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
>>          Interrupt:16 Memory:f8d00000-f8d01fff
>>
>> debian:~# ifconfig w2g1
>> w2g1      Link encap:Point-to-Point Protocol
>>          UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
>>          RX packets:792715 errors:0 dropped:0 overruns:0 frame:0
>>          TX packets:792715 errors:0 dropped:0 overruns:1 carrier:0
>>          collisions:0 txqueuelen:100
>>          RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
>>          Interrupt:16 Memory:f8d00000-f8d01fff
>>
>> debian:~# ifconfig w2g1
>> w2g1      Link encap:Point-to-Point Protocol
>>          UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
>>          RX packets:793219 errors:0 dropped:0 overruns:0 frame:0
>>          TX packets:793219 errors:0 dropped:0 overruns:1 carrier:0
>>          collisions:0 txqueuelen:100
>>          RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
>>          Interrupt:16 Memory:f8d00000-f8d01fff
>>
>>
>> wanpipemon -i w2g1 -c Ta
>>
>> ***** w2g1: E1 Alarms (Framer) *****
>>
>> ALOS:   OFF     | LOS:  OFF
>> RED:    OFF     | AIS:  OFF
>> LOF:    OFF     | RAI:  OFF
>>
>> ***** w2g1: E1 Alarms (LIU) *****
>>
>> Short Circuit:  OFF
>> Open Circuit:   OFF
>> Loss of Signal: OFF
>>
>>
>> ***** w2g1: E1 Performance Monitoring Counters *****
>>
>> Line Code Violation     : 0
>> Far End Block Errors    : 0
>> CRC4 Errors             : 0
>> FAS Errors              : 0
>>
>>
>> Rx Level        : > -2.5db
>>
>>
>> debian:~# wanpipemon -i w1g1 -c Ta
>>
>> ***** w1g1: E1 Alarms (Framer) *****
>>
>> ALOS:   OFF     | LOS:  OFF
>> RED:    OFF     | AIS:  OFF
>> LOF:    OFF     | RAI:  OFF
>>
>> ***** w1g1: E1 Alarms (LIU) *****
>>
>> Short Circuit:  OFF
>> Open Circuit:   OFF
>> Loss of Signal: OFF
>>
>>
>> ***** w1g1: E1 Performance Monitoring Counters *****
>>
>> Line Code Violation     : 371
>> Far End Block Errors    : 0
>> CRC4 Errors             : 0
>> FAS Errors              : 0
>>
>>
>> Rx Level        : > -2.5db
>>
>>
>> Jan  3 20:33:34 debian kernel: [55441.172179] wanpipe1: Shutting Down!
>> Jan  3 20:33:34 debian kernel: [55441.172179] wanpipe1: E1 Front End
>> unconfigation!
>> Jan  3 20:33:34 debian kernel: [55441.172179] wanpipe1: Unregister Wanpipe
>> device from Zaptel!
>> Jan  3 20:33:34 debian kernel: [55441.178246] wanpipe1: unregistering
>> 'w1g1'
>> Jan  3 20:33:34 debian kernel: [55441.248278] wanpipe1: TASKQ Not Running
>> Jan  3 20:33:34 debian kernel: [55441.248281] wanpipe1: E1 Front End
>> unconfigation!
>> Jan  3 20:33:34 debian kernel: [55441.248314] wanpipe1: AFT communications
>> disabled! (Dev Cnt: 1 Cause: Device Down)
>> Jan  3 20:33:34 debian kernel: [55441.248335] wanpipe1: E1 Front End
>> unconfigation!
>> Jan  3 20:33:34 debian kernel: [55441.248373] wanpipe1: AFT communications
>> disabled! (Dev Cnt: 1 Cause: Device Down)
>> Jan  3 20:33:34 debian kernel: [55441.248415] wanpipe1: Global Chip
>> Shutdown Usage=1
>> Jan  3 20:33:34 debian kernel: [55441.248419] wanpipe1: Global E1 Front
>> End unconfigation!
>> Jan  3 20:33:34 debian kernel: [55441.250485] wanpipe1: Master shutting
>> down
>> Jan  3 20:33:34 debian kernel: [55441.444571] wanpipe1: Starting WAN Setup
>> Jan  3 20:33:34 debian kernel: [55441.444575] wanpipe1: Locating:
>> A101/1D/A102/2D/4/4D/8 card, CPU A, PciSlot=4, PciBus=5
>> Jan  3 20:33:34 debian kernel: [55441.444579] wanpipe1: Found:
>> A101/1D/A102/2D/4/4D/8 card, CPU A, PciSlot=4, PciBus=5, Port=0
>> Jan  3 20:33:34 debian kernel: [55441.444599] wanpipe1: AFT PCI memory at
>> 0xD3200000
>> Jan  3 20:33:34 debian kernel: [55441.444600] wanpipe1: IRQ 16 allocated
>> to the AFT PCI card
>> Jan  3 20:33:34 debian kernel: [55441.444608] wanpipe1: Starting AFT 2/4/8
>> Hardware Init.
>> Jan  3 20:33:34 debian kernel: [55441.444614] wanpipe1: Enabling front end
>> link monitor
>> Jan  3 20:33:34 debian kernel: [55441.444616] wanpipe1: Global Chip
>> Configuration: used=1
>> Jan  3 20:33:34 debian kernel: [55441.448455] wanpipe1: Global E1 Front
>> End configuration
>> Jan  3 20:33:34 debian kernel: [55441.448455] wanpipe1: AFT Data Mux Bit
>> Map: 0x01234567
>> Jan  3 20:33:34 debian kernel: [55441.448455] wanpipe1: Defaulting E1 Rx
>> Sens. Gain= 43 db
>> Jan  3 20:33:34 debian kernel: [55441.448455] wanpipe1: Configuring DS
>> DS26528 E1 FE
>> Jan  3 20:33:34 debian kernel: [55441.448455] wanpipe1:    Port
>> 1,HDB3,non-CRC4,120OH
>> Jan  3 20:33:34 debian kernel: [55441.448455] wanpipe1:    Clk Normal:0,
>> Channels: FFFFFFFF
>> Jan  3 20:33:34 debian kernel: [55441.448455] wanpipe1:    Sig Mode CCS
>> Jan  3 20:33:34 debian kernel: [55441.456490] wanpipe1:    Rx Sensitivity
>> Gain 43dB (default).
>> Jan  3 20:33:34 debian kernel: [55441.457242] wanpipe1: Front end
>> successful
>> Jan  3 20:33:34 debian kernel: [55441.457469] wanpipe1: Front End
>> Interface Ready 0x40000000
>> Jan  3 20:33:34 debian kernel: [55441.457473] wanpipe1: WARNING: No Echo
>> Canceller channels are available!
>> Jan  3 20:33:34 debian kernel: [55441.457494] wanpipe1: Configuring Device
>>   :wanpipe1  FrmVr=39
>> Jan  3 20:33:34 debian kernel: [55441.457495] wanpipe1:    Global MTU
>>       = 1500
>> Jan  3 20:33:34 debian kernel: [55441.457496] wanpipe1:    Global MRU
>>       = 1500
>> Jan  3 20:33:34 debian kernel: [55441.457497] wanpipe1:    Data Mux Map
>>       = 0x01234567
>> Jan  3 20:33:34 debian kernel: [55441.457498] wanpipe1:    Rx CRC Bytes
>>       = 0
>> Jan  3 20:33:34 debian kernel: [55441.457499] wanpipe1:    Global TDM Int
>>       = Enabled
>> Jan  3 20:33:34 debian kernel: [55441.457500] wanpipe1:    Global TDM Ring
>>      = Enabled
>> Jan  3 20:33:34 debian kernel: [55441.457501] wanpipe1:    TDMV HW
>> DTMF/FAX     = Disabled/Disabled(0)
>> Jan  3 20:33:34 debian kernel: [55441.457502] wanpipe1:    TDMV Span
>>      = 1 : Enabled
>> Jan  3 20:33:34 debian kernel: [55441.457503] wanpipe1:    TDMV Dummy
>>   = Disabled
>> Jan  3 20:33:34 debian kernel: [55441.457505] wanpipe1:    RTP TAP
>>      = Disabled
>> Jan  3 20:33:34 debian kernel: [55441.457540] wanpipe1: Configuring
>> Interface: w1g1
>> Jan  3 20:33:34 debian kernel: [55441.457543] wanpipe1:w1g1: Running in
>> TDM Voice Zaptel Mode.
>> Jan  3 20:33:34 debian kernel: [55441.457547] wanpipe1:    Fifo Level
>> Map:0x01041040
>> Jan  3 20:33:34 debian kernel: [55441.457549] wanpipe1:    MRU
>> :8
>> Jan  3 20:33:34 debian kernel: [55441.457550] wanpipe1:    MTU
>> :8
>> Jan  3 20:33:34 debian kernel: [55441.457551] wanpipe1:    HDLC Eng
>>  :Off (Transparent) | N/A
>> Jan  3 20:33:34 debian kernel: [55441.457553] wanpipe1:    Data Mux Ctrl
>> :On
>> Jan  3 20:33:34 debian kernel: [55441.457554] wanpipe1:    Active Ch Map
>> :0x00000002
>> Jan  3 20:33:34 debian kernel: [55441.457555] wanpipe1:    First TSlot
>> :1
>> Jan  3 20:33:34 debian kernel: [55441.457568] wanpipe1:
>>  DMA/Len/Chain/EC :4/1024/Off/Off
>> Jan  3 20:33:34 debian kernel: [55441.457575] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.457581] wanpipe1:    Active Ch Map
>> :0x00000004
>> Jan  3 20:33:34 debian kernel: [55441.457582] wanpipe1:    First TSlot
>> :2
>> Jan  3 20:33:34 debian kernel: [55441.457597] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.457602] wanpipe1:    Active Ch Map
>> :0x00000008
>> Jan  3 20:33:34 debian kernel: [55441.457603] wanpipe1:    First TSlot
>> :3
>> Jan  3 20:33:34 debian kernel: [55441.457618] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.457623] wanpipe1:    Active Ch Map
>> :0x00000010
>> Jan  3 20:33:34 debian kernel: [55441.457624] wanpipe1:    First TSlot
>> :4
>> Jan  3 20:33:34 debian kernel: [55441.457637] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.457643] wanpipe1:    Active Ch Map
>> :0x00000020
>> Jan  3 20:33:34 debian kernel: [55441.457644] wanpipe1:    First TSlot
>> :5
>> Jan  3 20:33:34 debian kernel: [55441.457657] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.457663] wanpipe1:    Active Ch Map
>> :0x00000040
>> Jan  3 20:33:34 debian kernel: [55441.457664] wanpipe1:    First TSlot
>> :6
>> Jan  3 20:33:34 debian kernel: [55441.457678] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.457684] wanpipe1:    Active Ch Map
>> :0x00000080
>> Jan  3 20:33:34 debian kernel: [55441.457685] wanpipe1:    First TSlot
>> :7
>> Jan  3 20:33:34 debian kernel: [55441.457698] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.457705] wanpipe1:    Active Ch Map
>> :0x00000100
>> Jan  3 20:33:34 debian kernel: [55441.457706] wanpipe1:    First TSlot
>> :8
>> Jan  3 20:33:34 debian kernel: [55441.457719] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.457725] wanpipe1:    Active Ch Map
>> :0x00000200
>> Jan  3 20:33:34 debian kernel: [55441.457726] wanpipe1:    First TSlot
>> :9
>> Jan  3 20:33:34 debian kernel: [55441.457740] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.457746] wanpipe1:    Active Ch Map
>> :0x00000400
>> Jan  3 20:33:34 debian kernel: [55441.457747] wanpipe1:    First TSlot
>> :10
>> Jan  3 20:33:34 debian kernel: [55441.457760] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.457765] wanpipe1:    Active Ch Map
>> :0x00000800
>> Jan  3 20:33:34 debian kernel: [55441.457766] wanpipe1:    First TSlot
>> :11
>> Jan  3 20:33:34 debian kernel: [55441.457781] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.457786] wanpipe1:    Active Ch Map
>> :0x00001000
>> Jan  3 20:33:34 debian kernel: [55441.457787] wanpipe1:    First TSlot
>> :12
>> Jan  3 20:33:34 debian kernel: [55441.457801] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.457807] wanpipe1:    Active Ch Map
>> :0x00002000
>> Jan  3 20:33:34 debian kernel: [55441.457808] wanpipe1:    First TSlot
>> :13
>> Jan  3 20:33:34 debian kernel: [55441.457822] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.457828] wanpipe1:    Active Ch Map
>> :0x00004000
>> Jan  3 20:33:34 debian kernel: [55441.457829] wanpipe1:    First TSlot
>> :14
>> Jan  3 20:33:34 debian kernel: [55441.457842] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.457848] wanpipe1:    Active Ch Map
>> :0x00008000
>> Jan  3 20:33:34 debian kernel: [55441.457849] wanpipe1:    First TSlot
>> :15
>> Jan  3 20:33:34 debian kernel: [55441.457862] wanpipe1: Configuring
>> Interface: w1g1
>> Jan  3 20:33:34 debian kernel: [55441.457864] wanpipe1:w1g1: Running in
>> TDM DCHAN Voice Zaptel Mode.
>> Jan  3 20:33:34 debian kernel: [55441.457866] wanpipe1:    MRU
>> :1500
>> Jan  3 20:33:34 debian kernel: [55441.457867] wanpipe1:    MTU
>> :1500
>> Jan  3 20:33:34 debian kernel: [55441.457868] wanpipe1:    HDLC Eng
>>  :On | N/A
>> Jan  3 20:33:34 debian kernel: [55441.457869] wanpipe1:    Data Mux Ctrl
>> :Off
>> Jan  3 20:33:34 debian kernel: [55441.457870] wanpipe1:    Active Ch Map
>> :0x00010000
>> Jan  3 20:33:34 debian kernel: [55441.457871] wanpipe1:    First TSlot
>> :16
>> Jan  3 20:33:34 debian kernel: [55441.457881] wanpipe1:
>>  DMA/Len/Chain/EC :65/4096/On/Off
>> Jan  3 20:33:34 debian kernel: [55441.457905] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.457910] wanpipe1:    Active Ch Map
>> :0x00020000
>> Jan  3 20:33:34 debian kernel: [55441.457911] wanpipe1:    First TSlot
>> :17
>> Jan  3 20:33:34 debian kernel: [55441.457925] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.457931] wanpipe1:    Active Ch Map
>> :0x00040000
>> Jan  3 20:33:34 debian kernel: [55441.457932] wanpipe1:    First TSlot
>> :18
>> Jan  3 20:33:34 debian kernel: [55441.457945] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.457951] wanpipe1:    Active Ch Map
>> :0x00080000
>> Jan  3 20:33:34 debian kernel: [55441.457952] wanpipe1:    First TSlot
>> :19
>> Jan  3 20:33:34 debian kernel: [55441.457967] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.457973] wanpipe1:    Active Ch Map
>> :0x00100000
>> Jan  3 20:33:34 debian kernel: [55441.457974] wanpipe1:    First TSlot
>> :20
>> Jan  3 20:33:34 debian kernel: [55441.457987] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.457993] wanpipe1:    Active Ch Map
>> :0x00200000
>> Jan  3 20:33:34 debian kernel: [55441.457994] wanpipe1:    First TSlot
>> :21
>> Jan  3 20:33:34 debian kernel: [55441.458009] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.458015] wanpipe1:    Active Ch Map
>> :0x00400000
>> Jan  3 20:33:34 debian kernel: [55441.458016] wanpipe1:    First TSlot
>> :22
>> Jan  3 20:33:34 debian kernel: [55441.458030] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.458036] wanpipe1:    Active Ch Map
>> :0x00800000
>> Jan  3 20:33:34 debian kernel: [55441.458037] wanpipe1:    First TSlot
>> :23
>> Jan  3 20:33:34 debian kernel: [55441.458051] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.458057] wanpipe1:    Active Ch Map
>> :0x01000000
>> Jan  3 20:33:34 debian kernel: [55441.458058] wanpipe1:    First TSlot
>> :24
>> Jan  3 20:33:34 debian kernel: [55441.458072] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.458078] wanpipe1:    Active Ch Map
>> :0x02000000
>> Jan  3 20:33:34 debian kernel: [55441.458079] wanpipe1:    First TSlot
>> :25
>> Jan  3 20:33:34 debian kernel: [55441.458094] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.458100] wanpipe1:    Active Ch Map
>> :0x04000000
>> Jan  3 20:33:34 debian kernel: [55441.458101] wanpipe1:    First TSlot
>> :26
>> Jan  3 20:33:34 debian kernel: [55441.458116] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.458122] wanpipe1:    Active Ch Map
>> :0x08000000
>> Jan  3 20:33:34 debian kernel: [55441.458123] wanpipe1:    First TSlot
>> :27
>> Jan  3 20:33:34 debian kernel: [55441.458137] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.458143] wanpipe1:    Active Ch Map
>> :0x10000000
>> Jan  3 20:33:34 debian kernel: [55441.458144] wanpipe1:    First TSlot
>> :28
>> Jan  3 20:33:34 debian kernel: [55441.458160] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.458166] wanpipe1:    Active Ch Map
>> :0x20000000
>> Jan  3 20:33:34 debian kernel: [55441.458167] wanpipe1:    First TSlot
>> :29
>> Jan  3 20:33:34 debian kernel: [55441.458181] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.458187] wanpipe1:    Active Ch Map
>> :0x40000000
>> Jan  3 20:33:34 debian kernel: [55441.458188] wanpipe1:    First TSlot
>> :30
>> Jan  3 20:33:34 debian kernel: [55441.458202] wanpipe1: Configuring
>> Interface: w1g1 (log supress)
>> Jan  3 20:33:34 debian kernel: [55441.458205] wanpipe1: Configuring TDMV
>> Master dev w1g1
>> Jan  3 20:33:34 debian kernel: [55441.458209] wanpipe1:    Active Ch Map
>> :0x80000000
>> Jan  3 20:33:34 debian kernel: [55441.458210] wanpipe1:    First TSlot
>> :31
>> Jan  3 20:33:34 debian kernel: [55441.458226] wanpipe1: Enable Zaptel HW
>> DCHAN interface
>> Jan  3 20:33:34 debian kernel: [55441.458462] wanpipe1: Wanpipe device is
>> registered to Zaptel span # 1!
>> Jan  3 20:33:34 debian kernel: [55441.458760] wanpipe1: TDM Free Run
>> Timing Enabled 1 ms
>> Jan  3 20:33:34 debian kernel: [55441.464708] wanpipe1: Wanpipe Front End
>> Interrupt Restart Timeout
>> Jan  3 20:33:40 debian kernel: [55450.402880] wanpipe1: E1 connected!
>> Jan  3 20:33:40 debian kernel: [55450.407421] wanpipe1: AFT communications
>> enabled!
>> Jan  3 20:33:40 debian kernel: [55450.407421] wanpipe1: AFT Global TDM
>> Intr
>> Jan  3 20:33:40 debian kernel: [55450.409902] wanpipe1: Global TDM Ring
>> Resync
>> Jan  3 20:33:41 debian kernel: [55452.785733] wanpipe1: Enable E1 CCS
>> Signalling mode!
>>
>>
>> debian*CLI> core show version
>> Asterisk 1.6.2.0 built by root @ debian on a i686 running Linux on
>> 2011-01-03 08:57:50 UTC
>>
>>
>> debian*CLI> ss7 show linkset 1
>> SS7 linkset 1 status: Down
>>
>>
>> debian*CLI> ss7 set debug off linkset 1
>> Enabled debugging on linkset 1
>> Len = 4 [ ff ff 01 03 ]
>> FSN: 127 FIB 1
>> BSN: 127 BIB 1
>> <[0] LSSU SIOS
>>
>> Link state change: NOTALIGNED -> NOTALIGNED
>> Len = 4 [ ff ff 01 00 ]
>> FSN: 127 FIB 1
>> BSN: 127 BIB 1
>> <[0] LSSU SIO
>>
>> Link state change: NOTALIGNED -> ALIGNED
>> Len = 4 [ ff ff 01 02 ]
>> FSN: 127 FIB 1
>> BSN: 127 BIB 1
>> >[0] LSSU SIE
>>
>> Link state change: ALIGNED -> IDLE
>> Link state change: IDLE -> NOTALIGNED
>> Len = 4 [ ff ff 01 00 ]
>> FSN: 127 FIB 1
>> BSN: 127 BIB 1
>> >[0] LSSU SIO
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
> peace be upon you ...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Tue Jan  4 15:45:12 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 4 Jan 2011 12:45:12 +0000
Subject: [Freeswitch-users] conferencing
In-Reply-To: <AANLkTikuYNOL55j0rBFgG=FqFQnPWfxT1J13Q1M8ZMf6@mail.gmail.com>
References: <AANLkTikgBYfg_UiZnbc==8dgs32G8fDkSvak=-DBG_Ak@mail.gmail.com>
	<AANLkTino24LWO+V9egF70aPu0Kojj0sLTMiKBDY6E2tr@mail.gmail.com>
	<AANLkTinLij+oYGQ5Go1_Etg_kZvb_+wtKAQRg4krcwhU@mail.gmail.com>
	<AANLkTi=KPRnLDNLL8uUCxCvsFiRHrrJprxNVKjM7Kw=B@mail.gmail.com>
	<AANLkTikuYNOL55j0rBFgG=FqFQnPWfxT1J13Q1M8ZMf6@mail.gmail.com>
Message-ID: <AANLkTi=r3vAfBuYqvUqDivgDcyJ1+MTjaBjMk53vh=Qw@mail.gmail.com>

Each member's call runs in its own thread, each conference runs in its
own thread. Nothing to configure there.

-Steve



On 4 January 2011 10:45, Sam <u2nsam at gmail.com> wrote:
> calls would not be TDM it would be ip-ip calls.
> Ok G711 uses less CPU than other codecs,how can i utilized multiple
> threading/profiles or any other options to scale ? as it would be an
> incoming call always on FS.
>
> Regds
> Sam
>
> On Tue, Jan 4, 2011 at 3:53 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> That will still involve transcoding. FS must combine all speaking
>> channels into a single stream that can be sent to the conference
>> members. That can't be done in a compressed audio format, so it must
>> convert G711 to L16 for each speaking member, combine those pieces of
>> audio, then convert L16 back to G711 to send to the conference
>> members. Additionally, since the conference members are not
>> necessarily all using the same codec that last transcoding step will
>> occur once for each member, not just once for the conference. Your
>> best bet is to use a codec that uses as little CPU for transcoding as
>> possible - in software I'm not sure which is best there (G711 is
>> simple so possibly a good choice), in hardware Sangoma D100/D500 would
>> almost all the processing off the CPU.
>>
>> -Steve
>>
>>
>> On 4 January 2011 10:15, Sam <u2nsam at gmail.com> wrote:
>> > Ok if i remove transcoding out of the picture and all conf are running
>> > on
>> > g711, what rest would matters ?
>> >
>> >
>> > Regds
>> > Sam
>> >
>> >
>> > On Tue, Jan 4, 2011 at 2:56 PM, Avi Marcus <Avi at amarcus.com> wrote:
>> >>
>> >> It's the transcoding that will kill you. If that's truly needed, you
>> >> may
>> >> want to look into multiple machines handling transcoding and then
>> >> bridging
>> >> into the conference, or the new?transcoding?hardware from sangoma.
>> >> -Avi
>> >> On Tue, Jan 4, 2011 at 9:55 AM, Sam <u2nsam at gmail.com> wrote:
>> >>>
>> >>> hello,
>> >>>
>> >>> how to increase the scalability / performance of the conference .
>> >>>
>> >>> Are there any parameters to be observed that could do a trick ?
>> >>>
>> >>> The scenario would be multiple conferencing bridges and multiple
>> >>> codecs
>> >>> involved.
>> >>>
>> >>> Is is possible to multiple threading of profiles for conferences so
>> >>> that
>> >>> it gets the scalability ?
>> >>>
>> >>> Regards
>> >>> Sam
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From jmesquita at freeswitch.org  Tue Jan  4 16:19:28 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 4 Jan 2011 10:19:28 -0300
Subject: [Freeswitch-users] mod_conference with cluster ODBC
In-Reply-To: <AANLkTimSCf1UvCTpEv8QaiG4=f5EsH=qoUbYKw1F_UT+@mail.gmail.com>
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705>
	<AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com>
	<623BCB43D9ED45269190F7D08A802469@e1705>
	<AANLkTin+-cou_QLsFXqVJpQmyk8o7Ke6TjDQc6UL3tzh@mail.gmail.com>
	<AANLkTimSCf1UvCTpEv8QaiG4=f5EsH=qoUbYKw1F_UT+@mail.gmail.com>
Message-ID: <AANLkTik5=aVtxb2f-G_BNLWbKyKrcfrrs-LbMyuNhArZ@mail.gmail.com>

This is a feature that has not been implemented, it has nothing to do with
not being possible or not wanted. Patches or bounties should be accepted if
any of you is willing to implement the feature.

Regards,
Jo?o Mesquita


On Tue, Jan 4, 2011 at 6:28 AM, Avi Marcus <Avi at amarcus.com> wrote:

> The ESL implementation would be the same...
>
>
> On Tue, Jan 4, 2011 at 8:37 AM, Sam <u2nsam at gmail.com> wrote:
>
>> Can this be done with the primary database sqlite ?
>>
>> Regds
>> Sam
>>
>>
>> On Tue, Jan 4, 2011 at 10:44 AM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  ha ok,
>>> any example to listening conference with ESL ?
>>> are bridge 2 conferences made the same way as 2 users ?
>>>
>>> suggestion: as there is one DB with ODBC for all nodes why not
>>> create a conference table as sip_registration to manage the ip of every
>>> first user in a conference
>>> and redirect the other automatically to the right node ?
>>>
>>> Thanks
>>>
>>> ----- Original Message -----
>>> *From:* Jo?o Mesquita <jmesquita at freeswitch.org>
>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> *Sent:* Monday, January 03, 2011 11:08 PM
>>> *Subject:* Re: [Freeswitch-users] mod_conference with cluster ODBC
>>>
>>> They are indeed 2 completely different conferences. There's no
>>> implementation of making these 2 conferences bridge themselves
>>> automatically.
>>>
>>> The way I have solved this problem for now is have an ESL daemon
>>> "listening" on the conference creation events and bridging the 2 servers
>>> together when the one with the same name on the same domain is created. This
>>> solution might work for you and it's no too hard to implement.
>>>
>>> Regards,
>>> Jo?o Mesquita
>>>
>>>
>>> On Mon, Jan 3, 2011 at 11:11 PM, Madovsky <infos at madovsky.org> wrote:
>>>
>>>>  If I have a conference created on node A with name abc-domain at default
>>>> and another with the same name in node B it seems
>>>> that FS considers it as 2 distinct conferences, or maybe
>>>> I wrongly set any configuration ?
>>>>
>>>> Thanks
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>  ------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From edpimentl at gmail.com  Tue Jan  4 16:25:28 2011
From: edpimentl at gmail.com (EdPimentl)
Date: Tue, 4 Jan 2011 08:25:28 -0500
Subject: [Freeswitch-users] FreeSWITCH configuration for Mobile Phones
	Running SIP (sipdroid) Client, 
Message-ID: <AANLkTinewmpFPQMv4Mwrh=OZ1BfMsxD_LDR=9m2XYdga@mail.gmail.com>

Hello Everyone,

Can someone please share their experience on the best configuration/dial
plans for using Mobile SipDroid Client with FreeSwitch?


Thanks in advance,
-E
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From gmaruzz at gmail.com  Tue Jan  4 16:52:17 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 4 Jan 2011 14:52:17 +0100
Subject: [Freeswitch-users] skypopen error when making a call
In-Reply-To: <SNT109-DS23DD745C224DDC1DCC71ADC1080@phx.gbl>
References: <SNT109-DS23DD745C224DDC1DCC71ADC1080@phx.gbl>
Message-ID: <AANLkTi=UGPnPBsa=D2T-Ecvq4Zab6hB8LLZnu+UiL8Zk@mail.gmail.com>

Hi Norman,

that is not the way to make calls.

mod_skypopen follows the standard syntax of freeswitch

if you want to originate a call from the command line you can do
(backgrounding the origination):

> bgapi originate skypopen/interface1/hwnorman 5000

this will originate a skype call using the interface1 and connect
skype user hwnorman to the extension 5000 of the dialplan (the
standard IVR, if you have not changed the dialplan)

what you are doing in your post is passing directly commands to the
skypeclient connected to interface1 (the commands you are passing are
wrong and the skypeclient whines).

you don't need to use that syntax, use the standard FS syntax.

For more extensive infos, please refer to the skypopen wiki page ;).

-giovanni


On Tue, Jan 4, 2011 at 9:36 AM, Norman Lam <hwnorman at hotmail.com> wrote:
> Hi Everyone
>
>
>
> I am trying to call skypopen(skype network) from freeswitch console and I am
> getting this error
>
>
>
> freeswitch at jfp8> skypopen interface1 hwnorman
>
>
>
> Using interface: globals.SKYPOPEN_INTERFACES[1].name=|||interface1|||
>
>
>
> 2011-01-04 16:28:01.687500 [ERR] skypopen_protocol.c:259???? [git-] [ERRORA
>
> ? 259? ][interface1???? ][IDLE,IDLE] Skype got ERROR: |||ERROR 2 Unknown
> command
>
> |||
>
> 2011-01-04 16:28:01.687500 [ERR] skypopen_protocol.c:261???? [git-] [ERRORA
>
> ? 261? ][interface1???? ][IDLE,FNSHED] skype_call now is DOWN
>
>
>
> But my interface is O.K , I can make a call from skype to freeswitch
> destination to the x-lite is O.K
>
>
>
> Please advise
>
>
>
> Norman Lam
>
>
>
> Here is the skypopen interface
>
>
>
> freeswitch at jfp8> sk list
>
>
>
> sk console is NOT yet assigned
>
> F ID??????? Name??????? IB (F/T)??? OB (F/T)??? State?? CallFlw???????? UUID
>
> = ====? ??========????? =======???? =======???? ======? ============
> ======
>
> ? 1???? [interface1]????? 0/1??????? 0/0??????? IDLE??? IDLE
>
> ? 2???? [interface2]????? 0/0??????? 0/0??????? IDLE??? IDLE
>
> ? 3???? [interface3]????? 0/0??????? 0/0??????? IDLE??? IDLE
>
> ? 4???? [interface4]????? 0/0??????? 0/0??????? IDLE??? IDLE
>
>
>
> Total Interfaces: 4? IB Calls(Failed/Total): 0/1? OB Calls(Failed/Total):
> 0/0
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From u2nsam at gmail.com  Tue Jan  4 17:40:53 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 4 Jan 2011 20:10:53 +0530
Subject: [Freeswitch-users] conferencing
In-Reply-To: <AANLkTimRVUySdwVL7kNXqzZPqJfyR+XU=pd2dRJKmPhT@mail.gmail.com>
References: <AANLkTikgBYfg_UiZnbc==8dgs32G8fDkSvak=-DBG_Ak@mail.gmail.com>
	<AANLkTino24LWO+V9egF70aPu0Kojj0sLTMiKBDY6E2tr@mail.gmail.com>
	<AANLkTinLij+oYGQ5Go1_Etg_kZvb_+wtKAQRg4krcwhU@mail.gmail.com>
	<AANLkTi=KPRnLDNLL8uUCxCvsFiRHrrJprxNVKjM7Kw=B@mail.gmail.com>
	<AANLkTikuYNOL55j0rBFgG=FqFQnPWfxT1J13Q1M8ZMf6@mail.gmail.com>
	<AANLkTimRVUySdwVL7kNXqzZPqJfyR+XU=pd2dRJKmPhT@mail.gmail.com>
Message-ID: <AANLkTimmkqzo0i20x8-k1opV0twebWSTC0E2DBxdsFrr@mail.gmail.com>

Any thing on hardware & software front ?

Regds
Sam

On Tue, Jan 4, 2011 at 6:07 PM, Avi Marcus <Avi at amarcus.com> wrote:

> If you mute most of the callers, that would limit the audio that would need
> to be processed.
> -Avi
>
>
> On Tue, Jan 4, 2011 at 12:45 PM, Sam <u2nsam at gmail.com> wrote:
>
>> calls would not be TDM it would be ip-ip calls.
>> Ok G711 uses less CPU than other codecs,how can i utilized multiple
>> threading/profiles or any other options to scale ? as it would be an
>> incoming call always on FS.
>>
>> Regds
>> Sam
>>
>>
>> On Tue, Jan 4, 2011 at 3:53 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> That will still involve transcoding. FS must combine all speaking
>>> channels into a single stream that can be sent to the conference
>>> members. That can't be done in a compressed audio format, so it must
>>> convert G711 to L16 for each speaking member, combine those pieces of
>>> audio, then convert L16 back to G711 to send to the conference
>>> members. Additionally, since the conference members are not
>>> necessarily all using the same codec that last transcoding step will
>>> occur once for each member, not just once for the conference. Your
>>> best bet is to use a codec that uses as little CPU for transcoding as
>>> possible - in software I'm not sure which is best there (G711 is
>>> simple so possibly a good choice), in hardware Sangoma D100/D500 would
>>> almost all the processing off the CPU.
>>>
>>> -Steve
>>>
>>>
>>> On 4 January 2011 10:15, Sam <u2nsam at gmail.com> wrote:
>>> > Ok if i remove transcoding out of the picture and all conf are running
>>> on
>>> > g711, what rest would matters ?
>>> >
>>> >
>>> > Regds
>>> > Sam
>>> >
>>> >
>>> > On Tue, Jan 4, 2011 at 2:56 PM, Avi Marcus <Avi at amarcus.com> wrote:
>>> >>
>>> >> It's the transcoding that will kill you. If that's truly needed, you
>>> may
>>> >> want to look into multiple machines handling transcoding and then
>>> bridging
>>> >> into the conference, or the new transcoding hardware from sangoma.
>>> >> -Avi
>>> >> On Tue, Jan 4, 2011 at 9:55 AM, Sam <u2nsam at gmail.com> wrote:
>>> >>>
>>> >>> hello,
>>> >>>
>>> >>> how to increase the scalability / performance of the conference .
>>> >>>
>>> >>> Are there any parameters to be observed that could do a trick ?
>>> >>>
>>> >>> The scenario would be multiple conferencing bridges and multiple
>>> codecs
>>> >>> involved.
>>> >>>
>>> >>> Is is possible to multiple threading of profiles for conferences so
>>> that
>>> >>> it gets the scalability ?
>>> >>>
>>> >>> Regards
>>> >>> Sam
>>> >>>
>>> >>> _______________________________________________
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>>
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From thomas at chaschperli.ch  Tue Jan  4 17:41:42 2011
From: thomas at chaschperli.ch (Thomas Mueller)
Date: Tue, 04 Jan 2011 15:41:42 +0100
Subject: [Freeswitch-users] FreeSWITCH configuration for Mobile Phones
 Running SIP (sipdroid) Client, 
In-Reply-To: <AANLkTinewmpFPQMv4Mwrh=OZ1BfMsxD_LDR=9m2XYdga@mail.gmail.com>
References: <AANLkTinewmpFPQMv4Mwrh=OZ1BfMsxD_LDR=9m2XYdga@mail.gmail.com>
Message-ID: <4D2331A6.4060300@chaschperli.ch>

Hoi E

> Can someone please share their experience on the best 
> configuration/dial plans for using Mobile SipDroid Client with FreeSwitch?
>

sipdroid works nice with the default dialplan.

- Thomas


From steveayre at gmail.com  Tue Jan  4 18:10:05 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 4 Jan 2011 15:10:05 +0000
Subject: [Freeswitch-users] conferencing
In-Reply-To: <AANLkTimmkqzo0i20x8-k1opV0twebWSTC0E2DBxdsFrr@mail.gmail.com>
References: <AANLkTikgBYfg_UiZnbc==8dgs32G8fDkSvak=-DBG_Ak@mail.gmail.com>
	<AANLkTino24LWO+V9egF70aPu0Kojj0sLTMiKBDY6E2tr@mail.gmail.com>
	<AANLkTinLij+oYGQ5Go1_Etg_kZvb_+wtKAQRg4krcwhU@mail.gmail.com>
	<AANLkTi=KPRnLDNLL8uUCxCvsFiRHrrJprxNVKjM7Kw=B@mail.gmail.com>
	<AANLkTikuYNOL55j0rBFgG=FqFQnPWfxT1J13Q1M8ZMf6@mail.gmail.com>
	<AANLkTimRVUySdwVL7kNXqzZPqJfyR+XU=pd2dRJKmPhT@mail.gmail.com>
	<AANLkTimmkqzo0i20x8-k1opV0twebWSTC0E2DBxdsFrr@mail.gmail.com>
Message-ID: <AANLkTimwH1V-seR928w4o1kvsByuu-4JOjhE6QAkp+Z_@mail.gmail.com>

Well, the more processing power you have the more calls you can
handle. You should make sure you're using 64bit.

There's a few tips on
http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations
too.

-Steve


On 4 January 2011 14:40, Sam <u2nsam at gmail.com> wrote:
> Any thing on hardware & software front ?
>
> Regds
> Sam
>
> On Tue, Jan 4, 2011 at 6:07 PM, Avi Marcus <Avi at amarcus.com> wrote:
>>
>> If you mute most of the callers, that would limit the audio that would
>> need to be processed.
>> -Avi
>>
>> On Tue, Jan 4, 2011 at 12:45 PM, Sam <u2nsam at gmail.com> wrote:
>>>
>>> calls would not be TDM it would be ip-ip calls.
>>> Ok G711 uses less CPU than other codecs,how can i utilized multiple
>>> threading/profiles or any other options to scale ? as it would be an
>>> incoming call always on FS.
>>>
>>> Regds
>>> Sam
>>>
>>> On Tue, Jan 4, 2011 at 3:53 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>>>
>>>> That will still involve transcoding. FS must combine all speaking
>>>> channels into a single stream that can be sent to the conference
>>>> members. That can't be done in a compressed audio format, so it must
>>>> convert G711 to L16 for each speaking member, combine those pieces of
>>>> audio, then convert L16 back to G711 to send to the conference
>>>> members. Additionally, since the conference members are not
>>>> necessarily all using the same codec that last transcoding step will
>>>> occur once for each member, not just once for the conference. Your
>>>> best bet is to use a codec that uses as little CPU for transcoding as
>>>> possible - in software I'm not sure which is best there (G711 is
>>>> simple so possibly a good choice), in hardware Sangoma D100/D500 would
>>>> almost all the processing off the CPU.
>>>>
>>>> -Steve
>>>>
>>>>
>>>> On 4 January 2011 10:15, Sam <u2nsam at gmail.com> wrote:
>>>> > Ok if i remove transcoding out of the picture and all conf are running
>>>> > on
>>>> > g711, what rest would matters ?
>>>> >
>>>> >
>>>> > Regds
>>>> > Sam
>>>> >
>>>> >
>>>> > On Tue, Jan 4, 2011 at 2:56 PM, Avi Marcus <Avi at amarcus.com> wrote:
>>>> >>
>>>> >> It's the transcoding that will kill you. If that's truly needed, you
>>>> >> may
>>>> >> want to look into multiple machines handling transcoding and then
>>>> >> bridging
>>>> >> into the conference, or the new?transcoding?hardware from sangoma.
>>>> >> -Avi
>>>> >> On Tue, Jan 4, 2011 at 9:55 AM, Sam <u2nsam at gmail.com> wrote:
>>>> >>>
>>>> >>> hello,
>>>> >>>
>>>> >>> how to increase the scalability / performance of the conference .
>>>> >>>
>>>> >>> Are there any parameters to be observed that could do a trick ?
>>>> >>>
>>>> >>> The scenario would be multiple conferencing bridges and multiple
>>>> >>> codecs
>>>> >>> involved.
>>>> >>>
>>>> >>> Is is possible to multiple threading of profiles for conferences so
>>>> >>> that
>>>> >>> it gets the scalability ?
>>>> >>>
>>>> >>> Regards
>>>> >>> Sam
>>>> >>>
>>>> >>> _______________________________________________
>>>> >>> FreeSWITCH-users mailing list
>>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>>
>>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >>> http://www.freeswitch.org
>>>> >>>
>>>> >>
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>
>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >>
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >
>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From infos at madovsky.org  Tue Jan  4 19:26:07 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 4 Jan 2011 11:26:07 -0500
Subject: [Freeswitch-users] mod_conference with cluster ODBC
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705><AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com><623BCB43D9ED45269190F7D08A802469@e1705><AANLkTin+-cou_QLsFXqVJpQmyk8o7Ke6TjDQc6UL3tzh@mail.gmail.com><AANLkTimSCf1UvCTpEv8QaiG4=f5EsH=qoUbYKw1F_UT+@mail.gmail.com>
	<AANLkTik5=aVtxb2f-G_BNLWbKyKrcfrrs-LbMyuNhArZ@mail.gmail.com>
Message-ID: <78D666CC02CD4A71A7C95F91C34CC740@e1705>

Ok, I need to learn more C language so ;)...
  ----- Original Message ----- 
  From: Jo?o Mesquita 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, January 04, 2011 8:19 AM
  Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


  This is a feature that has not been implemented, it has nothing to do with not being possible or not wanted. Patches or bounties should be accepted if any of you is willing to implement the feature.


  Regards,
  Jo?o Mesquita



  On Tue, Jan 4, 2011 at 6:28 AM, Avi Marcus <Avi at amarcus.com> wrote:

    The ESL implementation would be the same...



    On Tue, Jan 4, 2011 at 8:37 AM, Sam <u2nsam at gmail.com> wrote:

      Can this be done with the primary database sqlite ?

      Regds
      Sam



      On Tue, Jan 4, 2011 at 10:44 AM, Madovsky <infos at madovsky.org> wrote:

        ha ok, 
        any example to listening conference with ESL ?
        are bridge 2 conferences made the same way as 2 users ?

        suggestion: as there is one DB with ODBC for all nodes why not
        create a conference table as sip_registration to manage the ip of every first user in a conference
        and redirect the other automatically to the right node ?

        Thanks
          ----- Original Message ----- 
          From: Jo?o Mesquita 
          To: FreeSWITCH Users Help 
          Sent: Monday, January 03, 2011 11:08 PM
          Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


          They are indeed 2 completely different conferences. There's no implementation of making these 2 conferences bridge themselves automatically. 


          The way I have solved this problem for now is have an ESL daemon "listening" on the conference creation events and bridging the 2 servers together when the one with the same name on the same domain is created. This solution might work for you and it's no too hard to implement.


          Regards,
          Jo?o Mesquita



          On Mon, Jan 3, 2011 at 11:11 PM, Madovsky <infos at madovsky.org> wrote:

            If I have a conference created on node A with name abc-domain at default
            and another with the same name in node B it seems
            that FS considers it as 2 distinct conferences, or maybe
            I wrongly set any configuration ?

            Thanks

            _______________________________________________
            FreeSWITCH-users mailing list
            FreeSWITCH-users at lists.freeswitch.org
            http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
            UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
            http://www.freeswitch.org






----------------------------------------------------------------------


          _______________________________________________
          FreeSWITCH-users mailing list
          FreeSWITCH-users at lists.freeswitch.org
          http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
          http://www.freeswitch.org


        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org





      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org





    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org






------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From lloyd.aloysius at gmail.com  Tue Jan  4 19:26:14 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 4 Jan 2011 11:26:14 -0500
Subject: [Freeswitch-users] sofia status profile internal - Output in Table
	view
Message-ID: <AANLkTikPhufZ0T80+9Wj9rS5nZcPqZ=2xZKRNsD9--KL@mail.gmail.com>

Hi All,

Is there any way to get the output from sofia status profile internal in a
table view.

Something like Asterisk "sip show peers"

Thanks
Lloyd
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From u2nsam at gmail.com  Tue Jan  4 19:30:10 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 4 Jan 2011 22:00:10 +0530
Subject: [Freeswitch-users] conferencing
In-Reply-To: <AANLkTimwH1V-seR928w4o1kvsByuu-4JOjhE6QAkp+Z_@mail.gmail.com>
References: <AANLkTikgBYfg_UiZnbc==8dgs32G8fDkSvak=-DBG_Ak@mail.gmail.com>
	<AANLkTino24LWO+V9egF70aPu0Kojj0sLTMiKBDY6E2tr@mail.gmail.com>
	<AANLkTinLij+oYGQ5Go1_Etg_kZvb_+wtKAQRg4krcwhU@mail.gmail.com>
	<AANLkTi=KPRnLDNLL8uUCxCvsFiRHrrJprxNVKjM7Kw=B@mail.gmail.com>
	<AANLkTikuYNOL55j0rBFgG=FqFQnPWfxT1J13Q1M8ZMf6@mail.gmail.com>
	<AANLkTimRVUySdwVL7kNXqzZPqJfyR+XU=pd2dRJKmPhT@mail.gmail.com>
	<AANLkTimmkqzo0i20x8-k1opV0twebWSTC0E2DBxdsFrr@mail.gmail.com>
	<AANLkTimwH1V-seR928w4o1kvsByuu-4JOjhE6QAkp+Z_@mail.gmail.com>
Message-ID: <AANLkTim7fshba_9OcHvn4Hm1uW65LyHQaBBJACjjv_1a@mail.gmail.com>

1 question from the link mentioned , there is a quote about "libsofia only
handles 1 thread per profile"

the inbound calls will look into which profile so that we can increase
threads by multiplying profiles ?

Regards
Sam


On Tue, Jan 4, 2011 at 8:40 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Well, the more processing power you have the more calls you can
> handle. You should make sure you're using 64bit.
>
> There's a few tips on
> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations
> too.
>
> -Steve
>
>
> On 4 January 2011 14:40, Sam <u2nsam at gmail.com> wrote:
> > Any thing on hardware & software front ?
> >
> > Regds
> > Sam
> >
> > On Tue, Jan 4, 2011 at 6:07 PM, Avi Marcus <Avi at amarcus.com> wrote:
> >>
> >> If you mute most of the callers, that would limit the audio that would
> >> need to be processed.
> >> -Avi
> >>
> >> On Tue, Jan 4, 2011 at 12:45 PM, Sam <u2nsam at gmail.com> wrote:
> >>>
> >>> calls would not be TDM it would be ip-ip calls.
> >>> Ok G711 uses less CPU than other codecs,how can i utilized multiple
> >>> threading/profiles or any other options to scale ? as it would be an
> >>> incoming call always on FS.
> >>>
> >>> Regds
> >>> Sam
> >>>
> >>> On Tue, Jan 4, 2011 at 3:53 PM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >>>>
> >>>> That will still involve transcoding. FS must combine all speaking
> >>>> channels into a single stream that can be sent to the conference
> >>>> members. That can't be done in a compressed audio format, so it must
> >>>> convert G711 to L16 for each speaking member, combine those pieces of
> >>>> audio, then convert L16 back to G711 to send to the conference
> >>>> members. Additionally, since the conference members are not
> >>>> necessarily all using the same codec that last transcoding step will
> >>>> occur once for each member, not just once for the conference. Your
> >>>> best bet is to use a codec that uses as little CPU for transcoding as
> >>>> possible - in software I'm not sure which is best there (G711 is
> >>>> simple so possibly a good choice), in hardware Sangoma D100/D500 would
> >>>> almost all the processing off the CPU.
> >>>>
> >>>> -Steve
> >>>>
> >>>>
> >>>> On 4 January 2011 10:15, Sam <u2nsam at gmail.com> wrote:
> >>>> > Ok if i remove transcoding out of the picture and all conf are
> running
> >>>> > on
> >>>> > g711, what rest would matters ?
> >>>> >
> >>>> >
> >>>> > Regds
> >>>> > Sam
> >>>> >
> >>>> >
> >>>> > On Tue, Jan 4, 2011 at 2:56 PM, Avi Marcus <Avi at amarcus.com> wrote:
> >>>> >>
> >>>> >> It's the transcoding that will kill you. If that's truly needed,
> you
> >>>> >> may
> >>>> >> want to look into multiple machines handling transcoding and then
> >>>> >> bridging
> >>>> >> into the conference, or the new transcoding hardware from sangoma.
> >>>> >> -Avi
> >>>> >> On Tue, Jan 4, 2011 at 9:55 AM, Sam <u2nsam at gmail.com> wrote:
> >>>> >>>
> >>>> >>> hello,
> >>>> >>>
> >>>> >>> how to increase the scalability / performance of the conference .
> >>>> >>>
> >>>> >>> Are there any parameters to be observed that could do a trick ?
> >>>> >>>
> >>>> >>> The scenario would be multiple conferencing bridges and multiple
> >>>> >>> codecs
> >>>> >>> involved.
> >>>> >>>
> >>>> >>> Is is possible to multiple threading of profiles for conferences
> so
> >>>> >>> that
> >>>> >>> it gets the scalability ?
> >>>> >>>
> >>>> >>> Regards
> >>>> >>> Sam
> >>>> >>>
> >>>> >>> _______________________________________________
> >>>> >>> FreeSWITCH-users mailing list
> >>>> >>> FreeSWITCH-users at lists.freeswitch.org
> >>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> >>>
> >>>> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> >>> http://www.freeswitch.org
> >>>> >>>
> >>>> >>
> >>>> >>
> >>>> >> _______________________________________________
> >>>> >> FreeSWITCH-users mailing list
> >>>> >> FreeSWITCH-users at lists.freeswitch.org
> >>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> >>
> >>>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> >> http://www.freeswitch.org
> >>>> >>
> >>>> >
> >>>> >
> >>>> > _______________________________________________
> >>>> > FreeSWITCH-users mailing list
> >>>> > FreeSWITCH-users at lists.freeswitch.org
> >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> >
> >>>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> > http://www.freeswitch.org
> >>>> >
> >>>> >
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Tue Jan  4 19:44:27 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 4 Jan 2011 11:44:27 -0500
Subject: [Freeswitch-users] mod_conference with cluster ODBC
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705><AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com><623BCB43D9ED45269190F7D08A802469@e1705><AANLkTin+-cou_QLsFXqVJpQmyk8o7Ke6TjDQc6UL3tzh@mail.gmail.com><AANLkTimSCf1UvCTpEv8QaiG4=f5EsH=qoUbYKw1F_UT+@mail.gmail.com>
	<AANLkTik5=aVtxb2f-G_BNLWbKyKrcfrrs-LbMyuNhArZ@mail.gmail.com>
Message-ID: <C4BB135D9E25434F818A9F085C3EC2E1@e1705>

and I created a bounty, 

thanks
  ----- Original Message ----- 
  From: Jo?o Mesquita 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, January 04, 2011 8:19 AM
  Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


  This is a feature that has not been implemented, it has nothing to do with not being possible or not wanted. Patches or bounties should be accepted if any of you is willing to implement the feature.


  Regards,
  Jo?o Mesquita



  On Tue, Jan 4, 2011 at 6:28 AM, Avi Marcus <Avi at amarcus.com> wrote:

    The ESL implementation would be the same...



    On Tue, Jan 4, 2011 at 8:37 AM, Sam <u2nsam at gmail.com> wrote:

      Can this be done with the primary database sqlite ?

      Regds
      Sam



      On Tue, Jan 4, 2011 at 10:44 AM, Madovsky <infos at madovsky.org> wrote:

        ha ok, 
        any example to listening conference with ESL ?
        are bridge 2 conferences made the same way as 2 users ?

        suggestion: as there is one DB with ODBC for all nodes why not
        create a conference table as sip_registration to manage the ip of every first user in a conference
        and redirect the other automatically to the right node ?

        Thanks
          ----- Original Message ----- 
          From: Jo?o Mesquita 
          To: FreeSWITCH Users Help 
          Sent: Monday, January 03, 2011 11:08 PM
          Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


          They are indeed 2 completely different conferences. There's no implementation of making these 2 conferences bridge themselves automatically. 


          The way I have solved this problem for now is have an ESL daemon "listening" on the conference creation events and bridging the 2 servers together when the one with the same name on the same domain is created. This solution might work for you and it's no too hard to implement.


          Regards,
          Jo?o Mesquita



          On Mon, Jan 3, 2011 at 11:11 PM, Madovsky <infos at madovsky.org> wrote:

            If I have a conference created on node A with name abc-domain at default
            and another with the same name in node B it seems
            that FS considers it as 2 distinct conferences, or maybe
            I wrongly set any configuration ?

            Thanks

            _______________________________________________
            FreeSWITCH-users mailing list
            FreeSWITCH-users at lists.freeswitch.org
            http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
            UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
            http://www.freeswitch.org






----------------------------------------------------------------------


          _______________________________________________
          FreeSWITCH-users mailing list
          FreeSWITCH-users at lists.freeswitch.org
          http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
          http://www.freeswitch.org


        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org





      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org





    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org






------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From dome at tel.co.th  Tue Jan  4 19:50:53 2011
From: dome at tel.co.th (dome at tel.co.th)
Date: Tue, 4 Jan 2011 23:50:53 +0700
Subject: [Freeswitch-users] FreeSWITCH configuration for Mobile Phones
 Running SIP (sipdroid) Client, 
In-Reply-To: <4D2331A6.4060300@chaschperli.ch>
References: <AANLkTinewmpFPQMv4Mwrh=OZ1BfMsxD_LDR=9m2XYdga@mail.gmail.com>
	<4D2331A6.4060300@chaschperli.ch>
Message-ID: <AANLkTikVgbukrdQ9_mWxhrHW4M8GjCV3GYJFBcZ+EevG@mail.gmail.com>

csipsimple also work fine.

Dome C.

2011/1/4 Thomas Mueller <thomas at chaschperli.ch>:
> Hoi E
>
>> Can someone please share their experience on the best
>> configuration/dial plans for using Mobile SipDroid Client with FreeSwitch?
>>
>
> sipdroid works nice with the default dialplan.
>
> - Thomas
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From lists at infosecurity.ch  Tue Jan  4 20:24:11 2011
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Tue, 04 Jan 2011 18:24:11 +0100
Subject: [Freeswitch-users] FreeSWITCH configuration for Mobile Phones
 Running SIP (sipdroid) Client, 
In-Reply-To: <AANLkTikVgbukrdQ9_mWxhrHW4M8GjCV3GYJFBcZ+EevG@mail.gmail.com>
References: <AANLkTinewmpFPQMv4Mwrh=OZ1BfMsxD_LDR=9m2XYdga@mail.gmail.com>	<4D2331A6.4060300@chaschperli.ch>
	<AANLkTikVgbukrdQ9_mWxhrHW4M8GjCV3GYJFBcZ+EevG@mail.gmail.com>
Message-ID: <4D2357BB.3060803@infosecurity.ch>

Some fine tuning:
- Use SIP/TCP or SIP/TLS transport to strongly save battery respect to
SIP/UDP
- Increase SIP timer to be able to work mobile networks with packet loss
(otherwise a single TCP congestion would fire
- Disable all ping checking from server (you must reduce keepalive traffic)
- Disable TCP kernel keepalive
- Use high ptime and ultra-narrowband codec (4-5kbit/s with VAD/DTX) to
work over 2.5 GPRS (edge/umts are much more powerful networks)

Those are some of the major improvements we have done while setting up
our server for PrivateGSM Professional SaaS (http://www.privatewave.com)
ZRTP VoIP encryption client for BB/Nokia/iPhone .

Fabio

On 04/01/11 17.50, dome at tel.co.th wrote:
> csipsimple also work fine.
>
> Dome C.
>
> 2011/1/4 Thomas Mueller <thomas at chaschperli.ch>:
>> Hoi E
>>
>>> Can someone please share their experience on the best
>>> configuration/dial plans for using Mobile SipDroid Client with FreeSwitch?
>>>
>> sipdroid works nice with the default dialplan.
>>
>> - Thomas
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Tue Jan  4 20:33:00 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 4 Jan 2011 11:33:00 -0600
Subject: [Freeswitch-users] sofia status profile internal - Output in
 Table view
In-Reply-To: <AANLkTikPhufZ0T80+9Wj9rS5nZcPqZ=2xZKRNsD9--KL@mail.gmail.com>
References: <AANLkTikPhufZ0T80+9Wj9rS5nZcPqZ=2xZKRNsD9--KL@mail.gmail.com>
Message-ID: <AANLkTin1R+5QvWp1YG5cOexjvHu5+qfB=zpC0_f1S1W4@mail.gmail.com>

there is "sofia xmlstatus profile internal"
then you can format it any way you want.


On Tue, Jan 4, 2011 at 10:26 AM, Aloysius Lloyd
<lloyd.aloysius at gmail.com> wrote:
> Hi All,
> Is there any way to get the output from?sofia status profile internal in a
> table view.
> Something like Asterisk "sip show peers"
>
> Thanks
> Lloyd
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From steveayre at gmail.com  Tue Jan  4 20:53:25 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 4 Jan 2011 17:53:25 +0000
Subject: [Freeswitch-users] conferencing
In-Reply-To: <AANLkTim7fshba_9OcHvn4Hm1uW65LyHQaBBJACjjv_1a@mail.gmail.com>
References: <AANLkTikgBYfg_UiZnbc==8dgs32G8fDkSvak=-DBG_Ak@mail.gmail.com>
	<AANLkTino24LWO+V9egF70aPu0Kojj0sLTMiKBDY6E2tr@mail.gmail.com>
	<AANLkTinLij+oYGQ5Go1_Etg_kZvb_+wtKAQRg4krcwhU@mail.gmail.com>
	<AANLkTi=KPRnLDNLL8uUCxCvsFiRHrrJprxNVKjM7Kw=B@mail.gmail.com>
	<AANLkTikuYNOL55j0rBFgG=FqFQnPWfxT1J13Q1M8ZMf6@mail.gmail.com>
	<AANLkTimRVUySdwVL7kNXqzZPqJfyR+XU=pd2dRJKmPhT@mail.gmail.com>
	<AANLkTimmkqzo0i20x8-k1opV0twebWSTC0E2DBxdsFrr@mail.gmail.com>
	<AANLkTimwH1V-seR928w4o1kvsByuu-4JOjhE6QAkp+Z_@mail.gmail.com>
	<AANLkTim7fshba_9OcHvn4Hm1uW65LyHQaBBJACjjv_1a@mail.gmail.com>
Message-ID: <AANLkTik2URCcAnM7zpRnKuEOZHBLEFZZhme8BnCKFB1P@mail.gmail.com>

Yes, each SIP profile is in its own thread. That processes the SIP
messages sent/received on the SIP profile. It's a relatively small
part of the call - RTP is handled in a different thread.

Some people have found on very busy servers that with a large volume
of SIP messages to process the profile's thread became a bottleneck
and adding a 2nd profile to handle some of the traffic helped. It
might help increase the CPS your server can cope with, but probably
won't make a difference to the maximum number of concurrent calls your
server can handle.

-Steve


On 4 January 2011 16:30, Sam <u2nsam at gmail.com> wrote:
> 1 question from the link mentioned , there is a quote about "libsofia only
> handles 1 thread per profile"
>
> the inbound calls will look into which profile so that we can increase
> threads by multiplying profiles ?
>
> Regards
> Sam
>
>
> On Tue, Jan 4, 2011 at 8:40 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> Well, the more processing power you have the more calls you can

We're in receipt of the details you recently sent us. Your order
will now be processed as normal.
>> handle. You should make sure you're using 64bit.
>>
>> There's a few tips on
>> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations
>> too.
>>
>> -Steve
>>
>>
>> On 4 January 2011 14:40, Sam <u2nsam at gmail.com> wrote:
>> > Any thing on hardware & software front ?
>> >
>> > Regds
>> > Sam
>> >
>> > On Tue, Jan 4, 2011 at 6:07 PM, Avi Marcus <Avi at amarcus.com> wrote:
>> >>
>> >> If you mute most of the callers, that would limit the audio that would
>> >> need to be processed.
>> >> -Avi
>> >>
>> >> On Tue, Jan 4, 2011 at 12:45 PM, Sam <u2nsam at gmail.com> wrote:
>> >>>
>> >>> calls would not be TDM it would be ip-ip calls.
>> >>> Ok G711 uses less CPU than other codecs,how can i utilized multiple
>> >>> threading/profiles or any other options to scale ? as it would be an
>> >>> incoming call always on FS.
>> >>>
>> >>> Regds
>> >>> Sam
>> >>>
>> >>> On Tue, Jan 4, 2011 at 3:53 PM, Steven Ayre <steveayre at gmail.com>
>> >>> wrote:
>> >>>>
>> >>>> That will still involve transcoding. FS must combine all speaking
>> >>>> channels into a single stream that can be sent to the conference
>> >>>> members. That can't be done in a compressed audio format, so it must
>> >>>> convert G711 to L16 for each speaking member, combine those pieces of
>> >>>> audio, then convert L16 back to G711 to send to the conference
>> >>>> members. Additionally, since the conference members are not
>> >>>> necessarily all using the same codec that last transcoding step will
>> >>>> occur once for each member, not just once for the conference. Your
>> >>>> best bet is to use a codec that uses as little CPU for transcoding as
>> >>>> possible - in software I'm not sure which is best there (G711 is
>> >>>> simple so possibly a good choice), in hardware Sangoma D100/D500
>> >>>> would
>> >>>> almost all the processing off the CPU.
>> >>>>
>> >>>> -Steve
>> >>>>
>> >>>>
>> >>>> On 4 January 2011 10:15, Sam <u2nsam at gmail.com> wrote:
>> >>>> > Ok if i remove transcoding out of the picture and all conf are
>> >>>> > running
>> >>>> > on
>> >>>> > g711, what rest would matters ?
>> >>>> >
>> >>>> >
>> >>>> > Regds
>> >>>> > Sam
>> >>>> >
>> >>>> >
>> >>>> > On Tue, Jan 4, 2011 at 2:56 PM, Avi Marcus <Avi at amarcus.com> wrote:
>> >>>> >>
>> >>>> >> It's the transcoding that will kill you. If that's truly needed,
>> >>>> >> you
>> >>>> >> may
>> >>>> >> want to look into multiple machines handling transcoding and then
>> >>>> >> bridging
>> >>>> >> into the conference, or the new?transcoding?hardware from sangoma.
>> >>>> >> -Avi
>> >>>> >> On Tue, Jan 4, 2011 at 9:55 AM, Sam <u2nsam at gmail.com> wrote:
>> >>>> >>>
>> >>>> >>> hello,
>> >>>> >>>
>> >>>> >>> how to increase the scalability / performance of the conference .
>> >>>> >>>
>> >>>> >>> Are there any parameters to be observed that could do a trick ?
>> >>>> >>>
>> >>>> >>> The scenario would be multiple conferencing bridges and multiple
>> >>>> >>> codecs
>> >>>> >>> involved.
>> >>>> >>>
>> >>>> >>> Is is possible to multiple threading of profiles for conferences
>> >>>> >>> so
>> >>>> >>> that
>> >>>> >>> it gets the scalability ?
>> >>>> >>>
>> >>>> >>> Regards
>> >>>> >>> Sam
>> >>>> >>>
>> >>>> >>> _______________________________________________
>> >>>> >>> FreeSWITCH-users mailing list
>> >>>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> >>>
>> >>>> >>>
>> >>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> >>> http://www.freeswitch.org
>> >>>> >>>
>> >>>> >>
>> >>>> >>
>> >>>> >> _______________________________________________
>> >>>> >> FreeSWITCH-users mailing list
>> >>>> >> FreeSWITCH-users at lists.freeswitch.org
>> >>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> >>
>> >>>> >>
>> >>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> >> http://www.freeswitch.org
>> >>>> >>
>> >>>> >
>> >>>> >
>> >>>> > _______________________________________________
>> >>>> > FreeSWITCH-users mailing list
>> >>>> > FreeSWITCH-users at lists.freeswitch.org
>> >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> >
>> >>>> >
>> >>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> > http://www.freeswitch.org
>> >>>> >
>> >>>> >
>> >>>>
>> >>>> _______________________________________________
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>
>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From edpimentl at gmail.com  Tue Jan  4 21:20:38 2011
From: edpimentl at gmail.com (EdPimentl)
Date: Tue, 4 Jan 2011 13:20:38 -0500
Subject: [Freeswitch-users] FreeSWITCH configuration for Mobile Phones
 Running SIP (sipdroid) Client, 
In-Reply-To: <4D2357BB.3060803@infosecurity.ch>
References: <AANLkTinewmpFPQMv4Mwrh=OZ1BfMsxD_LDR=9m2XYdga@mail.gmail.com>
	<4D2331A6.4060300@chaschperli.ch>
	<AANLkTikVgbukrdQ9_mWxhrHW4M8GjCV3GYJFBcZ+EevG@mail.gmail.com>
	<4D2357BB.3060803@infosecurity.ch>
Message-ID: <AANLkTik2enCP3KnmOP3K1+_UGNPr6zKqzuRXHrwRL=fR@mail.gmail.com>

Thanks!

-E




On Tue, Jan 4, 2011 at 12:24 PM, Fabio Pietrosanti (naif) <
lists at infosecurity.ch> wrote:

> Some fine tuning:
> - Use SIP/TCP or SIP/TLS transport to strongly save battery respect to
> SIP/UDP
> - Increase SIP timer to be able to work mobile networks with packet loss
> (otherwise a single TCP congestion would fire
> - Disable all ping checking from server (you must reduce keepalive traffic)
> - Disable TCP kernel keepalive
> - Use high ptime and ultra-narrowband codec (4-5kbit/s with VAD/DTX) to
> work over 2.5 GPRS (edge/umts are much more powerful networks)
>
> Those are some of the major improvements we have done while setting up
> our server for PrivateGSM Professional SaaS (http://www.privatewave.com)
> ZRTP VoIP encryption client for BB/Nokia/iPhone .
>
> Fabio
>
> On 04/01/11 17.50, dome at tel.co.th wrote:
> > csipsimple also work fine.
> >
> > Dome C.
> >
> > 2011/1/4 Thomas Mueller <thomas at chaschperli.ch>:
> >> Hoi E
> >>
> >>> Can someone please share their experience on the best
> >>> configuration/dial plans for using Mobile SipDroid Client with
> FreeSwitch?
> >>>
> >> sipdroid works nice with the default dialplan.
> >>
> >> - Thomas
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Jan  4 21:30:54 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 4 Jan 2011 10:30:54 -0800
Subject: [Freeswitch-users] no ringback tone
In-Reply-To: <AANLkTim6iJ6gFc1=21REAKkvz9z8ZgPK2ShHC7pBUpxH@mail.gmail.com>
References: <AANLkTinbd06YgZFWue=eQGG9A5JO2a550tJ+bo-yG2yv@mail.gmail.com>
	<AANLkTinA46HPKmk_pWaAiMpy-hdihc7STfY9_FuNkthx@mail.gmail.com>
	<AANLkTim6iJ6gFc1=21REAKkvz9z8ZgPK2ShHC7pBUpxH@mail.gmail.com>
Message-ID: <AANLkTikxZZdV-D3c0B9yCN8jsdg4JASkxxJtevbkNA=M@mail.gmail.com>

You don't have any control over whether they send you audio, no? Either they
send it or they don't. Unless there's something else that I'm missing...
-MC

On Mon, Jan 3, 2011 at 7:03 PM, Sam <u2nsam at gmail.com> wrote:

> Some mobile operators give VAS as ringback tone / caller tune to be set as
> mobile ringback,
> now when dialing out such mobile users who has ringback enabled i just get
> plain ring and no proceeding with 183 for the media to listen,
> how to enable such so that some one dialing out will hear ringback
> generated by the mobile operators.
>
> call flow is FS --> Sangoma --> mobile (ringback enabled by mobile operator
> )
>
> Regards
> Sam
>
>
> On Tue, Jan 4, 2011 at 12:02 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Can you clarify what the actual issue is?
>> -MC
>>
>> On Sun, Jan 2, 2011 at 8:30 PM, Sam <u2nsam at gmail.com> wrote:
>>
>>> Hi All,
>>>
>>> happy new you to you !
>>>
>>> using a sangoma card and when dialing a mobile number which is having
>>> ringback tune / caller tune ;
>>> but a plain ring is heard to the user dialing that mobile through the
>>> trunk.
>>>
>>> I am using below syntax to dial out:-
>>> <action application="bridge"
>>> data="{monitor_early_media_ring_total=3}freetdm/wp1/a/${destination_number}"/>
>>>
>>> Any suggestions.
>>>
>>> Regards
>>>  Sam
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kapil.rastogi at telemune.net  Mon Jan  3 10:17:44 2011
From: kapil.rastogi at telemune.net (kapil.rastogi)
Date: Sun, 2 Jan 2011 23:17:44 -0800 (PST)
Subject: [Freeswitch-users] Regarding mod_java application in freeswitch
Message-ID: <1294039064455-5880529.post@n2.nabble.com>


Hi,

I want to know how to use mod_java application in my freeswitch voice chat
code.

Can u plz send me a sample code of java with its configuration detail.


Thanks & Regards,
Kapil Rastogi
Telemune Software Solutions P Ltd.
kapil.rastogi at telemune.net
+919013204760
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Regarding-mod-java-application-in-freeswitch-tp5880529p5880529.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kapil.rastogi at telemune.net  Mon Jan  3 18:26:41 2011
From: kapil.rastogi at telemune.net (kapil.rastogi)
Date: Mon, 3 Jan 2011 07:26:41 -0800 (PST)
Subject: [Freeswitch-users] Regarding mod_java application in freeswitch
In-Reply-To: <1294065197457-5880847.post@n2.nabble.com>
References: <1294039064455-5880529.post@n2.nabble.com>
	<1294065197457-5880847.post@n2.nabble.com>
Message-ID: <1294068401368-5880910.post@n2.nabble.com>


I am using the same java example code and configuration. But still I am
getting the error
"NoClassDefFoundError" when I make a call to freeSWITCH.

can you please specify me about these dialplan value if the name of java
program file is PhoneTest.java and it is located at the
"/usr/local/freeswitch/scripts/" path.

<action application="java" data="fully.qualified.class.name arg1 arg2
arg3"/>

<action application="java" data="/path/to/file.jar
fully.qualified.class.name arg1 arg2 arg3"/>

-----
Regards,
Kapil Rastogi
Telemune Software Solutions P Ltd.
kapil.rastogi at telemune.net
+919013204760
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Regarding-mod-java-application-in-freeswitch-tp5880529p5880910.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kapil.rastogi at telemune.net  Tue Jan  4 10:44:02 2011
From: kapil.rastogi at telemune.net (kapil.rastogi)
Date: Mon, 3 Jan 2011 23:44:02 -0800 (PST)
Subject: [Freeswitch-users] How to set value of Arrays as global variables
 setting in FreeSWITCH
Message-ID: <1294127042677-5887859.post@n2.nabble.com>


Hi,

I am using array as global variable. I want to set the value of these aarays
globally in starting of application. Can U plz send me an example how to set
these arrays globally with their function description?

-----
Regards,
Kapil Rastogi
Telemune Software Solutions P Ltd.
kapil.rastogi at telemune.net
+919013204760
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-set-value-of-Arrays-as-global-variables-setting-in-FreeSWITCH-tp5887859p5887859.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From karpov at viva64.com  Sat Jan  1 20:41:39 2011
From: karpov at viva64.com (Andrey)
Date: Sat, 1 Jan 2011 17:41:39 +0000 (UTC)
Subject: [Freeswitch-users] Can't build on Windows 7 64bit C++ Express!?
References: <AANLkTincSQSt-Pg00T4qVG+GXfVw2XpE6FU+2Ffe253y@mail.gmail.com>
Message-ID: <loom.20110101T184008-31@post.gmane.org>

How to compile a 64-bit application with using Visual C++ Express 
2005/2008/2010? - http://www.viva64.com/en/k/0009/




From kapilrastogi.ipec at gmail.com  Sun Jan  2 15:39:55 2011
From: kapilrastogi.ipec at gmail.com (kapil rastogi)
Date: Sun, 2 Jan 2011 18:09:55 +0530
Subject: [Freeswitch-users] Help
In-Reply-To: <AANLkTinBFTAsgrszgeRq=gvaYimR3VBUQS4CAnvvkdSe@mail.gmail.com>
References: <AANLkTinBFTAsgrszgeRq=gvaYimR3VBUQS4CAnvvkdSe@mail.gmail.com>
Message-ID: <AANLkTikzWZR_RnmqOq9S27YMP6V6xrY1LkpRfA012ETe@mail.gmail.com>

Hi,

I am getting error when i call to freeswitch application. I am using
application as java.

The error is : NoClassDefFoundException

Please send me an example, how to use java as an application with its all
cofiguration.






-- 
Best Regards,

Kapil Rastogi
+919013204760
+919027382143
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From siobhan.pluggedin at gmail.com  Tue Jan  4 07:57:36 2011
From: siobhan.pluggedin at gmail.com (siobhan.pluggedin at gmail.com)
Date: Tue, 04 Jan 2011 04:57:36 +0000
Subject: [Freeswitch-users] Question About Conferencing Capabilities
Message-ID: <0016e68b68bc708ea00498fe1be7@google.com>

My company is building a VOIP application, and initially were just using a  
barebones OpenSIPS implementation to host one-on-one calls; however, we  
want to expand the functionality to conferencing (which, of course,  
OpenSIPS doesn't handle) and was looking into Freeswitch (the other option  
being Asterisk). I've been poring through the docs, and have even set up a  
test server myself, but there are some very specific things we are looking  
for that I can't figure out if Freeswitch can do or not.

We want to be able to do the following:
- Create dynamic, on-the-fly conferences that can remain active even when  
initiating user leaves
- Within a conference, give users the ability to mute and/or deaf  
individual users (which I know can already be done with the "relate"  
command)
- Give users the ability to enter a "whisper" mode with another user -  
where they are holding a private conversation that can only be heard by the  
two of them
- Allow users to be in two conferences at once; the user would most likely  
have one muted at any given time so as to hear the other one, but we want  
them to be able to switch back and forth easily

Could anyone advise me on whether Freeswitch can accomplish these needs, or  
perhaps what it might take to do so? We are not averse to doing some  
customization if we can find the people who know how to make it happen!

Thanks,
Siobhan Hamilton
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From siobhan.pluggedin at gmail.com  Tue Jan  4 15:55:06 2011
From: siobhan.pluggedin at gmail.com (Siobhan Hamilton)
Date: Tue, 4 Jan 2011 07:55:06 -0500
Subject: [Freeswitch-users] Question about Conferencing Capabilities
Message-ID: <AANLkTinLyb=M1-jOYqNebLv-nGwa--0LW8buXpk466S6@mail.gmail.com>

My company is building a VOIP application, and initially were just using a
barebones OpenSIPS implementation to host one-on-one calls; however, we want
to expand the functionality to conferencing (which, of course, OpenSIPS
doesn't handle) and was looking into Freeswitch (the other option being
Asterisk).  I've been poring through the docs, and have even set up a test
server myself, but there are some very specific things we are looking for
that I can't figure out if Freeswitch can do or not.

We want to be able to do the following:
- Create dynamic, on-the-fly conferences that can remain active even when
initiating user leaves
- Within a conference, give users the ability to mute and/or deaf individual
users (which I know can already be done with the "relate" command)
- Give users the ability to enter a "whisper" mode with another user - where
they are holding a private conversation that can only be heard by the two of
them
- Allow users to be in two conferences at once; the user would most likely
have one muted at any given time so as to hear the other one, but we want
them to be able to switch back and forth easily

Could anyone advise me on whether Freeswitch can accomplish these needs, or
perhaps what it might take to do so?  We are not averse to doing some
customization if we can find the people who know how to make it happen!

Thanks,
Siobhan Hamilton
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From siobhan at pluggedin-tech.com  Tue Jan  4 18:03:53 2011
From: siobhan at pluggedin-tech.com (Siobhan Hamilton)
Date: Tue, 4 Jan 2011 10:03:53 -0500
Subject: [Freeswitch-users] Conferencing/Whisper Mode Question
Message-ID: <CD965058-A169-46E5-9EB7-A83C81841B1A@pluggedin-tech.com>

My apologies for any duplication; I have tried to post this question several times to no avail (at least from my end)....

My company is building a VOIP application, and initially were just using a barebones OpenSIPS implementation to host one-on-one calls; however, we want to expand the functionality to conferencing (which, of course, OpenSIPS doesn't handle) and was looking into Freeswitch (the other option being Asterisk).  I've been poring through the docs, and have even set up a test server myself, but there are some very specific things we are looking for that I can't figure out if Freeswitch can do or not.

We want to be able to do the following:
- Create dynamic, on-the-fly conferences that can remain active even when initiating user leaves
- Within a conference, give users the ability to mute and/or deaf individual users (which I know can already be done with the "relate" command, so that's solved, pretty much)
- Give users the ability to enter a "whisper" mode with another user - where they are holding a private conversation that can only be heard by the two of them
- Allow users to be in two conferences at once; the user would most likely have one muted at any given time so as to hear the other one, but we want them to be able to switch back and forth easily

Could anyone advise me on whether Freeswitch can accomplish these needs, or perhaps what it might take to do so?  We are not averse to doing some customization if we can find the people who know how to make it happen!

Thanks,
Siobhan Hamilton
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From msc at freeswitch.org  Tue Jan  4 22:04:26 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 4 Jan 2011 11:04:26 -0800
Subject: [Freeswitch-users] Attack using 5843 and music account?
In-Reply-To: <AANLkTi=vdoj5D_KXD06JpM_QObeuieWVN6XGn_ia74_B@mail.gmail.com>
References: <AANLkTinNmW4OtL275UWzRoZof0U_YAq3vhfwNkJo3oOG@mail.gmail.com>
	<28AF5B89-AFB3-438E-AB7B-AF598CB18204@freeswitch.org>
	<AANLkTi=vdoj5D_KXD06JpM_QObeuieWVN6XGn_ia74_B@mail.gmail.com>
Message-ID: <AANLkTi=dkhw_ixau8esZsehd6TAbWCZ=v=ZXh7gz2OCf@mail.gmail.com>

Just curious, but did you notice that the IP address was internal?
192.168.0.6 - what IP address is that?

On Mon, Jan 3, 2011 at 9:34 PM, xuyan yang <xyangni at gmail.com> wrote:

> Got it. But if no failure log. fail2ban will not work. So how can we
> protect fs from this kind of attack besides manually setup firewall rules 1
> by 1 on discovery?
>
> On Tue, Jan 4, 2011 at 12:54 AM, Brian West <brian at freeswitch.org> wrote:
>
>> Chances are he never received the challenge.. thus never logs an auth
>> failure.
>>
>> /b
>>
>> On Jan 3, 2011, at 9:26 AM, xuyan yang wrote:
>>
>> 2011-01-03 15:19:32.360152 [WARNING] sofia_reg.c:1161 SIP auth failure
>> (REGISTER) on sofia profile 'internal' for [music at 192.168.0.3] from ip
>> 192.168.0.6
>>
>> So, how can this hacker successfully registered music account and avoid to
>> be baned? it is strange.
>>
>> Thanks
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Jan  4 22:20:41 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 4 Jan 2011 11:20:41 -0800
Subject: [Freeswitch-users] Conferencing/Whisper Mode Question
In-Reply-To: <CD965058-A169-46E5-9EB7-A83C81841B1A@pluggedin-tech.com>
References: <CD965058-A169-46E5-9EB7-A83C81841B1A@pluggedin-tech.com>
Message-ID: <AANLkTimS0szLouGanNrv8XJzk7QY0ScFm8AkvJ=KaX1A@mail.gmail.com>

Answers inline..

On Tue, Jan 4, 2011 at 7:03 AM, Siobhan Hamilton <siobhan at pluggedin-tech.com
> wrote:

> My apologies for any duplication; I have tried to post this question
> several times to no avail (at least from my end)....
>
Apologies... non-list members are automatically moderated.

>
> My company is building a VOIP application, and initially were just using a
> barebones OpenSIPS implementation to host one-on-one calls; however, we want
> to expand the functionality to conferencing (which, of course, OpenSIPS
> doesn't handle) and was looking into Freeswitch (the other option being
> Asterisk).  I've been poring through the docs, and have even set up a test
> server myself, but there are some very specific things we are looking for
> that I can't figure out if Freeswitch can do or not.
>
> We want to be able to do the following:
> - Create dynamic, on-the-fly conferences that can remain active even when
> initiating user leaves
>
Yes. In fact, this is the standard behavior of FS conferences


> - Within a conference, give users the ability to mute and/or deaf
> individual users (which I know can already be done with the "relate"
> command, so that's solved, pretty much)
>
Yes, you are correct. The challenge for you will be to create the external
process that manages users and permissions, so that certain users can
deaf/mute/kick other users and also process the DTMFs that users dial while
in the conference.


> - Give users the ability to enter a "whisper" mode with another user -
> where they are holding a private conversation that can only be heard by the
> two of them
>
Yes, this can also be handled with the relate command. The challenge will be
making sure that the two parties both know that they are in whisper mode and
you have to decide if you are going to mix the audio from the rest of the
conference into the whispering parties' private chat. In any case this is
definitely doable with a little work in an external control script/program.

- Allow users to be in two conferences at once; the user would most likely
> have one muted at any given time so as to hear the other one, but we want
> them to be able to switch back and forth easily
>
This is an interesting one. I'm sure it can be done, I just don't know the
most elegant way of doing it. A brute-force way of doing it would be to let
the user be in his own "personal" conference and from there have that
personal conference make outbound calls to the other conferences. From there
you could use the mute and/or relate commands to control the flow of audio.
Again, the big challenge there would be giving the user control over his
audio and finding a way to give the user audible indications as to which
conference his audio is flowing to, if at all. FreeSWITCH absolutely has the
tools to do this. Its conference app is probably the most versatile in the
telecom world - OSS or proprietary. Coupled with the event socket you can do
al sorts of interesting things, limited only by your imagination and
programming skills.

-MC


>
> Could anyone advise me on whether Freeswitch can accomplish these needs, or
> perhaps what it might take to do so?  We are not averse to doing some
> customization if we can find the people who know how to make it happen!
>
> Thanks,
> Siobhan Hamilton
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From siobhan.pluggedin at gmail.com  Tue Jan  4 22:28:27 2011
From: siobhan.pluggedin at gmail.com (Siobhan Hamilton)
Date: Tue, 4 Jan 2011 14:28:27 -0500
Subject: [Freeswitch-users] Conferencing/Whisper Mode Question
In-Reply-To: <AANLkTimS0szLouGanNrv8XJzk7QY0ScFm8AkvJ=KaX1A@mail.gmail.com>
References: <CD965058-A169-46E5-9EB7-A83C81841B1A@pluggedin-tech.com>
	<AANLkTimS0szLouGanNrv8XJzk7QY0ScFm8AkvJ=KaX1A@mail.gmail.com>
Message-ID: <AANLkTik4H-jFArJ+k0pPWmusvMC9jdgtdN4cgZHeec1F@mail.gmail.com>

Michael, thanks very much for your response - and as an aside to anyone out
there - we are looking to hire someone to build a server with these specs
and get it running efficiently for us, if anyone is qualified/interested...



On Tue, Jan 4, 2011 at 2:20 PM, Michael Collins <msc at freeswitch.org> wrote:

> Answers inline..
>
> On Tue, Jan 4, 2011 at 7:03 AM, Siobhan Hamilton <
> siobhan at pluggedin-tech.com> wrote:
>
>> My apologies for any duplication; I have tried to post this question
>> several times to no avail (at least from my end)....
>>
> Apologies... non-list members are automatically moderated.
>
>>
>> My company is building a VOIP application, and initially were just using a
>> barebones OpenSIPS implementation to host one-on-one calls; however, we want
>> to expand the functionality to conferencing (which, of course, OpenSIPS
>> doesn't handle) and was looking into Freeswitch (the other option being
>> Asterisk).  I've been poring through the docs, and have even set up a test
>> server myself, but there are some very specific things we are looking for
>> that I can't figure out if Freeswitch can do or not.
>>
>> We want to be able to do the following:
>> - Create dynamic, on-the-fly conferences that can remain active even when
>> initiating user leaves
>>
> Yes. In fact, this is the standard behavior of FS conferences
>
>
>> - Within a conference, give users the ability to mute and/or deaf
>> individual users (which I know can already be done with the "relate"
>> command, so that's solved, pretty much)
>>
> Yes, you are correct. The challenge for you will be to create the external
> process that manages users and permissions, so that certain users can
> deaf/mute/kick other users and also process the DTMFs that users dial while
> in the conference.
>
>
>> - Give users the ability to enter a "whisper" mode with another user -
>> where they are holding a private conversation that can only be heard by the
>> two of them
>>
> Yes, this can also be handled with the relate command. The challenge will
> be making sure that the two parties both know that they are in whisper mode
> and you have to decide if you are going to mix the audio from the rest of
> the conference into the whispering parties' private chat. In any case this
> is definitely doable with a little work in an external control
> script/program.
>
> - Allow users to be in two conferences at once; the user would most likely
>> have one muted at any given time so as to hear the other one, but we want
>> them to be able to switch back and forth easily
>>
> This is an interesting one. I'm sure it can be done, I just don't know the
> most elegant way of doing it. A brute-force way of doing it would be to let
> the user be in his own "personal" conference and from there have that
> personal conference make outbound calls to the other conferences. From there
> you could use the mute and/or relate commands to control the flow of audio.
> Again, the big challenge there would be giving the user control over his
> audio and finding a way to give the user audible indications as to which
> conference his audio is flowing to, if at all. FreeSWITCH absolutely has the
> tools to do this. Its conference app is probably the most versatile in the
> telecom world - OSS or proprietary. Coupled with the event socket you can do
> al sorts of interesting things, limited only by your imagination and
> programming skills.
>
> -MC
>
>
>>
>> Could anyone advise me on whether Freeswitch can accomplish these needs,
>> or perhaps what it might take to do so?  We are not averse to doing some
>> customization if we can find the people who know how to make it happen!
>>
>> Thanks,
>> Siobhan Hamilton
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Jan  4 22:43:45 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 4 Jan 2011 11:43:45 -0800
Subject: [Freeswitch-users] Conferencing/Whisper Mode Question
In-Reply-To: <AANLkTik4H-jFArJ+k0pPWmusvMC9jdgtdN4cgZHeec1F@mail.gmail.com>
References: <CD965058-A169-46E5-9EB7-A83C81841B1A@pluggedin-tech.com>
	<AANLkTimS0szLouGanNrv8XJzk7QY0ScFm8AkvJ=KaX1A@mail.gmail.com>
	<AANLkTik4H-jFArJ+k0pPWmusvMC9jdgtdN4cgZHeec1F@mail.gmail.com>
Message-ID: <AANLkTik3c5b11nEUSkdPsTcHCqVkT7v04TGLyOF5Q0hO@mail.gmail.com>

You might find a few people. :) If you want "the pros" then go to
www.freeswitchsolutions.com or email support at freeswitch.org. The guys who
wrote FreeSWITCH run that and they have a network of professional FS
consultants who can do all sorts of fun things for you!

-MC

On Tue, Jan 4, 2011 at 11:28 AM, Siobhan Hamilton <
siobhan.pluggedin at gmail.com> wrote:

> Michael, thanks very much for your response - and as an aside to anyone out
> there - we are looking to hire someone to build a server with these specs
> and get it running efficiently for us, if anyone is qualified/interested...
>
>
>
> On Tue, Jan 4, 2011 at 2:20 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Answers inline..
>>
>> On Tue, Jan 4, 2011 at 7:03 AM, Siobhan Hamilton <
>> siobhan at pluggedin-tech.com> wrote:
>>
>>> My apologies for any duplication; I have tried to post this question
>>> several times to no avail (at least from my end)....
>>>
>> Apologies... non-list members are automatically moderated.
>>
>>>
>>> My company is building a VOIP application, and initially were just using
>>> a barebones OpenSIPS implementation to host one-on-one calls; however, we
>>> want to expand the functionality to conferencing (which, of course, OpenSIPS
>>> doesn't handle) and was looking into Freeswitch (the other option being
>>> Asterisk).  I've been poring through the docs, and have even set up a test
>>> server myself, but there are some very specific things we are looking for
>>> that I can't figure out if Freeswitch can do or not.
>>>
>>> We want to be able to do the following:
>>> - Create dynamic, on-the-fly conferences that can remain active even when
>>> initiating user leaves
>>>
>> Yes. In fact, this is the standard behavior of FS conferences
>>
>>
>>> - Within a conference, give users the ability to mute and/or deaf
>>> individual users (which I know can already be done with the "relate"
>>> command, so that's solved, pretty much)
>>>
>>  Yes, you are correct. The challenge for you will be to create the
>> external process that manages users and permissions, so that certain users
>> can deaf/mute/kick other users and also process the DTMFs that users dial
>> while in the conference.
>>
>>
>>> - Give users the ability to enter a "whisper" mode with another user -
>>> where they are holding a private conversation that can only be heard by the
>>> two of them
>>>
>> Yes, this can also be handled with the relate command. The challenge will
>> be making sure that the two parties both know that they are in whisper mode
>> and you have to decide if you are going to mix the audio from the rest of
>> the conference into the whispering parties' private chat. In any case this
>> is definitely doable with a little work in an external control
>> script/program.
>>
>> - Allow users to be in two conferences at once; the user would most likely
>>> have one muted at any given time so as to hear the other one, but we want
>>> them to be able to switch back and forth easily
>>>
>> This is an interesting one. I'm sure it can be done, I just don't know the
>> most elegant way of doing it. A brute-force way of doing it would be to let
>> the user be in his own "personal" conference and from there have that
>> personal conference make outbound calls to the other conferences. From there
>> you could use the mute and/or relate commands to control the flow of audio.
>> Again, the big challenge there would be giving the user control over his
>> audio and finding a way to give the user audible indications as to which
>> conference his audio is flowing to, if at all. FreeSWITCH absolutely has the
>> tools to do this. Its conference app is probably the most versatile in the
>> telecom world - OSS or proprietary. Coupled with the event socket you can do
>> al sorts of interesting things, limited only by your imagination and
>> programming skills.
>>
>> -MC
>>
>>
>>>
>>> Could anyone advise me on whether Freeswitch can accomplish these needs,
>>> or perhaps what it might take to do so?  We are not averse to doing some
>>> customization if we can find the people who know how to make it happen!
>>>
>>> Thanks,
>>> Siobhan Hamilton
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Jan  4 22:55:52 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 4 Jan 2011 11:55:52 -0800
Subject: [Freeswitch-users] multi company
In-Reply-To: <AANLkTikNNi9J8oJOMzqwfuZNSPwEsLnkMe2sX5ZSxvUG@mail.gmail.com>
References: <AANLkTikjjhHMMxKmsEGpDYxXBpozjNycdimV6zZFmhL7@mail.gmail.com>
	<AANLkTikW=LmuBhtLZ3Kxqhaqx95oiz4VTs3EfFU2tNVh@mail.gmail.com>
	<AANLkTikNNi9J8oJOMzqwfuZNSPwEsLnkMe2sX5ZSxvUG@mail.gmail.com>
Message-ID: <AANLkTimsvoiQVG4+0ePzT3k8bJKnW2w4jYm_ZfrbcK+Z@mail.gmail.com>

I am unable to replicate this error. I added a test domain in the
default.xml file and it worked. I also created an entirely new file:
conf/directory/mikey.xml and it worked as well. For the record, here is my
file:

<include>

  <domain name="mikey">
    <params>
      <param name="dial-string" value="{presence_id=${dialed_user}@
${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
    </params>

    <variables>
      <variable name="record_stereo" value="true"/>
      <variable name="default_gateway" value="$${default_provider}"/>
      <variable name="default_areacode" value="$${default_areacode}"/>
      <variable name="transfer_fallback_extension" value="operator"/>
    </variables>
  </domain>

</include>

As you can see, I simply copied and pasted some stuff from the default
domain into the 'mikey' domain and it worked just fine.

My guess is that you've got an erroneous closing tag somewhere in your XML.
If you can't find it then I recommend pastebin the directory section and ask
others to have a look.
-MC

On Mon, Jan 3, 2011 at 7:05 PM, Sam <u2nsam at gmail.com> wrote:

> Did it exactly what they they have suggested in wiki but give that error.
>
> Regards
> Sam
>
>
> On Tue, Jan 4, 2011 at 12:07 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Just for confirmation, have you looked at these two pages?
>> http://wiki.freeswitch.org/wiki/Multi-tenant
>> http://wiki.freeswitch.org/wiki/Multiple_Domains
>>
>> There is quite a bit of information for you to try.
>> -MC
>>
>> On Mon, Jan 3, 2011 at 2:37 AM, Sam <u2nsam at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> Was using multi company setup,
>>>
>>> it gave an error while using below syntax
>>>
>>> <domain name="xyz">
>>>
>>> </domain>
>>>
>>> Cannot Initialize [[error near line 2868]: unexpected closing tag
>>> </domain>]
>>>
>>> And when i remove , it dont gives an error .
>>> the file is in directory/xyz.xml
>>>
>>>
>>> same happens on freeswitch.xml file
>>>
>>> when i remove the syntax
>>>
>>> <section name="directory" description="User Directory">
>>>     <X-PRE-PROCESS cmd="include" data="directory/*.xml"/>
>>>   </section>
>>>
>>> it works when removed the section other wise give error as unexpected
>>> closing tag </section>]
>>>
>>> Regds
>>> Sam
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gmaruzz at gmail.com  Wed Jan  5 00:54:37 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 4 Jan 2011 22:54:37 +0100
Subject: [Freeswitch-users] Attack using 5843 and music account?
In-Reply-To: <AANLkTi=dkhw_ixau8esZsehd6TAbWCZ=v=ZXh7gz2OCf@mail.gmail.com>
References: <AANLkTinNmW4OtL275UWzRoZof0U_YAq3vhfwNkJo3oOG@mail.gmail.com>
	<28AF5B89-AFB3-438E-AB7B-AF598CB18204@freeswitch.org>
	<AANLkTi=vdoj5D_KXD06JpM_QObeuieWVN6XGn_ia74_B@mail.gmail.com>
	<AANLkTi=dkhw_ixau8esZsehd6TAbWCZ=v=ZXh7gz2OCf@mail.gmail.com>
Message-ID: <AANLkTimH+ORtX_0Vvg1Qqn8AVBVRd5zi3gOwSfn7U+Aq@mail.gmail.com>

The internal address seems to be the one from which the OPoster tried
to be registered, and sent the challenge (and was rejected).
The original attempts, that were not rejected (probably because were
not sending challenges) were coming from external addresses.
-giovanni


On 1/4/11, Michael Collins <msc at freeswitch.org> wrote:
> Just curious, but did you notice that the IP address was internal?
> 192.168.0.6 - what IP address is that?
>
> On Mon, Jan 3, 2011 at 9:34 PM, xuyan yang <xyangni at gmail.com> wrote:
>
>> Got it. But if no failure log. fail2ban will not work. So how can we
>> protect fs from this kind of attack besides manually setup firewall rules
>> 1
>> by 1 on discovery?
>>
>> On Tue, Jan 4, 2011 at 12:54 AM, Brian West <brian at freeswitch.org> wrote:
>>
>>> Chances are he never received the challenge.. thus never logs an auth
>>> failure.
>>>
>>> /b
>>>
>>> On Jan 3, 2011, at 9:26 AM, xuyan yang wrote:
>>>
>>> 2011-01-03 15:19:32.360152 [WARNING] sofia_reg.c:1161 SIP auth failure
>>> (REGISTER) on sofia profile 'internal' for [music at 192.168.0.3] from ip
>>> 192.168.0.6
>>>
>>> So, how can this hacker successfully registered music account and avoid
>>> to
>>> be baned? it is strange.
>>>
>>> Thanks
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From Avi at aMarcus.com  Wed Jan  5 01:37:49 2011
From: Avi at aMarcus.com (Avi Marcus)
Date: Wed, 5 Jan 2011 00:37:49 +0200
Subject: [Freeswitch-users] SIP calls to Verizon FiOS?
Message-ID: <AANLkTikDOCr+FOUsof3Ap4B4rZzHKQkqTjQS3VOuoz-m@mail.gmail.com>

Hi, someone mentioned in IRC that they were finishing to submit paperwork
for peering with Verizon.
I have a few numbers on the FiOS network that I call often and would
appreciate info on making SIP calls to them.
Thanks!
-Avi
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From covici at ccs.covici.com  Wed Jan  5 02:15:20 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 04 Jan 2011 18:15:20 -0500
Subject: [Freeswitch-users] SIP calls to Verizon FiOS?
In-Reply-To: <AANLkTikDOCr+FOUsof3Ap4B4rZzHKQkqTjQS3VOuoz-m@mail.gmail.com>
References: <AANLkTikDOCr+FOUsof3Ap4B4rZzHKQkqTjQS3VOuoz-m@mail.gmail.com>
Message-ID: <31655.1294182920@ccs.covici.com>

I have no problems making calls to a box on fios.  The provider is not
Verizon, but that does not seem to matter.

Avi Marcus <Avi at aMarcus.com> wrote:

> Hi, someone mentioned in IRC that they were finishing to submit paperwork
> for peering with Verizon.
> I have a few numbers on the FiOS network that I call often and would
> appreciate info on making SIP calls to them.
> Thanks!
> -Avi
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From Avi at aMarcus.com  Wed Jan  5 02:28:36 2011
From: Avi at aMarcus.com (Avi Marcus)
Date: Wed, 5 Jan 2011 01:28:36 +0200
Subject: [Freeswitch-users] SIP calls to Verizon FiOS?
In-Reply-To: <31655.1294182920@ccs.covici.com>
References: <AANLkTikDOCr+FOUsof3Ap4B4rZzHKQkqTjQS3VOuoz-m@mail.gmail.com>
	<31655.1294182920@ccs.covici.com>
Message-ID: <AANLkTimhYz-SQqTU3pZef6AH8o9TtnGP4SGQ0R9u0y6e@mail.gmail.com>

Oh, I'm sorry, I'm referring to the actual FiOS Telephone. It's part of
their package - Verizon switches over your POTS telephone to a VOIP phone
when you sign up for the package. I'd like to make a sip call to that phone.
-Avi

On Wed, Jan 5, 2011 at 1:15 AM, <covici at ccs.covici.com> wrote:

> I have no problems making calls to a box on fios.  The provider is not
> Verizon, but that does not seem to matter.
>
> Avi Marcus <Avi at aMarcus.com> wrote:
>
> > Hi, someone mentioned in IRC that they were finishing to submit paperwork
> > for peering with Verizon.
> > I have a few numbers on the FiOS network that I call often and would
> > appreciate info on making SIP calls to them.
> > Thanks!
> > -Avi
> >
> > ----------------------------------------------------
> > Alternatives:
> >
> > ----------------------------------------------------
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From covici at ccs.covici.com  Wed Jan  5 02:40:46 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 04 Jan 2011 18:40:46 -0500
Subject: [Freeswitch-users] SIP calls to Verizon FiOS?
In-Reply-To: <AANLkTimhYz-SQqTU3pZef6AH8o9TtnGP4SGQ0R9u0y6e@mail.gmail.com>
References: <AANLkTikDOCr+FOUsof3Ap4B4rZzHKQkqTjQS3VOuoz-m@mail.gmail.com>
	<31655.1294182920@ccs.covici.com>
	<AANLkTimhYz-SQqTU3pZef6AH8o9TtnGP4SGQ0R9u0y6e@mail.gmail.com>
Message-ID: <32089.1294184446@ccs.covici.com>

That phone is on the pstn, not sip.  The FIOS fiber it uses is actually
not on the internet at all.

Avi Marcus <Avi at aMarcus.com> wrote:

> Oh, I'm sorry, I'm referring to the actual FiOS Telephone. It's part of
> their package - Verizon switches over your POTS telephone to a VOIP phone
> when you sign up for the package. I'd like to make a sip call to that phone.
> -Avi
> 
> On Wed, Jan 5, 2011 at 1:15 AM, <covici at ccs.covici.com> wrote:
> 
> > I have no problems making calls to a box on fios.  The provider is not
> > Verizon, but that does not seem to matter.
> >
> > Avi Marcus <Avi at aMarcus.com> wrote:
> >
> > > Hi, someone mentioned in IRC that they were finishing to submit paperwork
> > > for peering with Verizon.
> > > I have a few numbers on the FiOS network that I call often and would
> > > appreciate info on making SIP calls to them.
> > > Thanks!
> > > -Avi
> > >
> > > ----------------------------------------------------
> > > Alternatives:
> > >
> > > ----------------------------------------------------
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >         John Covici
> >         covici at ccs.covici.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From infos at madovsky.org  Wed Jan  5 04:33:35 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 4 Jan 2011 20:33:35 -0500
Subject: [Freeswitch-users] change nibble_account on answer
Message-ID: <5BB07D02D9ED46B284CA94565664BE02@e1705>

.....
<action application="set" data="nibble_account=000"/>
.....
<action application="answer"/>
<action application="set" data="nibble_account=111"/>
.......

it seems that nibblebill keeps the nibble reference before answer ?

Thanks
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From joaocarlosleme at gmail.com  Wed Jan  5 04:42:35 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Tue, 4 Jan 2011 17:42:35 -0800
Subject: [Freeswitch-users] Can't build on Windows 7 64bit C++ Express!?
In-Reply-To: <snt0-eas12908A012C361DEE1DF0DC7B9070@phx.gbl>
References: <snt0-eas12908A012C361DEE1DF0DC7B9070@phx.gbl>
Message-ID: <AANLkTimkeKAR+qrZ827y-1H_2jczQSPLhUR_qXQP=me2@mail.gmail.com>

Hi Jeff,
I just opened the solution, right click on the main solution node and
selected build. It is .NET Framework 4 targeting Win32 (default I guess).
Didn't to any changes.

Here is a log of the build

Build started 1/4/2011 5:14:04 PM.
Project "C:\Users\DEll
1\Documents\FreeSWITCH\Freeswitch3\src\mod\applications\mod_fsv\mod_fsv.2010.vcxproj"
on node 2 (build target(s)).
InitializeBuildStatus:
  Creating "Win32\Debug\mod_fsv.unsuccessfulbuild" because "AlwaysCreate"
was specified.
ClCompile:
  C:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\bin\CL.exe /c
/I"C:\Users\DEll 1\Documents\FreeSWITCH\Freeswitch3\\src\include" /Zi
/nologo /W4 /WX- /Od /Oy- /D _DEBUG /D DEBUG /D WIN32 /D _WINDOWS /D _USRDLL
/D MOD_EXPORTS /D _WINDLL /D _MBCS /Gm- /EHsc /RTC1 /MDd /GS /fp:precise
/Zc:wchar_t /Zc:forScope /Fo"Win32\Debug\\" /Fd"Win32\Debug\vc100.pdb" /Gd
/TC /analyze /errorReport:prompt mod_fsv.c
  mod_fsv.c
c:\users\dell
1\documents\freeswitch\freeswitch3\src\mod\applications\mod_fsv\mod_fsv.c(342):
warning C6246: Local declaration of 'data' hides declaration of the same
name in outer scope. For additional information, see previous declaration at
line '236' of 'c:\users\dell
1\documents\freeswitch\freeswitch3\src\mod\applications\mod_fsv\mod_fsv.c':
Lines: 236
Link:
  C:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\bin\link.exe
/ERRORREPORT:PROMPT /OUT:"C:\Users\DEll
1\Documents\FreeSWITCH\Freeswitch3\Win32\Debug\mod\mod_fsv.dll"
/INCREMENTAL:NO /NOLOGO kernel32.lib user32.lib gdi32.lib winspool.lib
comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib
odbc32.lib odbccp32.lib Ws2_32.lib Iphlpapi.lib Winmm.lib kernel32.lib
user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib
ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /MANIFEST
/ManifestFile:"Win32\Debug\mod_fsv.dll.intermediate.manifest"
/MANIFESTUAC:"level='asInvoker' uiAccess='false'" /DEBUG /PDB:"C:\Users\DEll
1\Documents\FreeSWITCH\Freeswitch3\Win32\Debug\mod\mod_fsv.pdb" /TLBID:1
/DYNAMICBASE:NO /IMPLIB:"C:\Users\DEll
1\Documents\FreeSWITCH\Freeswitch3\Win32\Debug\mod\mod_fsv.lib" /MACHINE:X86
/DLL Win32\Debug\mod_fsv.obj
  "C:\Users\DEll
1\Documents\FreeSWITCH\Freeswitch3\Win32\Debug\FreeSwitchCore.lib"
LINK : fatal error LNK1181: cannot open input file 'C:\Users\DEll
1\Documents\FreeSWITCH\Freeswitch3\Win32\Debug\FreeSwitchCore.lib'
Done Building Project "C:\Users\DEll
1\Documents\FreeSWITCH\Freeswitch3\src\mod\applications\mod_fsv\mod_fsv.2010.vcxproj"
(build target(s)) -- FAILED.

Build FAILED.

Time Elapsed 00:00:05.55

Lots of c1 : fatal error C1083: Cannot open source file:
'pcre_chartables.c': No such file or directory

On Sun, Jan 2, 2011 at 8:09 PM, Jeff Lenk <jeff at jefflenk.com> wrote:

>  What target are you trying to build? Post a log of the build or pastebin
> reference
>
> Sent from my Windows Phone
> ------------------------------
> From: Joao Leme
> Sent: Sunday, January 02, 2011 1:28 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Can't build on Windows 7 64bit C++
> Express!?
>
>
>
> > Ok Thanks. Any idea why all the errors and why i can't build? I've built
> > before on Windows Vista 64bit and VS2008Pro with no problems but can't
> get
> > it to work on Express edition.
> >
> > On Sat, Jan 1, 2011 at 9:51 PM, babak yakhchali
> ><babak.freeswitch at gmail.com>wrote:
> >
> >> you just need to build the project in
> >>
> >>
> D:\gitRepos\2010exp-freeswitch\freeswitch\src\mod\languages\mod_managed\managed
> >> using vc# express, others should just work using vc++
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sos at sokhapkin.dyndns.org  Wed Jan  5 04:44:44 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 4 Jan 2011 20:44:44 -0500
Subject: [Freeswitch-users] change nibble_account on answer
In-Reply-To: <5BB07D02D9ED46B284CA94565664BE02@e1705>
References: <5BB07D02D9ED46B284CA94565664BE02@e1705>
Message-ID: <201101042044.44553.sos@sokhapkin.dyndns.org>

Do you run mod_nibblebill on a or b leg?

On Tuesday 04 January 2011, Madovsky wrote:
> .....
> <action application="set" data="nibble_account=000"/>
> .....
> <action application="answer"/>
> <action application="set" data="nibble_account=111"/>
> .......
> 
> it seems that nibblebill keeps the nibble reference before answer ?
> 
> Thanks



From infos at madovsky.org  Wed Jan  5 05:03:37 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 4 Jan 2011 21:03:37 -0500
Subject: [Freeswitch-users] change nibble_account on answer
References: <5BB07D02D9ED46B284CA94565664BE02@e1705>
	<201101042044.44553.sos@sokhapkin.dyndns.org>
Message-ID: <10EEC07125C44DB28029DF1D396DEF65@e1705>

Hi Sergey, what's up ? ;)

yes legA

thanks

----- Original Message ----- 
From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, January 04, 2011 8:44 PM
Subject: Re: [Freeswitch-users] change nibble_account on answer


> Do you run mod_nibblebill on a or b leg?
> 
> On Tuesday 04 January 2011, Madovsky wrote:
>> .....
>> <action application="set" data="nibble_account=000"/>
>> .....
>> <action application="answer"/>
>> <action application="set" data="nibble_account=111"/>
>> .......
>> 
>> it seems that nibblebill keeps the nibble reference before answer ?
>> 
>> Thanks
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From sos at sokhapkin.dyndns.org  Wed Jan  5 05:22:11 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 4 Jan 2011 21:22:11 -0500
Subject: [Freeswitch-users] change nibble_account on answer
In-Reply-To: <10EEC07125C44DB28029DF1D396DEF65@e1705>
References: <5BB07D02D9ED46B284CA94565664BE02@e1705>
	<201101042044.44553.sos@sokhapkin.dyndns.org>
	<10EEC07125C44DB28029DF1D396DEF65@e1705>
Message-ID: <201101042122.11883.sos@sokhapkin.dyndns.org>

Why do you think mod_nibblebill do not change account? It reads channel 
variable nibble_account on every billing attempt. I also execute  "nibblebill" 
application, otherwise mod_nibblebill doesn't work for me on leg a.

On Tuesday 04 January 2011, Madovsky wrote:
> Hi Sergey, what's up ? ;)
> 
> yes legA
> 
> thanks
> 
> ----- Original Message -----
> From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, January 04, 2011 8:44 PM
> Subject: Re: [Freeswitch-users] change nibble_account on answer
> 
> > Do you run mod_nibblebill on a or b leg?
> > 
> > On Tuesday 04 January 2011, Madovsky wrote:
> >> .....
> >> <action application="set" data="nibble_account=000"/>
> >> .....
> >> <action application="answer"/>
> >> <action application="set" data="nibble_account=111"/>
> >> .......
> >> 
> >> it seems that nibblebill keeps the nibble reference before answer ?
> >> 
> >> Thanks
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From infos at madovsky.org  Wed Jan  5 05:27:35 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 4 Jan 2011 21:27:35 -0500
Subject: [Freeswitch-users] change nibble_account on answer
References: <5BB07D02D9ED46B284CA94565664BE02@e1705><201101042044.44553.sos@sokhapkin.dyndns.org>
	<10EEC07125C44DB28029DF1D396DEF65@e1705>
Message-ID: <EF401B18FB6F4B19A6487A435DDF9B69@e1705>

forgot to say it's for a leg A that enter in conference
so it's one leg only

----- Original Message ----- 
From: "Madovsky" <infos at madovsky.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, January 04, 2011 9:03 PM
Subject: Re: [Freeswitch-users] change nibble_account on answer


> Hi Sergey, what's up ? ;)
> 
> yes legA
> 
> thanks
> 
> ----- Original Message ----- 
> From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, January 04, 2011 8:44 PM
> Subject: Re: [Freeswitch-users] change nibble_account on answer
> 
> 
>> Do you run mod_nibblebill on a or b leg?
>> 
>> On Tuesday 04 January 2011, Madovsky wrote:
>>> .....
>>> <action application="set" data="nibble_account=000"/>
>>> .....
>>> <action application="answer"/>
>>> <action application="set" data="nibble_account=111"/>
>>> .......
>>> 
>>> it seems that nibblebill keeps the nibble reference before answer ?
>>> 
>>> Thanks
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From infos at madovsky.org  Wed Jan  5 05:29:04 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 4 Jan 2011 21:29:04 -0500
Subject: [Freeswitch-users] change nibble_account on answer
References: <5BB07D02D9ED46B284CA94565664BE02@e1705><201101042044.44553.sos@sokhapkin.dyndns.org><10EEC07125C44DB28029DF1D396DEF65@e1705>
	<201101042122.11883.sos@sokhapkin.dyndns.org>
Message-ID: <F9F2C602D35F48899EAF6EF74C17DD7C@e1705>

when answered,
even if I change nibble_account to another user,
the nibble_current_balance stays to the first user account set before answer

----- Original Message ----- 
From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, January 04, 2011 9:22 PM
Subject: Re: [Freeswitch-users] change nibble_account on answer


> Why do you think mod_nibblebill do not change account? It reads channel
> variable nibble_account on every billing attempt. I also execute 
> "nibblebill"
> application, otherwise mod_nibblebill doesn't work for me on leg a.
>
> On Tuesday 04 January 2011, Madovsky wrote:
>> Hi Sergey, what's up ? ;)
>>
>> yes legA
>>
>> thanks
>>
>> ----- Original Message -----
>> From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Tuesday, January 04, 2011 8:44 PM
>> Subject: Re: [Freeswitch-users] change nibble_account on answer
>>
>> > Do you run mod_nibblebill on a or b leg?
>> >
>> > On Tuesday 04 January 2011, Madovsky wrote:
>> >> .....
>> >> <action application="set" data="nibble_account=000"/>
>> >> .....
>> >> <action application="answer"/>
>> >> <action application="set" data="nibble_account=111"/>
>> >> .......
>> >>
>> >> it seems that nibblebill keeps the nibble reference before answer ?
>> >>
>> >> Thanks
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From infos at madovsky.org  Wed Jan  5 07:06:15 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 4 Jan 2011 23:06:15 -0500
Subject: [Freeswitch-users] change nibble_account on answer
References: <5BB07D02D9ED46B284CA94565664BE02@e1705><201101042044.44553.sos@sokhapkin.dyndns.org><10EEC07125C44DB28029DF1D396DEF65@e1705>
	<201101042122.11883.sos@sokhapkin.dyndns.org>
Message-ID: <CC3B0EFA536647ED801B04C8B35D578C@e1705>

argh, I don't know why now it works... I took vodka and now it's magical ;)

----- Original Message ----- 
From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, January 04, 2011 9:22 PM
Subject: Re: [Freeswitch-users] change nibble_account on answer


> Why do you think mod_nibblebill do not change account? It reads channel
> variable nibble_account on every billing attempt. I also execute 
> "nibblebill"
> application, otherwise mod_nibblebill doesn't work for me on leg a.
>
> On Tuesday 04 January 2011, Madovsky wrote:
>> Hi Sergey, what's up ? ;)
>>
>> yes legA
>>
>> thanks
>>
>> ----- Original Message -----
>> From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Tuesday, January 04, 2011 8:44 PM
>> Subject: Re: [Freeswitch-users] change nibble_account on answer
>>
>> > Do you run mod_nibblebill on a or b leg?
>> >
>> > On Tuesday 04 January 2011, Madovsky wrote:
>> >> .....
>> >> <action application="set" data="nibble_account=000"/>
>> >> .....
>> >> <action application="answer"/>
>> >> <action application="set" data="nibble_account=111"/>
>> >> .......
>> >>
>> >> it seems that nibblebill keeps the nibble reference before answer ?
>> >>
>> >> Thanks
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From u2nsam at gmail.com  Wed Jan  5 07:14:11 2011
From: u2nsam at gmail.com (Sam)
Date: Wed, 5 Jan 2011 09:44:11 +0530
Subject: [Freeswitch-users] no ringback tone
In-Reply-To: <AANLkTikxZZdV-D3c0B9yCN8jsdg4JASkxxJtevbkNA=M@mail.gmail.com>
References: <AANLkTinbd06YgZFWue=eQGG9A5JO2a550tJ+bo-yG2yv@mail.gmail.com>
	<AANLkTinA46HPKmk_pWaAiMpy-hdihc7STfY9_FuNkthx@mail.gmail.com>
	<AANLkTim6iJ6gFc1=21REAKkvz9z8ZgPK2ShHC7pBUpxH@mail.gmail.com>
	<AANLkTikxZZdV-D3c0B9yCN8jsdg4JASkxxJtevbkNA=M@mail.gmail.com>
Message-ID: <AANLkTi=m4DvTCSCNfmfPjPwBDiYQwT+_7M_vEpCTXnOC@mail.gmail.com>

But when i do it from asterisk, it gives back the audio but not when i do it
from freeswitch with sangoma.
is there any method to listen to the audio send from the operator with
combination with sangoma?

Regds
Sam


On Wed, Jan 5, 2011 at 12:00 AM, Michael Collins <msc at freeswitch.org> wrote:

> You don't have any control over whether they send you audio, no? Either
> they send it or they don't. Unless there's something else that I'm
> missing...
> -MC
>
>
> On Mon, Jan 3, 2011 at 7:03 PM, Sam <u2nsam at gmail.com> wrote:
>
>> Some mobile operators give VAS as ringback tone / caller tune to be set as
>> mobile ringback,
>> now when dialing out such mobile users who has ringback enabled i just get
>> plain ring and no proceeding with 183 for the media to listen,
>> how to enable such so that some one dialing out will hear ringback
>> generated by the mobile operators.
>>
>> call flow is FS --> Sangoma --> mobile (ringback enabled by mobile
>> operator )
>>
>> Regards
>> Sam
>>
>>
>> On Tue, Jan 4, 2011 at 12:02 AM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Can you clarify what the actual issue is?
>>> -MC
>>>
>>> On Sun, Jan 2, 2011 at 8:30 PM, Sam <u2nsam at gmail.com> wrote:
>>>
>>>> Hi All,
>>>>
>>>> happy new you to you !
>>>>
>>>> using a sangoma card and when dialing a mobile number which is having
>>>> ringback tune / caller tune ;
>>>> but a plain ring is heard to the user dialing that mobile through the
>>>> trunk.
>>>>
>>>> I am using below syntax to dial out:-
>>>> <action application="bridge"
>>>> data="{monitor_early_media_ring_total=3}freetdm/wp1/a/${destination_number}"/>
>>>>
>>>> Any suggestions.
>>>>
>>>> Regards
>>>>  Sam
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Jan  5 08:06:50 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 4 Jan 2011 21:06:50 -0800
Subject: [Freeswitch-users] no ringback tone
In-Reply-To: <AANLkTi=m4DvTCSCNfmfPjPwBDiYQwT+_7M_vEpCTXnOC@mail.gmail.com>
References: <AANLkTinbd06YgZFWue=eQGG9A5JO2a550tJ+bo-yG2yv@mail.gmail.com>
	<AANLkTinA46HPKmk_pWaAiMpy-hdihc7STfY9_FuNkthx@mail.gmail.com>
	<AANLkTim6iJ6gFc1=21REAKkvz9z8ZgPK2ShHC7pBUpxH@mail.gmail.com>
	<AANLkTikxZZdV-D3c0B9yCN8jsdg4JASkxxJtevbkNA=M@mail.gmail.com>
	<AANLkTi=m4DvTCSCNfmfPjPwBDiYQwT+_7M_vEpCTXnOC@mail.gmail.com>
Message-ID: <AANLkTim6OZAuLMBdTDmv_rmg1jcGC5VdD7RhsaEQW18r@mail.gmail.com>

What kind of phone service is this? If it's ISDN then you can compare
d-channel traces between the Asterisk and FreeSWITCH instances and see if
there are any differences. Otherwise I'll have to defer to Sangoma on this
one.

-MC

On Tue, Jan 4, 2011 at 8:14 PM, Sam <u2nsam at gmail.com> wrote:

> But when i do it from asterisk, it gives back the audio but not when i do
> it from freeswitch with sangoma.
> is there any method to listen to the audio send from the operator with
> combination with sangoma?
>
> Regds
> Sam
>
>
>
> On Wed, Jan 5, 2011 at 12:00 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> You don't have any control over whether they send you audio, no? Either
>> they send it or they don't. Unless there's something else that I'm
>> missing...
>> -MC
>>
>>
>> On Mon, Jan 3, 2011 at 7:03 PM, Sam <u2nsam at gmail.com> wrote:
>>
>>> Some mobile operators give VAS as ringback tone / caller tune to be set
>>> as mobile ringback,
>>> now when dialing out such mobile users who has ringback enabled i just
>>> get plain ring and no proceeding with 183 for the media to listen,
>>> how to enable such so that some one dialing out will hear ringback
>>> generated by the mobile operators.
>>>
>>> call flow is FS --> Sangoma --> mobile (ringback enabled by mobile
>>> operator )
>>>
>>> Regards
>>> Sam
>>>
>>>
>>> On Tue, Jan 4, 2011 at 12:02 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> Can you clarify what the actual issue is?
>>>> -MC
>>>>
>>>> On Sun, Jan 2, 2011 at 8:30 PM, Sam <u2nsam at gmail.com> wrote:
>>>>
>>>>> Hi All,
>>>>>
>>>>> happy new you to you !
>>>>>
>>>>> using a sangoma card and when dialing a mobile number which is having
>>>>> ringback tune / caller tune ;
>>>>> but a plain ring is heard to the user dialing that mobile through the
>>>>> trunk.
>>>>>
>>>>> I am using below syntax to dial out:-
>>>>> <action application="bridge"
>>>>> data="{monitor_early_media_ring_total=3}freetdm/wp1/a/${destination_number}"/>
>>>>>
>>>>> Any suggestions.
>>>>>
>>>>> Regards
>>>>>  Sam
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Wed Jan  5 08:21:49 2011
From: u2nsam at gmail.com (Sam)
Date: Wed, 5 Jan 2011 10:51:49 +0530
Subject: [Freeswitch-users] no ringback tone
In-Reply-To: <AANLkTim6OZAuLMBdTDmv_rmg1jcGC5VdD7RhsaEQW18r@mail.gmail.com>
References: <AANLkTinbd06YgZFWue=eQGG9A5JO2a550tJ+bo-yG2yv@mail.gmail.com>
	<AANLkTinA46HPKmk_pWaAiMpy-hdihc7STfY9_FuNkthx@mail.gmail.com>
	<AANLkTim6iJ6gFc1=21REAKkvz9z8ZgPK2ShHC7pBUpxH@mail.gmail.com>
	<AANLkTikxZZdV-D3c0B9yCN8jsdg4JASkxxJtevbkNA=M@mail.gmail.com>
	<AANLkTi=m4DvTCSCNfmfPjPwBDiYQwT+_7M_vEpCTXnOC@mail.gmail.com>
	<AANLkTim6OZAuLMBdTDmv_rmg1jcGC5VdD7RhsaEQW18r@mail.gmail.com>
Message-ID: <AANLkTimi9iT+BxWP1y7BnvYLxysu9tSuzf+S08z1bp-L@mail.gmail.com>

I have attached the isdn pcap to this mail of FS --> Sangoma --> pri

Regards
Sam

On Wed, Jan 5, 2011 at 10:36 AM, Michael Collins <msc at freeswitch.org> wrote:

> What kind of phone service is this? If it's ISDN then you can compare
> d-channel traces between the Asterisk and FreeSWITCH instances and see if
> there are any differences. Otherwise I'll have to defer to Sangoma on this
> one.
>
> -MC
>
>
> On Tue, Jan 4, 2011 at 8:14 PM, Sam <u2nsam at gmail.com> wrote:
>
>> But when i do it from asterisk, it gives back the audio but not when i do
>> it from freeswitch with sangoma.
>> is there any method to listen to the audio send from the operator with
>> combination with sangoma?
>>
>> Regds
>>  Sam
>>
>>
>>
>> On Wed, Jan 5, 2011 at 12:00 AM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> You don't have any control over whether they send you audio, no? Either
>>> they send it or they don't. Unless there's something else that I'm
>>> missing...
>>> -MC
>>>
>>>
>>> On Mon, Jan 3, 2011 at 7:03 PM, Sam <u2nsam at gmail.com> wrote:
>>>
>>>> Some mobile operators give VAS as ringback tone / caller tune to be set
>>>> as mobile ringback,
>>>> now when dialing out such mobile users who has ringback enabled i just
>>>> get plain ring and no proceeding with 183 for the media to listen,
>>>> how to enable such so that some one dialing out will hear ringback
>>>> generated by the mobile operators.
>>>>
>>>> call flow is FS --> Sangoma --> mobile (ringback enabled by mobile
>>>> operator )
>>>>
>>>> Regards
>>>> Sam
>>>>
>>>>
>>>> On Tue, Jan 4, 2011 at 12:02 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>> Can you clarify what the actual issue is?
>>>>> -MC
>>>>>
>>>>> On Sun, Jan 2, 2011 at 8:30 PM, Sam <u2nsam at gmail.com> wrote:
>>>>>
>>>>>> Hi All,
>>>>>>
>>>>>> happy new you to you !
>>>>>>
>>>>>> using a sangoma card and when dialing a mobile number which is having
>>>>>> ringback tune / caller tune ;
>>>>>> but a plain ring is heard to the user dialing that mobile through the
>>>>>> trunk.
>>>>>>
>>>>>> I am using below syntax to dial out:-
>>>>>> <action application="bridge"
>>>>>> data="{monitor_early_media_ring_total=3}freetdm/wp1/a/${destination_number}"/>
>>>>>>
>>>>>> Any suggestions.
>>>>>>
>>>>>> Regards
>>>>>>  Sam
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From hwnorman at hotmail.com  Wed Jan  5 10:42:32 2011
From: hwnorman at hotmail.com (Norman Lam)
Date: Wed, 5 Jan 2011 15:42:32 +0800
Subject: [Freeswitch-users] Iksemel msvs compiling
Message-ID: <SNT109-DS13313A1FA906527438E8CC1090@phx.gbl>

Hi Everyone 

 

I am trying to compile the  freeswitch with mod_dingaling and Iksemel to
work on Google voice or Gmail voice, I have read  

 

This  http://wiki.freeswitch.org/wiki/Iksemel_MSVS_project_example and the
http://wiki.freeswitch.org/wiki/Dingaling

 

But I am stuck at  this  clause  

 

Add HAVE_GNUTLS=1 the the preprocessor compile in the iksemel project

 

What does this mean and how to go about this 

 

Thanks in advance

 

Norman Lam 

 

 

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From thisjoy0528 at gmail.com  Wed Jan  5 11:15:10 2011
From: thisjoy0528 at gmail.com (joy this)
Date: Wed, 5 Jan 2011 16:15:10 +0800
Subject: [Freeswitch-users] mod_conference with cluster ODBC
In-Reply-To: <AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com>
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705>
	<AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com>
Message-ID: <AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com>

Would you please explain how to bridge two conference in different servers
together? I tried the uuid_bridge and uuid_transfer, but it doesn't work. I
could only use "originate sofia/gateway/gateway_name/extension" to bridge,
but I don't want the dailing appear.

Beside, I want to know that how to bridge two conference in the same server
together. I could use "conference cof_name transfer" to bridge, but when I
bridge the members in conference B to conference A, the conference B will be
destroyed. So I can not transfer the original conference B members back.

Sincerely yours,
thisjoy.

2011/1/4 Jo?o Mesquita <jmesquita at freeswitch.org>

> They are indeed 2 completely different conferences. There's no
> implementation of making these 2 conferences bridge themselves
> automatically.
>
> The way I have solved this problem for now is have an ESL daemon
> "listening" on the conference creation events and bridging the 2 servers
> together when the one with the same name on the same domain is created. This
> solution might work for you and it's no too hard to implement.
>
> Regards,
> Jo?o Mesquita
>
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From moises.silva at gmail.com  Wed Jan  5 17:24:27 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 5 Jan 2011 09:24:27 -0500
Subject: [Freeswitch-users] no ringback tone
In-Reply-To: <AANLkTimi9iT+BxWP1y7BnvYLxysu9tSuzf+S08z1bp-L@mail.gmail.com>
References: <AANLkTinbd06YgZFWue=eQGG9A5JO2a550tJ+bo-yG2yv@mail.gmail.com>
	<AANLkTinA46HPKmk_pWaAiMpy-hdihc7STfY9_FuNkthx@mail.gmail.com>
	<AANLkTim6iJ6gFc1=21REAKkvz9z8ZgPK2ShHC7pBUpxH@mail.gmail.com>
	<AANLkTikxZZdV-D3c0B9yCN8jsdg4JASkxxJtevbkNA=M@mail.gmail.com>
	<AANLkTi=m4DvTCSCNfmfPjPwBDiYQwT+_7M_vEpCTXnOC@mail.gmail.com>
	<AANLkTim6OZAuLMBdTDmv_rmg1jcGC5VdD7RhsaEQW18r@mail.gmail.com>
	<AANLkTimi9iT+BxWP1y7BnvYLxysu9tSuzf+S08z1bp-L@mail.gmail.com>
Message-ID: <AANLkTintG=H+TnRtMfyTvg85NNMh=xjgS-rLToL+SO_g@mail.gmail.com>

On Wed, Jan 5, 2011 at 12:21 AM, Sam <u2nsam at gmail.com> wrote:

> I have attached the isdn pcap to this mail of FS --> Sangoma --> pri
>
>
Thanks Sam,

The problem is we're not moving to PROGRESS_MEDIA state. I confirmed that
the ISDN pcap has the progress indicator in the PROGRESS message, therefore
we should be moving to PROGRESS_MEDIA.

Are you working with latest libsng_isdn and freetdm?

Latest version for sure should be moving to PROGRESS_MEDIA.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com
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From rupa at rupa.com  Wed Jan  5 18:20:27 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 5 Jan 2011 09:20:27 -0600
Subject: [Freeswitch-users] mod_conference with cluster ODBC
In-Reply-To: <AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com>
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705>
	<AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com>
	<AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com>
Message-ID: <AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com>

Use the api: conference <confname> dial [{dial string
options}]<endpoint_module_name>/<destination> [<callerid_number>
[<callerid_name>]]

To initiate the call from within conference A on server 1.  Have a
corresponding dialplan entry on server 2 to accept the call and add it into
the conference A on server 2.  You've now bridged the two conferences in the
two servers.

On Wed, Jan 5, 2011 at 2:15 AM, joy this <thisjoy0528 at gmail.com> wrote:

> Would you please explain how to bridge two conference in different servers
> together? I tried the uuid_bridge and uuid_transfer, but it doesn't work. I
> could only use "originate sofia/gateway/gateway_name/extension" to bridge,
> but I don't want the dailing appear.
>
> Beside, I want to know that how to bridge two conference in the same server
> together. I could use "conference cof_name transfer" to bridge, but when I
> bridge the members in conference B to conference A, the conference B will be
> destroyed. So I can not transfer the original conference B members back.
>
> Sincerely yours,
> thisjoy.
>
> 2011/1/4 Jo?o Mesquita <jmesquita at freeswitch.org>
>
> They are indeed 2 completely different conferences. There's no
>> implementation of making these 2 conferences bridge themselves
>> automatically.
>>
>> The way I have solved this problem for now is have an ESL daemon
>> "listening" on the conference creation events and bridging the 2 servers
>> together when the one with the same name on the same domain is created. This
>> solution might work for you and it's no too hard to implement.
>>
>> Regards,
>> Jo?o Mesquita
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From steveayre at gmail.com  Wed Jan  5 18:26:54 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 5 Jan 2011 15:26:54 +0000
Subject: [Freeswitch-users] mod_conference with cluster ODBC
In-Reply-To: <AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com>
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705>
	<AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com>
	<AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com>
	<AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com>
Message-ID: <AANLkTin0bcKTr7zHstcxEzffRhu3MkdNwB1DFX-P+vaC@mail.gmail.com>

Out of interest rupa, how are conference commands such as relate
implemented during a conf-conf bridge? For instance when specifying
member A on FS1 should be heard by member B on FS2 but not by member C
on FS2? I'm guessing it's only possible on the same server, not across
a cluster?

-Steve


On 5 January 2011 15:20, Rupa Schomaker <rupa at rupa.com> wrote:
> Use the api:?conference <confname> dial [{dial string
> options}]<endpoint_module_name>/<destination> [<callerid_number>
> [<callerid_name>]]
> To initiate the call from within conference A on server 1. ?Have a
> corresponding dialplan entry on server 2 to accept the call and add it into
> the conference A on server 2. ?You've now bridged the two conferences in the
> two servers.
> On Wed, Jan 5, 2011 at 2:15 AM, joy this <thisjoy0528 at gmail.com> wrote:
>>
>> Would you please explain how to bridge two conference in different servers
>> together? I tried the uuid_bridge and uuid_transfer, but it doesn't work. I
>> could only use "originate sofia/gateway/gateway_name/extension" to bridge,
>> but I don't want the dailing appear.
>>
>> Beside, I want to know that how to bridge two conference in the same
>> server together. I could use "conference cof_name transfer" to bridge, but
>> when I bridge the members in conference B to conference A, the conference B
>> will be destroyed. So I can not transfer the original conference B members
>> back.
>>
>> Sincerely yours,
>> thisjoy.
>>
>> 2011/1/4 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>
>>> They are indeed 2 completely different conferences. There's no
>>> implementation of making these 2 conferences bridge themselves
>>> automatically.
>>> The way I have solved this problem for now is have an ESL daemon
>>> "listening" on the conference creation events and bridging the 2 servers
>>> together when the one with the same name on the same domain is created. This
>>> solution might work for you and it's no too hard to implement.
>>> Regards,
>>> Jo?o Mesquita
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From peter.olsson at visionutveckling.se  Wed Jan  5 18:35:32 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 5 Jan 2011 16:35:32 +0100
Subject: [Freeswitch-users] mod_conference with cluster ODBC
In-Reply-To: <AANLkTin0bcKTr7zHstcxEzffRhu3MkdNwB1DFX-P+vaC@mail.gmail.com>
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705>
	<AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com>
	<AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com>
	<AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com>
	<AANLkTin0bcKTr7zHstcxEzffRhu3MkdNwB1DFX-P+vaC@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57ECE32952@cooper>

Yes,

Since there is only one voice channel between the servers, if you relate to that channel, everything from the other server will follow.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Steven Ayre
Skickat: den 5 januari 2011 16:27
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] mod_conference with cluster ODBC

Out of interest rupa, how are conference commands such as relate
implemented during a conf-conf bridge? For instance when specifying
member A on FS1 should be heard by member B on FS2 but not by member C
on FS2? I'm guessing it's only possible on the same server, not across
a cluster?

-Steve


On 5 January 2011 15:20, Rupa Schomaker <rupa at rupa.com> wrote:
> Use the api:?conference <confname> dial [{dial string
> options}]<endpoint_module_name>/<destination> [<callerid_number>
> [<callerid_name>]]
> To initiate the call from within conference A on server 1. ?Have a
> corresponding dialplan entry on server 2 to accept the call and add it into
> the conference A on server 2. ?You've now bridged the two conferences in the
> two servers.
> On Wed, Jan 5, 2011 at 2:15 AM, joy this <thisjoy0528 at gmail.com> wrote:
>>
>> Would you please explain how to bridge two conference in different servers
>> together? I tried the uuid_bridge and uuid_transfer, but it doesn't work. I
>> could only use "originate sofia/gateway/gateway_name/extension" to bridge,
>> but I don't want the dailing appear.
>>
>> Beside, I want to know that how to bridge two conference in the same
>> server together. I could use "conference cof_name transfer" to bridge, but
>> when I bridge the members in conference B to conference A, the conference B
>> will be destroyed. So I can not transfer the original conference B members
>> back.
>>
>> Sincerely yours,
>> thisjoy.
>>
>> 2011/1/4 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>
>>> They are indeed 2 completely different conferences. There's no
>>> implementation of making these 2 conferences bridge themselves
>>> automatically.
>>> The way I have solved this problem for now is have an ESL daemon
>>> "listening" on the conference creation events and bridging the 2 servers
>>> together when the one with the same name on the same domain is created. This
>>> solution might work for you and it's no too hard to implement.
>>> Regards,
>>> Jo?o Mesquita
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4d248eeb32762038977453!



From u2nsam at gmail.com  Wed Jan  5 19:12:51 2011
From: u2nsam at gmail.com (Sam)
Date: Wed, 5 Jan 2011 21:42:51 +0530
Subject: [Freeswitch-users] no ringback tone
In-Reply-To: <AANLkTintG=H+TnRtMfyTvg85NNMh=xjgS-rLToL+SO_g@mail.gmail.com>
References: <AANLkTinbd06YgZFWue=eQGG9A5JO2a550tJ+bo-yG2yv@mail.gmail.com>
	<AANLkTinA46HPKmk_pWaAiMpy-hdihc7STfY9_FuNkthx@mail.gmail.com>
	<AANLkTim6iJ6gFc1=21REAKkvz9z8ZgPK2ShHC7pBUpxH@mail.gmail.com>
	<AANLkTikxZZdV-D3c0B9yCN8jsdg4JASkxxJtevbkNA=M@mail.gmail.com>
	<AANLkTi=m4DvTCSCNfmfPjPwBDiYQwT+_7M_vEpCTXnOC@mail.gmail.com>
	<AANLkTim6OZAuLMBdTDmv_rmg1jcGC5VdD7RhsaEQW18r@mail.gmail.com>
	<AANLkTimi9iT+BxWP1y7BnvYLxysu9tSuzf+S08z1bp-L@mail.gmail.com>
	<AANLkTintG=H+TnRtMfyTvg85NNMh=xjgS-rLToL+SO_g@mail.gmail.com>
Message-ID: <AANLkTimiaXnvi_d_m6TTy5JURjsKLwQPWhjRVyCr_qe+@mail.gmail.com>

Hi,

We are using libsng_isdn-7.0.0.x86_64 and latest freetdm, also there is 1
more problem you could see on the pcap, that we are sending callerid but its
not presented to the callee instead of that the callee gets the presentation
as the pilot number of pri which is 67287000.

Regds
Sam

On Wed, Jan 5, 2011 at 7:54 PM, Moises Silva <moises.silva at gmail.com> wrote:

> On Wed, Jan 5, 2011 at 12:21 AM, Sam <u2nsam at gmail.com> wrote:
>
>> I have attached the isdn pcap to this mail of FS --> Sangoma --> pri
>>
>>
> Thanks Sam,
>
> The problem is we're not moving to PROGRESS_MEDIA state. I confirmed that
> the ISDN pcap has the progress indicator in the PROGRESS message, therefore
> we should be moving to PROGRESS_MEDIA.
>
> Are you working with latest libsng_isdn and freetdm?
>
> Latest version for sure should be moving to PROGRESS_MEDIA.
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R
> 9R6 Canada
> t. 1 905 474 1990 x128 | e. moy at sangoma.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Wed Jan  5 19:27:39 2011
From: u2nsam at gmail.com (Sam)
Date: Wed, 5 Jan 2011 21:57:39 +0530
Subject: [Freeswitch-users] mod_conference with cluster ODBC
In-Reply-To: <AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com>
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705>
	<AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com>
	<AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com>
	<AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com>
Message-ID: <AANLkTinQUSr9pckOgCUwUZqf03HsgbzE6d5h=7LM8wi-@mail.gmail.com>

Thats interesting ... need to try it !

Regards
Sam

On Wed, Jan 5, 2011 at 8:50 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> Use the api: conference <confname> dial [{dial string
> options}]<endpoint_module_name>/<destination> [<callerid_number>
> [<callerid_name>]]
>
> To initiate the call from within conference A on server 1.  Have a
> corresponding dialplan entry on server 2 to accept the call and add it into
> the conference A on server 2.  You've now bridged the two conferences in the
> two servers.
>
> On Wed, Jan 5, 2011 at 2:15 AM, joy this <thisjoy0528 at gmail.com> wrote:
>
>> Would you please explain how to bridge two conference in different servers
>> together? I tried the uuid_bridge and uuid_transfer, but it doesn't work. I
>> could only use "originate sofia/gateway/gateway_name/extension" to bridge,
>> but I don't want the dailing appear.
>>
>> Beside, I want to know that how to bridge two conference in the same
>> server together. I could use "conference cof_name transfer" to bridge, but
>> when I bridge the members in conference B to conference A, the conference B
>> will be destroyed. So I can not transfer the original conference B members
>> back.
>>
>> Sincerely yours,
>> thisjoy.
>>
>> 2011/1/4 Jo?o Mesquita <jmesquita at freeswitch.org>
>>
>> They are indeed 2 completely different conferences. There's no
>>> implementation of making these 2 conferences bridge themselves
>>> automatically.
>>>
>>> The way I have solved this problem for now is have an ESL daemon
>>> "listening" on the conference creation events and bridging the 2 servers
>>> together when the one with the same name on the same domain is created. This
>>> solution might work for you and it's no too hard to implement.
>>>
>>> Regards,
>>> Jo?o Mesquita
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Wed Jan  5 19:37:17 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 5 Jan 2011 11:37:17 -0500
Subject: [Freeswitch-users] mod_conference with cluster ODBC
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705><AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com><AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com><AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com>
	<AANLkTinQUSr9pckOgCUwUZqf03HsgbzE6d5h=7LM8wi-@mail.gmail.com>
Message-ID: <A9992761D94F4A0DB96895A62FE1E03A@e1705>

> Use the api: conference <confname> dial [{dial string options}]<endpoint_module_name>/<destination> [<callerid_number> [<callerid_name>]]
so destination is the name of the conference on server 2 ?
  ----- Original Message ----- 
  From: Sam 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, January 05, 2011 11:27 AM
  Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


  Thats interesting ... need to try it !

  Regards
  Sam


  On Wed, Jan 5, 2011 at 8:50 PM, Rupa Schomaker <rupa at rupa.com> wrote:

    Use the api: conference <confname> dial [{dial string options}]<endpoint_module_name>/<destination> [<callerid_number> [<callerid_name>]]


    To initiate the call from within conference A on server 1.  Have a corresponding dialplan entry on server 2 to accept the call and add it into the conference A on server 2.  You've now bridged the two conferences in the two servers.


    On Wed, Jan 5, 2011 at 2:15 AM, joy this <thisjoy0528 at gmail.com> wrote:

      Would you please explain how to bridge two conference in different servers together? I tried the uuid_bridge and uuid_transfer, but it doesn't work. I could only use "originate sofia/gateway/gateway_name/extension" to bridge, but I don't want the dailing appear.

      Beside, I want to know that how to bridge two conference in the same server together. I could use "conference cof_name transfer" to bridge, but when I bridge the members in conference B to conference A, the conference B will be destroyed. So I can not transfer the original conference B members back. 


      Sincerely yours,
      thisjoy.



      2011/1/4 Jo?o Mesquita <jmesquita at freeswitch.org>


        They are indeed 2 completely different conferences. There's no implementation of making these 2 conferences bridge themselves automatically.


        The way I have solved this problem for now is have an ESL daemon "listening" on the conference creation events and bridging the 2 servers together when the one with the same name on the same domain is created. This solution might work for you and it's no too hard to implement.


        Regards,
        Jo?o Mesquita




      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org





    -- 
    -Rupa


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org






------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From peter.olsson at visionutveckling.se  Wed Jan  5 19:49:28 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 5 Jan 2011 17:49:28 +0100
Subject: [Freeswitch-users] mod_conference with cluster ODBC
In-Reply-To: <A9992761D94F4A0DB96895A62FE1E03A@e1705>
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705><AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com><AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com><AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com>
	<AANLkTinQUSr9pckOgCUwUZqf03HsgbzE6d5h=7LM8wi-@mail.gmail.com>,
	<A9992761D94F4A0DB96895A62FE1E03A@e1705>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57EC81C4E2@cooper>

The destination can be whatever, as long as you handle that destination in the dialplan on the destination FS server. It will end up as a normal call, and then be parsed by the dialplan.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Madovsky [infos at madovsky.org]
Skickat: den 5 januari 2011 17:37
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] mod_conference with cluster ODBC

> Use the api: conference <confname> dial [{dial string options}]<endpoint_module_name>/<destination> [<callerid_number> [<callerid_name>]]
so destination is the name of the conference on server 2 ?
----- Original Message -----
From: Sam<mailto:u2nsam at gmail.com>
To: FreeSWITCH Users Help<mailto:freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, January 05, 2011 11:27 AM
Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC

Thats interesting ... need to try it !

Regards
Sam

On Wed, Jan 5, 2011 at 8:50 PM, Rupa Schomaker <rupa at rupa.com<mailto:rupa at rupa.com>> wrote:
Use the api: conference <confname> dial [{dial string options}]<endpoint_module_name>/<destination> [<callerid_number> [<callerid_name>]]

To initiate the call from within conference A on server 1.  Have a corresponding dialplan entry on server 2 to accept the call and add it into the conference A on server 2.  You've now bridged the two conferences in the two servers.

On Wed, Jan 5, 2011 at 2:15 AM, joy this <thisjoy0528 at gmail.com<mailto:thisjoy0528 at gmail.com>> wrote:
Would you please explain how to bridge two conference in different servers together? I tried the uuid_bridge and uuid_transfer, but it doesn't work. I could only use "originate sofia/gateway/gateway_name/extension" to bridge, but I don't want the dailing appear.

Beside, I want to know that how to bridge two conference in the same server together. I could use "conference cof_name transfer" to bridge, but when I bridge the members in conference B to conference A, the conference B will be destroyed. So I can not transfer the original conference B members back.

Sincerely yours,
thisjoy.

2011/1/4 Jo?o Mesquita <jmesquita at freeswitch.org<mailto:jmesquita at freeswitch.org>>

They are indeed 2 completely different conferences. There's no implementation of making these 2 conferences bridge themselves automatically.

The way I have solved this problem for now is have an ESL daemon "listening" on the conference creation events and bridging the 2 servers together when the one with the same name on the same domain is created. This solution might work for you and it's no too hard to implement.

Regards,
Jo?o Mesquita


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
-Rupa

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



________________________________

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
!DSPAM:4d249f7b32761448239528!


From infos at madovsky.org  Wed Jan  5 19:56:06 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 5 Jan 2011 11:56:06 -0500
Subject: [Freeswitch-users] mod_conference with cluster ODBC
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705><AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com><AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com><AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com><AANLkTinQUSr9pckOgCUwUZqf03HsgbzE6d5h=7LM8wi-@mail.gmail.com>,
	<A9992761D94F4A0DB96895A62FE1E03A@e1705>
	<549CFEF87AEDE841A38E9D15EAB4C04C57EC81C4E2@cooper>
Message-ID: <9258F3F2AE8F4D4CBC81B5717487F2AC@e1705>

haaaa ok, understood :D

----- Original Message ----- 
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, January 05, 2011 11:49 AM
Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


The destination can be whatever, as long as you handle that destination in 
the dialplan on the destination FS server. It will end up as a normal call, 
and then be parsed by the dialplan.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org 
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Madovsky 
[infos at madovsky.org]
Skickat: den 5 januari 2011 17:37
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] mod_conference with cluster ODBC

> Use the api: conference <confname> dial [{dial string 
> options}]<endpoint_module_name>/<destination> [<callerid_number> 
> [<callerid_name>]]
so destination is the name of the conference on server 2 ?
----- Original Message -----
From: Sam<mailto:u2nsam at gmail.com>
To: FreeSWITCH Users Help<mailto:freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, January 05, 2011 11:27 AM
Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC

Thats interesting ... need to try it !

Regards
Sam

On Wed, Jan 5, 2011 at 8:50 PM, Rupa Schomaker 
<rupa at rupa.com<mailto:rupa at rupa.com>> wrote:
Use the api: conference <confname> dial [{dial string 
options}]<endpoint_module_name>/<destination> [<callerid_number> 
[<callerid_name>]]

To initiate the call from within conference A on server 1.  Have a 
corresponding dialplan entry on server 2 to accept the call and add it into 
the conference A on server 2.  You've now bridged the two conferences in the 
two servers.

On Wed, Jan 5, 2011 at 2:15 AM, joy this 
<thisjoy0528 at gmail.com<mailto:thisjoy0528 at gmail.com>> wrote:
Would you please explain how to bridge two conference in different servers 
together? I tried the uuid_bridge and uuid_transfer, but it doesn't work. I 
could only use "originate sofia/gateway/gateway_name/extension" to bridge, 
but I don't want the dailing appear.

Beside, I want to know that how to bridge two conference in the same server 
together. I could use "conference cof_name transfer" to bridge, but when I 
bridge the members in conference B to conference A, the conference B will be 
destroyed. So I can not transfer the original conference B members back.

Sincerely yours,
thisjoy.

2011/1/4 Jo?o Mesquita 
<jmesquita at freeswitch.org<mailto:jmesquita at freeswitch.org>>

They are indeed 2 completely different conferences. There's no 
implementation of making these 2 conferences bridge themselves 
automatically.

The way I have solved this problem for now is have an ESL daemon "listening" 
on the conference creation events and bridging the 2 servers together when 
the one with the same name on the same domain is created. This solution 
might work for you and it's no too hard to implement.

Regards,
Jo?o Mesquita


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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--
-Rupa

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From moises.silva at gmail.com  Wed Jan  5 20:48:05 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 5 Jan 2011 12:48:05 -0500
Subject: [Freeswitch-users] no ringback tone
In-Reply-To: <AANLkTimiaXnvi_d_m6TTy5JURjsKLwQPWhjRVyCr_qe+@mail.gmail.com>
References: <AANLkTinbd06YgZFWue=eQGG9A5JO2a550tJ+bo-yG2yv@mail.gmail.com>
	<AANLkTinA46HPKmk_pWaAiMpy-hdihc7STfY9_FuNkthx@mail.gmail.com>
	<AANLkTim6iJ6gFc1=21REAKkvz9z8ZgPK2ShHC7pBUpxH@mail.gmail.com>
	<AANLkTikxZZdV-D3c0B9yCN8jsdg4JASkxxJtevbkNA=M@mail.gmail.com>
	<AANLkTi=m4DvTCSCNfmfPjPwBDiYQwT+_7M_vEpCTXnOC@mail.gmail.com>
	<AANLkTim6OZAuLMBdTDmv_rmg1jcGC5VdD7RhsaEQW18r@mail.gmail.com>
	<AANLkTimi9iT+BxWP1y7BnvYLxysu9tSuzf+S08z1bp-L@mail.gmail.com>
	<AANLkTintG=H+TnRtMfyTvg85NNMh=xjgS-rLToL+SO_g@mail.gmail.com>
	<AANLkTimiaXnvi_d_m6TTy5JURjsKLwQPWhjRVyCr_qe+@mail.gmail.com>
Message-ID: <AANLkTimkAYJzXrf8--8iZqwiGdbdkj+4pbPjb1AQXvmu@mail.gmail.com>

On Wed, Jan 5, 2011 at 11:12 AM, Sam <u2nsam at gmail.com> wrote:

> Hi,
>
> We are using libsng_isdn-7.0.0.x86_64 and latest freetdm, also there is 1
> more problem you could see on the pcap, that we are sending callerid but its
> not presented to the callee instead of that the callee gets the presentation
> as the pilot number of pri which is 67287000.
>
>
The log you posted has line numbers that do not match latest FreeTDM. Please
update, there is a chance this was fixed in any of the multiple changes done
in the past 2 weeks that were just merged yesterday into FreeSWITCH
repository.

As for the caller id, you mean the number 7001? and the callee (for which I
don't have a trace here) receives 67287000? That seems to be your telco not
liking our caller id (or where we send it) and putting the default (your PRI
line number).

May be your telco wants the caller id coming in a facility message?


If you have a working pcap (if working from Asterisk, take the pcap when
dialing from Asterisk) to see how the caller id is being sent, that would be
useful.

Please open a jira ticket to keep working on your issue. (
http://jira.freeswitch.org/)

In the jira ticket add a debug log and matching pcap (by matching I mean the
debug log and the pcap must be for the same call), and do not cut the debug
log until the call is completely done.

Thanks!

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com
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From msc at freeswitch.org  Wed Jan  5 20:50:58 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Jan 2011 09:50:58 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <AANLkTi=+tEK2wiCYyM1RmKfYxOgWcpa_33VvsY+J5evG@mail.gmail.com>

Hey all!

Today's agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2011_01_05

We have a few things to discuss. Also, if DRK's schedule permits he will be
dropping by to talk more about his billing stuff! Hope to talk to you soon.

-Michael
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From fabio.bigliardi at gmail.com  Wed Jan  5 18:20:40 2011
From: fabio.bigliardi at gmail.com (Fabio Bigliardi)
Date: Wed, 5 Jan 2011 16:20:40 +0100
Subject: [Freeswitch-users] How to select audio from the initiator of a
 conference and play it to all participants
In-Reply-To: <AANLkTin7cGtdpNwm+x3p9Am099t+2ormWnOate1ieQ2n@mail.gmail.com>
References: <AANLkTin7cGtdpNwm+x3p9Am099t+2ormWnOate1ieQ2n@mail.gmail.com>
Message-ID: <AANLkTimvcKE0GgBfknzm1ENE-ketw2QiH7FS1PQrERPe@mail.gmail.com>

Hi all,
I would like to configure a dialplan so that the initiator of a
conference can select an audio file and play it simultaneously to all
the participants in the conference.

 I tried the following:

<extension name="my_mad_boss">
      <condition field="destination_number" expression="^0916$">
        <action application="set"
data="conference_auto_outcall_caller_id_name=my mad boss"/>
        <action application="set"
data="conference_auto_outcall_caller_id_number=0916"/>
        <action application="set" data="conference_auto_outcall_timeout=60"/>
        <action application="set" data="conference_auto_outcall_flags=mute"/>
        <action application="set"
data="conference_auto_outcall_prefix={sip_auto_answer=true}"/>
          <action application="set" data="sip_exclude_contact=${network_addr}"/>
        <action application="conference_set_auto_outcall"
data="${group_call(sales)}"/>
        <action application="bind_digit_action"
data="cool,11,exec:playback,ivr/ivr-welcome_to_freeswitch.wav"/>
        <action application="bind_digit_action"
data="cool,12,exec:playback,ivr/ivr-thank_you.wav"/>
        <action application="digit_action_set_realm" data="cool"/>
        <action application="conference"
data="my_mad_boss at default+flags{endconf}"/>
      </condition>
 </extension>

but it plays the right audio file only to the initiator of the conference.
How to have it played to all members of the conference instead?

Thank you in advance for your answer.

F. Bigliardi


From steveayre at gmail.com  Wed Jan  5 20:58:23 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 5 Jan 2011 17:58:23 +0000
Subject: [Freeswitch-users] How to select audio from the initiator of a
 conference and play it to all participants
In-Reply-To: <AANLkTimvcKE0GgBfknzm1ENE-ketw2QiH7FS1PQrERPe@mail.gmail.com>
References: <AANLkTin7cGtdpNwm+x3p9Am099t+2ormWnOate1ieQ2n@mail.gmail.com>
	<AANLkTimvcKE0GgBfknzm1ENE-ketw2QiH7FS1PQrERPe@mail.gmail.com>
Message-ID: <AANLkTikQUaptjs9k4OUUGOyHcpVERqcjdOjMSpmTxESJ@mail.gmail.com>

There's an API command for mod_conference to play a file to a conference.

http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference - see play.

-Steve

On 5 January 2011 15:20, Fabio Bigliardi <fabio.bigliardi at gmail.com> wrote:
> Hi all,
> I would like to configure a dialplan so that the initiator of a
> conference can select an audio file and play it simultaneously to all
> the participants in the conference.
>
> ?I tried the following:
>
> <extension name="my_mad_boss">
> ? ? ?<condition field="destination_number" expression="^0916$">
> ? ? ? ?<action application="set"
> data="conference_auto_outcall_caller_id_name=my mad boss"/>
> ? ? ? ?<action application="set"
> data="conference_auto_outcall_caller_id_number=0916"/>
> ? ? ? ?<action application="set" data="conference_auto_outcall_timeout=60"/>
> ? ? ? ?<action application="set" data="conference_auto_outcall_flags=mute"/>
> ? ? ? ?<action application="set"
> data="conference_auto_outcall_prefix={sip_auto_answer=true}"/>
> ? ? ? ? ?<action application="set" data="sip_exclude_contact=${network_addr}"/>
> ? ? ? ?<action application="conference_set_auto_outcall"
> data="${group_call(sales)}"/>
> ? ? ? ?<action application="bind_digit_action"
> data="cool,11,exec:playback,ivr/ivr-welcome_to_freeswitch.wav"/>
> ? ? ? ?<action application="bind_digit_action"
> data="cool,12,exec:playback,ivr/ivr-thank_you.wav"/>
> ? ? ? ?<action application="digit_action_set_realm" data="cool"/>
> ? ? ? ?<action application="conference"
> data="my_mad_boss at default+flags{endconf}"/>
> ? ? ?</condition>
> ?</extension>
>
> but it plays the right audio file only to the initiator of the conference.
> How to have it played to all members of the conference instead?
>
> Thank you in advance for your answer.
>
> F. Bigliardi
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From jgalaz at yx.cl  Wed Jan  5 23:30:27 2011
From: jgalaz at yx.cl (Javier Galaz Jeria)
Date: Wed, 5 Jan 2011 17:30:27 -0300
Subject: [Freeswitch-users] Memory using
Message-ID: <20110105203027.GB28848@jgalaz-desktop>

Hello all

I'm a new user of freeswitch and I've a few questions,

1. I've searched the mail list and some forums about the switch "waste" and I 
   haven't found out what it does and how it does it. Anyone care to explain a
   little bit about it? The only thing that I've found is that it consumes more
   memory but in my tests [1] it doesn't change that much.

2. Do any of you guys have a rough estimate of what each call memory footprint
   should be?

3. About the ulimit -s 240, I've read that it limits the stack size, but I'm at
   a lost point as how to pick a suitable value. What is the criteria?

4. Are the tests that I've done [1] reasonable?

I'm trying to know this things because I'm planning to use FreeSWITCH on a
resource limited hardware.

Best Regards

Javier

[1] test done with SIPp, using 0 to 20 users calling simultaneously, connecting
    the call, and hanging up after 5 secs.
    statistics using linear interpolation.
    using debian's init script (ulimit -s 240):
    slope	898.7 kB
    y-intercept	15410 kB
    using /usr/local/freeswitch/bin/freeswitch -waste -nc
    slope	810.6 kB
    y-intercept	16273 kB



From brian at freeswitch.org  Wed Jan  5 23:57:27 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 Jan 2011 14:57:27 -0600
Subject: [Freeswitch-users] Memory using
In-Reply-To: <20110105203027.GB28848@jgalaz-desktop>
References: <20110105203027.GB28848@jgalaz-desktop>
Message-ID: <1DFDA531-ABEF-4AA1-8EB6-0846007D48EE@freeswitch.org>

240 is the suitable value... Its telling you.

/b

On Jan 5, 2011, at 2:30 PM, Javier Galaz Jeria wrote:

> 3. About the ulimit -s 240, I've read that it limits the stack size, but I'm at
>   a lost point as how to pick a suitable value. What is the criteria?



From steveayre at gmail.com  Thu Jan  6 00:10:30 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 5 Jan 2011 21:10:30 +0000
Subject: [Freeswitch-users] Memory using
In-Reply-To: <20110105203027.GB28848@jgalaz-desktop>
References: <20110105203027.GB28848@jgalaz-desktop>
Message-ID: <DF9234E8-9136-4F48-914F-7A23F91DA9F8@gmail.com>

Answers inline

Steve on iPhone

On 5 Jan 2011, at 20:30, Javier Galaz Jeria <jgalaz at yx.cl> wrote:

> Hello all
> 
> I'm a new user of freeswitch and I've a few questions,
> 
> 1. I've searched the mail list and some forums about the switch "waste" and I 
>   haven't found out what it does and how it does it. Anyone care to explain a
>   little bit about it? The only thing that I've found is that it consumes more
>   memory but in my tests [1] it doesn't change that much.

Run on 64bit. You can then ignore it.

It's to do with virtual memory space on 32bit - each thread gets 8mb by default which can lead to out of memory errors, you can either shrink that to 240 or use waste to suppress a warning message.

It doesn't acrtually use any more real memory, just virtual address space.

You should use 64 bit if you can.

> 
> 2. Do any of you guys have a rough estimate of what each call memory footprint
>   should be?

That depends on what the call is doing.

> 
> 3. About the ulimit -s 240, I've read that it limits the stack size, but I'm at
>   a lost point as how to pick a suitable value. What is the criteria?

See 1. On 64 bit virtual memory space is so large this won't matter, just use the default (but 240 is still fine).


> 
> 4. Are the tests that I've done [1] reasonable?

It doesn't acrtually use any more real memory, just virtual address space.

> 
> I'm trying to know this things because I'm planning to use FreeSWITCH on a
> resource limited hardware.
> 
> Best Regards
> 
> Javier
> 
> [1] test done with SIPp, using 0 to 20 users calling simultaneously, connecting
>    the call, and hanging up after 5 secs.
>    statistics using linear interpolation.
>    using debian's init script (ulimit -s 240):
>    slope    898.7 kB
>    y-intercept    15410 kB
>    using /usr/local/freeswitch/bin/freeswitch -waste -nc
>    slope    810.6 kB
>    y-intercept    16273 kB
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From infos at madovsky.org  Thu Jan  6 00:22:30 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 5 Jan 2011 16:22:30 -0500
Subject: [Freeswitch-users] mod_conference with cluster ODBC
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705><AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com><AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com><AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com><AANLkTinQUSr9pckOgCUwUZqf03HsgbzE6d5h=7LM8wi-@mail.gmail.com>,
	<A9992761D94F4A0DB96895A62FE1E03A@e1705>
	<549CFEF87AEDE841A38E9D15EAB4C04C57EC81C4E2@cooper>
Message-ID: <23085FF5D4A847EAABF19E8DD8F357C4@e1705>

I really don't know how to detect with ESL if there is
a conference with same name on different nodes, it should be a common point
like DB or other no ?


----- Original Message ----- 
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, January 05, 2011 11:49 AM
Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


The destination can be whatever, as long as you handle that destination in 
the dialplan on the destination FS server. It will end up as a normal call, 
and then be parsed by the dialplan.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org 
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Madovsky 
[infos at madovsky.org]
Skickat: den 5 januari 2011 17:37
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] mod_conference with cluster ODBC

> Use the api: conference <confname> dial [{dial string 
> options}]<endpoint_module_name>/<destination> [<callerid_number> 
> [<callerid_name>]]
so destination is the name of the conference on server 2 ?
----- Original Message -----
From: Sam<mailto:u2nsam at gmail.com>
To: FreeSWITCH Users Help<mailto:freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, January 05, 2011 11:27 AM
Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC

Thats interesting ... need to try it !

Regards
Sam

On Wed, Jan 5, 2011 at 8:50 PM, Rupa Schomaker 
<rupa at rupa.com<mailto:rupa at rupa.com>> wrote:
Use the api: conference <confname> dial [{dial string 
options}]<endpoint_module_name>/<destination> [<callerid_number> 
[<callerid_name>]]

To initiate the call from within conference A on server 1.  Have a 
corresponding dialplan entry on server 2 to accept the call and add it into 
the conference A on server 2.  You've now bridged the two conferences in the 
two servers.

On Wed, Jan 5, 2011 at 2:15 AM, joy this 
<thisjoy0528 at gmail.com<mailto:thisjoy0528 at gmail.com>> wrote:
Would you please explain how to bridge two conference in different servers 
together? I tried the uuid_bridge and uuid_transfer, but it doesn't work. I 
could only use "originate sofia/gateway/gateway_name/extension" to bridge, 
but I don't want the dailing appear.

Beside, I want to know that how to bridge two conference in the same server 
together. I could use "conference cof_name transfer" to bridge, but when I 
bridge the members in conference B to conference A, the conference B will be 
destroyed. So I can not transfer the original conference B members back.

Sincerely yours,
thisjoy.

2011/1/4 Jo?o Mesquita 
<jmesquita at freeswitch.org<mailto:jmesquita at freeswitch.org>>

They are indeed 2 completely different conferences. There's no 
implementation of making these 2 conferences bridge themselves 
automatically.

The way I have solved this problem for now is have an ESL daemon "listening" 
on the conference creation events and bridging the 2 servers together when 
the one with the same name on the same domain is created. This solution 
might work for you and it's no too hard to implement.

Regards,
Jo?o Mesquita


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
-Rupa

_______________________________________________
FreeSWITCH-users mailing list
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________________________________

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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!DSPAM:4d249f7b32761448239528!

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 



From mattdfong at gmail.com  Thu Jan  6 02:16:22 2011
From: mattdfong at gmail.com (Matthew Fong)
Date: Wed, 5 Jan 2011 15:16:22 -0800
Subject: [Freeswitch-users] inline string of commands rather than phrase:
	for greet-long ivr attribute
Message-ID: <AANLkTi=4xWErgY0r=iEHrK0NicS6BKvSaE9Sm95ggqsA@mail.gmail.com>

I am wondering if it is possible to use an inline style string of commands
like playback:play1.wav say:hello playback:play2.wav in the greet-long xml
ivr attribute rather than having to create a phrase macro. If it is possible
can I please get an example. thanks

--matt
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From jmesquita at freeswitch.org  Thu Jan  6 02:20:20 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 5 Jan 2011 20:20:20 -0300
Subject: [Freeswitch-users] mod_conference with cluster ODBC
In-Reply-To: <23085FF5D4A847EAABF19E8DD8F357C4@e1705>
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705>
	<AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com>
	<AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com>
	<AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com>
	<AANLkTinQUSr9pckOgCUwUZqf03HsgbzE6d5h=7LM8wi-@mail.gmail.com>
	<A9992761D94F4A0DB96895A62FE1E03A@e1705>
	<549CFEF87AEDE841A38E9D15EAB4C04C57EC81C4E2@cooper>
	<23085FF5D4A847EAABF19E8DD8F357C4@e1705>
Message-ID: <AANLkTik_1nf4p7H6-m_r+vpx1dEfTSJJCeMvJOovL=BS@mail.gmail.com>

Have you ever used esl? You can use the Action: add-member header in
conjunction with the Conference-Size: 1 header to know that a conference has
been created. Based on that, it is pretty straightforward.

Regards,
Jo?o Mesquita



On Wed, Jan 5, 2011 at 6:22 PM, Madovsky <infos at madovsky.org> wrote:

> I really don't know how to detect with ESL if there is
> a conference with same name on different nodes, it should be a common point
> like DB or other no ?
>
>
> ----- Original Message -----
> From: "Peter Olsson" <peter.olsson at visionutveckling.se>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, January 05, 2011 11:49 AM
> Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
>
>
> The destination can be whatever, as long as you handle that destination in
> the dialplan on the destination FS server. It will end up as a normal call,
> and then be parsed by the dialplan.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Madovsky
> [infos at madovsky.org]
> Skickat: den 5 januari 2011 17:37
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] mod_conference with cluster ODBC
>
> > Use the api: conference <confname> dial [{dial string
> > options}]<endpoint_module_name>/<destination> [<callerid_number>
> > [<callerid_name>]]
> so destination is the name of the conference on server 2 ?
> ----- Original Message -----
> From: Sam<mailto:u2nsam at gmail.com>
> To: FreeSWITCH Users Help<mailto:freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, January 05, 2011 11:27 AM
> Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
>
> Thats interesting ... need to try it !
>
> Regards
> Sam
>
> On Wed, Jan 5, 2011 at 8:50 PM, Rupa Schomaker
> <rupa at rupa.com<mailto:rupa at rupa.com>> wrote:
> Use the api: conference <confname> dial [{dial string
> options}]<endpoint_module_name>/<destination> [<callerid_number>
> [<callerid_name>]]
>
> To initiate the call from within conference A on server 1.  Have a
> corresponding dialplan entry on server 2 to accept the call and add it into
> the conference A on server 2.  You've now bridged the two conferences in
> the
> two servers.
>
> On Wed, Jan 5, 2011 at 2:15 AM, joy this
> <thisjoy0528 at gmail.com<mailto:thisjoy0528 at gmail.com>> wrote:
> Would you please explain how to bridge two conference in different servers
> together? I tried the uuid_bridge and uuid_transfer, but it doesn't work. I
> could only use "originate sofia/gateway/gateway_name/extension" to bridge,
> but I don't want the dailing appear.
>
> Beside, I want to know that how to bridge two conference in the same server
> together. I could use "conference cof_name transfer" to bridge, but when I
> bridge the members in conference B to conference A, the conference B will
> be
> destroyed. So I can not transfer the original conference B members back.
>
> Sincerely yours,
> thisjoy.
>
> 2011/1/4 Jo?o Mesquita
> <jmesquita at freeswitch.org<mailto:jmesquita at freeswitch.org>>
>
> They are indeed 2 completely different conferences. There's no
> implementation of making these 2 conferences bridge themselves
> automatically.
>
> The way I have solved this problem for now is have an ESL daemon
> "listening"
> on the conference creation events and bridging the 2 servers together when
> the one with the same name on the same domain is created. This solution
> might work for you and it's no too hard to implement.
>
> Regards,
> Jo?o Mesquita
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ________________________________
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> !DSPAM:4d249f7b32761448239528!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From infos at madovsky.org  Thu Jan  6 02:25:15 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 5 Jan 2011 18:25:15 -0500
Subject: [Freeswitch-users] mod_conference with cluster ODBC
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705><AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com><AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com><AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com><AANLkTinQUSr9pckOgCUwUZqf03HsgbzE6d5h=7LM8wi-@mail.gmail.com><A9992761D94F4A0DB96895A62FE1E03A@e1705><549CFEF87AEDE841A38E9D15EAB4C04C57EC81C4E2@cooper><23085FF5D4A847EAABF19E8DD8F357C4@e1705>
	<AANLkTik_1nf4p7H6-m_r+vpx1dEfTSJJCeMvJOovL=BS@mail.gmail.com>
Message-ID: <4D914F8428BB498F942E413F28F04BDF@e1705>

Not, never, it's like a chimere for me ;)
ok I will try

Obrigado
  ----- Original Message ----- 
  From: Jo?o Mesquita 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, January 05, 2011 6:20 PM
  Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


  Have you ever used esl? You can use the Action: add-member header in conjunction with the Conference-Size: 1 header to know that a conference has been created. Based on that, it is pretty straightforward.


  Regards,
  Jo?o Mesquita




  On Wed, Jan 5, 2011 at 6:22 PM, Madovsky <infos at madovsky.org> wrote:

    I really don't know how to detect with ESL if there is
    a conference with same name on different nodes, it should be a common point
    like DB or other no ?



    ----- Original Message -----
    From: "Peter Olsson" <peter.olsson at visionutveckling.se>

    To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
    Sent: Wednesday, January 05, 2011 11:49 AM
    Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


    The destination can be whatever, as long as you handle that destination in
    the dialplan on the destination FS server. It will end up as a normal call,
    and then be parsed by the dialplan.

    /Peter
    ________________________________________
    Fr?n: freeswitch-users-bounces at lists.freeswitch.org
    [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Madovsky
    [infos at madovsky.org]
    Skickat: den 5 januari 2011 17:37
    Till: FreeSWITCH Users Help
    ?mne: Re: [Freeswitch-users] mod_conference with cluster ODBC

    > Use the api: conference <confname> dial [{dial string
    > options}]<endpoint_module_name>/<destination> [<callerid_number>
    > [<callerid_name>]]
    so destination is the name of the conference on server 2 ?
    ----- Original Message -----
    From: Sam<mailto:u2nsam at gmail.com>
    To: FreeSWITCH Users Help<mailto:freeswitch-users at lists.freeswitch.org>
    Sent: Wednesday, January 05, 2011 11:27 AM
    Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC

    Thats interesting ... need to try it !

    Regards
    Sam

    On Wed, Jan 5, 2011 at 8:50 PM, Rupa Schomaker
    <rupa at rupa.com<mailto:rupa at rupa.com>> wrote:
    Use the api: conference <confname> dial [{dial string
    options}]<endpoint_module_name>/<destination> [<callerid_number>
    [<callerid_name>]]

    To initiate the call from within conference A on server 1.  Have a
    corresponding dialplan entry on server 2 to accept the call and add it into
    the conference A on server 2.  You've now bridged the two conferences in the
    two servers.

    On Wed, Jan 5, 2011 at 2:15 AM, joy this
    <thisjoy0528 at gmail.com<mailto:thisjoy0528 at gmail.com>> wrote:
    Would you please explain how to bridge two conference in different servers
    together? I tried the uuid_bridge and uuid_transfer, but it doesn't work. I
    could only use "originate sofia/gateway/gateway_name/extension" to bridge,
    but I don't want the dailing appear.

    Beside, I want to know that how to bridge two conference in the same server
    together. I could use "conference cof_name transfer" to bridge, but when I
    bridge the members in conference B to conference A, the conference B will be
    destroyed. So I can not transfer the original conference B members back.

    Sincerely yours,
    thisjoy.

    2011/1/4 Jo?o Mesquita
    <jmesquita at freeswitch.org<mailto:jmesquita at freeswitch.org>>

    They are indeed 2 completely different conferences. There's no
    implementation of making these 2 conferences bridge themselves
    automatically.

    The way I have solved this problem for now is have an ESL daemon "listening"
    on the conference creation events and bridging the 2 servers together when
    the one with the same name on the same domain is created. This solution
    might work for you and it's no too hard to implement.

    Regards,
    Jo?o Mesquita


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org




    --
    -Rupa

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org



    ________________________________

    _______________________________________________
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    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org
    !DSPAM:4d249f7b32761448239528!

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





------------------------------------------------------------------------------


  _______________________________________________
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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From msc at freeswitch.org  Thu Jan  6 03:09:17 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Jan 2011 16:09:17 -0800
Subject: [Freeswitch-users] inline string of commands rather than
 phrase: for greet-long ivr attribute
In-Reply-To: <AANLkTi=4xWErgY0r=iEHrK0NicS6BKvSaE9Sm95ggqsA@mail.gmail.com>
References: <AANLkTi=4xWErgY0r=iEHrK0NicS6BKvSaE9Sm95ggqsA@mail.gmail.com>
Message-ID: <AANLkTi=mho3LjsrY6WVVm_CUBhdt1Uba1SsNwEaSjq1Z@mail.gmail.com>

Yes:
greet-long="file_string://${sound_prefix}/ivr/8000/ivr-good_afternoon.wav!${sound_prefix}/ivr/8000/ivr-generic_greeting.wav"

Just make sure that mod_file_string is installed. Also, read the wiki entry
on mod_file_string, particularly on when to use absolute path names!

-MC

On Wed, Jan 5, 2011 at 3:16 PM, Matthew Fong <mattdfong at gmail.com> wrote:

> I am wondering if it is possible to use an inline style string of commands
> like playback:play1.wav say:hello playback:play2.wav in the greet-long xml
> ivr attribute rather than having to create a phrase macro. If it is possible
> can I please get an example. thanks
>
> --matt
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From larry at geekmax.org  Thu Jan  6 03:56:00 2011
From: larry at geekmax.org (Larry Wimble)
Date: Wed, 05 Jan 2011 19:56:00 -0500
Subject: [Freeswitch-users] Voicepulse destination_number
Message-ID: <4D251320.4000100@geekmax.org>


Greetings Freeswitch gurus from a FS Newbie...

I have a standard voicepulse account (4 channels) with 3 DIDs.  In 
setting up Freeswitch for use with this account today, I was unable to 
make it route based on the dialed number.

The relevant dialplan is this:

<extension name="VP-IN-TEMP" >
<condition field="context" expression="public"/>
<condition field="destination_number" expression="ZZZZZZZZZZZ">
<action application="ivr" data="DEMO"/>
</condition>
</extension>

When I set ZZZZZZZZZZZ to the number I'm dialing in on, it does NOT work.
When I set ZZZZZZZZZZZ to my voicepulse login (from the credentials 
page), it works fine.

It seems Voicepulse is transmitting my login ID where Freeswitch expects 
to see the dialed number.

When I dial in, this appears in the log:
2011-01-05 19:51:50.672153 [INFO] mod_dialplan_xml.c:331 Processing Cell 
Phone   FL <8135551212>->ZZZZZZZZZZZ in context public

...where ZZZZZZZZZZZ is my voicepulse login ID.

The problem with this is that I can't route based on the number dialed.  
How do I find out what number was dialed?

Any ideas?

TIA,
Larry Wimble



From brian at freeswitch.org  Thu Jan  6 05:22:26 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 Jan 2011 20:22:26 -0600
Subject: [Freeswitch-users] Voicepulse destination_number
In-Reply-To: <4D251320.4000100@geekmax.org>
References: <4D251320.4000100@geekmax.org>
Message-ID: <EF6788C7-6D1C-489A-9E86-433770FAC987@freeswitch.org>

Bet you if you replace destination_number with ${sip_to_user} it works fine.  And if you also set extension=auto_to_user on your gateway it will also do this for you.

/b

On Jan 5, 2011, at 6:56 PM, Larry Wimble wrote:

> <extension name="VP-IN-TEMP" >
> <condition field="context" expression="public"/>
> <condition field="destination_number" expression="ZZZZZZZZZZZ">
> <action application="ivr" data="DEMO"/>
> </condition>
> </extension>



From jeff at jefflenk.com  Thu Jan  6 05:26:06 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 5 Jan 2011 18:26:06 -0800 (PST)
Subject: [Freeswitch-users] Can't build on Windows 7 64bit C++ Express!?
In-Reply-To: <AANLkTimkeKAR+qrZ827y-1H_2jczQSPLhUR_qXQP=me2@mail.gmail.com>
References: <snt0-eas12908A012C361DEE1DF0DC7B9070@phx.gbl>
	<AANLkTimkeKAR+qrZ827y-1H_2jczQSPLhUR_qXQP=me2@mail.gmail.com>
Message-ID: <1294280766814-5894372.post@n2.nabble.com>


Follow the wiki - look for autocrlf problems with git - Peter already
mentioned this.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Re-Can-t-build-on-Windows-7-64bit-C-Express-tp5885019p5894372.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jeff at jefflenk.com  Thu Jan  6 05:31:17 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 5 Jan 2011 18:31:17 -0800 (PST)
Subject: [Freeswitch-users] Iksemel msvs compiling
In-Reply-To: <SNT109-DS13313A1FA906527438E8CC1090@phx.gbl>
References: <SNT109-DS13313A1FA906527438E8CC1090@phx.gbl>
Message-ID: <1294281077009-5894380.post@n2.nabble.com>


I corrected a small typo on that page. Just look at the preprocessor section
on the sample images shown.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Iksemel-msvs-compiling-tp5891263p5894380.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From lloyd.aloysius at gmail.com  Thu Jan  6 06:25:38 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Wed, 5 Jan 2011 22:25:38 -0500
Subject: [Freeswitch-users] sofia status profile internal - Output in
 Table view
In-Reply-To: <AANLkTin1R+5QvWp1YG5cOexjvHu5+qfB=zpC0_f1S1W4@mail.gmail.com>
References: <AANLkTikPhufZ0T80+9Wj9rS5nZcPqZ=2xZKRNsD9--KL@mail.gmail.com>
	<AANLkTin1R+5QvWp1YG5cOexjvHu5+qfB=zpC0_f1S1W4@mail.gmail.com>
Message-ID: <AANLkTinQ1AxUiMNWBhu9gnJ6RBtovcsL6+NFQx-_0ywj@mail.gmail.com>

Thanks Anthony.

How to format the output like "sip show peers"

Thanks
Lloyd


On Tue, Jan 4, 2011 at 12:33 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> there is "sofia xmlstatus profile internal"
> then you can format it any way you want.
>
>
> On Tue, Jan 4, 2011 at 10:26 AM, Aloysius Lloyd
> <lloyd.aloysius at gmail.com> wrote:
> > Hi All,
> > Is there any way to get the output from sofia status profile internal in
> a
> > table view.
> > Something like Asterisk "sip show peers"
> >
> > Thanks
> > Lloyd
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mattdfong at gmail.com  Thu Jan  6 10:26:23 2011
From: mattdfong at gmail.com (Matthew Fong)
Date: Wed, 5 Jan 2011 23:26:23 -0800
Subject: [Freeswitch-users] inline string of commands rather than
 phrase: for greet-long ivr attribute
In-Reply-To: <AANLkTi=mho3LjsrY6WVVm_CUBhdt1Uba1SsNwEaSjq1Z@mail.gmail.com>
References: <AANLkTi=4xWErgY0r=iEHrK0NicS6BKvSaE9Sm95ggqsA@mail.gmail.com>
	<AANLkTi=mho3LjsrY6WVVm_CUBhdt1Uba1SsNwEaSjq1Z@mail.gmail.com>
Message-ID: <AANLkTinR+hHr1VhPDc8-H48bMS8uB5VSRF-h=ZJbSLAK@mail.gmail.com>

is it also possible to combine tts like cepstral commands in this way? or
must they all be sound files?

On Wed, Jan 5, 2011 at 4:09 PM, Michael Collins <msc at freeswitch.org> wrote:

> Yes:
>
> greet-long="file_string://${sound_prefix}/ivr/8000/ivr-good_afternoon.wav!${sound_prefix}/ivr/8000/ivr-generic_greeting.wav"
>
> Just make sure that mod_file_string is installed. Also, read the wiki entry
> on mod_file_string, particularly on when to use absolute path names!
>
> -MC
>
> On Wed, Jan 5, 2011 at 3:16 PM, Matthew Fong <mattdfong at gmail.com> wrote:
>
>> I am wondering if it is possible to use an inline style string of commands
>> like playback:play1.wav say:hello playback:play2.wav in the greet-long xml
>> ivr attribute rather than having to create a phrase macro. If it is possible
>> can I please get an example. thanks
>>
>> --matt
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From thisjoy0528 at gmail.com  Thu Jan  6 10:51:17 2011
From: thisjoy0528 at gmail.com (joy this)
Date: Thu, 6 Jan 2011 15:51:17 +0800
Subject: [Freeswitch-users] mod_conference with cluster ODBC
In-Reply-To: <AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com>
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705>
	<AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com>
	<AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com>
	<AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com>
Message-ID: <AANLkTimutwOyeubcS=tax5EbDEniCD+Nc4iHMydQxpFb@mail.gmail.com>

It works. Thank you everyone.

2011/1/5 Rupa Schomaker <rupa at rupa.com>

> Use the api: conference <confname> dial [{dial string
> options}]<endpoint_module_name>/<destination> [<callerid_number>
> [<callerid_name>]]
>
> To initiate the call from within conference A on server 1.  Have a
> corresponding dialplan entry on server 2 to accept the call and add it into
> the conference A on server 2.  You've now bridged the two conferences in the
> two servers.
>
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From u2nsam at gmail.com  Thu Jan  6 11:14:40 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 6 Jan 2011 13:44:40 +0530
Subject: [Freeswitch-users] no ringback tone
In-Reply-To: <AANLkTimkAYJzXrf8--8iZqwiGdbdkj+4pbPjb1AQXvmu@mail.gmail.com>
References: <AANLkTinbd06YgZFWue=eQGG9A5JO2a550tJ+bo-yG2yv@mail.gmail.com>
	<AANLkTinA46HPKmk_pWaAiMpy-hdihc7STfY9_FuNkthx@mail.gmail.com>
	<AANLkTim6iJ6gFc1=21REAKkvz9z8ZgPK2ShHC7pBUpxH@mail.gmail.com>
	<AANLkTikxZZdV-D3c0B9yCN8jsdg4JASkxxJtevbkNA=M@mail.gmail.com>
	<AANLkTi=m4DvTCSCNfmfPjPwBDiYQwT+_7M_vEpCTXnOC@mail.gmail.com>
	<AANLkTim6OZAuLMBdTDmv_rmg1jcGC5VdD7RhsaEQW18r@mail.gmail.com>
	<AANLkTimi9iT+BxWP1y7BnvYLxysu9tSuzf+S08z1bp-L@mail.gmail.com>
	<AANLkTintG=H+TnRtMfyTvg85NNMh=xjgS-rLToL+SO_g@mail.gmail.com>
	<AANLkTimiaXnvi_d_m6TTy5JURjsKLwQPWhjRVyCr_qe+@mail.gmail.com>
	<AANLkTimkAYJzXrf8--8iZqwiGdbdkj+4pbPjb1AQXvmu@mail.gmail.com>
Message-ID: <AANLkTin4Jw4sgZqw6d+bzRxhZW4=0PJCsRKsshrGuQok@mail.gmail.com>

Tried installing the latest git-4272dcb 2011-01-05 20-12-19 -0600 it giving
the sangoma codec error while instaling.

Regards
Sam




On Wed, Jan 5, 2011 at 11:18 PM, Moises Silva <moises.silva at gmail.com>wrote:

>
> On Wed, Jan 5, 2011 at 11:12 AM, Sam <u2nsam at gmail.com> wrote:
>
>> Hi,
>>
>> We are using libsng_isdn-7.0.0.x86_64 and latest freetdm, also there is 1
>> more problem you could see on the pcap, that we are sending callerid but its
>> not presented to the callee instead of that the callee gets the presentation
>> as the pilot number of pri which is 67287000.
>>
>>
> The log you posted has line numbers that do not match latest FreeTDM.
> Please update, there is a chance this was fixed in any of the multiple
> changes done in the past 2 weeks that were just merged yesterday into
> FreeSWITCH repository.
>
> As for the caller id, you mean the number 7001? and the callee (for which I
> don't have a trace here) receives 67287000? That seems to be your telco not
> liking our caller id (or where we send it) and putting the default (your PRI
> line number).
>
> May be your telco wants the caller id coming in a facility message?
>
>
> If you have a working pcap (if working from Asterisk, take the pcap when
> dialing from Asterisk) to see how the caller id is being sent, that would be
> useful.
>
> Please open a jira ticket to keep working on your issue. (
> http://jira.freeswitch.org/)
>
> In the jira ticket add a debug log and matching pcap (by matching I mean
> the debug log and the pcap must be for the same call), and do not cut the
> debug log until the call is completely done.
>
> Thanks!
>
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R
> 9R6 Canada
> t. 1 905 474 1990 x128 | e. moy at sangoma.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Thu Jan  6 11:25:33 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 6 Jan 2011 13:55:33 +0530
Subject: [Freeswitch-users] old git
Message-ID: <AANLkTinfBQqv3i+PWAbH8SgiF0WWTEUCQ+86v=H8DJZG@mail.gmail.com>

Hi

how can i download old git , if i want to download old git of git-34a0ca5
2010-12-24 20-38-57 -0600

Regards
Sam
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From freeswitch at tlainvestments.com  Thu Jan  6 11:49:16 2011
From: freeswitch at tlainvestments.com (Troy Anderson)
Date: Thu, 6 Jan 2011 01:49:16 -0700
Subject: [Freeswitch-users] old git
In-Reply-To: <AANLkTinfBQqv3i+PWAbH8SgiF0WWTEUCQ+86v=H8DJZG@mail.gmail.com>
References: <AANLkTinfBQqv3i+PWAbH8SgiF0WWTEUCQ+86v=H8DJZG@mail.gmail.com>
Message-ID: <4D9ED7B2-28C7-4D6E-964B-0FEA6F0E2375@tlainvestments.com>

From within the freeswitch directory:

git checkout 34a0ca5

Once you've done so, to prove it to yourself, type:

git log

To get back on the mainline:

git checkout master

-Troy


On Jan 6, 2011, at 1:25 AM, Sam wrote:

> Hi
> 
> how can i download old git , if i want to download old git of git-34a0ca5 2010-12-24 20-38-57 -0600
> 
> Regards
> Sam
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From wstephen80 at gmail.com  Thu Jan  6 12:13:23 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Thu, 6 Jan 2011 10:13:23 +0100
Subject: [Freeswitch-users] Error in log with latest FreeTDM
Message-ID: <AANLkTik+pv3C3svaCRDUBE4P8OYGHj30tJ+q-1bqO9E2@mail.gmail.com>

I have tried to update to the latest git my Freeswitch but I have due to
revert to the previous version because with latest one
(96ac90adce931a3a28c768e102b863637c8ba98d, Jan 5 16:55:06 2011 + Sangoma
ISDN library 7.0.0) I have many warning errors and some critical errors in
the log.
No problem with my previous version
(715d250e171a94736b19019ac742f739899ad997, Dec 15 21:29:52 2010 + Sangoma
ISDN library 6.0.0).

Here an extraction of my log file:

2011-01-06 02:01:25.990974 [WARNING] mod_freetdm.c:434 [s23c22][23:22] Why
bother changing state from PROCEED to PROCEED
2011-01-06 02:01:25.994018 [CRIT] ftmod_sangoma_isdn_stack_rcv.c:988
sng_isdn->

date: 01/06/2011 time: 02:01:25
2011-01-06 02:01:25.994018 [CRIT] ftmod_sangoma_isdn_stack_rcv.c:988
sng_isdn->
mtss(posix): sw error:  ent: 010  inst: 000  proc id: 001
file: /usr/src/libsng_trillium-build/libsng_isdn/trillium/in/in_bdy1.c line:
2144 errcode: 46909632820939 errcls: ERRCLS_DEBUG
errval: 00001  errdesc: InUiIntConReq() Failed.
2011-01-06 02:01:25.994018 [WARNING] ftmod_sangoma_isdn_stack_hndl.c:862
[s10c1][10:1] STATUS CONFIRM (call_state:0 channel-state:DIALING cause:100)
(suId:1 suInstId:57 spInstId:0)
2011-01-06 02:01:31.068159 [WARNING] mod_freetdm.c:434 [s22c9][22:9] Why
bother changing state from PROCEED to PROCEED
2011-01-06 02:01:31.172106 [WARNING] ftdm_io.c:2250 [s22c9][22:9] Cannot
indicate PROGRESS_MEDIA in channel with indication PROCEED still pending in
state PROCEED
2011-01-06 02:01:37.465176 [WARNING] mod_freetdm.c:434 [s3c28][3:28] Why
bother changing state from PROCEED to PROCEED
2011-01-06 02:01:37.472130 [WARNING] ftdm_io.c:2250 [s3c28][3:28] Cannot
indicate RINGING in channel with indication PROCEED still pending in state
PROCEED
2011-01-06 02:01:43.705155 [WARNING] mod_freetdm.c:434 [s23c23][23:23] Why
bother changing state from PROCEED to PROCEED
2011-01-06 02:01:43.707124 [CRIT] ftmod_sangoma_isdn_stack_rcv.c:988
sng_isdn->

date: 01/06/2011 time: 02:01:43
2011-01-06 02:01:43.707124 [CRIT] ftmod_sangoma_isdn_stack_rcv.c:988
sng_isdn->
mtss(posix): sw error:  ent: 010  inst: 000  proc id: 001
file: /usr/src/libsng_trillium-build/libsng_isdn/trillium/in/in_bdy1.c line:
2144 errcode: 46909632820939 errcls: ERRCLS_DEBUG
errval: 00001  errdesc: InUiIntConReq() Failed.
2011-01-06 02:01:43.707124 [WARNING] ftmod_sangoma_isdn_stack_hndl.c:862
[s5c1][5:1] STATUS CONFIRM (call_state:0 channel-state:DIALING cause:100)
(suId:1 suInstId:61 spInstId:0)
2011-01-06 02:01:49.463158 [WARNING] mod_freetdm.c:434 [s16c14][16:14] Why
bother changing state from PROCEED to PROCEED
2011-01-06 02:01:49.470188 [WARNING] ftdm_io.c:2250 [s16c14][16:14] Cannot
indicate RINGING in channel with indication PROCEED still pending in state
PROCEED
2011-01-06 02:01:53.938264 [WARNING] mod_freetdm.c:434 [s1c1][1:1] Why
bother changing state from PROCEED to PROCEED
2011-01-06 02:01:53.944202 [WARNING] ftdm_io.c:2250 [s1c1][1:1] Cannot
indicate RINGING in channel with indication PROCEED still pending in state
PROCEED
2011-01-06 02:01:59.468295 [WARNING] mod_freetdm.c:434 [s8c3][8:3] Why
bother changing state from PROCEED to PROCEED
2011-01-06 02:01:59.476270 [WARNING] ftdm_io.c:2250 [s8c3][8:3] Cannot
indicate RINGING in channel with indication PROCEED still pending in state
PROCEED
2011-01-06 02:02:01.057245 [WARNING] mod_freetdm.c:434 [s1c2][1:2] Why
bother changing state from PROCEED to PROCEED
2011-01-06 02:02:01.083313 [WARNING] ftdm_io.c:2250 [s1c2][1:2] Cannot
indicate PROGRESS_MEDIA in channel with indication PROCEED still pending in
state PROCEED
2011-01-06 02:02:04.286328 [WARNING] mod_freetdm.c:434 [s20c24][20:24] Why
bother changing state from PROCEED to PROCEED
2011-01-06 02:02:14.525323 [WARNING] mod_freetdm.c:434 [s17c23][17:23] Why
bother changing state from PROCEED to PROCEED
2011-01-06 02:02:15.029458 [WARNING] ftdm_io.c:2250 [s17c23][17:23] Cannot
indicate PROGRESS_MEDIA in channel with indication PROCEED still pending in
state PROCEED
2011-01-06 02:02:19.789411 [WARNING] mod_freetdm.c:434 [s12c5][12:5] Why
bother changing state from PROCEED to PROCEED
2011-01-06 02:02:19.891438 [WARNING] ftdm_io.c:2250 [s12c5][12:5] Cannot
indicate PROGRESS_MEDIA in channel with indication PROCEED still pending in
state PROCEED
2011-01-06 02:02:23.764507 [WARNING] mod_freetdm.c:434 [s22c10][22:10] Why
bother changing state from PROCEED to PROCEED
2011-01-06 02:02:26.462538 [WARNING] mod_freetdm.c:434 [s19c24][19:24] Why
bother changing state from PROCEED to PROCEED
2011-01-06 02:02:29.171509 [WARNING] ftdm_io.c:2250 [s22c10][22:10] Cannot
indicate PROGRESS_MEDIA in channel with indication PROCEED still pending in
state PROCEED
2011-01-06 02:02:31.698471 [WARNING] mod_freetdm.c:434 [s23c24][23:24] Why
bother changing state from PROCEED to PROCEED
2011-01-06 02:02:31.724623 [WARNING] ftdm_io.c:2250 [s23c24][23:24] Cannot
indicate PROGRESS_MEDIA in channel with indication PROCEED still pending in
state PROCEED
2011-01-06 02:02:44.469700 [WARNING] mod_freetdm.c:434 [s6c21][6:21] Why
bother changing state from PROCEED to PROCEED
2011-01-06 02:02:44.476688 [WARNING] ftdm_io.c:2250 [s6c21][6:21] Cannot
indicate RINGING in channel with indication PROCEED still pending in state
PROCEED
2011-01-06 02:02:48.018716 [WARNING] mod_freetdm.c:434 [s20c25][20:25] Why
bother changing state from PROCEED to PROCEED
2011-01-06 02:02:48.046713 [WARNING] ftdm_io.c:2250 [s20c25][20:25] Cannot
indicate PROGRESS_MEDIA in channel with indication PROCEED still pending in
state PROCEED
2011-01-06 02:02:49.463708 [WARNING] mod_freetdm.c:434 [s8c4][8:4] Why
bother changing state from PROCEED to PROCEED
2011-01-06 02:02:54.462848 [WARNING] mod_freetdm.c:434 [s19c25][19:25] Why
bother changing state from PROCEED to PROCEED
2011-01-06 02:02:54.468754 [CRIT] ftmod_sangoma_isdn_stack_rcv.c:988
sng_isdn->

date: 01/06/2011 time: 02:02:54
2011-01-06 02:02:54.468754 [CRIT] ftmod_sangoma_isdn_stack_rcv.c:988
sng_isdn->
mtss(posix): sw error:  ent: 010  inst: 000  proc id: 001
file: /usr/src/libsng_trillium-build/libsng_isdn/trillium/in/in_bdy1.c line:
2144 errcode: 46909632820939 errcls: ERRCLS_DEBUG
errval: 00001  errdesc: InUiIntConReq() Failed.
2011-01-06 02:02:54.468754 [WARNING] ftmod_sangoma_isdn_stack_hndl.c:862
[s10c1][10:1] STATUS CONFIRM (call_state:0 channel-state:DIALING cause:100)
(suId:1 suInstId:81 spInstId:0)
2011-01-06 02:02:55.305075 [WARNING] ftdm_io.c:2250 [s8c4][8:4] Cannot
indicate PROGRESS_MEDIA in channel with indication PROCEED still pending in
state PROCEED
2011-01-06 02:02:57.785073 [WARNING] ftdm_io.c:2250 [s19c25][19:25] Cannot
indicate PROGRESS_MEDIA in channel with indication PROCEED still pending in
state PROCEED
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From u2nsam at gmail.com  Thu Jan  6 12:37:51 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 6 Jan 2011 15:07:51 +0530
Subject: [Freeswitch-users] Error in log with latest FreeTDM
In-Reply-To: <AANLkTik+pv3C3svaCRDUBE4P8OYGHj30tJ+q-1bqO9E2@mail.gmail.com>
References: <AANLkTik+pv3C3svaCRDUBE4P8OYGHj30tJ+q-1bqO9E2@mail.gmail.com>
Message-ID: <AANLkTikABrSrKJ6brR=eQCe0F0+F25W08A9pxO=SYj8b@mail.gmail.com>

For me to while compiling....


This was the error while installing latest git-4272dcb 2011-01-05 20-12-19
-0600







making all mod_sangoma_codec

Compiling
/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c...

quiet_libtool: compile:  gcc -I/usr/local/src/freeswitch/src/include
-I/usr/local/src/freeswitch/src/include
-I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
-g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE
-DHAVE_CONFIG_H -c
/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c
-fPIC -DPIC -o .libs/mod_sangoma_codec.o

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:37:31:
error: sng_tc/sngtc_node.h: No such file or directory

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:50:
error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?g_init_cfg?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:88:
error: ?SNGTC_CODEC_PCMU? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:88:
error: ?IANA_PCMU_A_8000_1? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:89:
error: ?SNGTC_CODEC_PCMA? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:89:
error: ?IANA_PCMA_A_8000_1? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:90:
error: ?SNGTC_CODEC_L16_1? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:90:
error: ?IANA_L16_A_8000_1? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:91:
error: ?SNGTC_CODEC_L16_2? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:91:
error: ?IANA_L16_A_16000_1? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:92:
error: ?SNGTC_CODEC_G729AB? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:92:
error: ?IANA_G729_AB_8000_1? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:93:
error: ?SNGTC_CODEC_G726_32? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:93:
error: ?IANA_G726_32_8000_1? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:94:
error: ?SNGTC_CODEC_G722? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:94:
error: ?IANA_G722_A_8000_1? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:97:
error: ?SNGTC_CODEC_GSM_FR? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:97:
error: ?IANA_GSM_A_8000_1? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:98:
error: ?SNGTC_CODEC_G723_1_63? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:98:
error: ?IANA_G723_A_8000_1? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:99:
error: ?SNGTC_CODEC_AMR_1220? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:99:
error: ?IANA_AMR_WB_16000_1? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:100:
error: ?SNGTC_CODEC_SIREN7_24? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:100:
error: ?IANA_SIREN7? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:101:
error: ?SNGTC_CODEC_SIREN7_32? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:102:
error: ?SNGTC_CODEC_ILBC_133? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:102:
error: ?IANA_ILBC_133_8000_1? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:103:
error: ?SNGTC_CODEC_ILBC_152? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:103:
error: ?IANA_ILBC_152_8000_1? undeclared here (not in a function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:124:
error: expected specifier-qualifier-list before ?sngtc_codec_request_t?

cc1: warnings being treated as errors

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:154:
error: struct has no members

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:
In function ?sangoma_create_rtp_port?:

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:224:
error: implicit declaration of function ?SNGTC_NIPV4?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:224:
error: too few arguments for format

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:
At top level:

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:230:
error: expected declaration specifiers or ?...? before
?sngtc_codec_request_leg_t?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:230:
error: expected declaration specifiers or ?...? before
?sngtc_codec_reply_leg_t?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:
In function ?sangoma_create_rtp?:

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:258:
error: ?codec_req_leg? undeclared (first use in this function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:258:
error: (Each undeclared identifier is reported only once

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:258:
error: for each function it appears in.)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:258:
error: ?vocallo_codec_t? has no member named ?host_udp_port?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:258:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:260:
error: ?vocallo_codec_t? has no member named ?host_ip?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:260:
error: initialization makes integer from pointer without a cast

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:262:
error: implicit declaration of function ?sngtc_codec_ipv4_hex_to_str?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:262:
error: ?codec_reply_leg? undeclared (first use in this function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:262:
error: ?vocallo_codec_t? has no member named ?codec_ip?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:266:
error: ?vocallo_codec_t? has no member named ?codec_udp_port?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:267:
error: ?vocallo_codec_t? has no member named ?ms?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:267:
error: invalid operands to binary * (have ?struct vocallo_codec_t *? and
?int?)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:267:
error: format ?%d? expects type ?int?, but argument 11 has type ?struct
vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:267:
error: format ?%d? expects type ?int?, but argument 14 has type ?struct
vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:271:
error: ?vocallo_codec_t? has no member named ?codec_udp_port?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:274:
error: ?vocallo_codec_t? has no member named ?ms?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:274:
error: invalid operands to binary * (have ?struct vocallo_codec_t *? and
?int?)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:275:
error: passing argument 4 of ?switch_rtp_new? makes integer from pointer
without a cast

/usr/local/src/freeswitch/src/include/switch_rtp.h:154: note: expected
?switch_port_t? but argument is of type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:275:
error: passing argument 7 of ?switch_rtp_new? makes integer from pointer
without a cast

/usr/local/src/freeswitch/src/include/switch_rtp.h:154: note: expected
?uint32_t? but argument is of type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:
In function ?switch_sangoma_init?:

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:318:
error: ?struct codec_data? has no member named ?lastrxseqno?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:318:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:319:
error: ?struct codec_data? has no member named ?lastrxseqno?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:319:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:329:
error: ?struct codec_data? has no member named ?request?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:329:
error: request for member ?usr_priv? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:329:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:330:
error: ?struct codec_data? has no member named ?request?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:330:
error: request for member ?a? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:330:
error: request for member ?host_ip? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:330:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:331:
error: ?struct codec_data? has no member named ?request?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:331:
error: request for member ?a? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:331:
error: request for member ?codec_id? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:332:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:333:
error: ?struct codec_data? has no member named ?request?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:333:
error: request for member ?a? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:333:
error: request for member ?ms? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:333:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:335:
error: ?struct codec_data? has no member named ?request?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:335:
error: request for member ?b? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:335:
error: request for member ?host_ip? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:335:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:336:
error: ?struct codec_data? has no member named ?request?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:336:
error: request for member ?b? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:336:
error: request for member ?codec_id? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:336:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:337:
error: ?struct codec_data? has no member named ?request?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:337:
error: request for member ?b? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:337:
error: request for member ?ms? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:337:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:341:
error: ?struct codec_data? has no member named ?request?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:341:
error: request for member ?usr_priv? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:341:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:342:
error: ?struct codec_data? has no member named ?request?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:342:
error: request for member ?a? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:342:
error: request for member ?host_ip? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:342:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:343:
error: ?struct codec_data? has no member named ?request?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:343:
error: request for member ?a? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:343:
error: request for member ?codec_id? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:343:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:344:
error: ?struct codec_data? has no member named ?request?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:344:
error: request for member ?a? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:344:
error: request for member ?ms? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:344:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:346:
error: ?struct codec_data? has no member named ?request?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:346:
error: request for member ?b? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:346:
error: request for member ?host_ip? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:346:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:347:
error: ?struct codec_data? has no member named ?request?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:347:
error: request for member ?b? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:347:
error: request for member ?codec_id? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:348:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:349:
error: ?struct codec_data? has no member named ?request?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:349:
error: request for member ?b? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:349:
error: request for member ?ms? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:349:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:
In function ?switch_sangoma_encode?:

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:446:
error: initialization makes integer from pointer without a cast

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:449:
error: ?struct codec_data? has no member named ?debug_timing?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:457:
error: ?struct codec_data? has no member named ?txrtp?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:460:
error: implicit declaration of function ?sngtc_create_transcoding_session?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:460:
error: ?struct codec_data? has no member named ?request?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:460:
error: ?struct codec_data? has no member named ?reply?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:467:
error: ?struct codec_data? has no member named ?txrtp?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:467:
error: ?struct codec_data? has no member named ?reply?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:467:
error: request for member ?tx_fd? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:467:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:468:
error: ?struct codec_data? has no member named ?rxrtp?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:468:
error: ?struct codec_data? has no member named ?reply?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:468:
error: request for member ?rx_fd? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:468:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:470:
error: ?struct codec_data? has no member named ?rxrtp?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:470:
error: passing argument 1 of ?flush_rtp? from incompatible pointer type

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:398:
note: expected ?struct switch_rtp_t *? but argument is of type ?struct
vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:473:
error: ?struct codec_data? has no member named ?debug_timing?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:473:
error: ?struct codec_data? has no member named ?last_func_call_time?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:474:
error: ?struct codec_data? has no member named ?last_func_call_time?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:474:
error: invalid operands to binary - (have ?switch_time_t? and ?struct
vocallo_codec_t *?)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:474:
error: assignment makes integer from pointer without a cast

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:493:
error: ?struct codec_data? has no member named ?txrtp?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:493:
error: passing argument 1 of ?switch_rtp_write_frame? from incompatible
pointer type

/usr/local/src/freeswitch/src/include/switch_rtp.h:403: note: expected
?struct switch_rtp_t *? but argument is of type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:504:
error: ?struct codec_data? has no member named ?tx?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:504:
error: lvalue required as increment operand

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:504:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:511:
error: ?struct codec_data? has no member named ?rxrtp?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:511:
error: passing argument 1 of ?switch_rtp_zerocopy_read_frame? from
incompatible pointer type

/usr/local/src/freeswitch/src/include/switch_rtp.h:366: note: expected
?struct switch_rtp_t *? but argument is of type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:540:
error: ?struct codec_data? has no member named ?request?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:540:
error: request for member ?b? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:540:
error: request for member ?codec_id? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:540:
error: ?struct codec_data? has no member named ?request?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:540:
error: request for member ?b? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:540:
error: request for member ?codec_id? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:543:
error: comparison between pointer and integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:543:
error: comparison between pointer and integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:545:
error: format ?%d? expects type ?int?, but argument 9 has type ?struct
vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:545:
error: format ?%d? expects type ?int?, but argument 10 has type ?struct
vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:555:
error: ?struct codec_data? has no member named ?queue_windex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:555:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:556:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:556:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:556:
error: array subscript is not an integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:556:
error: request for member ?datalen? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:558:
error: ?struct codec_data? has no member named ?rxdiscarded?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:558:
error: lvalue required as increment operand

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:558:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:560:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:560:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:560:
error: array subscript is not an integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:560:
error: request for member ?datalen? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:560:
error: ?struct codec_data? has no member named ?queue_windex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:560:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:560:
error: format ?%d? expects type ?int?, but argument 8 has type ?struct
vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:560:
error: format ?%d? expects type ?int?, but argument 10 has type ?struct
vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:560:
error: format ?%d? expects type ?int?, but argument 11 has type ?struct
vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:561:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:561:
error: lvalue required as increment operand

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:561:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:561:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:561:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:561:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:561:
error: invalid operands to binary / (have ?struct vocallo_codec_t *? and
?long unsigned int?)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:561:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:561:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:562:
error: ?struct codec_data? has no member named ?queue_size?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:562:
error: lvalue required as decrement operand

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:562:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:566:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:566:
error: ?struct codec_data? has no member named ?queue_windex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:566:
error: array subscript is not an integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:566:
error: request for member ?data? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:567:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:567:
error: ?struct codec_data? has no member named ?queue_windex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:567:
error: array subscript is not an integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:567:
error: request for member ?datalen? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:567:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:568:
error: ?struct codec_data? has no member named ?queue_windex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:568:
error: lvalue required as increment operand

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:568:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:568:
error: ?struct codec_data? has no member named ?queue_windex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:568:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:568:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:568:
error: invalid operands to binary / (have ?struct vocallo_codec_t *? and
?long unsigned int?)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:568:
error: ?struct codec_data? has no member named ?queue_windex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:568:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:571:
error: ?struct codec_data? has no member named ?queue_size?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:571:
error: lvalue required as increment operand

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:571:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:572:
error: ?struct codec_data? has no member named ?queue_size?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:572:
error: ?struct codec_data? has no member named ?queue_max_ever?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:573:
error: ?struct codec_data? has no member named ?queue_size?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:574:
error: ?struct codec_data? has no member named ?queue_windex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:574:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:574:
error: format ?%d? expects type ?int?, but argument 9 has type ?struct
vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:574:
error: format ?%d? expects type ?int?, but argument 10 has type ?struct
vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:574:
error: format ?%d? expects type ?int?, but argument 11 has type ?struct
vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:575:
error: ?struct codec_data? has no member named ?queue_max_ever?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:575:
error: ?struct codec_data? has no member named ?queue_size?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:575:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:580:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:580:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:580:
error: array subscript is not an integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:580:
error: request for member ?datalen? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:581:
error: ?struct codec_data? has no member named ?rx?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:581:
error: lvalue required as increment operand

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:581:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:583:
error: ?struct codec_data? has no member named ?last_rx_time?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:584:
error: ?struct codec_data? has no member named ?last_rx_time?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:584:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:586:
error: ?struct codec_data? has no member named ?last_rx_time?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:586:
error: invalid operands to binary - (have ?switch_time_t? and ?struct
vocallo_codec_t *?)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:586:
error: assignment makes integer from pointer without a cast

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:587:
error: ?struct codec_data? has no member named ?avgrxus?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:587:
error: ?struct codec_data? has no member named ?avgrxus?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:587:
error: ?struct codec_data? has no member named ?avgrxus?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:587:
error: invalid operands to binary / (have ?struct vocallo_codec_t *? and
?int?)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:587:
error: pointer/integer type mismatch in conditional expression

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:587:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:588:
error: ?struct codec_data? has no member named ?last_rx_time?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:588:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:592:
error: ?struct codec_data? has no member named ?lastrxseqno?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:592:
error: ordered comparison of pointer with integer zero

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:593:
error: ?struct codec_data? has no member named ?lastrxseqno?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:593:
error: comparison between pointer and integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:594:
error: ?struct codec_data? has no member named ?rxlost?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:594:
error: ?struct codec_data? has no member named ?lastrxseqno?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:594:
error: invalid operands to binary - (have ?int? and ?struct vocallo_codec_t
*?)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:594:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:597:
error: ?struct codec_data? has no member named ?lastrxseqno?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:597:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:600:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:600:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:600:
error: array subscript is not an integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:600:
error: request for member ?datalen? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:600:
error: assignment makes integer from pointer without a cast

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:601:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:601:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:601:
error: array subscript is not an integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:601:
error: request for member ?data? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:602:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:602:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:602:
error: array subscript is not an integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:602:
error: request for member ?datalen? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:602:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:603:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:603:
error: lvalue required as increment operand

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:603:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:603:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:603:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:603:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:603:
error: invalid operands to binary / (have ?struct vocallo_codec_t *? and
?long unsigned int?)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:603:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:603:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:604:
error: ?struct codec_data? has no member named ?queue_size?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:604:
error: lvalue required as decrement operand

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:604:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:612:
error: ?struct codec_data? has no member named ?debug_timing?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:618:
error: ?struct codec_data? has no member named ?last_func_call_time?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:618:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:
In function ?switch_sangoma_decode?:

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:645:
error: initialization makes integer from pointer without a cast

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:648:
error: ?struct codec_data? has no member named ?debug_timing?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:664:
error: ?struct codec_data? has no member named ?txrtp?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:667:
error: ?struct codec_data? has no member named ?request?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:667:
error: ?struct codec_data? has no member named ?reply?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:674:
error: ?struct codec_data? has no member named ?txrtp?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:674:
error: ?struct codec_data? has no member named ?reply?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:674:
error: request for member ?tx_fd? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:674:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:675:
error: ?struct codec_data? has no member named ?rxrtp?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:675:
error: ?struct codec_data? has no member named ?reply?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:675:
error: request for member ?rx_fd? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:675:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:677:
error: ?struct codec_data? has no member named ?rxrtp?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:677:
error: passing argument 1 of ?flush_rtp? from incompatible pointer type

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:398:
note: expected ?struct switch_rtp_t *? but argument is of type ?struct
vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:680:
error: ?struct codec_data? has no member named ?debug_timing?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:680:
error: ?struct codec_data? has no member named ?last_func_call_time?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:681:
error: ?struct codec_data? has no member named ?last_func_call_time?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:681:
error: invalid operands to binary - (have ?switch_time_t? and ?struct
vocallo_codec_t *?)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:681:
error: assignment makes integer from pointer without a cast

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:694:
error: ?struct codec_data? has no member named ?txrtp?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:694:
error: passing argument 1 of ?switch_rtp_write_frame? from incompatible
pointer type

/usr/local/src/freeswitch/src/include/switch_rtp.h:403: note: expected
?struct switch_rtp_t *? but argument is of type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:706:
error: ?struct codec_data? has no member named ?tx?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:706:
error: lvalue required as increment operand

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:706:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:713:
error: ?struct codec_data? has no member named ?rxrtp?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:713:
error: passing argument 1 of ?switch_rtp_zerocopy_read_frame? from
incompatible pointer type

/usr/local/src/freeswitch/src/include/switch_rtp.h:366: note: expected
?struct switch_rtp_t *? but argument is of type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:742:
error: ?struct codec_data? has no member named ?queue_windex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:742:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:743:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:743:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:743:
error: array subscript is not an integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:743:
error: request for member ?datalen? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:745:
error: ?struct codec_data? has no member named ?rxdiscarded?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:745:
error: lvalue required as increment operand

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:745:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:747:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:747:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:747:
error: array subscript is not an integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:747:
error: request for member ?datalen? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:747:
error: ?struct codec_data? has no member named ?queue_windex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:747:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:747:
error: format ?%d? expects type ?int?, but argument 8 has type ?struct
vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:747:
error: format ?%d? expects type ?int?, but argument 10 has type ?struct
vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:747:
error: format ?%d? expects type ?int?, but argument 11 has type ?struct
vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:748:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:748:
error: lvalue required as increment operand

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:748:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:748:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:748:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:748:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:748:
error: invalid operands to binary / (have ?struct vocallo_codec_t *? and
?long unsigned int?)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:748:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:748:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:749:
error: ?struct codec_data? has no member named ?queue_size?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:749:
error: lvalue required as decrement operand

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:749:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:754:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:754:
error: ?struct codec_data? has no member named ?queue_windex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:754:
error: array subscript is not an integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:754:
error: request for member ?data? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:759:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:759:
error: ?struct codec_data? has no member named ?queue_windex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:759:
error: array subscript is not an integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:759:
error: request for member ?datalen? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:759:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:761:
error: ?struct codec_data? has no member named ?queue_windex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:761:
error: lvalue required as increment operand

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:761:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:761:
error: ?struct codec_data? has no member named ?queue_windex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:761:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:761:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:761:
error: invalid operands to binary / (have ?struct vocallo_codec_t *? and
?long unsigned int?)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:761:
error: ?struct codec_data? has no member named ?queue_windex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:761:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:764:
error: ?struct codec_data? has no member named ?queue_size?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:764:
error: lvalue required as increment operand

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:764:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:765:
error: ?struct codec_data? has no member named ?queue_size?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:765:
error: ?struct codec_data? has no member named ?queue_max_ever?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:766:
error: ?struct codec_data? has no member named ?queue_size?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:767:
error: ?struct codec_data? has no member named ?queue_windex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:767:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:767:
error: format ?%d? expects type ?int?, but argument 9 has type ?struct
vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:767:
error: format ?%d? expects type ?int?, but argument 10 has type ?struct
vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:767:
error: format ?%d? expects type ?int?, but argument 11 has type ?struct
vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:768:
error: ?struct codec_data? has no member named ?queue_max_ever?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:768:
error: ?struct codec_data? has no member named ?queue_size?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:768:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:772:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:772:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:772:
error: array subscript is not an integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:772:
error: request for member ?datalen? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:774:
error: ?struct codec_data? has no member named ?rx?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:774:
error: lvalue required as increment operand

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:774:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:777:
error: ?struct codec_data? has no member named ?last_rx_time?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:778:
error: ?struct codec_data? has no member named ?last_rx_time?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:778:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:780:
error: ?struct codec_data? has no member named ?last_rx_time?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:780:
error: invalid operands to binary - (have ?switch_time_t? and ?struct
vocallo_codec_t *?)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:780:
error: assignment makes integer from pointer without a cast

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:781:
error: ?struct codec_data? has no member named ?avgrxus?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:781:
error: ?struct codec_data? has no member named ?avgrxus?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:781:
error: ?struct codec_data? has no member named ?avgrxus?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:781:
error: invalid operands to binary / (have ?struct vocallo_codec_t *? and
?int?)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:781:
error: pointer/integer type mismatch in conditional expression

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:781:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:782:
error: ?struct codec_data? has no member named ?last_rx_time?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:782:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:786:
error: ?struct codec_data? has no member named ?lastrxseqno?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:786:
error: ordered comparison of pointer with integer zero

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:787:
error: ?struct codec_data? has no member named ?lastrxseqno?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:787:
error: comparison between pointer and integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:788:
error: ?struct codec_data? has no member named ?rxlost?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:788:
error: ?struct codec_data? has no member named ?lastrxseqno?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:788:
error: invalid operands to binary - (have ?int? and ?struct vocallo_codec_t
*?)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:788:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:791:
error: ?struct codec_data? has no member named ?lastrxseqno?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:791:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:794:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:794:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:794:
error: array subscript is not an integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:794:
error: request for member ?data? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:794:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:794:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:794:
error: array subscript is not an integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:794:
error: request for member ?datalen? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:794:
error: passing argument 3 of ?memcpy? makes integer from pointer without a
cast

/usr/include/string.h:43: note: expected ?size_t? but argument is of type
?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:795:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:795:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:795:
error: array subscript is not an integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:795:
error: request for member ?datalen? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:795:
error: assignment makes integer from pointer without a cast

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:796:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:796:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:796:
error: array subscript is not an integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:796:
error: request for member ?datalen? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:796:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:797:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:797:
error: lvalue required as increment operand

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:797:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:797:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:797:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:797:
error: ?struct codec_data? has no member named ?rtp_queue?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:797:
error: invalid operands to binary / (have ?struct vocallo_codec_t *? and
?long unsigned int?)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:797:
error: ?struct codec_data? has no member named ?queue_rindex?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:797:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:798:
error: ?struct codec_data? has no member named ?queue_size?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:798:
error: lvalue required as decrement operand

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:798:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:808:
error: ?struct codec_data? has no member named ?debug_timing?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:814:
error: ?struct codec_data? has no member named ?last_func_call_time?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:814:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:
In function ?switch_sangoma_destroy?:

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:829:
error: ?struct codec_data? has no member named ?txrtp?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:830:
error: implicit declaration of function ?sngtc_free_transcoding_session?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:830:
error: ?struct codec_data? has no member named ?reply?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:833:
error: ?struct codec_data? has no member named ?txrtp?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:834:
error: ?struct codec_data? has no member named ?reply?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:
In function ?sangoma_function?:

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:918:
error: ?struct codec_data? has no member named ?tx?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:918:
error: format ?%lu? expects type ?long unsigned int?, but argument 4 has
type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:918:
error: format ?%lu? expects type ?long unsigned int?, but argument 4 has
type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:919:
error: ?struct codec_data? has no member named ?rx?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:919:
error: format ?%lu? expects type ?long unsigned int?, but argument 4 has
type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:919:
error: format ?%lu? expects type ?long unsigned int?, but argument 4 has
type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:920:
error: ?struct codec_data? has no member named ?tx?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:920:
error: format ?%lu? expects type ?long unsigned int?, but argument 4 has
type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:920:
error: format ?%lu? expects type ?long unsigned int?, but argument 4 has
type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:921:
error: ?struct codec_data? has no member named ?rx?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:921:
error: format ?%lu? expects type ?long unsigned int?, but argument 4 has
type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:921:
error: format ?%lu? expects type ?long unsigned int?, but argument 4 has
type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:922:
error: ?struct codec_data? has no member named ?rxlost?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:922:
error: format ?%lu? expects type ?long unsigned int?, but argument 4 has
type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:922:
error: format ?%lu? expects type ?long unsigned int?, but argument 4 has
type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:923:
error: ?struct codec_data? has no member named ?rxlost?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:923:
error: format ?%lu? expects type ?long unsigned int?, but argument 4 has
type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:923:
error: format ?%lu? expects type ?long unsigned int?, but argument 4 has
type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:924:
error: ?struct codec_data? has no member named ?avgrxus?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:924:
error: invalid operands to binary / (have ?struct vocallo_codec_t *? and
?int?)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:925:
error: ?struct codec_data? has no member named ?avgrxus?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:925:
error: invalid operands to binary / (have ?struct vocallo_codec_t *? and
?int?)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:931:
error: ?struct codec_data? has no member named ?txrtp?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:932:
error: ?struct codec_data? has no member named ?txrtp?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:966:
error: ?struct codec_data? has no member named ?rxrtp?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:967:
error: ?struct codec_data? has no member named ?rxrtp?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:967:
error: passing argument 1 of ?switch_rtp_get_stats? from incompatible
pointer type

/usr/local/src/freeswitch/src/include/switch_rtp.h:459: note: expected
?struct switch_rtp_t *? but argument is of type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:970:
error: ?struct codec_data? has no member named ?reply?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:970:
error: request for member ?a? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:970:
error: request for member ?host_ip? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:970:
error: ?struct codec_data? has no member named ?reply?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:970:
error: request for member ?a? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:970:
error: request for member ?host_udp_port? in something not a structure or
union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:971:
error: ?struct codec_data? has no member named ?reply?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:971:
error: request for member ?a? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:971:
error: request for member ?codec_ip? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:971:
error: ?struct codec_data? has no member named ?reply?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:971:
error: request for member ?a? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:971:
error: request for member ?codec_udp_port? in something not a structure or
union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:972:
error: ?struct codec_data? has no member named ?reply?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:972:
error: request for member ?b? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:972:
error: request for member ?host_ip? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:972:
error: ?struct codec_data? has no member named ?reply?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:972:
error: request for member ?b? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:972:
error: request for member ?host_udp_port? in something not a structure or
union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:973:
error: ?struct codec_data? has no member named ?reply?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:973:
error: request for member ?b? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:973:
error: request for member ?codec_ip? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:973:
error: ?struct codec_data? has no member named ?reply?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:973:
error: request for member ?b? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:973:
error: request for member ?codec_udp_port? in something not a structure or
union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:977:
error: ?struct codec_data? has no member named ?rxdiscarded?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:981:
error: ?struct codec_data? has no member named ?txrtp?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:981:
error: passing argument 1 of ?switch_rtp_get_stats? from incompatible
pointer type

/usr/local/src/freeswitch/src/include/switch_rtp.h:459: note: expected
?struct switch_rtp_t *? but argument is of type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:986:
error: ?struct codec_data? has no member named ?rxrtp?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:987:
error: ?struct codec_data? has no member named ?rxrtp?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:987:
error: passing argument 1 of ?switch_rtp_get_stats? from incompatible
pointer type

/usr/local/src/freeswitch/src/include/switch_rtp.h:459: note: expected
?struct switch_rtp_t *? but argument is of type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:990:
error: ?struct codec_data? has no member named ?reply?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:990:
error: request for member ?a? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:990:
error: request for member ?host_ip? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:990:
error: ?struct codec_data? has no member named ?reply?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:990:
error: request for member ?a? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:990:
error: request for member ?host_udp_port? in something not a structure or
union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:991:
error: ?struct codec_data? has no member named ?reply?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:991:
error: request for member ?a? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:991:
error: request for member ?codec_ip? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:991:
error: ?struct codec_data? has no member named ?reply?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:991:
error: request for member ?a? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:991:
error: request for member ?codec_udp_port? in something not a structure or
union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:992:
error: ?struct codec_data? has no member named ?reply?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:992:
error: request for member ?b? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:992:
error: request for member ?host_ip? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:992:
error: ?struct codec_data? has no member named ?reply?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:992:
error: request for member ?b? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:992:
error: request for member ?host_udp_port? in something not a structure or
union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:993:
error: ?struct codec_data? has no member named ?reply?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:993:
error: request for member ?b? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:993:
error: request for member ?codec_ip? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:993:
error: ?struct codec_data? has no member named ?reply?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:993:
error: request for member ?b? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:993:
error: request for member ?codec_udp_port? in something not a structure or
union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:996:
error: ?struct codec_data? has no member named ?rxdiscarded?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:999:
error: ?struct codec_data? has no member named ?txrtp?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:999:
error: passing argument 1 of ?switch_rtp_get_stats? from incompatible
pointer type

/usr/local/src/freeswitch/src/include/switch_rtp.h:459: note: expected
?struct switch_rtp_t *? but argument is of type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1021:
error: ?struct codec_data? has no member named ?debug_timing?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1021:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1022:
error: ?struct codec_data? has no member named ?debug_timing?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1022:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1042:
error: ?struct codec_data? has no member named ?debug_timing?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1042:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1043:
error: ?struct codec_data? has no member named ?debug_timing?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1043:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:
In function ?sangoma_logger?:

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1069:
error: ?SNGTC_LOGLEVEL_DEBUG? undeclared (first use in this function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1072:
error: ?SNGTC_LOGLEVEL_WARN? undeclared (first use in this function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1075:
error: ?SNGTC_LOGLEVEL_INFO? undeclared (first use in this function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1078:
error: ?SNGTC_LOGLEVEL_STATS? undeclared (first use in this function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1080:
error: ?SNGTC_LOGLEVEL_ERROR? undeclared (first use in this function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1083:
error: ?SNGTC_LOGLEVEL_CRIT? undeclared (first use in this function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:
In function ?sangoma_parse_config?:

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1108:
error: ?g_init_cfg? undeclared (first use in this function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1108:
error: passing argument 3 of ?memset? makes integer from pointer without a
cast

/usr/include/string.h:64: note: expected ?size_t? but argument is of type
?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1151:
error: request for member ?host_nic_vocallo_sz? in something not a structure
or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1151:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:
In function ?mod_sangoma_codec_load?:

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1179:
error: ?g_init_cfg? undeclared (first use in this function)

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1179:
error: request for member ?log? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1179:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1180:
error: request for member ?create_rtp? in something not a structure or union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1180:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1181:
error: request for member ?create_rtp_port? in something not a structure or
union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1181:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1182:
error: request for member ?destroy_rtp? in something not a structure or
union

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1182:
error: statement with no effect

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1184:
error: implicit declaration of function ?sngtc_detect_init_modules?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1189:
error: implicit declaration of function ?sngtc_activate_modules?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1197:
error: implicit declaration of function ?sngtc_set_soap_server_url?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1210:
error: comparison between pointer and integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1210:
error: comparison between pointer and integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1230:
error: comparison between pointer and integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1230:
error: comparison between pointer and integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1235:
error: comparison between pointer and integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1235:
error: comparison between pointer and integer

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1303:
error: passing argument 1 of ?get_codec_from_id? makes integer from pointer
without a cast

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:197:
note: expected ?int? but argument is of type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1323:
error: passing argument 1 of ?get_codec_from_id? makes integer from pointer
without a cast

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:197:
note: expected ?int? but argument is of type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1394:
error: passing argument 1 of ?get_codec_from_id? makes integer from pointer
without a cast

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:197:
note: expected ?int? but argument is of type ?struct vocallo_codec_t *?

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:1414:
error: passing argument 1 of ?get_codec_from_id? makes integer from pointer
without a cast

/usr/local/src/freeswitch/src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c:197:
note: expected ?int? but argument is of type ?struct vocallo_codec_t *?

make[5]: *** [mod_sangoma_codec.lo] Error 1

make[4]: *** [all] Error 1

make[3]: *** [mod_sangoma_codec-all] Error 1

make[2]: *** [all-recursive] Error 1

make[1]: *** [all-recursive] Error 1

make: *** [all] Error 2








Regds
Sam

On Thu, Jan 6, 2011 at 2:43 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> I have tried to update to the latest git my Freeswitch but I have due to
> revert to the previous version because with latest one
> (96ac90adce931a3a28c768e102b863637c8ba98d, Jan 5 16:55:06 2011 + Sangoma
> ISDN library 7.0.0) I have many warning errors and some critical errors in
> the log.
> No problem with my previous version
> (715d250e171a94736b19019ac742f739899ad997, Dec 15 21:29:52 2010 + Sangoma
> ISDN library 6.0.0).
>
> Here an extraction of my log file:
>
> 2011-01-06 02:01:25.990974 [WARNING] mod_freetdm.c:434 [s23c22][23:22] Why
> bother changing state from PROCEED to PROCEED
> 2011-01-06 02:01:25.994018 [CRIT] ftmod_sangoma_isdn_stack_rcv.c:988
> sng_isdn->
>
> date: 01/06/2011 time: 02:01:25
> 2011-01-06 02:01:25.994018 [CRIT] ftmod_sangoma_isdn_stack_rcv.c:988
> sng_isdn->
> mtss(posix): sw error:  ent: 010  inst: 000  proc id: 001
> file: /usr/src/libsng_trillium-build/libsng_isdn/trillium/in/in_bdy1.c
> line: 2144 errcode: 46909632820939 errcls: ERRCLS_DEBUG
> errval: 00001  errdesc: InUiIntConReq() Failed.
> 2011-01-06 02:01:25.994018 [WARNING] ftmod_sangoma_isdn_stack_hndl.c:862
> [s10c1][10:1] STATUS CONFIRM (call_state:0 channel-state:DIALING cause:100)
> (suId:1 suInstId:57 spInstId:0)
> 2011-01-06 02:01:31.068159 [WARNING] mod_freetdm.c:434 [s22c9][22:9] Why
> bother changing state from PROCEED to PROCEED
> 2011-01-06 02:01:31.172106 [WARNING] ftdm_io.c:2250 [s22c9][22:9] Cannot
> indicate PROGRESS_MEDIA in channel with indication PROCEED still pending in
> state PROCEED
> 2011-01-06 02:01:37.465176 [WARNING] mod_freetdm.c:434 [s3c28][3:28] Why
> bother changing state from PROCEED to PROCEED
> 2011-01-06 02:01:37.472130 [WARNING] ftdm_io.c:2250 [s3c28][3:28] Cannot
> indicate RINGING in channel with indication PROCEED still pending in state
> PROCEED
> 2011-01-06 02:01:43.705155 [WARNING] mod_freetdm.c:434 [s23c23][23:23] Why
> bother changing state from PROCEED to PROCEED
> 2011-01-06 02:01:43.707124 [CRIT] ftmod_sangoma_isdn_stack_rcv.c:988
> sng_isdn->
>
> date: 01/06/2011 time: 02:01:43
> 2011-01-06 02:01:43.707124 [CRIT] ftmod_sangoma_isdn_stack_rcv.c:988
> sng_isdn->
> mtss(posix): sw error:  ent: 010  inst: 000  proc id: 001
> file: /usr/src/libsng_trillium-build/libsng_isdn/trillium/in/in_bdy1.c
> line: 2144 errcode: 46909632820939 errcls: ERRCLS_DEBUG
> errval: 00001  errdesc: InUiIntConReq() Failed.
> 2011-01-06 02:01:43.707124 [WARNING] ftmod_sangoma_isdn_stack_hndl.c:862
> [s5c1][5:1] STATUS CONFIRM (call_state:0 channel-state:DIALING cause:100)
> (suId:1 suInstId:61 spInstId:0)
> 2011-01-06 02:01:49.463158 [WARNING] mod_freetdm.c:434 [s16c14][16:14] Why
> bother changing state from PROCEED to PROCEED
> 2011-01-06 02:01:49.470188 [WARNING] ftdm_io.c:2250 [s16c14][16:14] Cannot
> indicate RINGING in channel with indication PROCEED still pending in state
> PROCEED
> 2011-01-06 02:01:53.938264 [WARNING] mod_freetdm.c:434 [s1c1][1:1] Why
> bother changing state from PROCEED to PROCEED
> 2011-01-06 02:01:53.944202 [WARNING] ftdm_io.c:2250 [s1c1][1:1] Cannot
> indicate RINGING in channel with indication PROCEED still pending in state
> PROCEED
> 2011-01-06 02:01:59.468295 [WARNING] mod_freetdm.c:434 [s8c3][8:3] Why
> bother changing state from PROCEED to PROCEED
> 2011-01-06 02:01:59.476270 [WARNING] ftdm_io.c:2250 [s8c3][8:3] Cannot
> indicate RINGING in channel with indication PROCEED still pending in state
> PROCEED
> 2011-01-06 02:02:01.057245 [WARNING] mod_freetdm.c:434 [s1c2][1:2] Why
> bother changing state from PROCEED to PROCEED
> 2011-01-06 02:02:01.083313 [WARNING] ftdm_io.c:2250 [s1c2][1:2] Cannot
> indicate PROGRESS_MEDIA in channel with indication PROCEED still pending in
> state PROCEED
> 2011-01-06 02:02:04.286328 [WARNING] mod_freetdm.c:434 [s20c24][20:24] Why
> bother changing state from PROCEED to PROCEED
> 2011-01-06 02:02:14.525323 [WARNING] mod_freetdm.c:434 [s17c23][17:23] Why
> bother changing state from PROCEED to PROCEED
> 2011-01-06 02:02:15.029458 [WARNING] ftdm_io.c:2250 [s17c23][17:23] Cannot
> indicate PROGRESS_MEDIA in channel with indication PROCEED still pending in
> state PROCEED
> 2011-01-06 02:02:19.789411 [WARNING] mod_freetdm.c:434 [s12c5][12:5] Why
> bother changing state from PROCEED to PROCEED
> 2011-01-06 02:02:19.891438 [WARNING] ftdm_io.c:2250 [s12c5][12:5] Cannot
> indicate PROGRESS_MEDIA in channel with indication PROCEED still pending in
> state PROCEED
> 2011-01-06 02:02:23.764507 [WARNING] mod_freetdm.c:434 [s22c10][22:10] Why
> bother changing state from PROCEED to PROCEED
> 2011-01-06 02:02:26.462538 [WARNING] mod_freetdm.c:434 [s19c24][19:24] Why
> bother changing state from PROCEED to PROCEED
> 2011-01-06 02:02:29.171509 [WARNING] ftdm_io.c:2250 [s22c10][22:10] Cannot
> indicate PROGRESS_MEDIA in channel with indication PROCEED still pending in
> state PROCEED
> 2011-01-06 02:02:31.698471 [WARNING] mod_freetdm.c:434 [s23c24][23:24] Why
> bother changing state from PROCEED to PROCEED
> 2011-01-06 02:02:31.724623 [WARNING] ftdm_io.c:2250 [s23c24][23:24] Cannot
> indicate PROGRESS_MEDIA in channel with indication PROCEED still pending in
> state PROCEED
> 2011-01-06 02:02:44.469700 [WARNING] mod_freetdm.c:434 [s6c21][6:21] Why
> bother changing state from PROCEED to PROCEED
> 2011-01-06 02:02:44.476688 [WARNING] ftdm_io.c:2250 [s6c21][6:21] Cannot
> indicate RINGING in channel with indication PROCEED still pending in state
> PROCEED
> 2011-01-06 02:02:48.018716 [WARNING] mod_freetdm.c:434 [s20c25][20:25] Why
> bother changing state from PROCEED to PROCEED
> 2011-01-06 02:02:48.046713 [WARNING] ftdm_io.c:2250 [s20c25][20:25] Cannot
> indicate PROGRESS_MEDIA in channel with indication PROCEED still pending in
> state PROCEED
> 2011-01-06 02:02:49.463708 [WARNING] mod_freetdm.c:434 [s8c4][8:4] Why
> bother changing state from PROCEED to PROCEED
> 2011-01-06 02:02:54.462848 [WARNING] mod_freetdm.c:434 [s19c25][19:25] Why
> bother changing state from PROCEED to PROCEED
> 2011-01-06 02:02:54.468754 [CRIT] ftmod_sangoma_isdn_stack_rcv.c:988
> sng_isdn->
>
> date: 01/06/2011 time: 02:02:54
> 2011-01-06 02:02:54.468754 [CRIT] ftmod_sangoma_isdn_stack_rcv.c:988
> sng_isdn->
> mtss(posix): sw error:  ent: 010  inst: 000  proc id: 001
> file: /usr/src/libsng_trillium-build/libsng_isdn/trillium/in/in_bdy1.c
> line: 2144 errcode: 46909632820939 errcls: ERRCLS_DEBUG
> errval: 00001  errdesc: InUiIntConReq() Failed.
> 2011-01-06 02:02:54.468754 [WARNING] ftmod_sangoma_isdn_stack_hndl.c:862
> [s10c1][10:1] STATUS CONFIRM (call_state:0 channel-state:DIALING cause:100)
> (suId:1 suInstId:81 spInstId:0)
> 2011-01-06 02:02:55.305075 [WARNING] ftdm_io.c:2250 [s8c4][8:4] Cannot
> indicate PROGRESS_MEDIA in channel with indication PROCEED still pending in
> state PROCEED
> 2011-01-06 02:02:57.785073 [WARNING] ftdm_io.c:2250 [s19c25][19:25] Cannot
> indicate PROGRESS_MEDIA in channel with indication PROCEED still pending in
> state PROCEED
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Thu Jan  6 12:43:11 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 6 Jan 2011 15:13:11 +0530
Subject: [Freeswitch-users] codec error
Message-ID: <AANLkTi=cKTUOcN--hZm2yLXY8s-pWotek+_YA1Bxw2jw@mail.gmail.com>

I am getting the error after installing latest git from freeswitch, where as
it was working from old git


2011-01-06 15:10:19.113781 [ERR] mod_g729.c:102 This codec is only usable in
passthrough mode!
2011-01-06 15:10:19.113781 [ERR] switch_core_io.c:1042 Codec G.729 encoder
error!


Regards
Sam
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From steveayre at gmail.com  Thu Jan  6 12:50:35 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 6 Jan 2011 09:50:35 +0000
Subject: [Freeswitch-users] codec error
In-Reply-To: <AANLkTi=cKTUOcN--hZm2yLXY8s-pWotek+_YA1Bxw2jw@mail.gmail.com>
References: <AANLkTi=cKTUOcN--hZm2yLXY8s-pWotek+_YA1Bxw2jw@mail.gmail.com>
Message-ID: <AANLkTinLm5AP9kqoeHarqrCJGjsUq6Xo8=aOnY05HNHN@mail.gmail.com>

Some more information please...

G729 has patents that mean it can't be used unlicensed. FreeSWITCH
either has mod_com_g729 for a full featured licensed version, or
mod_g729 which operates in passthrough mode for G729-G729 calls (it
needs no license since for these calls there is no encoding/decoding
step, the already encoded data is just passed straight through). You
will never have had mod_g729 working in the past for a transcoding
call, so that won't have changed.

I would say that it's either:
- Your config files have changed
- The endpoints are offering different codecs from before
- Something changed in git about the codec negotiation

Can you pastebin a debug level log of the calls, and enable siptrace?
Those will show what codecs are being offered, selected and show the
codec negotiation. It'd also be useful to know what your sip profile
config files look like since there are several options that adjust how
the negotiation is done.

-Steve



On 6 January 2011 09:43, Sam <u2nsam at gmail.com> wrote:
> I am getting the error after installing latest git from freeswitch, where as
> it was working from old git
>
>
> 2011-01-06 15:10:19.113781 [ERR] mod_g729.c:102 This codec is only usable in
> passthrough mode!
> 2011-01-06 15:10:19.113781 [ERR] switch_core_io.c:1042 Codec G.729 encoder
> error!
>
>
> Regards
> Sam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From u2nsam at gmail.com  Thu Jan  6 13:10:45 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 6 Jan 2011 15:40:45 +0530
Subject: [Freeswitch-users] codec error
In-Reply-To: <AANLkTinLm5AP9kqoeHarqrCJGjsUq6Xo8=aOnY05HNHN@mail.gmail.com>
References: <AANLkTi=cKTUOcN--hZm2yLXY8s-pWotek+_YA1Bxw2jw@mail.gmail.com>
	<AANLkTinLm5AP9kqoeHarqrCJGjsUq6Xo8=aOnY05HNHN@mail.gmail.com>
Message-ID: <AANLkTikmHcMHfc3ydk+8b2SWEKB+sR=Awj0OVri6qdrK@mail.gmail.com>

Hi,

This was working earlier and the config file have not changed , only
upgraded to latest git.
have attached the file codec.txt .



Regards
Sam



On Thu, Jan 6, 2011 at 3:20 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Some more information please...
>
> G729 has patents that mean it can't be used unlicensed. FreeSWITCH
> either has mod_com_g729 for a full featured licensed version, or
> mod_g729 which operates in passthrough mode for G729-G729 calls (it
> needs no license since for these calls there is no encoding/decoding
> step, the already encoded data is just passed straight through). You
> will never have had mod_g729 working in the past for a transcoding
> call, so that won't have changed.
>
> I would say that it's either:
> - Your config files have changed
> - The endpoints are offering different codecs from before
> - Something changed in git about the codec negotiation
>
> Can you pastebin a debug level log of the calls, and enable siptrace?
> Those will show what codecs are being offered, selected and show the
> codec negotiation. It'd also be useful to know what your sip profile
> config files look like since there are several options that adjust how
> the negotiation is done.
>
> -Steve
>
>
>
> On 6 January 2011 09:43, Sam <u2nsam at gmail.com> wrote:
> > I am getting the error after installing latest git from freeswitch, where
> as
> > it was working from old git
> >
> >
> > 2011-01-06 15:10:19.113781 [ERR] mod_g729.c:102 This codec is only usable
> in
> > passthrough mode!
> > 2011-01-06 15:10:19.113781 [ERR] switch_core_io.c:1042 Codec G.729
> encoder
> > error!
> >
> >
> > Regards
> > Sam
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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called with g722-->


2011-01-06 15:30:43.040722 [NOTICE] switch_channel.c:808 New Channel sofia/internal/7001 at 192.168.2.190 [a97449df-688e-4bf9-9724-3d63338907fb]
2011-01-06 15:30:43.041645 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/7001 at 192.168.2.190) Running State Change CS_NEW
2011-01-06 15:30:43.041645 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/7001 at 192.168.2.190) State NEW
2011-01-06 15:30:43.041645 [DEBUG] sofia.c:4616 Channel sofia/internal/7001 at 192.168.2.190 entering state [received][100]
2011-01-06 15:30:43.041645 [DEBUG] sofia.c:4627 Remote SDP:
v=0
o=- 25952772 25952780 IN IP4 192.168.2.15
s=eyeBeam
c=IN IP4 192.168.2.15
t=0 0
m=audio 6398 RTP/AVP 110 101
a=rtpmap:110 g7222/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 1 : AC5CC22D 0000007A 192.168.2.15 6398
a=ptime:110 20

2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[G7221:115:32000:20:48000]
2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[G7221:107:16000:20:32000]
2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[PCMU:0:8000:20:64000]
2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[PCMA:8:8000:20:64000]
2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [g7222:110:16000:110:0]/[G729:18:8000:20:8000]
2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [telephone-event:101:8000:110:0]/[G7221:115:32000:20:48000]
2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [telephone-event:101:8000:110:0]/[G7221:107:16000:20:32000]
2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [telephone-event:101:8000:110:0]/[PCMU:0:8000:20:64000]
2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [telephone-event:101:8000:110:0]/[PCMA:8:8000:20:64000]
2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [telephone-event:101:8000:110:0]/[G729:18:8000:20:8000]
2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4507 Set 2833 dtmf send/recv payload to 101
2011-01-06 15:30:43.041645 [DEBUG] switch_channel.c:2535 (sofia/internal/7001 at 192.168.2.190) Callstate Change DOWN -> HANGUP
2011-01-06 15:30:43.041645 [NOTICE] sofia.c:4838 Hangup sofia/internal/7001 at 192.168.2.190 [CS_NEW] [INCOMPATIBLE_DESTINATION]
2011-01-06 15:30:43.041645 [DEBUG] switch_channel.c:2551 Send signal sofia/internal/7001 at 192.168.2.190 [KILL]
2011-01-06 15:30:43.041645 [DEBUG] switch_core_session.c:1083 Send signal sofia/internal/7001 at 192.168.2.190 [BREAK]
2011-01-06 15:30:43.042561 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/7001 at 192.168.2.190) Running State Change CS_HANGUP
2011-01-06 15:30:43.042561 [DEBUG] switch_core_state_machine.c:557 (sofia/internal/7001 at 192.168.2.190) State HANGUP
2011-01-06 15:30:43.042561 [DEBUG] mod_sofia.c:459 Channel sofia/internal/7001 at 192.168.2.190 hanging up, cause: INCOMPATIBLE_DESTINATION
2011-01-06 15:30:43.044377 [DEBUG] mod_sofia.c:521 Responding to INVITE with: 488
2011-01-06 15:30:43.044377 [DEBUG] switch_core_state_machine.c:46 sofia/internal/7001 at 192.168.2.190 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION
2011-01-06 15:30:43.044377 [DEBUG] switch_core_state_machine.c:557 (sofia/internal/7001 at 192.168.2.190) State HANGUP going to sleep
2011-01-06 15:30:43.044377 [DEBUG] switch_core_state_machine.c:351 (sofia/internal/7001 at 192.168.2.190) State Change CS_HANGUP -> CS_REPORTING
2011-01-06 15:30:43.044377 [DEBUG] switch_core_session.c:1083 Send signal sofia/internal/7001 at 192.168.2.190 [BREAK]
2011-01-06 15:30:43.044377 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/7001 at 192.168.2.190) Running State Change CS_REPORTING
2011-01-06 15:30:43.044377 [DEBUG] switch_core_state_machine.c:617 (sofia/internal/7001 at 192.168.2.190) State REPORTING
2011-01-06 15:30:43.044377 [DEBUG] switch_core_state_machine.c:53 sofia/internal/7001 at 192.168.2.190 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION
2011-01-06 15:30:43.044377 [DEBUG] switch_core_state_machine.c:617 (sofia/internal/7001 at 192.168.2.190) State REPORTING going to sleep
2011-01-06 15:30:43.045293 [DEBUG] switch_core_state_machine.c:345 (sofia/internal/7001 at 192.168.2.190) State Change CS_REPORTING -> CS_DESTROY
2011-01-06 15:30:43.045293 [DEBUG] switch_core_session.c:1083 Send signal sofia/internal/7001 at 192.168.2.190 [BREAK]
2011-01-06 15:30:43.045293 [DEBUG] switch_core_session.c:1255 Session 46 (sofia/internal/7001 at 192.168.2.190) Locked, Waiting on external entities
2011-01-06 15:30:43.045293 [NOTICE] switch_core_session.c:1273 Session 46 (sofia/internal/7001 at 192.168.2.190) Ended
2011-01-06 15:30:43.045293 [NOTICE] switch_core_session.c:1275 Close Channel sofia/internal/7001 at 192.168.2.190 [CS_DESTROY]
2011-01-06 15:30:43.045293 [DEBUG] switch_core_state_machine.c:449 (sofia/internal/7001 at 192.168.2.190) Callstate Change HANGUP -> DOWN
2011-01-06 15:30:43.045293 [DEBUG] switch_core_state_machine.c:452 (sofia/internal/7001 at 192.168.2.190) Running State Change CS_DESTROY
2011-01-06 15:30:43.045293 [DEBUG] switch_core_state_machine.c:462 (sofia/internal/7001 at 192.168.2.190) State DESTROY
2011-01-06 15:30:43.045293 [DEBUG] mod_sofia.c:364 sofia/internal/7001 at 192.168.2.190 SOFIA DESTROY
2011-01-06 15:30:43.045293 [DEBUG] switch_core_state_machine.c:60 sofia/internal/7001 at 192.168.2.190 Standard DESTROY
2011-01-06 15:30:43.045293 [DEBUG] switch_core_state_machine.c:462 (sofia/internal/7001 at 192.168.2.190) State DESTROY going to sleep



When sending with g729 -->




2011-01-06 15:36:16.594761 [NOTICE] switch_channel.c:808 New Channel sofia/internal/7001 at 192.168.2.190 [946f3be9-dc5f-43cc-948d-b090c21b0079]
2011-01-06 15:36:16.595675 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/7001 at 192.168.2.190) Running State Change CS_NEW
2011-01-06 15:36:16.595675 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/7001 at 192.168.2.190) State NEW
2011-01-06 15:36:16.595675 [DEBUG] sofia.c:4616 Channel sofia/internal/7001 at 192.168.2.190 entering state [received][100]
2011-01-06 15:36:16.595675 [DEBUG] sofia.c:4627 Remote SDP:
v=0
o=- 26286332 26286340 IN IP4 192.168.2.15
s=eyeBeam
c=IN IP4 192.168.2.15
t=0 0
m=audio 6398 RTP/AVP 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 1 : DB678F86 00000016 192.168.2.15 6398

2011-01-06 15:36:16.595675 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [G729:18:8000:0:8000]/[G7221:115:32000:20:48000]
2011-01-06 15:36:16.595675 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [G729:18:8000:0:8000]/[G7221:107:16000:20:32000]
2011-01-06 15:36:16.595675 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [G729:18:8000:0:8000]/[PCMU:0:8000:20:64000]
2011-01-06 15:36:16.595675 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [G729:18:8000:0:8000]/[PCMA:8:8000:20:64000]
2011-01-06 15:36:16.595675 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [G729:18:8000:0:8000]/[G729:18:8000:20:8000]
2011-01-06 15:36:16.595675 [DEBUG] sofia_glue.c:2750 Set Codec sofia/internal/7001 at 192.168.2.190 G729/8000 20 ms 160 samples 8000 bits
2011-01-06 15:36:16.596685 [DEBUG] sofia_glue.c:4507 Set 2833 dtmf send/recv payload to 101
2011-01-06 15:36:16.596685 [DEBUG] sofia.c:4794 (sofia/internal/7001 at 192.168.2.190) State Change CS_NEW -> CS_INIT
2011-01-06 15:36:16.596685 [DEBUG] switch_core_session.c:1083 Send signal sofia/internal/7001 at 192.168.2.190 [BREAK]
2011-01-06 15:36:16.596685 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/7001 at 192.168.2.190) Running State Change CS_INIT
2011-01-06 15:36:16.596685 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/7001 at 192.168.2.190) State INIT
2011-01-06 15:36:16.596685 [DEBUG] mod_sofia.c:86 sofia/internal/7001 at 192.168.2.190 SOFIA INIT
2011-01-06 15:36:16.596685 [DEBUG] mod_sofia.c:126 (sofia/internal/7001 at 192.168.2.190) State Change CS_INIT -> CS_ROUTING
2011-01-06 15:36:16.596685 [DEBUG] switch_core_session.c:1083 Send signal sofia/internal/7001 at 192.168.2.190 [BREAK]
2011-01-06 15:36:16.596685 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/7001 at 192.168.2.190) State INIT going to sleep
2011-01-06 15:36:16.596685 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/7001 at 192.168.2.190) Running State Change CS_ROUTING
2011-01-06 15:36:16.596685 [DEBUG] switch_channel.c:1657 (sofia/internal/7001 at 192.168.2.190) Callstate Change DOWN -> RINGING
2011-01-06 15:36:16.596685 [DEBUG] switch_core_state_machine.c:359 (sofia/internal/7001 at 192.168.2.190) State ROUTING
2011-01-06 15:36:16.596685 [DEBUG] mod_sofia.c:149 sofia/internal/7001 at 192.168.2.190 SOFIA ROUTING
2011-01-06 15:36:16.596685 [DEBUG] switch_core_state_machine.c:77 sofia/internal/7001 at 192.168.2.190 Standard ROUTING
2011-01-06 15:36:16.596685 [INFO] mod_dialplan_xml.c:331 Processing 7001 <7001>->7013 in context novanet
Dialplan: sofia/internal/7001 at 192.168.2.190 parsing [novanet->novanet_7011] continue=false
Dialplan: sofia/internal/7001 at 192.168.2.190 Regex (FAIL) [novanet_7011] destination_number(7013) =~ /^(7011)/ break=on-false
Dialplan: sofia/internal/7001 at 192.168.2.190 parsing [novanet->novanet_7012] continue=false
Dialplan: sofia/internal/7001 at 192.168.2.190 Regex (FAIL) [novanet_7012] destination_number(7013) =~ /^(7012)/ break=on-false
Dialplan: sofia/internal/7001 at 192.168.2.190 parsing [novanet->novanet_7013] continue=false
Dialplan: sofia/internal/7001 at 192.168.2.190 Regex (PASS) [novanet_7013] destination_number(7013) =~ /^(7013)/ break=on-false
Dialplan: sofia/internal/7001 at 192.168.2.190 Action set(dialed_extension=7013)
Dialplan: sofia/internal/7001 at 192.168.2.190 Action export(dialed_extension=7013)
Dialplan: sofia/internal/7001 at 192.168.2.190 Action set(transfer_ringback=local_stream://moh)
Dialplan: sofia/internal/7001 at 192.168.2.190 Action set(call_timeout=20)
Dialplan: sofia/internal/7001 at 192.168.2.190 Action set(export_vars=#,*)
Dialplan: sofia/internal/7001 at 192.168.2.190 Action export(#=true)
Dialplan: sofia/internal/7001 at 192.168.2.190 Action set(bind_meta_key=#)
Dialplan: sofia/internal/7001 at 192.168.2.190 Action bind_meta_app(1 ab s execute_extension::dx XML features)
Dialplan: sofia/internal/7001 at 192.168.2.190 Action bind_meta_app(2 ab s execute_extension::att_xfer XML features)
Dialplan: sofia/internal/7001 at 192.168.2.190 Action set(hangup_after_bridge=true)
Dialplan: sofia/internal/7001 at 192.168.2.190 Action set(continue_on_fail=true)
Dialplan: sofia/internal/7001 at 192.168.2.190 Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number})
Dialplan: sofia/internal/7001 at 192.168.2.190 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
Dialplan: sofia/internal/7001 at 192.168.2.190 Action hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid})
Dialplan: sofia/internal/7001 at 192.168.2.190 Action hash(insert/${domain_name}-last_dial_ext/global/${uuid})
Dialplan: sofia/internal/7001 at 192.168.2.190 Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)})
Dialplan: sofia/internal/7001 at 192.168.2.190 Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
Dialplan: sofia/internal/7001 at 192.168.2.190 Action bridge({sip_invite_domain=192.168.2.190}user/${dialed_extension}@${domain_name})
Dialplan: sofia/internal/7001 at 192.168.2.190 Action set(continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED)
Dialplan: sofia/internal/7001 at 192.168.2.190 Action bridge(freetdm/wp1/a/9833658320)
Dialplan: sofia/internal/7001 at 192.168.2.190 Action answer()
Dialplan: sofia/internal/7001 at 192.168.2.190 Action sleep(1000)
Dialplan: sofia/internal/7001 at 192.168.2.190 Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension})
2011-01-06 15:36:16.598554 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/7001 at 192.168.2.190) State Change CS_ROUTING -> CS_EXECUTE
2011-01-06 15:36:16.598554 [DEBUG] switch_core_session.c:1083 Send signal sofia/internal/7001 at 192.168.2.190 [BREAK]
2011-01-06 15:36:16.598554 [DEBUG] switch_core_state_machine.c:359 (sofia/internal/7001 at 192.168.2.190) State ROUTING going to sleep
2011-01-06 15:36:16.598554 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/7001 at 192.168.2.190) Running State Change CS_EXECUTE
2011-01-06 15:36:16.598554 [DEBUG] switch_core_state_machine.c:366 (sofia/internal/7001 at 192.168.2.190) State EXECUTE
2011-01-06 15:36:16.598554 [DEBUG] mod_sofia.c:242 sofia/internal/7001 at 192.168.2.190 SOFIA EXECUTE
2011-01-06 15:36:16.598554 [DEBUG] switch_core_state_machine.c:157 sofia/internal/7001 at 192.168.2.190 Standard EXECUTE
EXECUTE sofia/internal/7001 at 192.168.2.190 set(dialed_extension=7013)
2011-01-06 15:36:16.598554 [DEBUG] mod_dptools.c:1050 sofia/internal/7001 at 192.168.2.190 SET [dialed_extension]=[7013]
EXECUTE sofia/internal/7001 at 192.168.2.190 export(dialed_extension=7013)
2011-01-06 15:36:16.599470 [DEBUG] switch_channel.c:957 EXPORT (export_vars) [dialed_extension]=[7013]
EXECUTE sofia/internal/7001 at 192.168.2.190 set(transfer_ringback=local_stream://moh)
2011-01-06 15:36:16.599470 [DEBUG] mod_dptools.c:1050 sofia/internal/7001 at 192.168.2.190 SET [transfer_ringback]=[local_stream://moh]
EXECUTE sofia/internal/7001 at 192.168.2.190 set(call_timeout=20)
2011-01-06 15:36:16.599470 [DEBUG] mod_dptools.c:1050 sofia/internal/7001 at 192.168.2.190 SET [call_timeout]=[20]
EXECUTE sofia/internal/7001 at 192.168.2.190 set(export_vars=#,*)
2011-01-06 15:36:16.600385 [DEBUG] mod_dptools.c:1050 sofia/internal/7001 at 192.168.2.190 SET [export_vars]=[#,*]
EXECUTE sofia/internal/7001 at 192.168.2.190 export(#=true)
2011-01-06 15:36:16.600385 [DEBUG] switch_channel.c:957 EXPORT (export_vars) [#]=[true]
EXECUTE sofia/internal/7001 at 192.168.2.190 set(bind_meta_key=#)
2011-01-06 15:36:16.600385 [DEBUG] mod_dptools.c:1050 sofia/internal/7001 at 192.168.2.190 SET [bind_meta_key]=[#]
EXECUTE sofia/internal/7001 at 192.168.2.190 bind_meta_app(1 ab s execute_extension::dx XML features)
2011-01-06 15:36:16.601300 [INFO] switch_ivr_async.c:3004 Bound A-Leg: # execute_extension::dx XML features
2011-01-06 15:36:16.601300 [INFO] switch_ivr_async.c:3012 Bound B-Leg: # execute_extension::dx XML features
EXECUTE sofia/internal/7001 at 192.168.2.190 bind_meta_app(2 ab s execute_extension::att_xfer XML features)
2011-01-06 15:36:16.601300 [INFO] switch_ivr_async.c:3004 Bound A-Leg: # execute_extension::att_xfer XML features
2011-01-06 15:36:16.601300 [INFO] switch_ivr_async.c:3012 Bound B-Leg: # execute_extension::att_xfer XML features
EXECUTE sofia/internal/7001 at 192.168.2.190 set(hangup_after_bridge=true)
2011-01-06 15:36:16.602215 [DEBUG] mod_dptools.c:1050 sofia/internal/7001 at 192.168.2.190 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/7001 at 192.168.2.190 set(continue_on_fail=true)
2011-01-06 15:36:16.602215 [DEBUG] mod_dptools.c:1050 sofia/internal/7001 at 192.168.2.190 SET [continue_on_fail]=[true]
EXECUTE sofia/internal/7001 at 192.168.2.190 hash(insert/192.168.2.190-call_return/7013/7001)
EXECUTE sofia/internal/7001 at 192.168.2.190 hash(insert/192.168.2.190-last_dial_ext/7013/946f3be9-dc5f-43cc-948d-b090c21b0079)
EXECUTE sofia/internal/7001 at 192.168.2.190 hash(insert/192.168.2.190-last_dial_ext//946f3be9-dc5f-43cc-948d-b090c21b0079)
EXECUTE sofia/internal/7001 at 192.168.2.190 hash(insert/192.168.2.190-last_dial_ext/global/946f3be9-dc5f-43cc-948d-b090c21b0079)
EXECUTE sofia/internal/7001 at 192.168.2.190 set(called_party_callgroup=1)
2011-01-06 15:36:16.603130 [DEBUG] mod_dptools.c:1050 sofia/internal/7001 at 192.168.2.190 SET [called_party_callgroup]=[1]
EXECUTE sofia/internal/7001 at 192.168.2.190 hash(insert/192.168.2.190-last_dial/1/946f3be9-dc5f-43cc-948d-b090c21b0079)
EXECUTE sofia/internal/7001 at 192.168.2.190 bridge({sip_invite_domain=192.168.2.190}user/7013 at 192.168.2.190)
2011-01-06 15:36:16.603130 [DEBUG] switch_channel.c:914 sofia/internal/7001 at 192.168.2.190 EXPORTING[export_vars] [#]=[true] to event
2011-01-06 15:36:16.603130 [DEBUG] switch_channel.c:914 sofia/internal/7001 at 192.168.2.190 EXPORTING[export_vars] [#]=[true] to event
2011-01-06 15:36:16.603130 [DEBUG] switch_ivr_originate.c:1961 variable string 0 = [sip_invite_domain=192.168.2.190]
2011-01-06 15:36:16.605902 [DEBUG] switch_channel.c:914 sofia/internal/7001 at 192.168.2.190 EXPORTING[export_vars] [#]=[true] to event
2011-01-06 15:36:16.605902 [DEBUG] switch_channel.c:914 sofia/internal/7001 at 192.168.2.190 EXPORTING[export_vars] [#]=[true] to event
2011-01-06 15:36:16.605902 [DEBUG] switch_ivr_originate.c:1961 variable string 0 = [presence_id=7013 at 192.168.2.190]
2011-01-06 15:36:16.606823 [NOTICE] switch_channel.c:808 New Channel sofia/internal/sip:7013 at 192.168.2.42:5060 [94892779-a422-4b43-91d6-8666006edfbf]
2011-01-06 15:36:16.606823 [DEBUG] mod_sofia.c:4101 (sofia/internal/sip:7013 at 192.168.2.42:5060) State Change CS_NEW -> CS_INIT
2011-01-06 15:36:16.606823 [DEBUG] switch_core_session.c:1083 Send signal sofia/internal/sip:7013 at 192.168.2.42:5060 [BREAK]
2011-01-06 15:36:16.606823 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/sip:7013 at 192.168.2.42:5060) Running State Change CS_INIT
2011-01-06 15:36:16.606823 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/sip:7013 at 192.168.2.42:5060) State INIT
2011-01-06 15:36:16.606823 [DEBUG] mod_sofia.c:86 sofia/internal/sip:7013 at 192.168.2.42:5060 SOFIA INIT
2011-01-06 15:36:16.606823 [DEBUG] sofia_glue.c:2317 sip:7013 at 192.168.2.42:5060;transport=udp Setting proxy route to sofia/internal/sip:7013 at 192.168.2.42:5060
2011-01-06 15:36:16.606823 [DEBUG] mod_sofia.c:126 (sofia/internal/sip:7013 at 192.168.2.42:5060) State Change CS_INIT -> CS_ROUTING
2011-01-06 15:36:16.606823 [DEBUG] switch_core_session.c:1083 Send signal sofia/internal/sip:7013 at 192.168.2.42:5060 [BREAK]
2011-01-06 15:36:16.606823 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/sip:7013 at 192.168.2.42:5060) State INIT going to sleep
2011-01-06 15:36:16.607737 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/sip:7013 at 192.168.2.42:5060) Running State Change CS_ROUTING
2011-01-06 15:36:16.607737 [DEBUG] switch_channel.c:1657 (sofia/internal/sip:7013 at 192.168.2.42:5060) Callstate Change DOWN -> RINGING
2011-01-06 15:36:16.607737 [DEBUG] switch_core_state_machine.c:359 (sofia/internal/sip:7013 at 192.168.2.42:5060) State ROUTING
2011-01-06 15:36:16.607737 [DEBUG] mod_sofia.c:149 sofia/internal/sip:7013 at 192.168.2.42:5060 SOFIA ROUTING
2011-01-06 15:36:16.607737 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/sip:7013 at 192.168.2.42:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2011-01-06 15:36:16.607737 [DEBUG] switch_core_session.c:1083 Send signal sofia/internal/sip:7013 at 192.168.2.42:5060 [BREAK]
2011-01-06 15:36:16.607737 [DEBUG] switch_core_state_machine.c:359 (sofia/internal/sip:7013 at 192.168.2.42:5060) State ROUTING going to sleep
2011-01-06 15:36:16.607737 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/sip:7013 at 192.168.2.42:5060) Running State Change CS_CONSUME_MEDIA
2011-01-06 15:36:16.607737 [DEBUG] switch_core_state_machine.c:378 (sofia/internal/sip:7013 at 192.168.2.42:5060) State CONSUME_MEDIA
2011-01-06 15:36:16.607737 [DEBUG] switch_core_state_machine.c:378 (sofia/internal/sip:7013 at 192.168.2.42:5060) State CONSUME_MEDIA going to sleep
2011-01-06 15:36:16.607737 [DEBUG] sofia.c:4616 Channel sofia/internal/sip:7013 at 192.168.2.42:5060 entering state [calling][0]
2011-01-06 15:36:16.871955 [INFO] sofia.c:720 sofia/internal/sip:7013 at 192.168.2.42:5060 Update Callee ID to "Outbound Call" <7013>
2011-01-06 15:36:16.872867 [DEBUG] sofia.c:4616 Channel sofia/internal/sip:7013 at 192.168.2.42:5060 entering state [proceeding][180]
2011-01-06 15:36:16.872867 [NOTICE] sofia.c:4694 Ring-Ready sofia/internal/sip:7013 at 192.168.2.42:5060!
2011-01-06 15:36:16.873779 [NOTICE] mod_sofia.c:2176 Ring-Ready sofia/internal/7001 at 192.168.2.190!
2011-01-06 15:36:16.873779 [DEBUG] switch_core_session.c:676 Send signal sofia/internal/7001 at 192.168.2.190 [BREAK]
2011-01-06 15:36:16.873779 [DEBUG] sofia.c:4616 Channel sofia/internal/7001 at 192.168.2.190 entering state [early][180]
2011-01-06 15:36:16.873779 [NOTICE] switch_ivr_originate.c:479 Ring Ready sofia/internal/7001 at 192.168.2.190!
2011-01-06 15:36:17.337685 [WARNING] sofia_reg.c:1216 SIP auth challenge (REGISTER) on sofia profile 'internal' for [7030 at 192.168.2.190] from ip 192.168.2.52
2011-01-06 15:36:19.913658 [WARNING] sofia_reg.c:1216 SIP auth challenge (REGISTER) on sofia profile 'internal' for [7013 at 192.168.2.190] from ip 192.168.2.42
2011-01-06 15:36:20.246147 [DEBUG] sofia.c:4616 Channel sofia/internal/sip:7013 at 192.168.2.42:5060 entering state [completing][200]
2011-01-06 15:36:20.246147 [DEBUG] sofia.c:4627 Remote SDP:
v=0
o=Cisco-SIPUA 22233 0 IN IP4 192.168.2.42
s=SIP Call
t=0 0
m=audio 29494 RTP/AVP 0 101
c=IN IP4 192.168.2.42
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

2011-01-06 15:36:20.246147 [DEBUG] sofia.c:4616 Channel sofia/internal/sip:7013 at 192.168.2.42:5060 entering state [ready][200]
2011-01-06 15:36:20.246147 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [PCMU:0:8000:0:64000]/[G729:18:8000:20:8000]
2011-01-06 15:36:20.246147 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [PCMU:0:8000:0:64000]/[G7221:115:32000:20:48000]
2011-01-06 15:36:20.246147 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [PCMU:0:8000:0:64000]/[G7221:107:16000:20:32000]
2011-01-06 15:36:20.246147 [DEBUG] sofia_glue.c:4403 Audio Codec Compare [PCMU:0:8000:0:64000]/[PCMU:0:8000:20:64000]
2011-01-06 15:36:20.246147 [DEBUG] sofia_glue.c:2750 Set Codec sofia/internal/sip:7013 at 192.168.2.42:5060 PCMU/8000 20 ms 160 samples 64000 bits
2011-01-06 15:36:20.247179 [DEBUG] sofia_glue.c:4501 Set 2833 dtmf send payload to 101
2011-01-06 15:36:20.247179 [DEBUG] sofia_glue.c:2980 AUDIO RTP [sofia/internal/sip:7013 at 192.168.2.42:5060] 192.168.2.190 port 29432 -> 192.168.2.42 port 29494 codec: 0 ms: 20
2011-01-06 15:36:20.247179 [DEBUG] switch_rtp.c:1427 Starting timer [soft] 160 bytes per 20ms
2011-01-06 15:36:20.248111 [DEBUG] sofia_glue.c:3221 Set 2833 dtmf send payload to 101
2011-01-06 15:36:20.248111 [DEBUG] sofia_glue.c:3226 Set 2833 dtmf receive payload to 101
2011-01-06 15:36:20.248111 [DEBUG] switch_channel.c:2782 (sofia/internal/sip:7013 at 192.168.2.42:5060) Callstate Change RINGING -> ACTIVE
2011-01-06 15:36:20.248111 [DEBUG] switch_channel.c:2794 Send signal sofia/internal/7001 at 192.168.2.190 [BREAK]
2011-01-06 15:36:20.248111 [NOTICE] sofia.c:5200 Channel [sofia/internal/sip:7013 at 192.168.2.42:5060] has been answered
2011-01-06 15:36:20.248111 [DEBUG] sofia_glue.c:2980 AUDIO RTP [sofia/internal/7001 at 192.168.2.190] 192.168.2.190 port 32534 -> 192.168.2.15 port 6398 codec: 18 ms: 20
2011-01-06 15:36:20.248111 [DEBUG] switch_rtp.c:1427 Starting timer [soft] 160 bytes per 20ms
2011-01-06 15:36:20.249967 [DEBUG] sofia_glue.c:3221 Set 2833 dtmf send payload to 101
2011-01-06 15:36:20.249967 [DEBUG] sofia_glue.c:3226 Set 2833 dtmf receive payload to 101
2011-01-06 15:36:20.249967 [DEBUG] mod_sofia.c:683 Local SDP sofia/internal/7001 at 192.168.2.190:
v=0
o=FreeSWITCH 1294275846 1294275847 IN IP4 192.168.2.190
s=FreeSWITCH
c=IN IP4 192.168.2.190
t=0 0
m=audio 32534 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2011-01-06 15:36:20.249967 [DEBUG] switch_core_session.c:676 Send signal sofia/internal/7001 at 192.168.2.190 [BREAK]
2011-01-06 15:36:20.249967 [DEBUG] switch_channel.c:2782 (sofia/internal/7001 at 192.168.2.190) Callstate Change RINGING -> ACTIVE
2011-01-06 15:36:20.250883 [NOTICE] switch_ivr_originate.c:3317 Channel [sofia/internal/7001 at 192.168.2.190] has been answered
2011-01-06 15:36:20.250883 [DEBUG] switch_ivr_originate.c:3362 Originate Resulted in Success: [sofia/internal/sip:7013 at 192.168.2.42:5060]
2011-01-06 15:36:20.250883 [DEBUG] sofia.c:4616 Channel sofia/internal/7001 at 192.168.2.190 entering state [completed][200]
2011-01-06 15:36:20.250883 [DEBUG] switch_ivr_originate.c:3362 Originate Resulted in Success: [sofia/internal/sip:7013 at 192.168.2.42:5060]
2011-01-06 15:36:20.251763 [DEBUG] switch_core_session.c:676 Send signal sofia/internal/sip:7013 at 192.168.2.42:5060 [BREAK]
2011-01-06 15:36:20.251763 [DEBUG] switch_core_session.c:676 Send signal sofia/internal/7001 at 192.168.2.190 [BREAK]
2011-01-06 15:36:20.251763 [DEBUG] switch_ivr_bridge.c:1234 (sofia/internal/sip:7013 at 192.168.2.42:5060) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2011-01-06 15:36:20.251763 [DEBUG] switch_core_session.c:1083 Send signal sofia/internal/sip:7013 at 192.168.2.42:5060 [BREAK]
2011-01-06 15:36:20.251763 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/sip:7013 at 192.168.2.42:5060) Running State Change CS_EXCHANGE_MEDIA
2011-01-06 15:36:20.251763 [DEBUG] switch_core_state_machine.c:369 (sofia/internal/sip:7013 at 192.168.2.42:5060) State EXCHANGE_MEDIA
2011-01-06 15:36:20.251763 [DEBUG] mod_sofia.c:554 SOFIA EXCHANGE_MEDIA
2011-01-06 15:36:20.252711 [DEBUG] sofia.c:4616 Channel sofia/internal/7001 at 192.168.2.190 entering state [ready][200]
2011-01-06 15:36:20.252711 [DEBUG] switch_core_session.c:738 Send signal sofia/internal/sip:7013 at 192.168.2.42:5060 [BREAK]
2011-01-06 15:36:20.252711 [DEBUG] switch_core_session.c:738 Send signal sofia/internal/7001 at 192.168.2.190 [BREAK]
2011-01-06 15:36:20.293303 [DEBUG] switch_rtp.c:2681 Correct ip/port confirmed.
2011-01-06 15:36:20.293303 [ERR] mod_g729.c:145 This codec is only usable in passthrough mode!
2011-01-06 15:36:20.293303 [ERR] switch_core_io.c:882 Codec G.729 decoder error!
2011-01-06 15:36:20.293303 [DEBUG] switch_ivr_bridge.c:494 sofia/internal/sip:7013 at 192.168.2.42:5060 ending bridge by request from write function
2011-01-06 15:36:20.294237 [DEBUG] switch_ivr_bridge.c:581 BRIDGE THREAD DONE [sofia/internal/7001 at 192.168.2.190]
2011-01-06 15:36:20.294237 [DEBUG] switch_ivr_bridge.c:601 Send signal sofia/internal/sip:7013 at 192.168.2.42:5060 [BREAK]
2011-01-06 15:36:20.294237 [DEBUG] switch_ivr_bridge.c:581 BRIDGE THREAD DONE [sofia/internal/sip:7013 at 192.168.2.42:5060]
2011-01-06 15:36:20.294237 [DEBUG] switch_ivr_bridge.c:601 Send signal sofia/internal/7001 at 192.168.2.190 [BREAK]
2011-01-06 15:36:20.294237 [DEBUG] switch_channel.c:2535 (sofia/internal/sip:7013 at 192.168.2.42:5060) Callstate Change ACTIVE -> HANGUP
2011-01-06 15:36:20.294237 [NOTICE] switch_ivr_bridge.c:653 Hangup sofia/internal/sip:7013 at 192.168.2.42:5060 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2011-01-06 15:36:20.294237 [DEBUG] switch_channel.c:2551 Send signal sofia/internal/sip:7013 at 192.168.2.42:5060 [KILL]
2011-01-06 15:36:20.294237 [DEBUG] switch_core_session.c:1083 Send signal sofia/internal/sip:7013 at 192.168.2.42:5060 [BREAK]
2011-01-06 15:36:20.294237 [DEBUG] switch_core_state_machine.c:369 (sofia/internal/sip:7013 at 192.168.2.42:5060) State EXCHANGE_MEDIA going to sleep
2011-01-06 15:36:20.294237 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/sip:7013 at 192.168.2.42:5060) Running State Change CS_HANGUP
2011-01-06 15:36:20.294237 [DEBUG] switch_core_state_machine.c:557 (sofia/internal/sip:7013 at 192.168.2.42:5060) State HANGUP
2011-01-06 15:36:20.295216 [DEBUG] mod_sofia.c:459 Channel sofia/internal/sip:7013 at 192.168.2.42:5060 hanging up, cause: NORMAL_CLEARING
2011-01-06 15:36:20.295216 [DEBUG] switch_ivr_bridge.c:1305 sofia/internal/sip:7013 at 192.168.2.42:5060 skip receive message [UNBRIDGE] (channel is hungup already)
2011-01-06 15:36:20.295216 [DEBUG] switch_core_session.c:676 Send signal sofia/internal/7001 at 192.168.2.190 [BREAK]
2011-01-06 15:36:20.295216 [DEBUG] switch_channel.c:2535 (sofia/internal/7001 at 192.168.2.190) Callstate Change ACTIVE -> HANGUP
2011-01-06 15:36:20.295216 [NOTICE] switch_ivr_bridge.c:1328 Hangup sofia/internal/7001 at 192.168.2.190 [CS_EXECUTE] [NORMAL_CLEARING]
2011-01-06 15:36:20.295216 [DEBUG] switch_channel.c:2551 Send signal sofia/internal/7001 at 192.168.2.190 [KILL]
2011-01-06 15:36:20.295216 [DEBUG] switch_core_session.c:1083 Send signal sofia/internal/7001 at 192.168.2.190 [BREAK]
2011-01-06 15:36:20.295216 [DEBUG] switch_core_session.c:2012 sofia/internal/7001 at 192.168.2.190 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2011-01-06 15:36:20.295216 [DEBUG] switch_core_state_machine.c:366 (sofia/internal/7001 at 192.168.2.190) State EXECUTE going to sleep
2011-01-06 15:36:20.295216 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/7001 at 192.168.2.190) Running State Change CS_HANGUP
2011-01-06 15:36:20.296200 [DEBUG] switch_core_state_machine.c:557 (sofia/internal/7001 at 192.168.2.190) State HANGUP
2011-01-06 15:36:20.296200 [DEBUG] mod_sofia.c:459 Channel sofia/internal/7001 at 192.168.2.190 hanging up, cause: NORMAL_CLEARING
2011-01-06 15:36:20.296200 [DEBUG] mod_sofia.c:502 Sending BYE to sofia/internal/sip:7013 at 192.168.2.42:5060
2011-01-06 15:36:20.297185 [DEBUG] mod_sofia.c:502 Sending BYE to sofia/internal/7001 at 192.168.2.190
2011-01-06 15:36:20.297185 [DEBUG] switch_core_state_machine.c:46 sofia/internal/sip:7013 at 192.168.2.42:5060 Standard HANGUP, cause: NORMAL_CLEARING
2011-01-06 15:36:20.297185 [DEBUG] switch_core_state_machine.c:557 (sofia/internal/sip:7013 at 192.168.2.42:5060) State HANGUP going to sleep
2011-01-06 15:36:20.297185 [DEBUG] switch_core_state_machine.c:351 (sofia/internal/sip:7013 at 192.168.2.42:5060) State Change CS_HANGUP -> CS_REPORTING
2011-01-06 15:36:20.297185 [DEBUG] switch_core_session.c:1083 Send signal sofia/internal/sip:7013 at 192.168.2.42:5060 [BREAK]
2011-01-06 15:36:20.297185 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/sip:7013 at 192.168.2.42:5060) Running State Change CS_REPORTING
2011-01-06 15:36:20.297185 [DEBUG] switch_core_state_machine.c:617 (sofia/internal/sip:7013 at 192.168.2.42:5060) State REPORTING
2011-01-06 15:36:20.297185 [DEBUG] switch_core_state_machine.c:53 sofia/internal/sip:7013 at 192.168.2.42:5060 Standard REPORTING, cause: NORMAL_CLEARING
2011-01-06 15:36:20.297185 [DEBUG] switch_core_state_machine.c:617 (sofia/internal/sip:7013 at 192.168.2.42:5060) State REPORTING going to sleep
2011-01-06 15:36:20.297185 [DEBUG] switch_core_state_machine.c:345 (sofia/internal/sip:7013 at 192.168.2.42:5060) State Change CS_REPORTING -> CS_DESTROY
2011-01-06 15:36:20.297185 [DEBUG] switch_core_session.c:1083 Send signal sofia/internal/sip:7013 at 192.168.2.42:5060 [BREAK]
2011-01-06 15:36:20.297185 [DEBUG] switch_core_session.c:1255 Session 55 (sofia/internal/sip:7013 at 192.168.2.42:5060) Locked, Waiting on external entities
2011-01-06 15:36:20.297185 [NOTICE] switch_core_session.c:1273 Session 55 (sofia/internal/sip:7013 at 192.168.2.42:5060) Ended
2011-01-06 15:36:20.297185 [NOTICE] switch_core_session.c:1275 Close Channel sofia/internal/sip:7013 at 192.168.2.42:5060 [CS_DESTROY]
2011-01-06 15:36:20.297185 [DEBUG] switch_core_state_machine.c:46 sofia/internal/7001 at 192.168.2.190 Standard HANGUP, cause: NORMAL_CLEARING
2011-01-06 15:36:20.297185 [DEBUG] switch_core_state_machine.c:557 (sofia/internal/7001 at 192.168.2.190) State HANGUP going to sleep
2011-01-06 15:36:20.297185 [DEBUG] switch_core_state_machine.c:449 (sofia/internal/sip:7013 at 192.168.2.42:5060) Callstate Change HANGUP -> DOWN
2011-01-06 15:36:20.297185 [DEBUG] switch_core_state_machine.c:351 (sofia/internal/7001 at 192.168.2.190) State Change CS_HANGUP -> CS_REPORTING
2011-01-06 15:36:20.297185 [DEBUG] switch_core_session.c:1083 Send signal sofia/internal/7001 at 192.168.2.190 [BREAK]
2011-01-06 15:36:20.297185 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/7001 at 192.168.2.190) Running State Change CS_REPORTING
2011-01-06 15:36:20.297185 [DEBUG] switch_core_state_machine.c:452 (sofia/internal/sip:7013 at 192.168.2.42:5060) Running State Change CS_DESTROY
2011-01-06 15:36:20.297185 [DEBUG] switch_core_state_machine.c:617 (sofia/internal/7001 at 192.168.2.190) State REPORTING
2011-01-06 15:36:20.297185 [DEBUG] switch_core_state_machine.c:462 (sofia/internal/sip:7013 at 192.168.2.42:5060) State DESTROY
2011-01-06 15:36:20.297185 [DEBUG] mod_sofia.c:364 sofia/internal/sip:7013 at 192.168.2.42:5060 SOFIA DESTROY
2011-01-06 15:36:20.297185 [DEBUG] switch_core_state_machine.c:60 sofia/internal/sip:7013 at 192.168.2.42:5060 Standard DESTROY
2011-01-06 15:36:20.297185 [DEBUG] switch_core_state_machine.c:462 (sofia/internal/sip:7013 at 192.168.2.42:5060) State DESTROY going to sleep
2011-01-06 15:36:20.297185 [DEBUG] switch_core_state_machine.c:53 sofia/internal/7001 at 192.168.2.190 Standard REPORTING, cause: NORMAL_CLEARING
2011-01-06 15:36:20.297185 [DEBUG] switch_core_state_machine.c:617 (sofia/internal/7001 at 192.168.2.190) State REPORTING going to sleep
2011-01-06 15:36:20.298177 [DEBUG] switch_core_state_machine.c:345 (sofia/internal/7001 at 192.168.2.190) State Change CS_REPORTING -> CS_DESTROY
2011-01-06 15:36:20.298177 [DEBUG] switch_core_session.c:1083 Send signal sofia/internal/7001 at 192.168.2.190 [BREAK]
2011-01-06 15:36:20.298177 [DEBUG] switch_core_session.c:1255 Session 54 (sofia/internal/7001 at 192.168.2.190) Locked, Waiting on external entities
2011-01-06 15:36:20.298177 [NOTICE] switch_core_session.c:1273 Session 54 (sofia/internal/7001 at 192.168.2.190) Ended
2011-01-06 15:36:20.298177 [NOTICE] switch_core_session.c:1275 Close Channel sofia/internal/7001 at 192.168.2.190 [CS_DESTROY]
2011-01-06 15:36:20.298177 [DEBUG] switch_core_state_machine.c:449 (sofia/internal/7001 at 192.168.2.190) Callstate Change HANGUP -> DOWN
2011-01-06 15:36:20.298177 [DEBUG] switch_core_state_machine.c:452 (sofia/internal/7001 at 192.168.2.190) Running State Change CS_DESTROY
2011-01-06 15:36:20.298177 [DEBUG] switch_core_state_machine.c:462 (sofia/internal/7001 at 192.168.2.190) State DESTROY
2011-01-06 15:36:20.298177 [DEBUG] mod_sofia.c:364 sofia/internal/7001 at 192.168.2.190 SOFIA DESTROY
2011-01-06 15:36:20.298177 [DEBUG] switch_core_state_machine.c:60 sofia/internal/7001 at 192.168.2.190 Standard DESTROY
2011-01-06 15:36:20.298177 [DEBUG] switch_core_state_machine.c:462 (sofia/internal/7001 at 192.168.2.190) State DESTROY going to sleep


From steveayre at gmail.com  Thu Jan  6 13:30:28 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 6 Jan 2011 10:30:28 +0000
Subject: [Freeswitch-users] codec error
In-Reply-To: <AANLkTikmHcMHfc3ydk+8b2SWEKB+sR=Awj0OVri6qdrK@mail.gmail.com>
References: <AANLkTi=cKTUOcN--hZm2yLXY8s-pWotek+_YA1Bxw2jw@mail.gmail.com>
	<AANLkTinLm5AP9kqoeHarqrCJGjsUq6Xo8=aOnY05HNHN@mail.gmail.com>
	<AANLkTikmHcMHfc3ydk+8b2SWEKB+sR=Awj0OVri6qdrK@mail.gmail.com>
Message-ID: <AANLkTi=e4sVBpb2M5HSni-1ZXViExxkxocvmnC=jBBi_@mail.gmail.com>

G7222 call fails with incompatible destination because it's not
enabled on the server.

G729 call starts a bridge to user/7013 at 192.168.2.190. They only offer
G711 ulaw. That means it's a G729 -> G711 call requiring transcoding.
There is no way that can work using mod_g729.

If you believe that that user has G729 enabled, repeat the test with
sip trace enabled.
'sofia global siptrace on'

That'll let you see the INVITE w/SDP sent to the user. If G729 isn't
in the outgoing SDP it's a config problem on FreeSWITCH. If G729 is in
the SDP and the user is only responding with G711 then it's a problem
on the Cisco endpoint.

-Steve



On 6 January 2011 10:10, Sam <u2nsam at gmail.com> wrote:
> Hi,
>
> This was working earlier and the config file have not changed , only
> upgraded to latest git.
> have attached the file codec.txt .
>
>
>
> Regards
> Sam
>
>
>
> On Thu, Jan 6, 2011 at 3:20 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> Some more information please...
>>
>> G729 has patents that mean it can't be used unlicensed. FreeSWITCH
>> either has mod_com_g729 for a full featured licensed version, or
>> mod_g729 which operates in passthrough mode for G729-G729 calls (it
>> needs no license since for these calls there is no encoding/decoding
>> step, the already encoded data is just passed straight through). You
>> will never have had mod_g729 working in the past for a transcoding
>> call, so that won't have changed.
>>
>> I would say that it's either:
>> - Your config files have changed
>> - The endpoints are offering different codecs from before
>> - Something changed in git about the codec negotiation
>>
>> Can you pastebin a debug level log of the calls, and enable siptrace?
>> Those will show what codecs are being offered, selected and show the
>> codec negotiation. It'd also be useful to know what your sip profile
>> config files look like since there are several options that adjust how
>> the negotiation is done.
>>
>> -Steve
>>
>>
>>
>> On 6 January 2011 09:43, Sam <u2nsam at gmail.com> wrote:
>> > I am getting the error after installing latest git from freeswitch,
>> > where as
>> > it was working from old git
>> >
>> >
>> > 2011-01-06 15:10:19.113781 [ERR] mod_g729.c:102 This codec is only
>> > usable in
>> > passthrough mode!
>> > 2011-01-06 15:10:19.113781 [ERR] switch_core_io.c:1042 Codec G.729
>> > encoder
>> > error!
>> >
>> >
>> > Regards
>> > Sam
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From u2nsam at gmail.com  Thu Jan  6 13:45:47 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 6 Jan 2011 16:15:47 +0530
Subject: [Freeswitch-users] codec error
In-Reply-To: <AANLkTi=e4sVBpb2M5HSni-1ZXViExxkxocvmnC=jBBi_@mail.gmail.com>
References: <AANLkTi=cKTUOcN--hZm2yLXY8s-pWotek+_YA1Bxw2jw@mail.gmail.com>
	<AANLkTinLm5AP9kqoeHarqrCJGjsUq6Xo8=aOnY05HNHN@mail.gmail.com>
	<AANLkTikmHcMHfc3ydk+8b2SWEKB+sR=Awj0OVri6qdrK@mail.gmail.com>
	<AANLkTi=e4sVBpb2M5HSni-1ZXViExxkxocvmnC=jBBi_@mail.gmail.com>
Message-ID: <AANLkTikGM8fQzqXaFoOd=m3LaVLae6B+O7qBPVAYmzL2@mail.gmail.com>

http://pastebin.freeswitch.org/14934
http://pastebin.freeswitch.org/14935

These are the paste bins where calls with g722 and G729 call fails but g729
calls rings and gets disconnected.
In past it was working with both the codec on the same server as no there is
no config change.


Regds
Sam



On Thu, Jan 6, 2011 at 4:00 PM, Steven Ayre <steveayre at gmail.com> wrote:

> G7222 call fails with incompatible destination because it's not
> enabled on the server.
>
> G729 call starts a bridge to user/7013 at 192.168.2.190. They only offer
> G711 ulaw. That means it's a G729 -> G711 call requiring transcoding.
> There is no way that can work using mod_g729.
>
> If you believe that that user has G729 enabled, repeat the test with
> sip trace enabled.
> 'sofia global siptrace on'
>
> That'll let you see the INVITE w/SDP sent to the user. If G729 isn't
> in the outgoing SDP it's a config problem on FreeSWITCH. If G729 is in
> the SDP and the user is only responding with G711 then it's a problem
> on the Cisco endpoint.
>
> -Steve
>
>
>
> On 6 January 2011 10:10, Sam <u2nsam at gmail.com> wrote:
> > Hi,
> >
> > This was working earlier and the config file have not changed , only
> > upgraded to latest git.
> > have attached the file codec.txt .
> >
> >
> >
> > Regards
> > Sam
> >
> >
> >
> > On Thu, Jan 6, 2011 at 3:20 PM, Steven Ayre <steveayre at gmail.com> wrote:
> >>
> >> Some more information please...
> >>
> >> G729 has patents that mean it can't be used unlicensed. FreeSWITCH
> >> either has mod_com_g729 for a full featured licensed version, or
> >> mod_g729 which operates in passthrough mode for G729-G729 calls (it
> >> needs no license since for these calls there is no encoding/decoding
> >> step, the already encoded data is just passed straight through). You
> >> will never have had mod_g729 working in the past for a transcoding
> >> call, so that won't have changed.
> >>
> >> I would say that it's either:
> >> - Your config files have changed
> >> - The endpoints are offering different codecs from before
> >> - Something changed in git about the codec negotiation
> >>
> >> Can you pastebin a debug level log of the calls, and enable siptrace?
> >> Those will show what codecs are being offered, selected and show the
> >> codec negotiation. It'd also be useful to know what your sip profile
> >> config files look like since there are several options that adjust how
> >> the negotiation is done.
> >>
> >> -Steve
> >>
> >>
> >>
> >> On 6 January 2011 09:43, Sam <u2nsam at gmail.com> wrote:
> >> > I am getting the error after installing latest git from freeswitch,
> >> > where as
> >> > it was working from old git
> >> >
> >> >
> >> > 2011-01-06 15:10:19.113781 [ERR] mod_g729.c:102 This codec is only
> >> > usable in
> >> > passthrough mode!
> >> > 2011-01-06 15:10:19.113781 [ERR] switch_core_io.c:1042 Codec G.729
> >> > encoder
> >> > error!
> >> >
> >> >
> >> > Regards
> >> > Sam
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Thu Jan  6 13:59:23 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 6 Jan 2011 10:59:23 +0000
Subject: [Freeswitch-users] codec error
In-Reply-To: <AANLkTikGM8fQzqXaFoOd=m3LaVLae6B+O7qBPVAYmzL2@mail.gmail.com>
References: <AANLkTi=cKTUOcN--hZm2yLXY8s-pWotek+_YA1Bxw2jw@mail.gmail.com>
	<AANLkTinLm5AP9kqoeHarqrCJGjsUq6Xo8=aOnY05HNHN@mail.gmail.com>
	<AANLkTikmHcMHfc3ydk+8b2SWEKB+sR=Awj0OVri6qdrK@mail.gmail.com>
	<AANLkTi=e4sVBpb2M5HSni-1ZXViExxkxocvmnC=jBBi_@mail.gmail.com>
	<AANLkTikGM8fQzqXaFoOd=m3LaVLae6B+O7qBPVAYmzL2@mail.gmail.com>
Message-ID: <AANLkTim8Bfsr7=hX=RBJJEo4TKZqUZgn5y7yUbh9d1k1@mail.gmail.com>

G722.2
======

This codec is not enabled on your server.

2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
Compare [g7222:110:16000:110:0]/[G7221:115:32000:20:48000]
2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
Compare [g7222:110:16000:110:0]/[G7221:107:16000:20:32000]
2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
Compare [g7222:110:16000:110:0]/[PCMU:0:8000:20:64000]
2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
Compare [g7222:110:16000:110:0]/[PCMA:8:8000:20:64000]
2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
Compare [g7222:110:16000:110:0]/[G729:18:8000:20:8000]

You are calling with G722.2.

G722.1 is enabled, but G722.2 is not.

G729
====

It rings because Cisco returns 180 Ringing with no SDP. At that time
no codec has been selected. Anything you hear will be generated by
your SIP client.

The codec is selected in the 200 OK w/SDP. That is the time the codec is picked.

FreeSWITCH offers G729 to Cisco in the INVITE, plus several other codecs:
   m=audio 24636 RTP/AVP 18 98 99 0 8 101 13
The Cisco only responds with PCMU:
   m=audio 31600 RTP/AVP 0 101
( 18=G729 0=PCMU 8=PCMA 98/99=G722.1 )

Either your FS config has changed so that is offering more codecs than
before, or the Cisco's config has changed to prefer G711 over G729.

-Steve


On 6 January 2011 10:45, Sam <u2nsam at gmail.com> wrote:
> http://pastebin.freeswitch.org/14934
> http://pastebin.freeswitch.org/14935
>
> These are the paste bins where calls with g722 and G729 call fails but g729
> calls rings and gets disconnected.
> In past it was working with both the codec on the same server as no there is
> no config change.
>
>
> Regds
> Sam
>
>
>
> On Thu, Jan 6, 2011 at 4:00 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> G7222 call fails with incompatible destination because it's not
>> enabled on the server.
>>
>> G729 call starts a bridge to user/7013 at 192.168.2.190. They only offer
>> G711 ulaw. That means it's a G729 -> G711 call requiring transcoding.
>> There is no way that can work using mod_g729.
>>
>> If you believe that that user has G729 enabled, repeat the test with
>> sip trace enabled.
>> 'sofia global siptrace on'
>>
>> That'll let you see the INVITE w/SDP sent to the user. If G729 isn't
>> in the outgoing SDP it's a config problem on FreeSWITCH. If G729 is in
>> the SDP and the user is only responding with G711 then it's a problem
>> on the Cisco endpoint.
>>
>> -Steve
>>
>>
>>
>> On 6 January 2011 10:10, Sam <u2nsam at gmail.com> wrote:
>> > Hi,
>> >
>> > This was working earlier and the config file have not changed , only
>> > upgraded to latest git.
>> > have attached the file codec.txt .
>> >
>> >
>> >
>> > Regards
>> > Sam
>> >
>> >
>> >
>> > On Thu, Jan 6, 2011 at 3:20 PM, Steven Ayre <steveayre at gmail.com> wrote:
>> >>
>> >> Some more information please...
>> >>
>> >> G729 has patents that mean it can't be used unlicensed. FreeSWITCH
>> >> either has mod_com_g729 for a full featured licensed version, or
>> >> mod_g729 which operates in passthrough mode for G729-G729 calls (it
>> >> needs no license since for these calls there is no encoding/decoding
>> >> step, the already encoded data is just passed straight through). You
>> >> will never have had mod_g729 working in the past for a transcoding
>> >> call, so that won't have changed.
>> >>
>> >> I would say that it's either:
>> >> - Your config files have changed
>> >> - The endpoints are offering different codecs from before
>> >> - Something changed in git about the codec negotiation
>> >>
>> >> Can you pastebin a debug level log of the calls, and enable siptrace?
>> >> Those will show what codecs are being offered, selected and show the
>> >> codec negotiation. It'd also be useful to know what your sip profile
>> >> config files look like since there are several options that adjust how
>> >> the negotiation is done.
>> >>
>> >> -Steve
>> >>
>> >>
>> >>
>> >> On 6 January 2011 09:43, Sam <u2nsam at gmail.com> wrote:
>> >> > I am getting the error after installing latest git from freeswitch,
>> >> > where as
>> >> > it was working from old git
>> >> >
>> >> >
>> >> > 2011-01-06 15:10:19.113781 [ERR] mod_g729.c:102 This codec is only
>> >> > usable in
>> >> > passthrough mode!
>> >> > 2011-01-06 15:10:19.113781 [ERR] switch_core_io.c:1042 Codec G.729
>> >> > encoder
>> >> > error!
>> >> >
>> >> >
>> >> > Regards
>> >> > Sam
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steveayre at gmail.com  Thu Jan  6 14:04:14 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 6 Jan 2011 11:04:14 +0000
Subject: [Freeswitch-users] codec error
In-Reply-To: <AANLkTim8Bfsr7=hX=RBJJEo4TKZqUZgn5y7yUbh9d1k1@mail.gmail.com>
References: <AANLkTi=cKTUOcN--hZm2yLXY8s-pWotek+_YA1Bxw2jw@mail.gmail.com>
	<AANLkTinLm5AP9kqoeHarqrCJGjsUq6Xo8=aOnY05HNHN@mail.gmail.com>
	<AANLkTikmHcMHfc3ydk+8b2SWEKB+sR=Awj0OVri6qdrK@mail.gmail.com>
	<AANLkTi=e4sVBpb2M5HSni-1ZXViExxkxocvmnC=jBBi_@mail.gmail.com>
	<AANLkTikGM8fQzqXaFoOd=m3LaVLae6B+O7qBPVAYmzL2@mail.gmail.com>
	<AANLkTim8Bfsr7=hX=RBJJEo4TKZqUZgn5y7yUbh9d1k1@mail.gmail.com>
Message-ID: <AANLkTi=KdMjptEK5_sAWZ4DcXU1zJ9DstXnA_29rEx_k@mail.gmail.com>

This sip profile parameter would allow the G729 call to work,
providing G729 is enabled on the Cisco:
<param name="disable-transcoding" value="true"/>

This will mean the INVITE sent to the Cisco on the bleg only includes
the codec from the aleg. The aleg is already using G729, so the Cisco
will only be offered G729. It is then forced to either accept G729 or
fail the call with Incompatible Destination.

-Steve


On 6 January 2011 10:59, Steven Ayre <steveayre at gmail.com> wrote:
> G722.2
> ======
>
> This codec is not enabled on your server.
>
> 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
> Compare [g7222:110:16000:110:0]/[G7221:115:32000:20:48000]
> 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
> Compare [g7222:110:16000:110:0]/[G7221:107:16000:20:32000]
> 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
> Compare [g7222:110:16000:110:0]/[PCMU:0:8000:20:64000]
> 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
> Compare [g7222:110:16000:110:0]/[PCMA:8:8000:20:64000]
> 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
> Compare [g7222:110:16000:110:0]/[G729:18:8000:20:8000]
>
> You are calling with G722.2.
>
> G722.1 is enabled, but G722.2 is not.
>
> G729
> ====
>
> It rings because Cisco returns 180 Ringing with no SDP. At that time
> no codec has been selected. Anything you hear will be generated by
> your SIP client.
>
> The codec is selected in the 200 OK w/SDP. That is the time the codec is picked.
>
> FreeSWITCH offers G729 to Cisco in the INVITE, plus several other codecs:
> ? m=audio 24636 RTP/AVP 18 98 99 0 8 101 13
> The Cisco only responds with PCMU:
> ? m=audio 31600 RTP/AVP 0 101
> ( 18=G729 0=PCMU 8=PCMA 98/99=G722.1 )
>
> Either your FS config has changed so that is offering more codecs than
> before, or the Cisco's config has changed to prefer G711 over G729.
>
> -Steve
>
>
> On 6 January 2011 10:45, Sam <u2nsam at gmail.com> wrote:
>> http://pastebin.freeswitch.org/14934
>> http://pastebin.freeswitch.org/14935
>>
>> These are the paste bins where calls with g722 and G729 call fails but g729
>> calls rings and gets disconnected.
>> In past it was working with both the codec on the same server as no there is
>> no config change.
>>
>>
>> Regds
>> Sam
>>
>>
>>
>> On Thu, Jan 6, 2011 at 4:00 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>>
>>> G7222 call fails with incompatible destination because it's not
>>> enabled on the server.
>>>
>>> G729 call starts a bridge to user/7013 at 192.168.2.190. They only offer
>>> G711 ulaw. That means it's a G729 -> G711 call requiring transcoding.
>>> There is no way that can work using mod_g729.
>>>
>>> If you believe that that user has G729 enabled, repeat the test with
>>> sip trace enabled.
>>> 'sofia global siptrace on'
>>>
>>> That'll let you see the INVITE w/SDP sent to the user. If G729 isn't
>>> in the outgoing SDP it's a config problem on FreeSWITCH. If G729 is in
>>> the SDP and the user is only responding with G711 then it's a problem
>>> on the Cisco endpoint.
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 6 January 2011 10:10, Sam <u2nsam at gmail.com> wrote:
>>> > Hi,
>>> >
>>> > This was working earlier and the config file have not changed , only
>>> > upgraded to latest git.
>>> > have attached the file codec.txt .
>>> >
>>> >
>>> >
>>> > Regards
>>> > Sam
>>> >
>>> >
>>> >
>>> > On Thu, Jan 6, 2011 at 3:20 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>> >>
>>> >> Some more information please...
>>> >>
>>> >> G729 has patents that mean it can't be used unlicensed. FreeSWITCH
>>> >> either has mod_com_g729 for a full featured licensed version, or
>>> >> mod_g729 which operates in passthrough mode for G729-G729 calls (it
>>> >> needs no license since for these calls there is no encoding/decoding
>>> >> step, the already encoded data is just passed straight through). You
>>> >> will never have had mod_g729 working in the past for a transcoding
>>> >> call, so that won't have changed.
>>> >>
>>> >> I would say that it's either:
>>> >> - Your config files have changed
>>> >> - The endpoints are offering different codecs from before
>>> >> - Something changed in git about the codec negotiation
>>> >>
>>> >> Can you pastebin a debug level log of the calls, and enable siptrace?
>>> >> Those will show what codecs are being offered, selected and show the
>>> >> codec negotiation. It'd also be useful to know what your sip profile
>>> >> config files look like since there are several options that adjust how
>>> >> the negotiation is done.
>>> >>
>>> >> -Steve
>>> >>
>>> >>
>>> >>
>>> >> On 6 January 2011 09:43, Sam <u2nsam at gmail.com> wrote:
>>> >> > I am getting the error after installing latest git from freeswitch,
>>> >> > where as
>>> >> > it was working from old git
>>> >> >
>>> >> >
>>> >> > 2011-01-06 15:10:19.113781 [ERR] mod_g729.c:102 This codec is only
>>> >> > usable in
>>> >> > passthrough mode!
>>> >> > 2011-01-06 15:10:19.113781 [ERR] switch_core_io.c:1042 Codec G.729
>>> >> > encoder
>>> >> > error!
>>> >> >
>>> >> >
>>> >> > Regards
>>> >> > Sam
>>> >> >
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >
>>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> > http://www.freeswitch.org
>>> >> >
>>> >> >
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>


From ovvenkatesan at gmail.com  Thu Jan  6 14:32:04 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Thu, 6 Jan 2011 17:02:04 +0530
Subject: [Freeswitch-users] hardware requirements
Message-ID: <AANLkTi=eXk=tRHK+Y8McrnUdYGvqHXthyYEOk3P0TJnZ@mail.gmail.com>

Hi to all,

I have to handle 72,000 inbound calls per day ( 6 - 8 hours ),
Anyone please suggest me the
hardware requirement for the same.

Like, What kind of Server and how many PRI or BRI cards will fulfill the
needs.

* Its Simple IVR, Call will be landing from mobile phones.


-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
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From steveayre at gmail.com  Thu Jan  6 14:42:27 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 6 Jan 2011 11:42:27 +0000
Subject: [Freeswitch-users] hardware requirements
In-Reply-To: <AANLkTi=eXk=tRHK+Y8McrnUdYGvqHXthyYEOk3P0TJnZ@mail.gmail.com>
References: <AANLkTi=eXk=tRHK+Y8McrnUdYGvqHXthyYEOk3P0TJnZ@mail.gmail.com>
Message-ID: <AANLkTinU_AqicisxSjOS-SJDGgfSpr1DYEeDZxQLmYp5@mail.gmail.com>

What do you expect the peak CPS and number of concurrent calls to be?

Any current Xeon server should be able to do 3cps fine (72000 in 6
hours). What really will dictate your hardware requirements is what
the maximum load you want to handle at any one time is.

-Steve


On 6 January 2011 11:32, ovvenkat <ovvenkatesan at gmail.com> wrote:
> Hi to all,
>
> I have to handle 72,000 inbound calls per day ( 6 - 8 hours ),
> Anyone please suggest me the
> hardware requirement for the same.
>
> Like, What kind of Server and how many PRI or BRI cards will fulfill the
> needs.
>
> * Its Simple IVR, Call will be landing from mobile phones.
>
>
> --
>
> If you have come to help me, you are wasting your time.
> If you have come to because your liberation is bound up in mine, we can work
> together.
>
>
> Regards
> Venkatesan OV.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From dyatsin at sangoma.com  Thu Jan  6 15:24:14 2011
From: dyatsin at sangoma.com (David Yat Sin)
Date: Thu, 06 Jan 2011 07:24:14 -0500
Subject: [Freeswitch-users]  Error in log with latest FreeTDM
In-Reply-To: <AANLkTik+pv3C3svaCRDUBE4P8OYGHj30tJ+q-1bqO9E2@mail.gmail.com>
References: <AANLkTik+pv3C3svaCRDUBE4P8OYGHj30tJ+q-1bqO9E2@mail.gmail.com>
Message-ID: <4D25B46E.1000309@sangoma.com>

Hi Stephen,
Can you email me your full FS log before getting this message.

Thanks

*David Yat Sin, BEng*
*/Senior Software Engineer/*
Sangoma Technologies
100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada

	



t.   +1 800 388 2475 x119
t.   +1 905 474 1990 x119
f.   +1 905 474 9223

	

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On 1/6/2011 4:13 AM, Stephen Wilde wrote:
> I have tried to update to the latest git my Freeswitch but I have due 
> to revert to the previous version because with latest one 
> (96ac90adce931a3a28c768e102b863637c8ba98d, Jan 5 16:55:06 2011 + 
> Sangoma ISDN library 7.0.0) I have many warning errors and some 
> critical errors in the log.
> No problem with my previous version 
> (715d250e171a94736b19019ac742f739899ad997, Dec 15 21:29:52 2010 + 
> Sangoma ISDN library 6.0.0).
>
> Here an extraction of my log file:
>
> 2011-01-06 02:01:25.990974 [WARNING] mod_freetdm.c:434 [s23c22][23:22] 
> Why bother changing state from PROCEED to PROCEED
> 2011-01-06 02:01:25.994018 [CRIT] ftmod_sangoma_isdn_stack_rcv.c:988 
> sng_isdn->
>
> date: 01/06/2011 time: 02:01:25
> 2011-01-06 02:01:25.994018 [CRIT] ftmod_sangoma_isdn_stack_rcv.c:988 
> sng_isdn->
> mtss(posix): sw error:  ent: 010  inst: 000  proc id: 001
> file: /usr/src/libsng_trillium-build/libsng_isdn/trillium/in/in_bdy1.c 
> line: 2144 errcode: 46909632820939 errcls: ERRCLS_DEBUG
> errval: 00001  errdesc: InUiIntConReq() Failed.
> 2011-01-06 02:01:25.994018 [WARNING] 
> ftmod_sangoma_isdn_stack_hndl.c:862 [s10c1][10:1] STATUS CONFIRM 
> (call_state:0 channel-state:DIALING cause:100) (suId:1 suInstId:57 
> spInstId:0)
> 2011-01-06 02:01:31.068159 [WARNING] mod_freetdm.c:434 [s22c9][22:9] 
> Why bother changing state from PROCEED to PROCEED
> 2011-01-06 02:01:31.172106 [WARNING] ftdm_io.c:2250 [s22c9][22:9] 
> Cannot indicate PROGRESS_MEDIA in channel with indication PROCEED 
> still pending in state PROCEED
> 2011-01-06 02:01:37.465176 [WARNING] mod_freetdm.c:434 [s3c28][3:28] 
> Why bother changing state from PROCEED to PROCEED
> 2011-01-06 02:01:37.472130 [WARNING] ftdm_io.c:2250 [s3c28][3:28] 
> Cannot indicate RINGING in channel with indication PROCEED still 
> pending in state PROCEED
> 2011-01-06 02:01:43.705155 [WARNING] mod_freetdm.c:434 [s23c23][23:23] 
> Why bother changing state from PROCEED to PROCEED
> 2011-01-06 02:01:43.707124 [CRIT] ftmod_sangoma_isdn_stack_rcv.c:988 
> sng_isdn->
>
> date: 01/06/2011 time: 02:01:43
> 2011-01-06 02:01:43.707124 [CRIT] ftmod_sangoma_isdn_stack_rcv.c:988 
> sng_isdn->
> mtss(posix): sw error:  ent: 010  inst: 000  proc id: 001
> file: /usr/src/libsng_trillium-build/libsng_isdn/trillium/in/in_bdy1.c 
> line: 2144 errcode: 46909632820939 errcls: ERRCLS_DEBUG
> errval: 00001  errdesc: InUiIntConReq() Failed.
> 2011-01-06 02:01:43.707124 [WARNING] 
> ftmod_sangoma_isdn_stack_hndl.c:862 [s5c1][5:1] STATUS CONFIRM 
> (call_state:0 channel-state:DIALING cause:100) (suId:1 suInstId:61 
> spInstId:0)
> 2011-01-06 02:01:49.463158 [WARNING] mod_freetdm.c:434 [s16c14][16:14] 
> Why bother changing state from PROCEED to PROCEED
> 2011-01-06 02:01:49.470188 [WARNING] ftdm_io.c:2250 [s16c14][16:14] 
> Cannot indicate RINGING in channel with indication PROCEED still 
> pending in state PROCEED
> 2011-01-06 02:01:53.938264 [WARNING] mod_freetdm.c:434 [s1c1][1:1] Why 
> bother changing state from PROCEED to PROCEED
> 2011-01-06 02:01:53.944202 [WARNING] ftdm_io.c:2250 [s1c1][1:1] Cannot 
> indicate RINGING in channel with indication PROCEED still pending in 
> state PROCEED
> 2011-01-06 02:01:59.468295 [WARNING] mod_freetdm.c:434 [s8c3][8:3] Why 
> bother changing state from PROCEED to PROCEED
> 2011-01-06 02:01:59.476270 [WARNING] ftdm_io.c:2250 [s8c3][8:3] Cannot 
> indicate RINGING in channel with indication PROCEED still pending in 
> state PROCEED
> 2011-01-06 02:02:01.057245 [WARNING] mod_freetdm.c:434 [s1c2][1:2] Why 
> bother changing state from PROCEED to PROCEED
> 2011-01-06 02:02:01.083313 [WARNING] ftdm_io.c:2250 [s1c2][1:2] Cannot 
> indicate PROGRESS_MEDIA in channel with indication PROCEED still 
> pending in state PROCEED
> 2011-01-06 02:02:04.286328 [WARNING] mod_freetdm.c:434 [s20c24][20:24] 
> Why bother changing state from PROCEED to PROCEED
> 2011-01-06 02:02:14.525323 [WARNING] mod_freetdm.c:434 [s17c23][17:23] 
> Why bother changing state from PROCEED to PROCEED
> 2011-01-06 02:02:15.029458 [WARNING] ftdm_io.c:2250 [s17c23][17:23] 
> Cannot indicate PROGRESS_MEDIA in channel with indication PROCEED 
> still pending in state PROCEED
> 2011-01-06 02:02:19.789411 [WARNING] mod_freetdm.c:434 [s12c5][12:5] 
> Why bother changing state from PROCEED to PROCEED
> 2011-01-06 02:02:19.891438 [WARNING] ftdm_io.c:2250 [s12c5][12:5] 
> Cannot indicate PROGRESS_MEDIA in channel with indication PROCEED 
> still pending in state PROCEED
> 2011-01-06 02:02:23.764507 [WARNING] mod_freetdm.c:434 [s22c10][22:10] 
> Why bother changing state from PROCEED to PROCEED
> 2011-01-06 02:02:26.462538 [WARNING] mod_freetdm.c:434 [s19c24][19:24] 
> Why bother changing state from PROCEED to PROCEED
> 2011-01-06 02:02:29.171509 [WARNING] ftdm_io.c:2250 [s22c10][22:10] 
> Cannot indicate PROGRESS_MEDIA in channel with indication PROCEED 
> still pending in state PROCEED
> 2011-01-06 02:02:31.698471 [WARNING] mod_freetdm.c:434 [s23c24][23:24] 
> Why bother changing state from PROCEED to PROCEED
> 2011-01-06 02:02:31.724623 [WARNING] ftdm_io.c:2250 [s23c24][23:24] 
> Cannot indicate PROGRESS_MEDIA in channel with indication PROCEED 
> still pending in state PROCEED
> 2011-01-06 02:02:44.469700 [WARNING] mod_freetdm.c:434 [s6c21][6:21] 
> Why bother changing state from PROCEED to PROCEED
> 2011-01-06 02:02:44.476688 [WARNING] ftdm_io.c:2250 [s6c21][6:21] 
> Cannot indicate RINGING in channel with indication PROCEED still 
> pending in state PROCEED
> 2011-01-06 02:02:48.018716 [WARNING] mod_freetdm.c:434 [s20c25][20:25] 
> Why bother changing state from PROCEED to PROCEED
> 2011-01-06 02:02:48.046713 [WARNING] ftdm_io.c:2250 [s20c25][20:25] 
> Cannot indicate PROGRESS_MEDIA in channel with indication PROCEED 
> still pending in state PROCEED
> 2011-01-06 02:02:49.463708 [WARNING] mod_freetdm.c:434 [s8c4][8:4] Why 
> bother changing state from PROCEED to PROCEED
> 2011-01-06 02:02:54.462848 [WARNING] mod_freetdm.c:434 [s19c25][19:25] 
> Why bother changing state from PROCEED to PROCEED
> 2011-01-06 02:02:54.468754 [CRIT] ftmod_sangoma_isdn_stack_rcv.c:988 
> sng_isdn->
>
> date: 01/06/2011 time: 02:02:54
> 2011-01-06 02:02:54.468754 [CRIT] ftmod_sangoma_isdn_stack_rcv.c:988 
> sng_isdn->
> mtss(posix): sw error:  ent: 010  inst: 000  proc id: 001
> file: /usr/src/libsng_trillium-build/libsng_isdn/trillium/in/in_bdy1.c 
> line: 2144 errcode: 46909632820939 errcls: ERRCLS_DEBUG
> errval: 00001  errdesc: InUiIntConReq() Failed.
> 2011-01-06 02:02:54.468754 [WARNING] 
> ftmod_sangoma_isdn_stack_hndl.c:862 [s10c1][10:1] STATUS CONFIRM 
> (call_state:0 channel-state:DIALING cause:100) (suId:1 suInstId:81 
> spInstId:0)
> 2011-01-06 02:02:55.305075 [WARNING] ftdm_io.c:2250 [s8c4][8:4] Cannot 
> indicate PROGRESS_MEDIA in channel with indication PROCEED still 
> pending in state PROCEED
> 2011-01-06 02:02:57.785073 [WARNING] ftdm_io.c:2250 [s19c25][19:25] 
> Cannot indicate PROGRESS_MEDIA in channel with indication PROCEED 
> still pending in state PROCEED
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From ovvenkatesan at gmail.com  Thu Jan  6 15:47:07 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Thu, 6 Jan 2011 18:17:07 +0530
Subject: [Freeswitch-users] hardware requirements
In-Reply-To: <AANLkTinU_AqicisxSjOS-SJDGgfSpr1DYEeDZxQLmYp5@mail.gmail.com>
References: <AANLkTi=eXk=tRHK+Y8McrnUdYGvqHXthyYEOk3P0TJnZ@mail.gmail.com>
	<AANLkTinU_AqicisxSjOS-SJDGgfSpr1DYEeDZxQLmYp5@mail.gmail.com>
Message-ID: <AANLkTi=W7yh2gZvzRG9mQs-qrSvrSw_eDw+PErp0dHWH@mail.gmail.com>

Thanks Steve,

As of now, I am planning to handle 120 concurrent calls.
( If client wants, I may increase concurrent calls )
Each call duration will be 90 - 120 seconds.

will  freeSwitch support OpenVox D410P 4-port E1/T1/J1 card?

Call will be landing from the mobile phones, Do I need to use any codec?

Regards,
Venkat.




On Thu, Jan 6, 2011 at 5:12 PM, Steven Ayre <steveayre at gmail.com> wrote:

> What do you expect the peak CPS and number of concurrent calls to be?
>
> Any current Xeon server should be able to do 3cps fine (72000 in 6
> hours). What really will dictate your hardware requirements is what
> the maximum load you want to handle at any one time is.
>
> -Steve
>
>
> On 6 January 2011 11:32, ovvenkat <ovvenkatesan at gmail.com> wrote:
> > Hi to all,
> >
> > I have to handle 72,000 inbound calls per day ( 6 - 8 hours ),
> > Anyone please suggest me the
> > hardware requirement for the same.
> >
> > Like, What kind of Server and how many PRI or BRI cards will fulfill the
> > needs.
> >
> > * Its Simple IVR, Call will be landing from mobile phones.
> >
> >
> > --
> >
> > If you have come to help me, you are wasting your time.
> > If you have come to because your liberation is bound up in mine, we can
> work
> > together.
> >
> >
> > Regards
> > Venkatesan OV.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
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From steveayre at gmail.com  Thu Jan  6 16:04:44 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 6 Jan 2011 13:04:44 +0000
Subject: [Freeswitch-users] hardware requirements
In-Reply-To: <AANLkTi=W7yh2gZvzRG9mQs-qrSvrSw_eDw+PErp0dHWH@mail.gmail.com>
References: <AANLkTi=eXk=tRHK+Y8McrnUdYGvqHXthyYEOk3P0TJnZ@mail.gmail.com>
	<AANLkTinU_AqicisxSjOS-SJDGgfSpr1DYEeDZxQLmYp5@mail.gmail.com>
	<AANLkTi=W7yh2gZvzRG9mQs-qrSvrSw_eDw+PErp0dHWH@mail.gmail.com>
Message-ID: <AANLkTimWTCAjkdPeDM5=OyZx5ctSvbjXMA3+r1jrcc96@mail.gmail.com>

On 6 January 2011 12:47, ovvenkat <ovvenkatesan at gmail.com> wrote:
> Thanks Steve,
>
> As of now, I am planning to handle 120 concurrent calls.
> ( If client wants, I may increase concurrent calls )
> Each call duration will be 90 - 120 seconds.

Not sure what the minimum requirements are, but I have 24month old
servers that are dual 4core Xeon E5405 with 8GB RAM. They wouldn't
find 120 calls taxing at all.

Load will depend what you're doing though - transcoding uses CPU, and
things like IVR, recording, voicemail, conferencing will involve far
more processing than just bridging a call.

>
> will? freeSwitch support OpenVox D410P 4-port E1/T1/J1 card?

I haven't used FreeSWITCH with TDM personally, but AFAIK any zaptel
compatible card will work with freetdm. That card appears to be zaptel
compatible.

Perhaps someone else can chime in to confirm?

>
> Call will be landing from the mobile phones, Do I need to use any codec?

Calls always use a codec... you just need to pick which is best.
You'll want to avoid using too much bandwidth so you can run over
2/2.5G networks. There's a few that would suit, including Speex, GSM,
G729. G729 would need a $10/channel license, the rest are free.

>
> Regards,
> Venkat.
>
>
>
>
> On Thu, Jan 6, 2011 at 5:12 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> What do you expect the peak CPS and number of concurrent calls to be?
>>
>> Any current Xeon server should be able to do 3cps fine (72000 in 6
>> hours). What really will dictate your hardware requirements is what
>> the maximum load you want to handle at any one time is.
>>
>> -Steve
>>
>>
>> On 6 January 2011 11:32, ovvenkat <ovvenkatesan at gmail.com> wrote:
>> > Hi to all,
>> >
>> > I have to handle 72,000 inbound calls per day ( 6 - 8 hours ),
>> > Anyone please suggest me the
>> > hardware requirement for the same.
>> >
>> > Like, What kind of Server and how many PRI or BRI cards will fulfill the
>> > needs.
>> >
>> > * Its Simple IVR, Call will be landing from mobile phones.
>> >
>> >
>> > --
>> >
>> > If you have come to help me, you are wasting your time.
>> > If you have come to because your liberation is bound up in mine, we can
>> > work
>> > together.
>> >
>> >
>> > Regards
>> > Venkatesan OV.
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
>
> If you have come to help me, you are wasting your time.
> If you have come to because your liberation is bound up in mine, we can work
> together.
>
>
> Regards
> Venkatesan OV.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From bernhard.suttner at winet.ch  Thu Jan  6 16:07:15 2011
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Thu, 6 Jan 2011 14:07:15 +0100
Subject: [Freeswitch-users] different mail-from/subject within voicemail
Message-ID: <f83f4d9e-5e43-4499-968e-66d940ecfd38@winet.ch>

Hi,

is it somehow possible to specify the "email-from" address and subject used within the voicemail application within the directory (per user different)?

Best regards,
Bernhard




From moises.silva at gmail.com  Thu Jan  6 18:25:13 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 6 Jan 2011 10:25:13 -0500
Subject: [Freeswitch-users] no ringback tone
In-Reply-To: <AANLkTin4Jw4sgZqw6d+bzRxhZW4=0PJCsRKsshrGuQok@mail.gmail.com>
References: <AANLkTinbd06YgZFWue=eQGG9A5JO2a550tJ+bo-yG2yv@mail.gmail.com>
	<AANLkTinA46HPKmk_pWaAiMpy-hdihc7STfY9_FuNkthx@mail.gmail.com>
	<AANLkTim6iJ6gFc1=21REAKkvz9z8ZgPK2ShHC7pBUpxH@mail.gmail.com>
	<AANLkTikxZZdV-D3c0B9yCN8jsdg4JASkxxJtevbkNA=M@mail.gmail.com>
	<AANLkTi=m4DvTCSCNfmfPjPwBDiYQwT+_7M_vEpCTXnOC@mail.gmail.com>
	<AANLkTim6OZAuLMBdTDmv_rmg1jcGC5VdD7RhsaEQW18r@mail.gmail.com>
	<AANLkTimi9iT+BxWP1y7BnvYLxysu9tSuzf+S08z1bp-L@mail.gmail.com>
	<AANLkTintG=H+TnRtMfyTvg85NNMh=xjgS-rLToL+SO_g@mail.gmail.com>
	<AANLkTimiaXnvi_d_m6TTy5JURjsKLwQPWhjRVyCr_qe+@mail.gmail.com>
	<AANLkTimkAYJzXrf8--8iZqwiGdbdkj+4pbPjb1AQXvmu@mail.gmail.com>
	<AANLkTin4Jw4sgZqw6d+bzRxhZW4=0PJCsRKsshrGuQok@mail.gmail.com>
Message-ID: <AANLkTi=hGhLWYmYmXm5AO=Lqf+14-8=fQnv4_n1KHoDa@mail.gmail.com>

On Thu, Jan 6, 2011 at 3:14 AM, Sam <u2nsam at gmail.com> wrote:

> Tried installing the latest git-4272dcb 2011-01-05 20-12-19 -0600 it giving
> the sangoma codec error while instaling.
>
>
Huh?? What does your PRI issue has to do, at all, with Sangoma codec module?

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com
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From moises.silva at gmail.com  Thu Jan  6 18:30:03 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 6 Jan 2011 10:30:03 -0500
Subject: [Freeswitch-users] Error in log with latest FreeTDM
In-Reply-To: <AANLkTikABrSrKJ6brR=eQCe0F0+F25W08A9pxO=SYj8b@mail.gmail.com>
References: <AANLkTik+pv3C3svaCRDUBE4P8OYGHj30tJ+q-1bqO9E2@mail.gmail.com>
	<AANLkTikABrSrKJ6brR=eQCe0F0+F25W08A9pxO=SYj8b@mail.gmail.com>
Message-ID: <AANLkTikUnaFAJgxmMK=P1cE3QR+ayAOGSr=TTJFC=s3Y@mail.gmail.com>

On Thu, Jan 6, 2011 at 4:37 AM, Sam <u2nsam at gmail.com> wrote:

> For me to while compiling....
>
>
> This was the error while installing latest git-4272dcb 2011-01-05 20-12-19
> -0600
>
>
You are trying to compile the codec module without the proper supporting
library + headers. Unless you have a D-series transcoding card, why are you
trying to compile this module at all? if you do really need this module,
follow the instructions at
http://wiki.sangoma.com/sangoma-media-transcoding

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com
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From sfrippiat at dti-be.com  Thu Jan  6 18:40:34 2011
From: sfrippiat at dti-be.com (=?ISO-8859-1?Q?S=E9bastien_Frippiat?=)
Date: Thu, 06 Jan 2011 16:40:34 +0100
Subject: [Freeswitch-users] Originate command freezes Grandstream phone
Message-ID: <4D25E272.1040509@dti-be.com>

Hello.

I already posted about this problem without much answers 
(http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066132.html) 
and I thought it was the time to ask again, with more informations.

First, I posted several parts of my configuration 
(http://pastebin.freeswitch.org/14936) as well as the full logs of my 
two tests (http://pastebin.freeswitch.org/14937).

I use two phones (26 & 27 are their sip id):
- 26 = Grandstream GXP2000
- 27 = Snom M3

The first test freezes the Grandstream phone but the second one works 
fine. The only thing changing in these two tests is the direction of the 
origination. Note that everything is on the same internal network.

1) originate {{origination_caller_id_name='Test originate 
1',origination_caller_id_number='27',originate_timeout=20,effective_caller_id_name='User1',effective_caller_id_number='26',allow_outside_calls=true,outside_calls_gateway=xxxxxxxxx}}sofia/internal/26% 
27
2) originate {{origination_caller_id_name='Test originate 
2',origination_caller_id_number='26',originate_timeout=20,effective_caller_id_name='User2',effective_caller_id_number='27',allow_outside_calls=true,outside_calls_gateway=xxxxxxxxx}}sofia/internal/27% 
26

1) Grandstream GXP2000 -> Snom M3 => KO (Grandstream completely freezes 
with horrible sounds on the speaker)
2) Snom M3 -> Grandstream GXP2000 => OK (no phones/audio problems)

It was previously working with a version from 2010/09/09 and when we 
updated to 2010/12/09 (git-e680c82 2010-12-09 08-59-06 -0600), it wasn't 
working anymore.
Latest version doesn't solve the issue.

Note that the caller name show on the phone is "Outbound call" in this 
case but the correct name is shown when directly calling from a phone to 
another one.

Anybody can help ?


From sfrippiat at dti-be.com  Thu Jan  6 18:44:17 2011
From: sfrippiat at dti-be.com (=?ISO-8859-1?Q?S=E9bastien_Frippiat?=)
Date: Thu, 06 Jan 2011 16:44:17 +0100
Subject: [Freeswitch-users] 10s delay between dialing and the ringing on an
	internal call
Message-ID: <4D25E351.4020805@dti-be.com>

Hello.

You can find some parts of my configuration here: 
http://pastebin.freeswitch.org/14936

Most of the times, internal calls work fine but I recently encountered 
odd problems. When I pickup my phone and dial a number (an internal 
number, the other phone is on the same network, registered to the same 
freeswitch instance), it sometimes takes up to 10s between the moment I 
dial the number and the moment the other phone starts ringing.

I included a full log of a call with the problem 
(http://pastebin.freeswitch.org/14938) as well as the log of a call 
without the problem (http://pastebin.freeswitch.org/14939).

Could the problem come from the "Rejected by acl "domains"" message" ?

Anybody can help ?


From rupa at rupa.com  Thu Jan  6 19:19:47 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 6 Jan 2011 10:19:47 -0600
Subject: [Freeswitch-users] 10s delay between dialing and the ringing on
 an internal call
In-Reply-To: <4D25E351.4020805@dti-be.com>
References: <4D25E351.4020805@dti-be.com>
Message-ID: <AANLkTimKJMd_iJyadZySB-QiVo3eT07zJV3XuHatp1BL@mail.gmail.com>

In the example given, the call with a problem has timestamps that go from

2011-01-06?15:39:29.862713
to
2011-01-06 15:40:11.324337

Which is... umm...  less than 1s.  So, whatever is introducing the 10s
delay is prior to the call hitting the acl.

You might try turning up SIP logging to see any delays in processing
the SIP traffic.

On Thu, Jan 6, 2011 at 9:44 AM, S?bastien Frippiat <sfrippiat at dti-be.com> wrote:
>
> Hello.
>
> You can find some parts of my configuration here:
> http://pastebin.freeswitch.org/14936
>
> Most of the times, internal calls work fine but I recently encountered
> odd problems. When I pickup my phone and dial a number (an internal
> number, the other phone is on the same network, registered to the same
> freeswitch instance), it sometimes takes up to 10s between the moment I
> dial the number and the moment the other phone starts ringing.
>
> I included a full log of a call with the problem
> (http://pastebin.freeswitch.org/14938) as well as the log of a call
> without the problem (http://pastebin.freeswitch.org/14939).
>
> Could the problem come from the "Rejected by acl "domains"" message" ?
>
> Anybody can help ?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



--
-Rupa


From brian at freeswitch.org  Thu Jan  6 20:16:03 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 6 Jan 2011 11:16:03 -0600
Subject: [Freeswitch-users] 10s delay between dialing and the ringing on
	an internal call
In-Reply-To: <AANLkTimKJMd_iJyadZySB-QiVo3eT07zJV3XuHatp1BL@mail.gmail.com>
References: <4D25E351.4020805@dti-be.com>
	<AANLkTimKJMd_iJyadZySB-QiVo3eT07zJV3XuHatp1BL@mail.gmail.com>
Message-ID: <17559C4F-7A22-4B5D-B38F-316A1D276E2A@freeswitch.org>

I'm going to guess he's using a cisco.

/b

On Jan 6, 2011, at 10:19 AM, Rupa Schomaker wrote:

> In the example given, the call with a problem has timestamps that go from
> 
> 2011-01-06 15:39:29.862713
> to
> 2011-01-06 15:40:11.324337
> 
> Which is... umm...  less than 1s.  So, whatever is introducing the 10s
> delay is prior to the call hitting the acl.
> 
> You might try turning up SIP logging to see any delays in processing
> the SIP traffic.



From kilburna at gmail.com  Thu Jan  6 14:59:32 2011
From: kilburna at gmail.com (Kilburn Abrahams)
Date: Thu, 06 Jan 2011 22:59:32 +1100
Subject: [Freeswitch-users] Register warnings and Errors
Message-ID: <4D25AEA4.4040408@gmail.com>

Hi

I upgraded to git-a90b4fe 2011-01-04 23-51-47 +0100. Getting this in the
console when a REGISTER is recorded. I think I also upgraded sqlite a
few days ago. Has anything changed to cause these warnings and errors.

2011-01-06 22:44:41.896726 [WARNING] sofia_reg.c:1216 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [1000 at 192.168.1.230] from ip
192.168.1.111
2011-01-06 22:44:42.693350 [ERR] sofia_reg.c:1317 DELETE PRESENCE SQL:
delete from sip_presence where sip_user='1000' and
sip_host='192.168.1.230' and profile_name='internal' and
open_closed='closed'

Thanks
Kilburn
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From djbinter at gmail.com  Thu Jan  6 20:45:18 2011
From: djbinter at gmail.com (DJB International)
Date: Thu, 6 Jan 2011 09:45:18 -0800
Subject: [Freeswitch-users] Register warnings and Errors
In-Reply-To: <4D25AEA4.4040408@gmail.com>
References: <4D25AEA4.4040408@gmail.com>
Message-ID: <AANLkTim-Vs0esnsjX_CSKHejR40WW+HaxTzK9+U2zEoB@mail.gmail.com>

This issue (FS-2961) has been corrected on Jan. 5 git
2c595a6c<http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=2c595a6ce04c714b51fc8986315ff68cd463fb35>

-djbinter

On Thu, Jan 6, 2011 at 3:59 AM, Kilburn Abrahams <kilburna at gmail.com> wrote:

>  Hi
>
> I upgraded to git-a90b4fe 2011-01-04 23-51-47 +0100. Getting this in the
> console when a REGISTER is recorded. I think I also upgraded sqlite a few
> days ago. Has anything changed to cause these warnings and errors.
>
> 2011-01-06 22:44:41.896726 [WARNING] sofia_reg.c:1216 SIP auth challenge
> (REGISTER) on sofia profile 'internal' for [1000 at 192.168.1.230] from ip
> 192.168.1.111
> 2011-01-06 22:44:42.693350 [ERR] sofia_reg.c:1317 DELETE PRESENCE SQL:
> delete from sip_presence where sip_user='1000' and sip_host='192.168.1.230'
> and profile_name='internal' and open_closed='closed'
>
> Thanks
> Kilburn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From edpimentl at gmail.com  Thu Jan  6 20:49:32 2011
From: edpimentl at gmail.com (EdPimentl)
Date: Thu, 6 Jan 2011 12:49:32 -0500
Subject: [Freeswitch-users] Get a CLUE an (ANSI C compiler targeting high
 level languages) LUA, JS, Perl, C, Java and Common Lisp
Message-ID: <AANLkTi=u76Yk7WM=Kc0AVwb63tD77WTJB3nTdKKcUsTm@mail.gmail.com>

FYIhttp://cluecc.sourceforge.net/

Clue: an ANSI C compiler targeting high level languages

*New! v0.5 released!* Now we support Java --- at about 40% of the speed of
native code!
What?

Clue is an ANSI C compiler (C89, some C99) that targets high-level languages
such as Lua, Javascript or Perl (and some low-level ones). It supports the
entire C language, including pointer arithmetic, and can be used to run
arbitrary pure-C programs.

Clue currently supports the following targets:

   - Lua 5.1.3
   - Javascript
   - Perl 5
   - C
   - Java
   - Common Lisp


Sincerely,
-E
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From msc at freeswitch.org  Thu Jan  6 22:07:12 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Jan 2011 11:07:12 -0800
Subject: [Freeswitch-users] sofia status profile internal - Output in
 Table view
In-Reply-To: <AANLkTinQ1AxUiMNWBhu9gnJ6RBtovcsL6+NFQx-_0ywj@mail.gmail.com>
References: <AANLkTikPhufZ0T80+9Wj9rS5nZcPqZ=2xZKRNsD9--KL@mail.gmail.com>
	<AANLkTin1R+5QvWp1YG5cOexjvHu5+qfB=zpC0_f1S1W4@mail.gmail.com>
	<AANLkTinQ1AxUiMNWBhu9gnJ6RBtovcsL6+NFQx-_0ywj@mail.gmail.com>
Message-ID: <AANLkTimyQoogsfydBmTdjU8DpN=a9PY71M29B4+CsK1-@mail.gmail.com>

You don't format it like "sip show peers" - you take the format in sofia
xmlstatus and do with it whatever you wish on your end. The XML output of
the command is ready for you to do whatever you want with it.

-MC

On Wed, Jan 5, 2011 at 7:25 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Thanks Anthony.
>
> How to format the output like "sip show peers"
>
> Thanks
> Lloyd
>
>
> On Tue, Jan 4, 2011 at 12:33 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> there is "sofia xmlstatus profile internal"
>> then you can format it any way you want.
>>
>>
>> On Tue, Jan 4, 2011 at 10:26 AM, Aloysius Lloyd
>> <lloyd.aloysius at gmail.com> wrote:
>> > Hi All,
>> > Is there any way to get the output from sofia status profile internal in
>> a
>> > table view.
>> > Something like Asterisk "sip show peers"
>> >
>> > Thanks
>> > Lloyd
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ayhkor at gmail.com  Thu Jan  6 22:09:40 2011
From: ayhkor at gmail.com (Ayhan Koroglu)
Date: Thu, 6 Jan 2011 14:09:40 -0500
Subject: [Freeswitch-users] mod_conference pin
Message-ID: <AANLkTiku8cBTrKRJXhqDR8=RkTAH1oeZJc4gLTk=Uhtt@mail.gmail.com>

Hi All

 Using the web based conferencing software
I am able to dial in (phone call) to conference  using pin number.
But the problem is  sip client(voice conf) is also asking that same
pin number.
in dialplan
confname at profile+[pin]
or in profile
<param name="pin" value="pin_number">
How should I set my dial plan so that only phone dialing asks the pin
number.

Thanks
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From infos at madovsky.org  Thu Jan  6 22:40:13 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 6 Jan 2011 14:40:13 -0500
Subject: [Freeswitch-users] mod_conference with cluster ODBC
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705><AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com><AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com><AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com>
	<AANLkTimutwOyeubcS=tax5EbDEniCD+Nc4iHMydQxpFb@mail.gmail.com>
Message-ID: <37761574FEE44D13BEBE4A7DE0089083@e1705>

in case of you have 8 servers you have to do it for each ?

Thanks
  ----- Original Message ----- 
  From: joy this 
  To: FreeSWITCH Users Help 
  Sent: Thursday, January 06, 2011 2:51 AM
  Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


  It works. Thank you everyone.


  2011/1/5 Rupa Schomaker <rupa at rupa.com>

    Use the api: conference <confname> dial [{dial string options}]<endpoint_module_name>/<destination> [<callerid_number> [<callerid_name>]]


    To initiate the call from within conference A on server 1.  Have a corresponding dialplan entry on server 2 to accept the call and add it into the conference A on server 2.  You've now bridged the two conferences in the two servers.


------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org

-- 
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.

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From rupa at rupa.com  Thu Jan  6 23:01:10 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 6 Jan 2011 14:01:10 -0600
Subject: [Freeswitch-users] mod_conference with cluster ODBC
In-Reply-To: <37761574FEE44D13BEBE4A7DE0089083@e1705>
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705>
	<AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com>
	<AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com>
	<AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com>
	<AANLkTimutwOyeubcS=tax5EbDEniCD+Nc4iHMydQxpFb@mail.gmail.com>
	<37761574FEE44D13BEBE4A7DE0089083@e1705>
Message-ID: <AANLkTi=VZ8JZ+PuyXoz_EYNVovY-AOuPyz9+yfKoUc3k@mail.gmail.com>

Yes

On Thu, Jan 6, 2011 at 1:40 PM, Madovsky <infos at madovsky.org> wrote:
> in case of you have 8 servers you have to do it for each ?
>
> Thanks
>
> ----- Original Message -----
> From: joy this
> To: FreeSWITCH Users Help
> Sent: Thursday, January 06, 2011 2:51 AM
> Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
> It works. Thank you everyone.
>
> 2011/1/5 Rupa Schomaker <rupa at rupa.com>
>>
>> Use the api:?conference <confname> dial [{dial string
>> options}]<endpoint_module_name>/<destination> [<callerid_number>
>> [<callerid_name>]]
>> To initiate the call from within conference A on server 1. ?Have a
>> corresponding dialplan entry on server 2 to accept the call and add it into
>> the conference A on server 2. ?You've now bridged the two conferences in the
>> two servers.
>
> ________________________________
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> --
> This message has been scanned for viruses and
> dangerous content by MailScanner, and is
> believed to be clean.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From infos at madovsky.org  Fri Jan  7 00:25:15 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 6 Jan 2011 16:25:15 -0500
Subject: [Freeswitch-users] mod_conference with cluster ODBC
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705><AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com><AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com><AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com><AANLkTimutwOyeubcS=tax5EbDEniCD+Nc4iHMydQxpFb@mail.gmail.com><37761574FEE44D13BEBE4A7DE0089083@e1705>
	<AANLkTi=VZ8JZ+PuyXoz_EYNVovY-AOuPyz9+yfKoUc3k@mail.gmail.com>
Message-ID: <337274A8CA2343259DF18D8135AF2DCE@e1705>

Rupa,

I don't want bother anyone with this thread but why not
to manage conference as SIP user ?
if someone from server A call an other who is registered on server B, so
FS do it automatically, why not with conference ? Or maybe create a param
in mod_conference that let the choice of the admin to manage unique name in 
all cluster or not.
like <param name="cluster_unique_name" value="true"/>
I will try to understand the C code to hack something like this...

----- Original Message ----- 
From: "Rupa Schomaker" <rupa at rupa.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, January 06, 2011 3:01 PM
Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


Yes

On Thu, Jan 6, 2011 at 1:40 PM, Madovsky <infos at madovsky.org> wrote:
> in case of you have 8 servers you have to do it for each ?
>
> Thanks
>
> ----- Original Message -----
> From: joy this
> To: FreeSWITCH Users Help
> Sent: Thursday, January 06, 2011 2:51 AM
> Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
> It works. Thank you everyone.
>
> 2011/1/5 Rupa Schomaker <rupa at rupa.com>
>>
>> Use the api: conference <confname> dial [{dial string
>> options}]<endpoint_module_name>/<destination> [<callerid_number>
>> [<callerid_name>]]
>> To initiate the call from within conference A on server 1. Have a
>> corresponding dialplan entry on server 2 to accept the call and add it 
>> into
>> the conference A on server 2. You've now bridged the two conferences in 
>> the
>> two servers.
>
> ________________________________
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> --
> This message has been scanned for viruses and
> dangerous content by MailScanner, and is
> believed to be clean.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 



From jmesquita at freeswitch.org  Fri Jan  7 00:33:36 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 6 Jan 2011 18:33:36 -0300
Subject: [Freeswitch-users] mod_conference with cluster ODBC
In-Reply-To: <337274A8CA2343259DF18D8135AF2DCE@e1705>
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705>
	<AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com>
	<AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com>
	<AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com>
	<AANLkTimutwOyeubcS=tax5EbDEniCD+Nc4iHMydQxpFb@mail.gmail.com>
	<37761574FEE44D13BEBE4A7DE0089083@e1705>
	<AANLkTi=VZ8JZ+PuyXoz_EYNVovY-AOuPyz9+yfKoUc3k@mail.gmail.com>
	<337274A8CA2343259DF18D8135AF2DCE@e1705>
Message-ID: <AANLkTimVaxdbwVHCTqpYC5w3q+Vi+t7-oMekrkxu4Kfd@mail.gmail.com>

Ok, let me see if I can get this into your head. (giggles)

A conference means that the audio needs to mixed in together so that all
participants can talk/hear each other, right? If you implement something in
C on mod_conference, you are going to essentially do the same as what an ESL
app does. You _need_ to call in from one server to the other so that you can
mix the audio of all the participants. The real advantage would be the
management API being only one for everything and the challenge is exactly
that. How to mute certain users on a conference that is spanning over 10
servers or deaf them, etc...

A SIP "user" is easier because you don't have to bridge audio from another
server necessarily. Got it?

Regards,
Jo?o Mesquita


On Thu, Jan 6, 2011 at 6:25 PM, Madovsky <infos at madovsky.org> wrote:

> Rupa,
>
> I don't want bother anyone with this thread but why not
> to manage conference as SIP user ?
> if someone from server A call an other who is registered on server B, so
> FS do it automatically, why not with conference ? Or maybe create a param
> in mod_conference that let the choice of the admin to manage unique name in
> all cluster or not.
> like <param name="cluster_unique_name" value="true"/>
> I will try to understand the C code to hack something like this...
>
> ----- Original Message -----
> From: "Rupa Schomaker" <rupa at rupa.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Thursday, January 06, 2011 3:01 PM
> Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
>
>
> Yes
>
> On Thu, Jan 6, 2011 at 1:40 PM, Madovsky <infos at madovsky.org> wrote:
> > in case of you have 8 servers you have to do it for each ?
> >
> > Thanks
> >
> > ----- Original Message -----
> > From: joy this
> > To: FreeSWITCH Users Help
> > Sent: Thursday, January 06, 2011 2:51 AM
> > Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
> > It works. Thank you everyone.
> >
> > 2011/1/5 Rupa Schomaker <rupa at rupa.com>
> >>
> >> Use the api: conference <confname> dial [{dial string
> >> options}]<endpoint_module_name>/<destination> [<callerid_number>
> >> [<callerid_name>]]
> >> To initiate the call from within conference A on server 1. Have a
> >> corresponding dialplan entry on server 2 to accept the call and add it
> >> into
> >> the conference A on server 2. You've now bridged the two conferences in
> >> the
> >> two servers.
> >
> > ________________________________
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > --
> > This message has been scanned for viruses and
> > dangerous content by MailScanner, and is
> > believed to be clean.
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tuyanozipek at gmail.com  Fri Jan  7 01:00:25 2011
From: tuyanozipek at gmail.com (=?ISO-8859-1?Q?Tuyan_=D6zipek?=)
Date: Thu, 6 Jan 2011 17:00:25 -0500
Subject: [Freeswitch-users] 10s delay between dialing and the ringing on
 an internal call
In-Reply-To: <17559C4F-7A22-4B5D-B38F-316A1D276E2A@freeswitch.org>
References: <4D25E351.4020805@dti-be.com>
	<AANLkTimKJMd_iJyadZySB-QiVo3eT07zJV3XuHatp1BL@mail.gmail.com>
	<17559C4F-7A22-4B5D-B38F-316A1D276E2A@freeswitch.org>
Message-ID: <AANLkTikRSz_tcsFhH9BmCW69A+L2rfpeuMafcv6ni_RT@mail.gmail.com>

Hi,
Answers are inline...

On Thu, Jan 6, 2011 at 12:16 PM, Brian West <brian at freeswitch.org> wrote:
> I'm going to guess he's using a cisco.
>
> /b
No clue..

>
> On Jan 6, 2011, at 10:19 AM, Rupa Schomaker wrote:
>
>> In the example given, the call with a problem has timestamps that go from
>>
>> 2011-01-06 15:39:29.862713
>> to
>> 2011-01-06 15:40:11.324337
>>
>> Which is... umm... ?less than 1s. ?So, whatever is introducing the 10s
>> delay is prior to the call hitting the acl.

Please correct me if i am wrong.. but ;
>From 15:39:29 to 15:40:11 its about 41 seconds , and there is like 10
s delay between

2011-01-06 15:39:47.692045 [DEBUG] switch_core_state_machine.c:378
(sofia/internal/sip:27 at 10.1.1.178) State CONSUME_MEDIA going to sleep
2011-01-06 15:39:57.160503 [DEBUG] sofia.c:4616 Channel
sofia/internal/sip:27 at 10.1.1.178 entering state [calling][0]

maybe i am too much decaffeinated...

Cheers,

/tyn

>>
>> You might try turning up SIP logging to see any delays in processing
>> the SIP traffic.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From msc at freeswitch.org  Fri Jan  7 01:12:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Jan 2011 14:12:02 -0800
Subject: [Freeswitch-users] inline string of commands rather than
 phrase: for greet-long ivr attribute
In-Reply-To: <AANLkTinR+hHr1VhPDc8-H48bMS8uB5VSRF-h=ZJbSLAK@mail.gmail.com>
References: <AANLkTi=4xWErgY0r=iEHrK0NicS6BKvSaE9Sm95ggqsA@mail.gmail.com>
	<AANLkTi=mho3LjsrY6WVVm_CUBhdt1Uba1SsNwEaSjq1Z@mail.gmail.com>
	<AANLkTinR+hHr1VhPDc8-H48bMS8uB5VSRF-h=ZJbSLAK@mail.gmail.com>
Message-ID: <AANLkTinm52ryMMDw9QNgrmxN3rcnAtG15JtVoGmkj_ah@mail.gmail.com>

No, that's just for sound files. Phrases are what you should use if you need
to mix and match sound files and TTS.
-MC

On Wed, Jan 5, 2011 at 11:26 PM, Matthew Fong <mattdfong at gmail.com> wrote:

> is it also possible to combine tts like cepstral commands in this way? or
> must they all be sound files?
>
>
> On Wed, Jan 5, 2011 at 4:09 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Yes:
>>
>> greet-long="file_string://${sound_prefix}/ivr/8000/ivr-good_afternoon.wav!${sound_prefix}/ivr/8000/ivr-generic_greeting.wav"
>>
>> Just make sure that mod_file_string is installed. Also, read the wiki
>> entry on mod_file_string, particularly on when to use absolute path names!
>>
>> -MC
>>
>> On Wed, Jan 5, 2011 at 3:16 PM, Matthew Fong <mattdfong at gmail.com> wrote:
>>
>>> I am wondering if it is possible to use an inline style string of
>>> commands like playback:play1.wav say:hello playback:play2.wav in the
>>> greet-long xml ivr attribute rather than having to create a phrase macro. If
>>> it is possible can I please get an example. thanks
>>>
>>> --matt
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri Jan  7 01:18:33 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Jan 2011 14:18:33 -0800
Subject: [Freeswitch-users] hardware requirements
In-Reply-To: <AANLkTimWTCAjkdPeDM5=OyZx5ctSvbjXMA3+r1jrcc96@mail.gmail.com>
References: <AANLkTi=eXk=tRHK+Y8McrnUdYGvqHXthyYEOk3P0TJnZ@mail.gmail.com>
	<AANLkTinU_AqicisxSjOS-SJDGgfSpr1DYEeDZxQLmYp5@mail.gmail.com>
	<AANLkTi=W7yh2gZvzRG9mQs-qrSvrSw_eDw+PErp0dHWH@mail.gmail.com>
	<AANLkTimWTCAjkdPeDM5=OyZx5ctSvbjXMA3+r1jrcc96@mail.gmail.com>
Message-ID: <AANLkTikQdyO4eaNibbxtgcKdCdFZsjbpsoWiNw_XBeYF@mail.gmail.com>

FreeSWITCH works just fine with TDM cards. I tinkered with an old P4 and a
4-port Tormenta2 clone and easily handled dozens of concurrent calls, and
that was before the FreeTDM stuff made things more reliable.

At this point hardware is cheap, so get yourself a 4- or 8- core box with
8GB RAM and some fast hard drives. For the best experience with TDM I
recommend a Sangoma card over the Zaptel clones.

-MC

On Thu, Jan 6, 2011 at 5:04 AM, Steven Ayre <steveayre at gmail.com> wrote:

> On 6 January 2011 12:47, ovvenkat <ovvenkatesan at gmail.com> wrote:
> > Thanks Steve,
> >
> > As of now, I am planning to handle 120 concurrent calls.
> > ( If client wants, I may increase concurrent calls )
> > Each call duration will be 90 - 120 seconds.
>
> Not sure what the minimum requirements are, but I have 24month old
> servers that are dual 4core Xeon E5405 with 8GB RAM. They wouldn't
> find 120 calls taxing at all.
>
> Load will depend what you're doing though - transcoding uses CPU, and
> things like IVR, recording, voicemail, conferencing will involve far
> more processing than just bridging a call.
>
> >
> > will  freeSwitch support OpenVox D410P 4-port E1/T1/J1 card?
>
> I haven't used FreeSWITCH with TDM personally, but AFAIK any zaptel
> compatible card will work with freetdm. That card appears to be zaptel
> compatible.
>
> Perhaps someone else can chime in to confirm?
>
> >
> > Call will be landing from the mobile phones, Do I need to use any codec?
>
> Calls always use a codec... you just need to pick which is best.
> You'll want to avoid using too much bandwidth so you can run over
> 2/2.5G networks. There's a few that would suit, including Speex, GSM,
> G729. G729 would need a $10/channel license, the rest are free.
>
> >
> > Regards,
> > Venkat.
> >
> >
> >
> >
> > On Thu, Jan 6, 2011 at 5:12 PM, Steven Ayre <steveayre at gmail.com> wrote:
> >>
> >> What do you expect the peak CPS and number of concurrent calls to be?
> >>
> >> Any current Xeon server should be able to do 3cps fine (72000 in 6
> >> hours). What really will dictate your hardware requirements is what
> >> the maximum load you want to handle at any one time is.
> >>
> >> -Steve
> >>
> >>
> >> On 6 January 2011 11:32, ovvenkat <ovvenkatesan at gmail.com> wrote:
> >> > Hi to all,
> >> >
> >> > I have to handle 72,000 inbound calls per day ( 6 - 8 hours ),
> >> > Anyone please suggest me the
> >> > hardware requirement for the same.
> >> >
> >> > Like, What kind of Server and how many PRI or BRI cards will fulfill
> the
> >> > needs.
> >> >
> >> > * Its Simple IVR, Call will be landing from mobile phones.
> >> >
> >> >
> >> > --
> >> >
> >> > If you have come to help me, you are wasting your time.
> >> > If you have come to because your liberation is bound up in mine, we
> can
> >> > work
> >> > together.
> >> >
> >> >
> >> > Regards
> >> > Venkatesan OV.
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> >
> > If you have come to help me, you are wasting your time.
> > If you have come to because your liberation is bound up in mine, we can
> work
> > together.
> >
> >
> > Regards
> > Venkatesan OV.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jmesquita at freeswitch.org  Fri Jan  7 01:31:18 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 6 Jan 2011 19:31:18 -0300
Subject: [Freeswitch-users] hardware requirements
In-Reply-To: <AANLkTikQdyO4eaNibbxtgcKdCdFZsjbpsoWiNw_XBeYF@mail.gmail.com>
References: <AANLkTi=eXk=tRHK+Y8McrnUdYGvqHXthyYEOk3P0TJnZ@mail.gmail.com>
	<AANLkTinU_AqicisxSjOS-SJDGgfSpr1DYEeDZxQLmYp5@mail.gmail.com>
	<AANLkTi=W7yh2gZvzRG9mQs-qrSvrSw_eDw+PErp0dHWH@mail.gmail.com>
	<AANLkTimWTCAjkdPeDM5=OyZx5ctSvbjXMA3+r1jrcc96@mail.gmail.com>
	<AANLkTikQdyO4eaNibbxtgcKdCdFZsjbpsoWiNw_XBeYF@mail.gmail.com>
Message-ID: <AANLkTikwx-j-5oo4A0L4oj_vWrpqYbs3uG69_SNwVD71@mail.gmail.com>

That's because you've never tried Khomp. I would say it would be a very good
battle to say the least!

Jo?o Mesquita


On Thu, Jan 6, 2011 at 7:18 PM, Michael Collins <msc at freeswitch.org> wrote:

> FreeSWITCH works just fine with TDM cards. I tinkered with an old P4 and a
> 4-port Tormenta2 clone and easily handled dozens of concurrent calls, and
> that was before the FreeTDM stuff made things more reliable.
>
> At this point hardware is cheap, so get yourself a 4- or 8- core box with
> 8GB RAM and some fast hard drives. For the best experience with TDM I
> recommend a Sangoma card over the Zaptel clones.
>
> -MC
>
>
> On Thu, Jan 6, 2011 at 5:04 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> On 6 January 2011 12:47, ovvenkat <ovvenkatesan at gmail.com> wrote:
>> > Thanks Steve,
>> >
>> > As of now, I am planning to handle 120 concurrent calls.
>> > ( If client wants, I may increase concurrent calls )
>> > Each call duration will be 90 - 120 seconds.
>>
>> Not sure what the minimum requirements are, but I have 24month old
>> servers that are dual 4core Xeon E5405 with 8GB RAM. They wouldn't
>> find 120 calls taxing at all.
>>
>> Load will depend what you're doing though - transcoding uses CPU, and
>> things like IVR, recording, voicemail, conferencing will involve far
>> more processing than just bridging a call.
>>
>> >
>> > will  freeSwitch support OpenVox D410P 4-port E1/T1/J1 card?
>>
>> I haven't used FreeSWITCH with TDM personally, but AFAIK any zaptel
>> compatible card will work with freetdm. That card appears to be zaptel
>> compatible.
>>
>> Perhaps someone else can chime in to confirm?
>>
>> >
>> > Call will be landing from the mobile phones, Do I need to use any codec?
>>
>> Calls always use a codec... you just need to pick which is best.
>> You'll want to avoid using too much bandwidth so you can run over
>> 2/2.5G networks. There's a few that would suit, including Speex, GSM,
>> G729. G729 would need a $10/channel license, the rest are free.
>>
>> >
>> > Regards,
>> > Venkat.
>> >
>> >
>> >
>> >
>> > On Thu, Jan 6, 2011 at 5:12 PM, Steven Ayre <steveayre at gmail.com>
>> wrote:
>> >>
>> >> What do you expect the peak CPS and number of concurrent calls to be?
>> >>
>> >> Any current Xeon server should be able to do 3cps fine (72000 in 6
>> >> hours). What really will dictate your hardware requirements is what
>> >> the maximum load you want to handle at any one time is.
>> >>
>> >> -Steve
>> >>
>> >>
>> >> On 6 January 2011 11:32, ovvenkat <ovvenkatesan at gmail.com> wrote:
>> >> > Hi to all,
>> >> >
>> >> > I have to handle 72,000 inbound calls per day ( 6 - 8 hours ),
>> >> > Anyone please suggest me the
>> >> > hardware requirement for the same.
>> >> >
>> >> > Like, What kind of Server and how many PRI or BRI cards will fulfill
>> the
>> >> > needs.
>> >> >
>> >> > * Its Simple IVR, Call will be landing from mobile phones.
>> >> >
>> >> >
>> >> > --
>> >> >
>> >> > If you have come to help me, you are wasting your time.
>> >> > If you have come to because your liberation is bound up in mine, we
>> can
>> >> > work
>> >> > together.
>> >> >
>> >> >
>> >> > Regards
>> >> > Venkatesan OV.
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> >
>> > If you have come to help me, you are wasting your time.
>> > If you have come to because your liberation is bound up in mine, we can
>> work
>> > together.
>> >
>> >
>> > Regards
>> > Venkatesan OV.
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rupa at rupa.com  Fri Jan  7 01:33:17 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 6 Jan 2011 16:33:17 -0600
Subject: [Freeswitch-users] 10s delay between dialing and the ringing on
 an internal call
In-Reply-To: <AANLkTikRSz_tcsFhH9BmCW69A+L2rfpeuMafcv6ni_RT@mail.gmail.com>
References: <4D25E351.4020805@dti-be.com>
	<AANLkTimKJMd_iJyadZySB-QiVo3eT07zJV3XuHatp1BL@mail.gmail.com>
	<17559C4F-7A22-4B5D-B38F-316A1D276E2A@freeswitch.org>
	<AANLkTikRSz_tcsFhH9BmCW69A+L2rfpeuMafcv6ni_RT@mail.gmail.com>
Message-ID: <AANLkTinYtCbNYQWaask4zvYLkWt2-A6aSDbJAOkpCw6-@mail.gmail.com>

Ok, yer right - the time is more.  dunno what I was thinking.  So...
no answer from me either.  Dunno what is going on.  You could try
increasing the sip logs maybe?

On Thu, Jan 6, 2011 at 4:00 PM, Tuyan ?zipek <tuyanozipek at gmail.com> wrote:
> Hi,
> Answers are inline...
>
> On Thu, Jan 6, 2011 at 12:16 PM, Brian West <brian at freeswitch.org> wrote:
>> I'm going to guess he's using a cisco.
>>
>> /b
> No clue..
>
>>
>> On Jan 6, 2011, at 10:19 AM, Rupa Schomaker wrote:
>>
>>> In the example given, the call with a problem has timestamps that go from
>>>
>>> 2011-01-06 15:39:29.862713
>>> to
>>> 2011-01-06 15:40:11.324337
>>>
>>> Which is... umm... ?less than 1s. ?So, whatever is introducing the 10s
>>> delay is prior to the call hitting the acl.
>
> Please correct me if i am wrong.. but ;
> >From 15:39:29 to 15:40:11 its about 41 seconds , and there is like 10
> s delay between
>
> 2011-01-06 15:39:47.692045 [DEBUG] switch_core_state_machine.c:378
> (sofia/internal/sip:27 at 10.1.1.178) State CONSUME_MEDIA going to sleep
> 2011-01-06 15:39:57.160503 [DEBUG] sofia.c:4616 Channel
> sofia/internal/sip:27 at 10.1.1.178 entering state [calling][0]
>
> maybe i am too much decaffeinated...
>
> Cheers,
>
> /tyn
>
>>>
>>> You might try turning up SIP logging to see any delays in processing
>>> the SIP traffic.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa


From brian at freeswitch.org  Fri Jan  7 01:49:21 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 6 Jan 2011 16:49:21 -0600
Subject: [Freeswitch-users] 10s delay between dialing and the ringing on
	an internal call
In-Reply-To: <AANLkTinYtCbNYQWaask4zvYLkWt2-A6aSDbJAOkpCw6-@mail.gmail.com>
References: <4D25E351.4020805@dti-be.com>
	<AANLkTimKJMd_iJyadZySB-QiVo3eT07zJV3XuHatp1BL@mail.gmail.com>
	<17559C4F-7A22-4B5D-B38F-316A1D276E2A@freeswitch.org>
	<AANLkTikRSz_tcsFhH9BmCW69A+L2rfpeuMafcv6ni_RT@mail.gmail.com>
	<AANLkTinYtCbNYQWaask4zvYLkWt2-A6aSDbJAOkpCw6-@mail.gmail.com>
Message-ID: <4291AA5D-E846-4C08-90B7-3A893D7ACC49@freeswitch.org>

Unless the OP says the device he is using ... this is a rather pointless discussion eh?

/b

On Jan 6, 2011, at 4:33 PM, Rupa Schomaker wrote:

> Ok, yer right - the time is more.  dunno what I was thinking.  So...
> no answer from me either.  Dunno what is going on.  You could try
> increasing the sip logs maybe?
> 
> On Thu, Jan 6, 2011 at 4:00 PM, Tuyan ?zipek <tuyanozipek at gmail.com> wrote:
>> Hi,
>> Answers are inline...
>> 
>> On Thu, Jan 6, 2011 at 12:16 PM, Brian West <brian at freeswitch.org> wrote:
>>> I'm going to guess he's using a cisco.
>>> 
>>> /b
>> No clue..
>> 
>>> 
>>> On Jan 6, 2011, at 10:19 AM, Rupa Schomaker wrote:
>>> 
>>>> In the example given, the call with a problem has timestamps that go from
>>>> 
>>>> 2011-01-06 15:39:29.862713
>>>> to
>>>> 2011-01-06 15:40:11.324337
>>>> 
>>>> Which is... umm...  less than 1s.  So, whatever is introducing the 10s
>>>> delay is prior to the call hitting the acl.
>> 
>> Please correct me if i am wrong.. but ;
>>> From 15:39:29 to 15:40:11 its about 41 seconds , and there is like 10
>> s delay between
>> 
>> 2011-01-06 15:39:47.692045 [DEBUG] switch_core_state_machine.c:378
>> (sofia/internal/sip:27 at 10.1.1.178) State CONSUME_MEDIA going to sleep
>> 2011-01-06 15:39:57.160503 [DEBUG] sofia.c:4616 Channel
>> sofia/internal/sip:27 at 10.1.1.178 entering state [calling][0]
>> 
>> maybe i am too much decaffeinated...
>> 
>> Cheers,
>> 
>> /tyn
>> 
>>>> 
>>>> You might try turning up SIP logging to see any delays in processing
>>>> the SIP traffic.
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> -- 
> -Rupa
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Fri Jan  7 02:59:30 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Jan 2011 15:59:30 -0800
Subject: [Freeswitch-users] hardware requirements
In-Reply-To: <AANLkTikwx-j-5oo4A0L4oj_vWrpqYbs3uG69_SNwVD71@mail.gmail.com>
References: <AANLkTi=eXk=tRHK+Y8McrnUdYGvqHXthyYEOk3P0TJnZ@mail.gmail.com>
	<AANLkTinU_AqicisxSjOS-SJDGgfSpr1DYEeDZxQLmYp5@mail.gmail.com>
	<AANLkTi=W7yh2gZvzRG9mQs-qrSvrSw_eDw+PErp0dHWH@mail.gmail.com>
	<AANLkTimWTCAjkdPeDM5=OyZx5ctSvbjXMA3+r1jrcc96@mail.gmail.com>
	<AANLkTikQdyO4eaNibbxtgcKdCdFZsjbpsoWiNw_XBeYF@mail.gmail.com>
	<AANLkTikwx-j-5oo4A0L4oj_vWrpqYbs3uG69_SNwVD71@mail.gmail.com>
Message-ID: <AANLkTimo4v8WVhG7R=7WU43-TkgK+f=wmkO+UX17dTH8@mail.gmail.com>

True enough. Send me a card and I will give it a test drive! :)
-MC

2011/1/6 Jo?o Mesquita <jmesquita at freeswitch.org>

> That's because you've never tried Khomp. I would say it would be a very
> good battle to say the least!
>
> Jo?o Mesquita
>
>
>
> On Thu, Jan 6, 2011 at 7:18 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> FreeSWITCH works just fine with TDM cards. I tinkered with an old P4 and a
>> 4-port Tormenta2 clone and easily handled dozens of concurrent calls, and
>> that was before the FreeTDM stuff made things more reliable.
>>
>> At this point hardware is cheap, so get yourself a 4- or 8- core box with
>> 8GB RAM and some fast hard drives. For the best experience with TDM I
>> recommend a Sangoma card over the Zaptel clones.
>>
>>  -MC
>>
>>
>> On Thu, Jan 6, 2011 at 5:04 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> On 6 January 2011 12:47, ovvenkat <ovvenkatesan at gmail.com> wrote:
>>> > Thanks Steve,
>>> >
>>> > As of now, I am planning to handle 120 concurrent calls.
>>> > ( If client wants, I may increase concurrent calls )
>>> > Each call duration will be 90 - 120 seconds.
>>>
>>> Not sure what the minimum requirements are, but I have 24month old
>>> servers that are dual 4core Xeon E5405 with 8GB RAM. They wouldn't
>>> find 120 calls taxing at all.
>>>
>>> Load will depend what you're doing though - transcoding uses CPU, and
>>> things like IVR, recording, voicemail, conferencing will involve far
>>> more processing than just bridging a call.
>>>
>>> >
>>> > will  freeSwitch support OpenVox D410P 4-port E1/T1/J1 card?
>>>
>>> I haven't used FreeSWITCH with TDM personally, but AFAIK any zaptel
>>> compatible card will work with freetdm. That card appears to be zaptel
>>> compatible.
>>>
>>> Perhaps someone else can chime in to confirm?
>>>
>>> >
>>> > Call will be landing from the mobile phones, Do I need to use any
>>> codec?
>>>
>>> Calls always use a codec... you just need to pick which is best.
>>> You'll want to avoid using too much bandwidth so you can run over
>>> 2/2.5G networks. There's a few that would suit, including Speex, GSM,
>>> G729. G729 would need a $10/channel license, the rest are free.
>>>
>>> >
>>> > Regards,
>>> > Venkat.
>>> >
>>> >
>>> >
>>> >
>>> > On Thu, Jan 6, 2011 at 5:12 PM, Steven Ayre <steveayre at gmail.com>
>>> wrote:
>>> >>
>>> >> What do you expect the peak CPS and number of concurrent calls to be?
>>> >>
>>> >> Any current Xeon server should be able to do 3cps fine (72000 in 6
>>> >> hours). What really will dictate your hardware requirements is what
>>> >> the maximum load you want to handle at any one time is.
>>> >>
>>> >> -Steve
>>> >>
>>> >>
>>> >> On 6 January 2011 11:32, ovvenkat <ovvenkatesan at gmail.com> wrote:
>>> >> > Hi to all,
>>> >> >
>>> >> > I have to handle 72,000 inbound calls per day ( 6 - 8 hours ),
>>> >> > Anyone please suggest me the
>>> >> > hardware requirement for the same.
>>> >> >
>>> >> > Like, What kind of Server and how many PRI or BRI cards will fulfill
>>> the
>>> >> > needs.
>>> >> >
>>> >> > * Its Simple IVR, Call will be landing from mobile phones.
>>> >> >
>>> >> >
>>> >> > --
>>> >> >
>>> >> > If you have come to help me, you are wasting your time.
>>> >> > If you have come to because your liberation is bound up in mine, we
>>> can
>>> >> > work
>>> >> > together.
>>> >> >
>>> >> >
>>> >> > Regards
>>> >> > Venkatesan OV.
>>> >> >
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> > http://www.freeswitch.org
>>> >> >
>>> >> >
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> >
>>> > --
>>> >
>>> > If you have come to help me, you are wasting your time.
>>> > If you have come to because your liberation is bound up in mine, we can
>>> work
>>> > together.
>>> >
>>> >
>>> > Regards
>>> > Venkatesan OV.
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Fri Jan  7 05:05:19 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 6 Jan 2011 23:05:19 -0300
Subject: [Freeswitch-users] hardware requirements
In-Reply-To: <AANLkTimo4v8WVhG7R=7WU43-TkgK+f=wmkO+UX17dTH8@mail.gmail.com>
References: <AANLkTi=eXk=tRHK+Y8McrnUdYGvqHXthyYEOk3P0TJnZ@mail.gmail.com>
	<AANLkTinU_AqicisxSjOS-SJDGgfSpr1DYEeDZxQLmYp5@mail.gmail.com>
	<AANLkTi=W7yh2gZvzRG9mQs-qrSvrSw_eDw+PErp0dHWH@mail.gmail.com>
	<AANLkTimWTCAjkdPeDM5=OyZx5ctSvbjXMA3+r1jrcc96@mail.gmail.com>
	<AANLkTikQdyO4eaNibbxtgcKdCdFZsjbpsoWiNw_XBeYF@mail.gmail.com>
	<AANLkTikwx-j-5oo4A0L4oj_vWrpqYbs3uG69_SNwVD71@mail.gmail.com>
	<AANLkTimo4v8WVhG7R=7WU43-TkgK+f=wmkO+UX17dTH8@mail.gmail.com>
Message-ID: <AANLkTimrucLmYkib+5YzRpoc0XdZWMvEZjvTa8MBzT66@mail.gmail.com>

That can be arranged. Are any of you going to be at ITExpo Miami?

Regards,
Jo?o Mesquita



On Thu, Jan 6, 2011 at 8:59 PM, Michael Collins <msc at freeswitch.org> wrote:

> True enough. Send me a card and I will give it a test drive! :)
> -MC
>
> 2011/1/6 Jo?o Mesquita <jmesquita at freeswitch.org>
>
> That's because you've never tried Khomp. I would say it would be a very
>> good battle to say the least!
>>
>> Jo?o Mesquita
>>
>>
>>
>> On Thu, Jan 6, 2011 at 7:18 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> FreeSWITCH works just fine with TDM cards. I tinkered with an old P4 and
>>> a 4-port Tormenta2 clone and easily handled dozens of concurrent calls, and
>>> that was before the FreeTDM stuff made things more reliable.
>>>
>>> At this point hardware is cheap, so get yourself a 4- or 8- core box with
>>> 8GB RAM and some fast hard drives. For the best experience with TDM I
>>> recommend a Sangoma card over the Zaptel clones.
>>>
>>>  -MC
>>>
>>>
>>> On Thu, Jan 6, 2011 at 5:04 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>>
>>>> On 6 January 2011 12:47, ovvenkat <ovvenkatesan at gmail.com> wrote:
>>>> > Thanks Steve,
>>>> >
>>>> > As of now, I am planning to handle 120 concurrent calls.
>>>> > ( If client wants, I may increase concurrent calls )
>>>> > Each call duration will be 90 - 120 seconds.
>>>>
>>>> Not sure what the minimum requirements are, but I have 24month old
>>>> servers that are dual 4core Xeon E5405 with 8GB RAM. They wouldn't
>>>> find 120 calls taxing at all.
>>>>
>>>> Load will depend what you're doing though - transcoding uses CPU, and
>>>> things like IVR, recording, voicemail, conferencing will involve far
>>>> more processing than just bridging a call.
>>>>
>>>> >
>>>> > will  freeSwitch support OpenVox D410P 4-port E1/T1/J1 card?
>>>>
>>>> I haven't used FreeSWITCH with TDM personally, but AFAIK any zaptel
>>>> compatible card will work with freetdm. That card appears to be zaptel
>>>> compatible.
>>>>
>>>> Perhaps someone else can chime in to confirm?
>>>>
>>>> >
>>>> > Call will be landing from the mobile phones, Do I need to use any
>>>> codec?
>>>>
>>>> Calls always use a codec... you just need to pick which is best.
>>>> You'll want to avoid using too much bandwidth so you can run over
>>>> 2/2.5G networks. There's a few that would suit, including Speex, GSM,
>>>> G729. G729 would need a $10/channel license, the rest are free.
>>>>
>>>> >
>>>> > Regards,
>>>> > Venkat.
>>>> >
>>>> >
>>>> >
>>>> >
>>>> > On Thu, Jan 6, 2011 at 5:12 PM, Steven Ayre <steveayre at gmail.com>
>>>> wrote:
>>>> >>
>>>> >> What do you expect the peak CPS and number of concurrent calls to be?
>>>> >>
>>>> >> Any current Xeon server should be able to do 3cps fine (72000 in 6
>>>> >> hours). What really will dictate your hardware requirements is what
>>>> >> the maximum load you want to handle at any one time is.
>>>> >>
>>>> >> -Steve
>>>> >>
>>>> >>
>>>> >> On 6 January 2011 11:32, ovvenkat <ovvenkatesan at gmail.com> wrote:
>>>> >> > Hi to all,
>>>> >> >
>>>> >> > I have to handle 72,000 inbound calls per day ( 6 - 8 hours ),
>>>> >> > Anyone please suggest me the
>>>> >> > hardware requirement for the same.
>>>> >> >
>>>> >> > Like, What kind of Server and how many PRI or BRI cards will
>>>> fulfill the
>>>> >> > needs.
>>>> >> >
>>>> >> > * Its Simple IVR, Call will be landing from mobile phones.
>>>> >> >
>>>> >> >
>>>> >> > --
>>>> >> >
>>>> >> > If you have come to help me, you are wasting your time.
>>>> >> > If you have come to because your liberation is bound up in mine, we
>>>> can
>>>> >> > work
>>>> >> > together.
>>>> >> >
>>>> >> >
>>>> >> > Regards
>>>> >> > Venkatesan OV.
>>>> >> >
>>>> >> > _______________________________________________
>>>> >> > FreeSWITCH-users mailing list
>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> > http://www.freeswitch.org
>>>> >> >
>>>> >> >
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >
>>>> >
>>>> >
>>>> > --
>>>> >
>>>> > If you have come to help me, you are wasting your time.
>>>> > If you have come to because your liberation is bound up in mine, we
>>>> can work
>>>> > together.
>>>> >
>>>> >
>>>> > Regards
>>>> > Venkatesan OV.
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From hwnorman at hotmail.com  Fri Jan  7 06:22:40 2011
From: hwnorman at hotmail.com (Norman Lam)
Date: Fri, 7 Jan 2011 11:22:40 +0800
Subject: [Freeswitch-users] Iksemel msvs compiling
Message-ID: <SNT109-DS1155D401A0B9742CF63E13C10B0@phx.gbl>

Hi everyone 

 

After several time, I am still getting this error  compiling using Visual C
++ 2008 express ,  can any body shed some light in this 

 

I have try using the gnutls-2.10.1, also tried gnutls-2.9.9

 

Thanks in advance 

 

Norman Lam 

 

:

:

18>------ Build started: Project: iksemel, Configuration: Debug Win32 ------

18>Compiling...

18>dom.c

18>filter.c

16>ecp_nist.c

18>iks.c

18>ikstack.c

16>ecp_mont.c

18>io-posix.c

16>eck_prn.c

18>jabber.c

18>md5.c

16>ec_print.c

18>sax.c

18>sha.c

16>ec_pmeth.c

18>stream.c

18>..\..\iksemel\src\stream.c(11) : fatal error C1083: Cannot open include
file: 'pthread.h': No such file or directory

18>utility.c

18>base64.c

18>Generating Code...

18>Build log was saved at
"file://c:\FS_GIT2\libs\win32\iksemel\Debug\BuildLog.htm"

18>iksemel - 1 error(s), 0 warning(s)

:

:

:

111>------ Build started: Project: mod_dingaling, Configuration: Debug Win32
------

111>Compiling...

111>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

111>mod_dingaling.c

112>------ Build started: Project: mod_ilbc, Configuration: Debug Win32
------

112>Compiling...

112>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

112>mod_ilbc.c

111>Linking...

111>LINK : fatal error LNK1181: cannot open input file
'..\..\..\..\libs\win32\iksemel\debug\iksemel.lib'

112>Linking...

111>Build log was saved at
"file://c:\FS_GIT2\src\mod\endpoints\mod_dingaling\Win32\Debug\BuildLog.htm"

111>mod_dingaling - 1 error(s), 1 warning(s)

:

:

========== Build: 120 succeeded, 2 failed, 15 up-to-date, 3 skipped
==========

 

 

 

From: Norman Lam [mailto:hwnorman at hotmail.com] 
Sent: Wednesday, January 05, 2011 3:43 PM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: Iksemel msvs compiling

 

Hi Everyone 

 

I am trying to compile the  freeswitch with mod_dingaling and Iksemel to
work on Google voice or Gmail voice, I have read  

 

This  http://wiki.freeswitch.org/wiki/Iksemel_MSVS_project_example and the
http://wiki.freeswitch.org/wiki/Dingaling

 

But I am stuck at  this  clause  

 

Add HAVE_GNUTLS=1 the the preprocessor compile in the iksemel project

 

What does this mean and how to go about this 

 

Thanks in advance

 

Norman Lam 

 

 

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From ayhkor at gmail.com  Fri Jan  7 05:41:58 2011
From: ayhkor at gmail.com (deniro)
Date: Thu, 6 Jan 2011 21:41:58 -0500
Subject: [Freeswitch-users] mod_conference context
Message-ID: <AANLkTi=xmncvv3=kF9WmkjrenGAO9usJs8MHq1yTe826@mail.gmail.com>

in using conferencing software, how would I change or seperate context for
voip call and phone call?  Currently both use public
context(dialplan/public.xml).
I want phone dialing use a seperate context and provide pin number.
--phone calling use a pin number, voip call not asking any pin.
 I am looking for some guidence
thx
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From msc at freeswitch.org  Fri Jan  7 07:01:05 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Jan 2011 20:01:05 -0800
Subject: [Freeswitch-users] hardware requirements
In-Reply-To: <AANLkTimrucLmYkib+5YzRpoc0XdZWMvEZjvTa8MBzT66@mail.gmail.com>
References: <AANLkTi=eXk=tRHK+Y8McrnUdYGvqHXthyYEOk3P0TJnZ@mail.gmail.com>
	<AANLkTinU_AqicisxSjOS-SJDGgfSpr1DYEeDZxQLmYp5@mail.gmail.com>
	<AANLkTi=W7yh2gZvzRG9mQs-qrSvrSw_eDw+PErp0dHWH@mail.gmail.com>
	<AANLkTimWTCAjkdPeDM5=OyZx5ctSvbjXMA3+r1jrcc96@mail.gmail.com>
	<AANLkTikQdyO4eaNibbxtgcKdCdFZsjbpsoWiNw_XBeYF@mail.gmail.com>
	<AANLkTikwx-j-5oo4A0L4oj_vWrpqYbs3uG69_SNwVD71@mail.gmail.com>
	<AANLkTimo4v8WVhG7R=7WU43-TkgK+f=wmkO+UX17dTH8@mail.gmail.com>
	<AANLkTimrucLmYkib+5YzRpoc0XdZWMvEZjvTa8MBzT66@mail.gmail.com>
Message-ID: <AANLkTinRpSNKz0QNG9qa3wSJjLAMoBWs6P=M15m=o-Tg@mail.gmail.com>

Nope. However, I hear that we'll all be in Chicago this August! ;)

-MC

2011/1/6 Jo?o Mesquita <jmesquita at freeswitch.org>

> That can be arranged. Are any of you going to be at ITExpo Miami?
>
> Regards,
> Jo?o Mesquita
>
>
>
>
> On Thu, Jan 6, 2011 at 8:59 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> True enough. Send me a card and I will give it a test drive! :)
>> -MC
>>
>> 2011/1/6 Jo?o Mesquita <jmesquita at freeswitch.org>
>>
>> That's because you've never tried Khomp. I would say it would be a very
>>> good battle to say the least!
>>>
>>> Jo?o Mesquita
>>>
>>>
>>>
>>> On Thu, Jan 6, 2011 at 7:18 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> FreeSWITCH works just fine with TDM cards. I tinkered with an old P4 and
>>>> a 4-port Tormenta2 clone and easily handled dozens of concurrent calls, and
>>>> that was before the FreeTDM stuff made things more reliable.
>>>>
>>>> At this point hardware is cheap, so get yourself a 4- or 8- core box
>>>> with 8GB RAM and some fast hard drives. For the best experience with TDM I
>>>> recommend a Sangoma card over the Zaptel clones.
>>>>
>>>>  -MC
>>>>
>>>>
>>>> On Thu, Jan 6, 2011 at 5:04 AM, Steven Ayre <steveayre at gmail.com>wrote:
>>>>
>>>>> On 6 January 2011 12:47, ovvenkat <ovvenkatesan at gmail.com> wrote:
>>>>> > Thanks Steve,
>>>>> >
>>>>> > As of now, I am planning to handle 120 concurrent calls.
>>>>> > ( If client wants, I may increase concurrent calls )
>>>>> > Each call duration will be 90 - 120 seconds.
>>>>>
>>>>> Not sure what the minimum requirements are, but I have 24month old
>>>>> servers that are dual 4core Xeon E5405 with 8GB RAM. They wouldn't
>>>>> find 120 calls taxing at all.
>>>>>
>>>>> Load will depend what you're doing though - transcoding uses CPU, and
>>>>> things like IVR, recording, voicemail, conferencing will involve far
>>>>> more processing than just bridging a call.
>>>>>
>>>>> >
>>>>> > will  freeSwitch support OpenVox D410P 4-port E1/T1/J1 card?
>>>>>
>>>>> I haven't used FreeSWITCH with TDM personally, but AFAIK any zaptel
>>>>> compatible card will work with freetdm. That card appears to be zaptel
>>>>> compatible.
>>>>>
>>>>> Perhaps someone else can chime in to confirm?
>>>>>
>>>>> >
>>>>> > Call will be landing from the mobile phones, Do I need to use any
>>>>> codec?
>>>>>
>>>>> Calls always use a codec... you just need to pick which is best.
>>>>> You'll want to avoid using too much bandwidth so you can run over
>>>>> 2/2.5G networks. There's a few that would suit, including Speex, GSM,
>>>>> G729. G729 would need a $10/channel license, the rest are free.
>>>>>
>>>>> >
>>>>> > Regards,
>>>>> > Venkat.
>>>>> >
>>>>> >
>>>>> >
>>>>> >
>>>>> > On Thu, Jan 6, 2011 at 5:12 PM, Steven Ayre <steveayre at gmail.com>
>>>>> wrote:
>>>>> >>
>>>>> >> What do you expect the peak CPS and number of concurrent calls to
>>>>> be?
>>>>> >>
>>>>> >> Any current Xeon server should be able to do 3cps fine (72000 in 6
>>>>> >> hours). What really will dictate your hardware requirements is what
>>>>> >> the maximum load you want to handle at any one time is.
>>>>> >>
>>>>> >> -Steve
>>>>> >>
>>>>> >>
>>>>> >> On 6 January 2011 11:32, ovvenkat <ovvenkatesan at gmail.com> wrote:
>>>>> >> > Hi to all,
>>>>> >> >
>>>>> >> > I have to handle 72,000 inbound calls per day ( 6 - 8 hours ),
>>>>> >> > Anyone please suggest me the
>>>>> >> > hardware requirement for the same.
>>>>> >> >
>>>>> >> > Like, What kind of Server and how many PRI or BRI cards will
>>>>> fulfill the
>>>>> >> > needs.
>>>>> >> >
>>>>> >> > * Its Simple IVR, Call will be landing from mobile phones.
>>>>> >> >
>>>>> >> >
>>>>> >> > --
>>>>> >> >
>>>>> >> > If you have come to help me, you are wasting your time.
>>>>> >> > If you have come to because your liberation is bound up in mine,
>>>>> we can
>>>>> >> > work
>>>>> >> > together.
>>>>> >> >
>>>>> >> >
>>>>> >> > Regards
>>>>> >> > Venkatesan OV.
>>>>> >> >
>>>>> >> > _______________________________________________
>>>>> >> > FreeSWITCH-users mailing list
>>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> > http://www.freeswitch.org
>>>>> >> >
>>>>> >> >
>>>>> >>
>>>>> >> _______________________________________________
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> http://www.freeswitch.org
>>>>> >
>>>>> >
>>>>> >
>>>>> > --
>>>>> >
>>>>> > If you have come to help me, you are wasting your time.
>>>>> > If you have come to because your liberation is bound up in mine, we
>>>>> can work
>>>>> > together.
>>>>> >
>>>>> >
>>>>> > Regards
>>>>> > Venkatesan OV.
>>>>> >
>>>>> > _______________________________________________
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>> >
>>>>> >
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From msc at freeswitch.org  Fri Jan  7 07:06:10 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Jan 2011 20:06:10 -0800
Subject: [Freeswitch-users] Originate command freezes Grandstream phone
In-Reply-To: <4D25E272.1040509@dti-be.com>
References: <4D25E272.1040509@dti-be.com>
Message-ID: <AANLkTinVPd5mTT5DKXhObkYg-k9Y7nrjZFfMmLbQXS4A@mail.gmail.com>

Get a SIP trace on the working version and then get one on the latest that
doesn't work. Pastebin them so we can see if there are any obvious
differences. My guess is that GS is just being stupid and barfing on
something completely legal, but without the SIP traces there isn't much to
go on.

-MC

On Thu, Jan 6, 2011 at 7:40 AM, S?bastien Frippiat <sfrippiat at dti-be.com>wrote:

> Hello.
>
> I already posted about this problem without much answers
> (
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066132.html
> )
> and I thought it was the time to ask again, with more informations.
>
> First, I posted several parts of my configuration
> (http://pastebin.freeswitch.org/14936) as well as the full logs of my
> two tests (http://pastebin.freeswitch.org/14937).
>
> I use two phones (26 & 27 are their sip id):
> - 26 = Grandstream GXP2000
> - 27 = Snom M3
>
> The first test freezes the Grandstream phone but the second one works
> fine. The only thing changing in these two tests is the direction of the
> origination. Note that everything is on the same internal network.
>
> 1) originate {{origination_caller_id_name='Test originate
>
> 1',origination_caller_id_number='27',originate_timeout=20,effective_caller_id_name='User1',effective_caller_id_number='26',allow_outside_calls=true,outside_calls_gateway=xxxxxxxxx}}sofia/internal/26%
> 27
> 2) originate {{origination_caller_id_name='Test originate
>
> 2',origination_caller_id_number='26',originate_timeout=20,effective_caller_id_name='User2',effective_caller_id_number='27',allow_outside_calls=true,outside_calls_gateway=xxxxxxxxx}}sofia/internal/27%
> 26
>
> 1) Grandstream GXP2000 -> Snom M3 => KO (Grandstream completely freezes
> with horrible sounds on the speaker)
> 2) Snom M3 -> Grandstream GXP2000 => OK (no phones/audio problems)
>
> It was previously working with a version from 2010/09/09 and when we
> updated to 2010/12/09 (git-e680c82 2010-12-09 08-59-06 -0600), it wasn't
> working anymore.
> Latest version doesn't solve the issue.
>
> Note that the caller name show on the phone is "Outbound call" in this
> case but the correct name is shown when directly calling from a phone to
> another one.
>
> Anybody can help ?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Fri Jan  7 07:16:52 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Jan 2011 20:16:52 -0800
Subject: [Freeswitch-users] mod_conference pin
In-Reply-To: <AANLkTiku8cBTrKRJXhqDR8=RkTAH1oeZJc4gLTk=Uhtt@mail.gmail.com>
References: <AANLkTiku8cBTrKRJXhqDR8=RkTAH1oeZJc4gLTk=Uhtt@mail.gmail.com>
Message-ID: <AANLkTinMJeQaHiMnUFr2KXQfY-UwUtgYt=qS4Ejf-_nB@mail.gmail.com>

This is pretty easy once you know what to do. Is it safe to assume that your
outside call comes in on the public context and then gets transferred to the
extension that then sends the caller into the conference? If so, just create
a separate extension for the external caller and call the conference with
the +[pin]. Then for the "internal" extension that your auth'd users dial
into don't have the +[pin] on the conference action.

Alternatively, you could set a chan var when the caller comes in, something
like outside_caller=true, and then have your conference extension check for
that value. Either method would probably work for you. It's just a matter of
what fits your scenario the best and if you need it to scale, etc.

-MC

On Thu, Jan 6, 2011 at 11:09 AM, Ayhan Koroglu <ayhkor at gmail.com> wrote:

> Hi All
>
>  Using the web based conferencing software
> I am able to dial in (phone call) to conference  using pin number.
> But the problem is  sip client(voice conf) is also asking that same
> pin number.
> in dialplan
> confname at profile+[pin]
> or in profile
> <param name="pin" value="pin_number">
> How should I set my dial plan so that only phone dialing asks the pin
> number.
>
> Thanks
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri Jan  7 07:18:04 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Jan 2011 20:18:04 -0800
Subject: [Freeswitch-users] mod_conference context
In-Reply-To: <AANLkTi=xmncvv3=kF9WmkjrenGAO9usJs8MHq1yTe826@mail.gmail.com>
References: <AANLkTi=xmncvv3=kF9WmkjrenGAO9usJs8MHq1yTe826@mail.gmail.com>
Message-ID: <AANLkTinxaWBy6bBPrDMqxWHqwt5snQ1=SJS=tsaZS+rX@mail.gmail.com>

See my response to your other email...
-MC

On Thu, Jan 6, 2011 at 6:41 PM, deniro <ayhkor at gmail.com> wrote:

> in using conferencing software, how would I change or seperate context for
> voip call and phone call?  Currently both use public
> context(dialplan/public.xml).
> I want phone dialing use a seperate context and provide pin number.
> --phone calling use a pin number, voip call not asking any pin.
>  I am looking for some guidence
> thx
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jeff at jefflenk.com  Fri Jan  7 07:43:58 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 6 Jan 2011 20:43:58 -0800 (PST)
Subject: [Freeswitch-users] Iksemel msvs compiling
In-Reply-To: <SNT109-DS1155D401A0B9742CF63E13C10B0@phx.gbl>
References: <SNT109-DS13313A1FA906527438E8CC1090@phx.gbl>
	<SNT109-DS1155D401A0B9742CF63E13C10B0@phx.gbl>
Message-ID: <1294375438447-5898181.post@n2.nabble.com>


try adding ..\..\pthreads-w32-2-7-0-release; 
to the include path and see what happens
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Iksemel-msvs-compiling-tp5891263p5898181.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From u2nsam at gmail.com  Fri Jan  7 07:57:26 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 7 Jan 2011 10:27:26 +0530
Subject: [Freeswitch-users] Error in log with latest FreeTDM
In-Reply-To: <AANLkTikUnaFAJgxmMK=P1cE3QR+ayAOGSr=TTJFC=s3Y@mail.gmail.com>
References: <AANLkTik+pv3C3svaCRDUBE4P8OYGHj30tJ+q-1bqO9E2@mail.gmail.com>
	<AANLkTikABrSrKJ6brR=eQCe0F0+F25W08A9pxO=SYj8b@mail.gmail.com>
	<AANLkTikUnaFAJgxmMK=P1cE3QR+ayAOGSr=TTJFC=s3Y@mail.gmail.com>
Message-ID: <AANLkTikpHHquQEU1StYGwWXm_0jV+UuTPAb014xtQ0wB@mail.gmail.com>

When we tried calling through pri it gives this error without compiling with
the module sangoma codec.

2011-01-07 10:23:28.140119 [ERR] switch_core_session.c:380 Could not locate
channel type freetdm
2011-01-07 10:23:28.140119 [ERR] switch_ivr_originate.c:2614 Cannot create
outgoing channel of type [freetdm] cause: [CHAN_NOT_IMPLEMENTED]
2011-01-07 10:23:28.140119 [DEBUG] switch_ivr_originate.c:3435 Originate
Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
2011-01-07 10:23:28.140119 [INFO] mod_dptools.c:2599 Originate Failed.
Cause: CHAN_NOT_IMPLEMENTED


Regards
Sam

On Thu, Jan 6, 2011 at 9:00 PM, Moises Silva <moises.silva at gmail.com> wrote:

>
> On Thu, Jan 6, 2011 at 4:37 AM, Sam <u2nsam at gmail.com> wrote:
>
>> For me to while compiling....
>>
>>
>> This was the error while installing latest git-4272dcb 2011-01-05 20-12-19
>> -0600
>>
>>
> You are trying to compile the codec module without the proper supporting
> library + headers. Unless you have a D-series transcoding card, why are you
> trying to compile this module at all? if you do really need this module,
> follow the instructions at
> http://wiki.sangoma.com/sangoma-media-transcoding
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R
> 9R6 Canada
> t. 1 905 474 1990 x128 | e. moy at sangoma.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Fri Jan  7 09:37:48 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 7 Jan 2011 12:07:48 +0530
Subject: [Freeswitch-users] codec error
In-Reply-To: <AANLkTi=KdMjptEK5_sAWZ4DcXU1zJ9DstXnA_29rEx_k@mail.gmail.com>
References: <AANLkTi=cKTUOcN--hZm2yLXY8s-pWotek+_YA1Bxw2jw@mail.gmail.com>
	<AANLkTinLm5AP9kqoeHarqrCJGjsUq6Xo8=aOnY05HNHN@mail.gmail.com>
	<AANLkTikmHcMHfc3ydk+8b2SWEKB+sR=Awj0OVri6qdrK@mail.gmail.com>
	<AANLkTi=e4sVBpb2M5HSni-1ZXViExxkxocvmnC=jBBi_@mail.gmail.com>
	<AANLkTikGM8fQzqXaFoOd=m3LaVLae6B+O7qBPVAYmzL2@mail.gmail.com>
	<AANLkTim8Bfsr7=hX=RBJJEo4TKZqUZgn5y7yUbh9d1k1@mail.gmail.com>
	<AANLkTi=KdMjptEK5_sAWZ4DcXU1zJ9DstXnA_29rEx_k@mail.gmail.com>
Message-ID: <AANLkTinSByL-0TA+VthcBeiiewcHP3ZZEsR73FoMRts4@mail.gmail.com>

Where does it compare these codecs,

2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
[G722:9:8000:0:64000]/[G7221:115:32000:20:48000]
2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
[G722:9:8000:0:64000]/[G7221:107:16000:20:32000]
2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
[G722:9:8000:0:64000]/[PCMU:0:8000:20:64000]
2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
[G722:9:8000:0:64000]/[PCMA:8:8000:20:64000]
2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
[G722:9:8000:0:64000]/[G729:18:8000:20:8000]
2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
[telephone-event:127:8000:0:0]/[G7221:115:32000:20:48000]
2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
[telephone-event:127:8000:0:0]/[G7221:107:16000:20:32000]
2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
[telephone-event:127:8000:0:0]/[PCMU:0:8000:20:64000]
2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
[telephone-event:127:8000:0:0]/[PCMA:8:8000:20:64000]
2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
[telephone-event:127:8000:0:0]/[G729:18:8000:20:8000]


I need to add g722

Regds
Sam

On Thu, Jan 6, 2011 at 4:34 PM, Steven Ayre <steveayre at gmail.com> wrote:

> This sip profile parameter would allow the G729 call to work,
> providing G729 is enabled on the Cisco:
> <param name="disable-transcoding" value="true"/>
>
> This will mean the INVITE sent to the Cisco on the bleg only includes
> the codec from the aleg. The aleg is already using G729, so the Cisco
> will only be offered G729. It is then forced to either accept G729 or
> fail the call with Incompatible Destination.
>
> -Steve
>
>
> On 6 January 2011 10:59, Steven Ayre <steveayre at gmail.com> wrote:
> > G722.2
> > ======
> >
> > This codec is not enabled on your server.
> >
> > 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
> > Compare [g7222:110:16000:110:0]/[G7221:115:32000:20:48000]
> > 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
> > Compare [g7222:110:16000:110:0]/[G7221:107:16000:20:32000]
> > 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
> > Compare [g7222:110:16000:110:0]/[PCMU:0:8000:20:64000]
> > 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
> > Compare [g7222:110:16000:110:0]/[PCMA:8:8000:20:64000]
> > 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
> > Compare [g7222:110:16000:110:0]/[G729:18:8000:20:8000]
> >
> > You are calling with G722.2.
> >
> > G722.1 is enabled, but G722.2 is not.
> >
> > G729
> > ====
> >
> > It rings because Cisco returns 180 Ringing with no SDP. At that time
> > no codec has been selected. Anything you hear will be generated by
> > your SIP client.
> >
> > The codec is selected in the 200 OK w/SDP. That is the time the codec is
> picked.
> >
> > FreeSWITCH offers G729 to Cisco in the INVITE, plus several other codecs:
> >   m=audio 24636 RTP/AVP 18 98 99 0 8 101 13
> > The Cisco only responds with PCMU:
> >   m=audio 31600 RTP/AVP 0 101
> > ( 18=G729 0=PCMU 8=PCMA 98/99=G722.1 )
> >
> > Either your FS config has changed so that is offering more codecs than
> > before, or the Cisco's config has changed to prefer G711 over G729.
> >
> > -Steve
> >
> >
> > On 6 January 2011 10:45, Sam <u2nsam at gmail.com> wrote:
> >> http://pastebin.freeswitch.org/14934
> >> http://pastebin.freeswitch.org/14935
> >>
> >> These are the paste bins where calls with g722 and G729 call fails but
> g729
> >> calls rings and gets disconnected.
> >> In past it was working with both the codec on the same server as no
> there is
> >> no config change.
> >>
> >>
> >> Regds
> >> Sam
> >>
> >>
> >>
> >> On Thu, Jan 6, 2011 at 4:00 PM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >>>
> >>> G7222 call fails with incompatible destination because it's not
> >>> enabled on the server.
> >>>
> >>> G729 call starts a bridge to user/7013 at 192.168.2.190. They only offer
> >>> G711 ulaw. That means it's a G729 -> G711 call requiring transcoding.
> >>> There is no way that can work using mod_g729.
> >>>
> >>> If you believe that that user has G729 enabled, repeat the test with
> >>> sip trace enabled.
> >>> 'sofia global siptrace on'
> >>>
> >>> That'll let you see the INVITE w/SDP sent to the user. If G729 isn't
> >>> in the outgoing SDP it's a config problem on FreeSWITCH. If G729 is in
> >>> the SDP and the user is only responding with G711 then it's a problem
> >>> on the Cisco endpoint.
> >>>
> >>> -Steve
> >>>
> >>>
> >>>
> >>> On 6 January 2011 10:10, Sam <u2nsam at gmail.com> wrote:
> >>> > Hi,
> >>> >
> >>> > This was working earlier and the config file have not changed , only
> >>> > upgraded to latest git.
> >>> > have attached the file codec.txt .
> >>> >
> >>> >
> >>> >
> >>> > Regards
> >>> > Sam
> >>> >
> >>> >
> >>> >
> >>> > On Thu, Jan 6, 2011 at 3:20 PM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >>> >>
> >>> >> Some more information please...
> >>> >>
> >>> >> G729 has patents that mean it can't be used unlicensed. FreeSWITCH
> >>> >> either has mod_com_g729 for a full featured licensed version, or
> >>> >> mod_g729 which operates in passthrough mode for G729-G729 calls (it
> >>> >> needs no license since for these calls there is no encoding/decoding
> >>> >> step, the already encoded data is just passed straight through). You
> >>> >> will never have had mod_g729 working in the past for a transcoding
> >>> >> call, so that won't have changed.
> >>> >>
> >>> >> I would say that it's either:
> >>> >> - Your config files have changed
> >>> >> - The endpoints are offering different codecs from before
> >>> >> - Something changed in git about the codec negotiation
> >>> >>
> >>> >> Can you pastebin a debug level log of the calls, and enable
> siptrace?
> >>> >> Those will show what codecs are being offered, selected and show the
> >>> >> codec negotiation. It'd also be useful to know what your sip profile
> >>> >> config files look like since there are several options that adjust
> how
> >>> >> the negotiation is done.
> >>> >>
> >>> >> -Steve
> >>> >>
> >>> >>
> >>> >>
> >>> >> On 6 January 2011 09:43, Sam <u2nsam at gmail.com> wrote:
> >>> >> > I am getting the error after installing latest git from
> freeswitch,
> >>> >> > where as
> >>> >> > it was working from old git
> >>> >> >
> >>> >> >
> >>> >> > 2011-01-06 15:10:19.113781 [ERR] mod_g729.c:102 This codec is only
> >>> >> > usable in
> >>> >> > passthrough mode!
> >>> >> > 2011-01-06 15:10:19.113781 [ERR] switch_core_io.c:1042 Codec G.729
> >>> >> > encoder
> >>> >> > error!
> >>> >> >
> >>> >> >
> >>> >> > Regards
> >>> >> > Sam
> >>> >> >
> >>> >> > _______________________________________________
> >>> >> > FreeSWITCH-users mailing list
> >>> >> > FreeSWITCH-users at lists.freeswitch.org
> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> >
> >>> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> > http://www.freeswitch.org
> >>> >> >
> >>> >> >
> >>> >>
> >>> >> _______________________________________________
> >>> >> FreeSWITCH-users mailing list
> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>
> >>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> http://www.freeswitch.org
> >>> >
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From u2nsam at gmail.com  Fri Jan  7 09:58:18 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 7 Jan 2011 12:28:18 +0530
Subject: [Freeswitch-users] codec error
In-Reply-To: <AANLkTinSByL-0TA+VthcBeiiewcHP3ZZEsR73FoMRts4@mail.gmail.com>
References: <AANLkTi=cKTUOcN--hZm2yLXY8s-pWotek+_YA1Bxw2jw@mail.gmail.com>
	<AANLkTinLm5AP9kqoeHarqrCJGjsUq6Xo8=aOnY05HNHN@mail.gmail.com>
	<AANLkTikmHcMHfc3ydk+8b2SWEKB+sR=Awj0OVri6qdrK@mail.gmail.com>
	<AANLkTi=e4sVBpb2M5HSni-1ZXViExxkxocvmnC=jBBi_@mail.gmail.com>
	<AANLkTikGM8fQzqXaFoOd=m3LaVLae6B+O7qBPVAYmzL2@mail.gmail.com>
	<AANLkTim8Bfsr7=hX=RBJJEo4TKZqUZgn5y7yUbh9d1k1@mail.gmail.com>
	<AANLkTi=KdMjptEK5_sAWZ4DcXU1zJ9DstXnA_29rEx_k@mail.gmail.com>
	<AANLkTinSByL-0TA+VthcBeiiewcHP3ZZEsR73FoMRts4@mail.gmail.com>
Message-ID: <AANLkTi=X=RObn1xHxZpBR49Xpk+X3YLkRt3c=Aejo9j7@mail.gmail.com>

in show codec i see ;

show codecs
type,name,ikey
codec,AMR,mod_amr
codec,G.711 alaw,CORE_PCM_MODULE
codec,G.711 ulaw,CORE_PCM_MODULE
codec,G.723.1 6.3k,mod_g723_1
codec,G.729,mod_g729
codec,H.261 Video (passthru),mod_h26x
codec,H.263 Video (passthru),mod_h26x
codec,H.263+ Video (passthru),mod_h26x
codec,H.263++ Video (passthru),mod_h26x
codec,H.264 Video (passthru),mod_h26x
codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
codec,Polycom(R) G722.1/G722.1C,mod_siren
codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
codec,Speex,mod_speex
codec,iLBC,mod_ilbc

i have this in vars.xml

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h
,G7222,PCMU,PCMA,G729"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,G7222,G7221 at 32000h
,G7221 at 16000h,PCMU,PCMA,G729"/>



Regds
Sam



On Fri, Jan 7, 2011 at 12:07 PM, Sam <u2nsam at gmail.com> wrote:

> Where does it compare these codecs,
>
> 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
> [G722:9:8000:0:64000]/[G7221:115:32000:20:48000]
> 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
> [G722:9:8000:0:64000]/[G7221:107:16000:20:32000]
> 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
> [G722:9:8000:0:64000]/[PCMU:0:8000:20:64000]
> 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
> [G722:9:8000:0:64000]/[PCMA:8:8000:20:64000]
> 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
> [G722:9:8000:0:64000]/[G729:18:8000:20:8000]
> 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
> [telephone-event:127:8000:0:0]/[G7221:115:32000:20:48000]
> 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
> [telephone-event:127:8000:0:0]/[G7221:107:16000:20:32000]
> 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
> [telephone-event:127:8000:0:0]/[PCMU:0:8000:20:64000]
> 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
> [telephone-event:127:8000:0:0]/[PCMA:8:8000:20:64000]
> 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
> [telephone-event:127:8000:0:0]/[G729:18:8000:20:8000]
>
>
> I need to add g722
>
> Regds
> Sam
>
>
> On Thu, Jan 6, 2011 at 4:34 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> This sip profile parameter would allow the G729 call to work,
>> providing G729 is enabled on the Cisco:
>> <param name="disable-transcoding" value="true"/>
>>
>> This will mean the INVITE sent to the Cisco on the bleg only includes
>> the codec from the aleg. The aleg is already using G729, so the Cisco
>> will only be offered G729. It is then forced to either accept G729 or
>> fail the call with Incompatible Destination.
>>
>> -Steve
>>
>>
>> On 6 January 2011 10:59, Steven Ayre <steveayre at gmail.com> wrote:
>> > G722.2
>> > ======
>> >
>> > This codec is not enabled on your server.
>> >
>> > 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
>> > Compare [g7222:110:16000:110:0]/[G7221:115:32000:20:48000]
>> > 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
>> > Compare [g7222:110:16000:110:0]/[G7221:107:16000:20:32000]
>> > 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
>> > Compare [g7222:110:16000:110:0]/[PCMU:0:8000:20:64000]
>> > 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
>> > Compare [g7222:110:16000:110:0]/[PCMA:8:8000:20:64000]
>> > 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
>> > Compare [g7222:110:16000:110:0]/[G729:18:8000:20:8000]
>> >
>> > You are calling with G722.2.
>> >
>> > G722.1 is enabled, but G722.2 is not.
>> >
>> > G729
>> > ====
>> >
>> > It rings because Cisco returns 180 Ringing with no SDP. At that time
>> > no codec has been selected. Anything you hear will be generated by
>> > your SIP client.
>> >
>> > The codec is selected in the 200 OK w/SDP. That is the time the codec is
>> picked.
>> >
>> > FreeSWITCH offers G729 to Cisco in the INVITE, plus several other
>> codecs:
>> >   m=audio 24636 RTP/AVP 18 98 99 0 8 101 13
>> > The Cisco only responds with PCMU:
>> >   m=audio 31600 RTP/AVP 0 101
>> > ( 18=G729 0=PCMU 8=PCMA 98/99=G722.1 )
>> >
>> > Either your FS config has changed so that is offering more codecs than
>> > before, or the Cisco's config has changed to prefer G711 over G729.
>> >
>> > -Steve
>> >
>> >
>> > On 6 January 2011 10:45, Sam <u2nsam at gmail.com> wrote:
>> >> http://pastebin.freeswitch.org/14934
>> >> http://pastebin.freeswitch.org/14935
>> >>
>> >> These are the paste bins where calls with g722 and G729 call fails but
>> g729
>> >> calls rings and gets disconnected.
>> >> In past it was working with both the codec on the same server as no
>> there is
>> >> no config change.
>> >>
>> >>
>> >> Regds
>> >> Sam
>> >>
>> >>
>> >>
>> >> On Thu, Jan 6, 2011 at 4:00 PM, Steven Ayre <steveayre at gmail.com>
>> wrote:
>> >>>
>> >>> G7222 call fails with incompatible destination because it's not
>> >>> enabled on the server.
>> >>>
>> >>> G729 call starts a bridge to user/7013 at 192.168.2.190. They only offer
>> >>> G711 ulaw. That means it's a G729 -> G711 call requiring transcoding.
>> >>> There is no way that can work using mod_g729.
>> >>>
>> >>> If you believe that that user has G729 enabled, repeat the test with
>> >>> sip trace enabled.
>> >>> 'sofia global siptrace on'
>> >>>
>> >>> That'll let you see the INVITE w/SDP sent to the user. If G729 isn't
>> >>> in the outgoing SDP it's a config problem on FreeSWITCH. If G729 is in
>> >>> the SDP and the user is only responding with G711 then it's a problem
>> >>> on the Cisco endpoint.
>> >>>
>> >>> -Steve
>> >>>
>> >>>
>> >>>
>> >>> On 6 January 2011 10:10, Sam <u2nsam at gmail.com> wrote:
>> >>> > Hi,
>> >>> >
>> >>> > This was working earlier and the config file have not changed , only
>> >>> > upgraded to latest git.
>> >>> > have attached the file codec.txt .
>> >>> >
>> >>> >
>> >>> >
>> >>> > Regards
>> >>> > Sam
>> >>> >
>> >>> >
>> >>> >
>> >>> > On Thu, Jan 6, 2011 at 3:20 PM, Steven Ayre <steveayre at gmail.com>
>> wrote:
>> >>> >>
>> >>> >> Some more information please...
>> >>> >>
>> >>> >> G729 has patents that mean it can't be used unlicensed. FreeSWITCH
>> >>> >> either has mod_com_g729 for a full featured licensed version, or
>> >>> >> mod_g729 which operates in passthrough mode for G729-G729 calls (it
>> >>> >> needs no license since for these calls there is no
>> encoding/decoding
>> >>> >> step, the already encoded data is just passed straight through).
>> You
>> >>> >> will never have had mod_g729 working in the past for a transcoding
>> >>> >> call, so that won't have changed.
>> >>> >>
>> >>> >> I would say that it's either:
>> >>> >> - Your config files have changed
>> >>> >> - The endpoints are offering different codecs from before
>> >>> >> - Something changed in git about the codec negotiation
>> >>> >>
>> >>> >> Can you pastebin a debug level log of the calls, and enable
>> siptrace?
>> >>> >> Those will show what codecs are being offered, selected and show
>> the
>> >>> >> codec negotiation. It'd also be useful to know what your sip
>> profile
>> >>> >> config files look like since there are several options that adjust
>> how
>> >>> >> the negotiation is done.
>> >>> >>
>> >>> >> -Steve
>> >>> >>
>> >>> >>
>> >>> >>
>> >>> >> On 6 January 2011 09:43, Sam <u2nsam at gmail.com> wrote:
>> >>> >> > I am getting the error after installing latest git from
>> freeswitch,
>> >>> >> > where as
>> >>> >> > it was working from old git
>> >>> >> >
>> >>> >> >
>> >>> >> > 2011-01-06 15:10:19.113781 [ERR] mod_g729.c:102 This codec is
>> only
>> >>> >> > usable in
>> >>> >> > passthrough mode!
>> >>> >> > 2011-01-06 15:10:19.113781 [ERR] switch_core_io.c:1042 Codec
>> G.729
>> >>> >> > encoder
>> >>> >> > error!
>> >>> >> >
>> >>> >> >
>> >>> >> > Regards
>> >>> >> > Sam
>> >>> >> >
>> >>> >> > _______________________________________________
>> >>> >> > FreeSWITCH-users mailing list
>> >>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >> >
>> >>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >> > http://www.freeswitch.org
>> >>> >> >
>> >>> >> >
>> >>> >>
>> >>> >> _______________________________________________
>> >>> >> FreeSWITCH-users mailing list
>> >>> >> FreeSWITCH-users at lists.freeswitch.org
>> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>
>> >>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >> http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From steveayre at gmail.com  Fri Jan  7 11:15:56 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 7 Jan 2011 08:15:56 +0000
Subject: [Freeswitch-users] codec error
In-Reply-To: <AANLkTi=X=RObn1xHxZpBR49Xpk+X3YLkRt3c=Aejo9j7@mail.gmail.com>
References: <AANLkTi=cKTUOcN--hZm2yLXY8s-pWotek+_YA1Bxw2jw@mail.gmail.com>
	<AANLkTinLm5AP9kqoeHarqrCJGjsUq6Xo8=aOnY05HNHN@mail.gmail.com>
	<AANLkTikmHcMHfc3ydk+8b2SWEKB+sR=Awj0OVri6qdrK@mail.gmail.com>
	<AANLkTi=e4sVBpb2M5HSni-1ZXViExxkxocvmnC=jBBi_@mail.gmail.com>
	<AANLkTikGM8fQzqXaFoOd=m3LaVLae6B+O7qBPVAYmzL2@mail.gmail.com>
	<AANLkTim8Bfsr7=hX=RBJJEo4TKZqUZgn5y7yUbh9d1k1@mail.gmail.com>
	<AANLkTi=KdMjptEK5_sAWZ4DcXU1zJ9DstXnA_29rEx_k@mail.gmail.com>
	<AANLkTinSByL-0TA+VthcBeiiewcHP3ZZEsR73FoMRts4@mail.gmail.com>
	<AANLkTi=X=RObn1xHxZpBR49Xpk+X3YLkRt3c=Aejo9j7@mail.gmail.com>
Message-ID: <AANLkTi=7XX0gWgHAG6VZocUi29Oj9XroPp-TMzFSouNn@mail.gmail.com>

show codecs
type,name,ikey
codec,AMR,mod_amr
codec,G.711 alaw,CORE_PCM_MODULE
codec,G.711 ulaw,CORE_PCM_MODULE
codec,G.723.1 6.3k,mod_g723_1
codec,G.729,mod_g729
codec,H.261 Video (passthru),mod_h26x
codec,H.263 Video (passthru),mod_h26x
codec,H.263+ Video (passthru),mod_h26x
codec,H.263++ Video (passthru),mod_h26x
codec,H.264 Video (passthru),mod_h26x
codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
codec,Polycom(R) G722.1/G722.1C,mod_siren
codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
codec,Speex,mod_speex
codec,iLBC,mod_ilbc

You have no module loaded that gives G722. Add mod_spandsp to modules.conf.xml

<X-PRE-PROCESS cmd="set"
data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G7222,PCMU,PCMA,G729"/>
<X-PRE-PROCESS cmd="set"
data="outbound_codec_prefs=G722,G7222,G7221 at 32000h,G7221 at 16000h,PCMU,PCMA,G729"/>

That sets a variable. The actual definition of which codecs to use is
on the sip profile, which could use the variable that's set above, or
specify its own list. That should be fine as long as the profile uses
the variable, but they'll only be used if the codec is loaded - see
above.

-Steve


On 7 January 2011 06:58, Sam <u2nsam at gmail.com> wrote:
> in show codec i see ;
>
> show codecs
> type,name,ikey
> codec,AMR,mod_amr
> codec,G.711 alaw,CORE_PCM_MODULE
> codec,G.711 ulaw,CORE_PCM_MODULE
> codec,G.723.1 6.3k,mod_g723_1
> codec,G.729,mod_g729
> codec,H.261 Video (passthru),mod_h26x
> codec,H.263 Video (passthru),mod_h26x
> codec,H.263+ Video (passthru),mod_h26x
> codec,H.263++ Video (passthru),mod_h26x
> codec,H.264 Video (passthru),mod_h26x
> codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
> codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
> codec,Polycom(R) G722.1/G722.1C,mod_siren
> codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
> codec,Speex,mod_speex
> codec,iLBC,mod_ilbc
>
> i have this in vars.xml
>
> <X-PRE-PROCESS cmd="set"
> data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G7222,PCMU,PCMA,G729"/>
> <X-PRE-PROCESS cmd="set"
> data="outbound_codec_prefs=G722,G7222,G7221 at 32000h,G7221 at 16000h,PCMU,PCMA,G729"/>
>
>
>
> Regds
> Sam
>
>
>
> On Fri, Jan 7, 2011 at 12:07 PM, Sam <u2nsam at gmail.com> wrote:
>>
>> Where does it compare these codecs,
>>
>> 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
>> [G722:9:8000:0:64000]/[G7221:115:32000:20:48000]
>> 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
>> [G722:9:8000:0:64000]/[G7221:107:16000:20:32000]
>> 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
>> [G722:9:8000:0:64000]/[PCMU:0:8000:20:64000]
>> 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
>> [G722:9:8000:0:64000]/[PCMA:8:8000:20:64000]
>> 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
>> [G722:9:8000:0:64000]/[G729:18:8000:20:8000]
>> 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
>> [telephone-event:127:8000:0:0]/[G7221:115:32000:20:48000]
>> 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
>> [telephone-event:127:8000:0:0]/[G7221:107:16000:20:32000]
>> 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
>> [telephone-event:127:8000:0:0]/[PCMU:0:8000:20:64000]
>> 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
>> [telephone-event:127:8000:0:0]/[PCMA:8:8000:20:64000]
>> 2011-01-07 12:05:12.420182 [DEBUG] sofia_glue.c:4401 Audio Codec Compare
>> [telephone-event:127:8000:0:0]/[G729:18:8000:20:8000]
>>
>>
>> I need to add g722
>>
>> Regds
>> Sam
>>
>> On Thu, Jan 6, 2011 at 4:34 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>>
>>> This sip profile parameter would allow the G729 call to work,
>>> providing G729 is enabled on the Cisco:
>>> <param name="disable-transcoding" value="true"/>
>>>
>>> This will mean the INVITE sent to the Cisco on the bleg only includes
>>> the codec from the aleg. The aleg is already using G729, so the Cisco
>>> will only be offered G729. It is then forced to either accept G729 or
>>> fail the call with Incompatible Destination.
>>>
>>> -Steve
>>>
>>>
>>> On 6 January 2011 10:59, Steven Ayre <steveayre at gmail.com> wrote:
>>> > G722.2
>>> > ======
>>> >
>>> > This codec is not enabled on your server.
>>> >
>>> > 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
>>> > Compare [g7222:110:16000:110:0]/[G7221:115:32000:20:48000]
>>> > 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
>>> > Compare [g7222:110:16000:110:0]/[G7221:107:16000:20:32000]
>>> > 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
>>> > Compare [g7222:110:16000:110:0]/[PCMU:0:8000:20:64000]
>>> > 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
>>> > Compare [g7222:110:16000:110:0]/[PCMA:8:8000:20:64000]
>>> > 2011-01-06 15:30:43.041645 [DEBUG] sofia_glue.c:4403 Audio Codec
>>> > Compare [g7222:110:16000:110:0]/[G729:18:8000:20:8000]
>>> >
>>> > You are calling with G722.2.
>>> >
>>> > G722.1 is enabled, but G722.2 is not.
>>> >
>>> > G729
>>> > ====
>>> >
>>> > It rings because Cisco returns 180 Ringing with no SDP. At that time
>>> > no codec has been selected. Anything you hear will be generated by
>>> > your SIP client.
>>> >
>>> > The codec is selected in the 200 OK w/SDP. That is the time the codec
>>> > is picked.
>>> >
>>> > FreeSWITCH offers G729 to Cisco in the INVITE, plus several other
>>> > codecs:
>>> > ? m=audio 24636 RTP/AVP 18 98 99 0 8 101 13
>>> > The Cisco only responds with PCMU:
>>> > ? m=audio 31600 RTP/AVP 0 101
>>> > ( 18=G729 0=PCMU 8=PCMA 98/99=G722.1 )
>>> >
>>> > Either your FS config has changed so that is offering more codecs than
>>> > before, or the Cisco's config has changed to prefer G711 over G729.
>>> >
>>> > -Steve
>>> >
>>> >
>>> > On 6 January 2011 10:45, Sam <u2nsam at gmail.com> wrote:
>>> >> http://pastebin.freeswitch.org/14934
>>> >> http://pastebin.freeswitch.org/14935
>>> >>
>>> >> These are the paste bins where calls with g722 and G729 call fails but
>>> >> g729
>>> >> calls rings and gets disconnected.
>>> >> In past it was working with both the codec on the same server as no
>>> >> there is
>>> >> no config change.
>>> >>
>>> >>
>>> >> Regds
>>> >> Sam
>>> >>
>>> >>
>>> >>
>>> >> On Thu, Jan 6, 2011 at 4:00 PM, Steven Ayre <steveayre at gmail.com>
>>> >> wrote:
>>> >>>
>>> >>> G7222 call fails with incompatible destination because it's not
>>> >>> enabled on the server.
>>> >>>
>>> >>> G729 call starts a bridge to user/7013 at 192.168.2.190. They only offer
>>> >>> G711 ulaw. That means it's a G729 -> G711 call requiring transcoding.
>>> >>> There is no way that can work using mod_g729.
>>> >>>
>>> >>> If you believe that that user has G729 enabled, repeat the test with
>>> >>> sip trace enabled.
>>> >>> 'sofia global siptrace on'
>>> >>>
>>> >>> That'll let you see the INVITE w/SDP sent to the user. If G729 isn't
>>> >>> in the outgoing SDP it's a config problem on FreeSWITCH. If G729 is
>>> >>> in
>>> >>> the SDP and the user is only responding with G711 then it's a problem
>>> >>> on the Cisco endpoint.
>>> >>>
>>> >>> -Steve
>>> >>>
>>> >>>
>>> >>>
>>> >>> On 6 January 2011 10:10, Sam <u2nsam at gmail.com> wrote:
>>> >>> > Hi,
>>> >>> >
>>> >>> > This was working earlier and the config file have not changed ,
>>> >>> > only
>>> >>> > upgraded to latest git.
>>> >>> > have attached the file codec.txt .
>>> >>> >
>>> >>> >
>>> >>> >
>>> >>> > Regards
>>> >>> > Sam
>>> >>> >
>>> >>> >
>>> >>> >
>>> >>> > On Thu, Jan 6, 2011 at 3:20 PM, Steven Ayre <steveayre at gmail.com>
>>> >>> > wrote:
>>> >>> >>
>>> >>> >> Some more information please...
>>> >>> >>
>>> >>> >> G729 has patents that mean it can't be used unlicensed. FreeSWITCH
>>> >>> >> either has mod_com_g729 for a full featured licensed version, or
>>> >>> >> mod_g729 which operates in passthrough mode for G729-G729 calls
>>> >>> >> (it
>>> >>> >> needs no license since for these calls there is no
>>> >>> >> encoding/decoding
>>> >>> >> step, the already encoded data is just passed straight through).
>>> >>> >> You
>>> >>> >> will never have had mod_g729 working in the past for a transcoding
>>> >>> >> call, so that won't have changed.
>>> >>> >>
>>> >>> >> I would say that it's either:
>>> >>> >> - Your config files have changed
>>> >>> >> - The endpoints are offering different codecs from before
>>> >>> >> - Something changed in git about the codec negotiation
>>> >>> >>
>>> >>> >> Can you pastebin a debug level log of the calls, and enable
>>> >>> >> siptrace?
>>> >>> >> Those will show what codecs are being offered, selected and show
>>> >>> >> the
>>> >>> >> codec negotiation. It'd also be useful to know what your sip
>>> >>> >> profile
>>> >>> >> config files look like since there are several options that adjust
>>> >>> >> how
>>> >>> >> the negotiation is done.
>>> >>> >>
>>> >>> >> -Steve
>>> >>> >>
>>> >>> >>
>>> >>> >>
>>> >>> >> On 6 January 2011 09:43, Sam <u2nsam at gmail.com> wrote:
>>> >>> >> > I am getting the error after installing latest git from
>>> >>> >> > freeswitch,
>>> >>> >> > where as
>>> >>> >> > it was working from old git
>>> >>> >> >
>>> >>> >> >
>>> >>> >> > 2011-01-06 15:10:19.113781 [ERR] mod_g729.c:102 This codec is
>>> >>> >> > only
>>> >>> >> > usable in
>>> >>> >> > passthrough mode!
>>> >>> >> > 2011-01-06 15:10:19.113781 [ERR] switch_core_io.c:1042 Codec
>>> >>> >> > G.729
>>> >>> >> > encoder
>>> >>> >> > error!
>>> >>> >> >
>>> >>> >> >
>>> >>> >> > Regards
>>> >>> >> > Sam
>>> >>> >> >
>>> >>> >> > _______________________________________________
>>> >>> >> > FreeSWITCH-users mailing list
>>> >>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> >> >
>>> >>> >> >
>>> >>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> >> > http://www.freeswitch.org
>>> >>> >> >
>>> >>> >> >
>>> >>> >>
>>> >>> >> _______________________________________________
>>> >>> >> FreeSWITCH-users mailing list
>>> >>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> >>
>>> >>> >>
>>> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> >> http://www.freeswitch.org
>>> >>> >
>>> >>> >
>>> >>> > _______________________________________________
>>> >>> > FreeSWITCH-users mailing list
>>> >>> > FreeSWITCH-users at lists.freeswitch.org
>>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> >
>>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> > http://www.freeswitch.org
>>> >>> >
>>> >>> >
>>> >>>
>>> >>> _______________________________________________
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>
>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >>
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steveayre at gmail.com  Fri Jan  7 11:16:53 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 7 Jan 2011 08:16:53 +0000
Subject: [Freeswitch-users] Error in log with latest FreeTDM
In-Reply-To: <AANLkTikpHHquQEU1StYGwWXm_0jV+UuTPAb014xtQ0wB@mail.gmail.com>
References: <AANLkTik+pv3C3svaCRDUBE4P8OYGHj30tJ+q-1bqO9E2@mail.gmail.com>
	<AANLkTikABrSrKJ6brR=eQCe0F0+F25W08A9pxO=SYj8b@mail.gmail.com>
	<AANLkTikUnaFAJgxmMK=P1cE3QR+ayAOGSr=TTJFC=s3Y@mail.gmail.com>
	<AANLkTikpHHquQEU1StYGwWXm_0jV+UuTPAb014xtQ0wB@mail.gmail.com>
Message-ID: <AANLkTi=qNi1A1Uxx=i_qvMw5W5HkxHs7yY0g2XcHx_SS@mail.gmail.com>

You do not have mod_freetdm loaded, so the freetdm/ dialstring prefix
is not registered.

-Steve


On 7 January 2011 04:57, Sam <u2nsam at gmail.com> wrote:
> When we tried calling through pri it gives this error without compiling with
> the module sangoma codec.
>
> 2011-01-07 10:23:28.140119 [ERR] switch_core_session.c:380 Could not locate
> channel type freetdm
> 2011-01-07 10:23:28.140119 [ERR] switch_ivr_originate.c:2614 Cannot create
> outgoing channel of type [freetdm] cause: [CHAN_NOT_IMPLEMENTED]
> 2011-01-07 10:23:28.140119 [DEBUG] switch_ivr_originate.c:3435 Originate
> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
> 2011-01-07 10:23:28.140119 [INFO] mod_dptools.c:2599 Originate Failed.
> Cause: CHAN_NOT_IMPLEMENTED
>
>
> Regards
> Sam
>
> On Thu, Jan 6, 2011 at 9:00 PM, Moises Silva <moises.silva at gmail.com> wrote:
>>
>> On Thu, Jan 6, 2011 at 4:37 AM, Sam <u2nsam at gmail.com> wrote:
>>>
>>> For me to while compiling....
>>>
>>>
>>> This was the error while installing latest git-4272dcb 2011-01-05
>>> 20-12-19 -0600
>>
>> You are trying to compile the codec module without the proper supporting
>> library + headers. Unless you have a D-series transcoding card, why are you
>> trying to compile this module at all? if you do really need this module,
>> follow the instructions at
>> ?http://wiki.sangoma.com/sangoma-media-transcoding
>> Moises Silva
>> Senior Software Engineer
>> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R
>> 9R6 Canada
>> t. 1 905 474 1990 x128 | e.?moy at sangoma.com
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From ross at ossiantelecom.co.uk  Fri Jan  7 14:04:57 2011
From: ross at ossiantelecom.co.uk (Ross McKillop)
Date: Fri, 7 Jan 2011 11:04:57 +0000
Subject: [Freeswitch-users] Using "Reason" from hangup
In-Reply-To: <AANLkTim8ZoTJBXUJu59cWLCQqr99kWaX9V+8utMpKQ0g@mail.gmail.com>
References: <D522FF78-D722-43DD-9D4A-5382472457E7@ossiantelecom.co.uk>
	<AANLkTim8ZoTJBXUJu59cWLCQqr99kWaX9V+8utMpKQ0g@mail.gmail.com>
Message-ID: <2FD902BB-6288-4FAF-9AD7-9AB92CC57865@ossiantelecom.co.uk>

Happy New Year All,

Sorry to drag this one up again..  I thought I had resolved this but I haven't.. 

The dialplan is now as follows;

> <action application="set" data="continue_on_fail=true" />
> <action application="set" data="hangup_after_bridge=false" />
> <action application="set" data="sip_ignore_remote_cause=true" />
> <action application="bridge" data="{bypass_media=true}sofia/external/NUMBER at 10.0.44.1" />
> <action application="log" data="WARNING HC : ${hangup_cause} / BHC: ${bridge_hangup_cause} / OD: ${originate_disposition}" />
> <action application="hangup" data="${bridge_hangup_cause}" />

(have also tried <action application="hangup" data="${originate_disposition}" /> as well
as not setting the sip_ignore_remote_cause variable)

Of interest, perhaps, is that sometimes early media is returned with the 183 Session
Progress (usually a recorded message from the remote party explaining the fault) 
before the call is dropped by the TDM switch with a 480 and Reason header containing
an appropriate cause code.

I simply want FreeSWITCH to pass the Reason header from the bridge attempt back
to the A party unchanged.  I had limited success using ${originate_disposition} in the
hangup application however this does not work where there's early media (and in any
event doesn't return the same q.850 code but it does return something indicative of
an error (usually) "NO_USER_RESPONSE" but still with a code 16.

SIP Trace (altered to remove some host-identifying information) is available at;

	http://pastebin.freeswitch.org/14949

Regards,
Ross 

On 10 Nov 2010, at 17:03, Eduardo Nunes Pereira wrote:

> If you use hangup after a bridge FreeSWITCH overrides the cause with
> the cause recieved from the bridge application. You can avoid this
> using sip_ignore_remote_cause=true
> 
> On Wed, Nov 10, 2010 at 12:25 PM, Ross McKillop
> <ross at ossiantelecom.co.uk> wrote:
>> I have a FreeSWITCH box acting as a gateway between other servers
>> and a TDM switch, primarily for accounting purposes - The TDM switch
>> accurately sets a Reason header as follows;
>> 
>>> SIP/2.0 480 Temporarily Unavailable.
>>> Reason: Q.850;cause=31.
>> 
>> This is then passed by FreeSWITCH back to the A party as
>> 
>>> SIP/2.0 480 Temporarily Unavailable.
>>> Reason: Q.850;cause=16;text="NORMAL_CLEARING".
>> 
>> Is there any way to get FreeSWITCH to use the Reason header from
>> the failed B leg when replying to the A party ?
>> 
>> I tried setting continue_on_fail=true and hangup_after_bridge=false
>> and then adding a line after the bridge in the dialplan to hangup with
>> the bridge hangup cause e.g.;
>> 
>>> <action application="hangup" data="${bridge_hangup_cause}" />
>> 
>> 
>> But that hasn't affected the behaviour at all.
>> 
>> Regards,
>> Ross



From ovvenkatesan at gmail.com  Fri Jan  7 17:39:08 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Fri, 7 Jan 2011 20:09:08 +0530
Subject: [Freeswitch-users] call getting hangup after git update
Message-ID: <AANLkTi=z5KOP0sHwdYg+=tiF38tDS_X6QY1nk7ehuuLQ@mail.gmail.com>

Hi,

Today, I have updated freeSwitch to latest git.
After that, When I am trying to do outbound call,
Call is getting hangup. When I check the fs_cli
logs, its showing that,

*mod_dptools.c:2610 Originate Failed.  Cause: NO_ROUTE_DESTINATION*

Here is the logs for the same
http://pastebin.freeswitch.org/14951

I have added STD code , before the number,
Still no luck. I am getting error like

*Originate Failed.  Cause: NORMAL_UNSPECIFIED*


Here is the log after adding STD code

http://pastebin.freeswitch.org/14952


Regards,
Venkat.




-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
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From jmesquita at freeswitch.org  Fri Jan  7 18:12:31 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 7 Jan 2011 12:12:31 -0300
Subject: [Freeswitch-users] hardware requirements
In-Reply-To: <AANLkTinRpSNKz0QNG9qa3wSJjLAMoBWs6P=M15m=o-Tg@mail.gmail.com>
References: <AANLkTi=eXk=tRHK+Y8McrnUdYGvqHXthyYEOk3P0TJnZ@mail.gmail.com>
	<AANLkTinU_AqicisxSjOS-SJDGgfSpr1DYEeDZxQLmYp5@mail.gmail.com>
	<AANLkTi=W7yh2gZvzRG9mQs-qrSvrSw_eDw+PErp0dHWH@mail.gmail.com>
	<AANLkTimWTCAjkdPeDM5=OyZx5ctSvbjXMA3+r1jrcc96@mail.gmail.com>
	<AANLkTikQdyO4eaNibbxtgcKdCdFZsjbpsoWiNw_XBeYF@mail.gmail.com>
	<AANLkTikwx-j-5oo4A0L4oj_vWrpqYbs3uG69_SNwVD71@mail.gmail.com>
	<AANLkTimo4v8WVhG7R=7WU43-TkgK+f=wmkO+UX17dTH8@mail.gmail.com>
	<AANLkTimrucLmYkib+5YzRpoc0XdZWMvEZjvTa8MBzT66@mail.gmail.com>
	<AANLkTinRpSNKz0QNG9qa3wSJjLAMoBWs6P=M15m=o-Tg@mail.gmail.com>
Message-ID: <AANLkTim2eyF+pW2g9AQ79Cge77yA7rbQ=TtohKq7J0m6@mail.gmail.com>

This year, for sure.

I am already being a bitch at everyone! LOL

Jo?o Mesquita


On Fri, Jan 7, 2011 at 1:01 AM, Michael Collins <msc at freeswitch.org> wrote:

> Nope. However, I hear that we'll all be in Chicago this August! ;)
>
> -MC
>
> 2011/1/6 Jo?o Mesquita <jmesquita at freeswitch.org>
>
>> That can be arranged. Are any of you going to be at ITExpo Miami?
>>
>> Regards,
>> Jo?o Mesquita
>>
>>
>>
>>
>> On Thu, Jan 6, 2011 at 8:59 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> True enough. Send me a card and I will give it a test drive! :)
>>> -MC
>>>
>>> 2011/1/6 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>
>>> That's because you've never tried Khomp. I would say it would be a very
>>>> good battle to say the least!
>>>>
>>>> Jo?o Mesquita
>>>>
>>>>
>>>>
>>>> On Thu, Jan 6, 2011 at 7:18 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>> FreeSWITCH works just fine with TDM cards. I tinkered with an old P4
>>>>> and a 4-port Tormenta2 clone and easily handled dozens of concurrent calls,
>>>>> and that was before the FreeTDM stuff made things more reliable.
>>>>>
>>>>> At this point hardware is cheap, so get yourself a 4- or 8- core box
>>>>> with 8GB RAM and some fast hard drives. For the best experience with TDM I
>>>>> recommend a Sangoma card over the Zaptel clones.
>>>>>
>>>>>  -MC
>>>>>
>>>>>
>>>>> On Thu, Jan 6, 2011 at 5:04 AM, Steven Ayre <steveayre at gmail.com>wrote:
>>>>>
>>>>>> On 6 January 2011 12:47, ovvenkat <ovvenkatesan at gmail.com> wrote:
>>>>>> > Thanks Steve,
>>>>>> >
>>>>>> > As of now, I am planning to handle 120 concurrent calls.
>>>>>> > ( If client wants, I may increase concurrent calls )
>>>>>> > Each call duration will be 90 - 120 seconds.
>>>>>>
>>>>>> Not sure what the minimum requirements are, but I have 24month old
>>>>>> servers that are dual 4core Xeon E5405 with 8GB RAM. They wouldn't
>>>>>> find 120 calls taxing at all.
>>>>>>
>>>>>> Load will depend what you're doing though - transcoding uses CPU, and
>>>>>> things like IVR, recording, voicemail, conferencing will involve far
>>>>>> more processing than just bridging a call.
>>>>>>
>>>>>> >
>>>>>> > will  freeSwitch support OpenVox D410P 4-port E1/T1/J1 card?
>>>>>>
>>>>>> I haven't used FreeSWITCH with TDM personally, but AFAIK any zaptel
>>>>>> compatible card will work with freetdm. That card appears to be zaptel
>>>>>> compatible.
>>>>>>
>>>>>> Perhaps someone else can chime in to confirm?
>>>>>>
>>>>>> >
>>>>>> > Call will be landing from the mobile phones, Do I need to use any
>>>>>> codec?
>>>>>>
>>>>>> Calls always use a codec... you just need to pick which is best.
>>>>>> You'll want to avoid using too much bandwidth so you can run over
>>>>>> 2/2.5G networks. There's a few that would suit, including Speex, GSM,
>>>>>> G729. G729 would need a $10/channel license, the rest are free.
>>>>>>
>>>>>> >
>>>>>> > Regards,
>>>>>> > Venkat.
>>>>>> >
>>>>>> >
>>>>>> >
>>>>>> >
>>>>>> > On Thu, Jan 6, 2011 at 5:12 PM, Steven Ayre <steveayre at gmail.com>
>>>>>> wrote:
>>>>>> >>
>>>>>> >> What do you expect the peak CPS and number of concurrent calls to
>>>>>> be?
>>>>>> >>
>>>>>> >> Any current Xeon server should be able to do 3cps fine (72000 in 6
>>>>>> >> hours). What really will dictate your hardware requirements is what
>>>>>> >> the maximum load you want to handle at any one time is.
>>>>>> >>
>>>>>> >> -Steve
>>>>>> >>
>>>>>> >>
>>>>>> >> On 6 January 2011 11:32, ovvenkat <ovvenkatesan at gmail.com> wrote:
>>>>>> >> > Hi to all,
>>>>>> >> >
>>>>>> >> > I have to handle 72,000 inbound calls per day ( 6 - 8 hours ),
>>>>>> >> > Anyone please suggest me the
>>>>>> >> > hardware requirement for the same.
>>>>>> >> >
>>>>>> >> > Like, What kind of Server and how many PRI or BRI cards will
>>>>>> fulfill the
>>>>>> >> > needs.
>>>>>> >> >
>>>>>> >> > * Its Simple IVR, Call will be landing from mobile phones.
>>>>>> >> >
>>>>>> >> >
>>>>>> >> > --
>>>>>> >> >
>>>>>> >> > If you have come to help me, you are wasting your time.
>>>>>> >> > If you have come to because your liberation is bound up in mine,
>>>>>> we can
>>>>>> >> > work
>>>>>> >> > together.
>>>>>> >> >
>>>>>> >> >
>>>>>> >> > Regards
>>>>>> >> > Venkatesan OV.
>>>>>> >> >
>>>>>> >> > _______________________________________________
>>>>>> >> > FreeSWITCH-users mailing list
>>>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >> > UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> >> > http://www.freeswitch.org
>>>>>> >> >
>>>>>> >> >
>>>>>> >>
>>>>>> >> _______________________________________________
>>>>>> >> FreeSWITCH-users mailing list
>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> >> http://www.freeswitch.org
>>>>>> >
>>>>>> >
>>>>>> >
>>>>>> > --
>>>>>> >
>>>>>> > If you have come to help me, you are wasting your time.
>>>>>> > If you have come to because your liberation is bound up in mine, we
>>>>>> can work
>>>>>> > together.
>>>>>> >
>>>>>> >
>>>>>> > Regards
>>>>>> > Venkatesan OV.
>>>>>> >
>>>>>> > _______________________________________________
>>>>>> > FreeSWITCH-users mailing list
>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> > UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> > http://www.freeswitch.org
>>>>>> >
>>>>>> >
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Fri Jan  7 18:12:46 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 7 Jan 2011 09:12:46 -0600
Subject: [Freeswitch-users] Using "Reason" from hangup
In-Reply-To: <2FD902BB-6288-4FAF-9AD7-9AB92CC57865@ossiantelecom.co.uk>
References: <D522FF78-D722-43DD-9D4A-5382472457E7@ossiantelecom.co.uk>
	<AANLkTim8ZoTJBXUJu59cWLCQqr99kWaX9V+8utMpKQ0g@mail.gmail.com>
	<2FD902BB-6288-4FAF-9AD7-9AB92CC57865@ossiantelecom.co.uk>
Message-ID: <026C72DF-B796-4BCC-B502-269221A80430@freeswitch.org>

don't call hangup at all...  Just let it pass it back.

/b

On Jan 7, 2011, at 5:04 AM, Ross McKillop wrote:

> (have also tried <action application="hangup" data="${originate_disposition}" /> as well
> as not setting the sip_ignore_remote_cause variable)



From brian at freeswitch.org  Fri Jan  7 18:13:47 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 7 Jan 2011 09:13:47 -0600
Subject: [Freeswitch-users] codec error
In-Reply-To: <AANLkTi=X=RObn1xHxZpBR49Xpk+X3YLkRt3c=Aejo9j7@mail.gmail.com>
References: <AANLkTi=cKTUOcN--hZm2yLXY8s-pWotek+_YA1Bxw2jw@mail.gmail.com>
	<AANLkTinLm5AP9kqoeHarqrCJGjsUq6Xo8=aOnY05HNHN@mail.gmail.com>
	<AANLkTikmHcMHfc3ydk+8b2SWEKB+sR=Awj0OVri6qdrK@mail.gmail.com>
	<AANLkTi=e4sVBpb2M5HSni-1ZXViExxkxocvmnC=jBBi_@mail.gmail.com>
	<AANLkTikGM8fQzqXaFoOd=m3LaVLae6B+O7qBPVAYmzL2@mail.gmail.com>
	<AANLkTim8Bfsr7=hX=RBJJEo4TKZqUZgn5y7yUbh9d1k1@mail.gmail.com>
	<AANLkTi=KdMjptEK5_sAWZ4DcXU1zJ9DstXnA_29rEx_k@mail.gmail.com>
	<AANLkTinSByL-0TA+VthcBeiiewcHP3ZZEsR73FoMRts4@mail.gmail.com>
	<AANLkTi=X=RObn1xHxZpBR49Xpk+X3YLkRt3c=Aejo9j7@mail.gmail.com>
Message-ID: <2E41E910-1DD0-4F24-B613-96A3A5BE0AF6@freeswitch.org>

For one you have G7222 ... secondly you do not have mod_spandsp loaded.

/b

On Jan 7, 2011, at 12:58 AM, Sam wrote:

> in show codec i see ;
> 
> show codecs
> type,name,ikey
> codec,AMR,mod_amr
> codec,G.711 alaw,CORE_PCM_MODULE
> codec,G.711 ulaw,CORE_PCM_MODULE
> codec,G.723.1 6.3k,mod_g723_1
> codec,G.729,mod_g729
> codec,H.261 Video (passthru),mod_h26x
> codec,H.263 Video (passthru),mod_h26x
> codec,H.263+ Video (passthru),mod_h26x
> codec,H.263++ Video (passthru),mod_h26x
> codec,H.264 Video (passthru),mod_h26x
> codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
> codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
> codec,Polycom(R) G722.1/G722.1C,mod_siren
> codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
> codec,Speex,mod_speex
> codec,iLBC,mod_ilbc
> 
> i have this in vars.xml
> 
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G7222,PCMU,PCMA,G729"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,G7222,G7221 at 32000h,G7221 at 16000h,PCMU,PCMA,G729"/>
> 
> 
> 
> Regds
> Sam
> 



From brian at freeswitch.org  Fri Jan  7 18:14:45 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 7 Jan 2011 09:14:45 -0600
Subject: [Freeswitch-users] call getting hangup after git update
In-Reply-To: <AANLkTi=z5KOP0sHwdYg+=tiF38tDS_X6QY1nk7ehuuLQ@mail.gmail.com>
References: <AANLkTi=z5KOP0sHwdYg+=tiF38tDS_X6QY1nk7ehuuLQ@mail.gmail.com>
Message-ID: <2B571B49-1886-445D-8473-C34CC9039301@freeswitch.org>

You're getting no route back from your freetdm circuit.

/b

On Jan 7, 2011, at 8:39 AM, ovvenkat wrote:

> 
> Hi, 
> 
> Today, I have updated freeSwitch to latest git. 
> After that, When I am trying to do outbound call,
> Call is getting hangup. When I check the fs_cli
> logs, its showing that, 
> 
> mod_dptools.c:2610 Originate Failed.  Cause: NO_ROUTE_DESTINATION
> 
> Here is the logs for the same 
> http://pastebin.freeswitch.org/14951
> 
> I have added STD code , before the number, 
> Still no luck. I am getting error like
> 
> Originate Failed.  Cause: NORMAL_UNSPECIFIED
> 
> 
> Here is the log after adding STD code
> 
> http://pastebin.freeswitch.org/14952
> 
> 
> Regards,
> Venkat.
> 
> 

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From ovvenkatesan at gmail.com  Fri Jan  7 18:20:56 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Fri, 7 Jan 2011 20:50:56 +0530
Subject: [Freeswitch-users] call getting hangup after git update
In-Reply-To: <2B571B49-1886-445D-8473-C34CC9039301@freeswitch.org>
References: <AANLkTi=z5KOP0sHwdYg+=tiF38tDS_X6QY1nk7ehuuLQ@mail.gmail.com>
	<2B571B49-1886-445D-8473-C34CC9039301@freeswitch.org>
Message-ID: <AANLkTikPa33av7_JwFHZ3rfFhKsiYR-arnvZ_fO0EaDX@mail.gmail.com>

Hi Brian,


I am not getting it. Could you plz elaborate it.
How it can be resolved?

I am still newbie in freeSwitch :)

Regards,
Venkat.



On Fri, Jan 7, 2011 at 8:44 PM, Brian West <brian at freeswitch.org> wrote:

> You're getting no route back from your freetdm circuit.
>
> /b
>
> On Jan 7, 2011, at 8:39 AM, ovvenkat wrote:
>
>
> Hi,
>
> Today, I have updated freeSwitch to latest git.
> After that, When I am trying to do outbound call,
> Call is getting hangup. When I check the fs_cli
> logs, its showing that,
>
> *mod_dptools.c:2610 Originate Failed.  Cause: NO_ROUTE_DESTINATION*
>
> Here is the logs for the same
> http://pastebin.freeswitch.org/14951
>
> I have added STD code , before the number,
> Still no luck. I am getting error like
>
> *Originate Failed.  Cause: NORMAL_UNSPECIFIED*
>
>
> Here is the log after adding STD code
>
> http://pastebin.freeswitch.org/14952
>
>
> Regards,
> Venkat.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
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From lloyd.aloysius at gmail.com  Fri Jan  7 19:05:49 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 7 Jan 2011 11:05:49 -0500
Subject: [Freeswitch-users] FreeSWITCH - Ploycom phone in hosted environment
Message-ID: <AANLkTimiG_GyWLO_JM+HYMM6bB0izxci1hZ=HosQwJsL@mail.gmail.com>

Hi All,

Ploycom phone's not supporting rport. How to use Ploycom phone in the Hosted
environment. Polycom phone in a NAT environment and FreeSWITCH hosted in the
Datacenter with public IP.

I could not make it work. Any help is appreciated.

Thanks
Lloyd
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From ross at ossiantelecom.co.uk  Fri Jan  7 20:00:54 2011
From: ross at ossiantelecom.co.uk (Ross McKillop)
Date: Fri, 7 Jan 2011 17:00:54 +0000
Subject: [Freeswitch-users] Using "Reason" from hangup
In-Reply-To: <026C72DF-B796-4BCC-B502-269221A80430@freeswitch.org>
References: <D522FF78-D722-43DD-9D4A-5382472457E7@ossiantelecom.co.uk>
	<AANLkTim8ZoTJBXUJu59cWLCQqr99kWaX9V+8utMpKQ0g@mail.gmail.com>
	<2FD902BB-6288-4FAF-9AD7-9AB92CC57865@ossiantelecom.co.uk>
	<026C72DF-B796-4BCC-B502-269221A80430@freeswitch.org>
Message-ID: <AC17FDFB-B739-41EE-A310-F8418E19CEF2@ossiantelecom.co.uk>


On 7 Jan 2011, at 15:12, Brian West wrote:

> don't call hangup at all...  Just let it pass it back.

Do you need hangup_after_bridge=true for this ? 

Previously (with hangup_after_bridge true + no hangup application call) it still returned 16 "NORMAL_CLEARING" even when the B party ended with cause 31.

> On Jan 7, 2011, at 5:04 AM, Ross McKillop wrote:
> 
>> (have also tried <action application="hangup" data="${originate_disposition}" /> as well
>> as not setting the sip_ignore_remote_cause variable)
> 



From ovvenkatesan at gmail.com  Fri Jan  7 20:25:10 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Fri, 7 Jan 2011 22:55:10 +0530
Subject: [Freeswitch-users] call getting hangup after git update
In-Reply-To: <2B571B49-1886-445D-8473-C34CC9039301@freeswitch.org>
References: <AANLkTi=z5KOP0sHwdYg+=tiF38tDS_X6QY1nk7ehuuLQ@mail.gmail.com>
	<2B571B49-1886-445D-8473-C34CC9039301@freeswitch.org>
Message-ID: <AANLkTikqr10zdWni96-RUWyRrpzus+J1God_yh7JiH9A@mail.gmail.com>

Hi Brian,

On Fri, Jan 7, 2011 at 8:44 PM, Brian West <brian at freeswitch.org> wrote:

> You're getting no route back from your freetdm circuit.
>
>

When I am calling to mobile number, it works fine.
Its giving the problem only with fixed line numbers.
Can you please tell , why its so?




> /b
>
> On Jan 7, 2011, at 8:39 AM, ovvenkat wrote:
>
>
> Hi,
>
> Today, I have updated freeSwitch to latest git.
> After that, When I am trying to do outbound call,
> Call is getting hangup. When I check the fs_cli
> logs, its showing that,
>
> *mod_dptools.c:2610 Originate Failed.  Cause: NO_ROUTE_DESTINATION*
>
> Here is the logs for the same
> http://pastebin.freeswitch.org/14951
>
> I have added STD code , before the number,
> Still no luck. I am getting error like
>
> *Originate Failed.  Cause: NORMAL_UNSPECIFIED*
>
>
> Here is the log after adding STD code
>
> http://pastebin.freeswitch.org/14952
>
>
> Regards,
> Venkat.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
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From brian at freeswitch.org  Fri Jan  7 20:30:08 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 7 Jan 2011 11:30:08 -0600
Subject: [Freeswitch-users] call getting hangup after git update
In-Reply-To: <AANLkTikqr10zdWni96-RUWyRrpzus+J1God_yh7JiH9A@mail.gmail.com>
References: <AANLkTi=z5KOP0sHwdYg+=tiF38tDS_X6QY1nk7ehuuLQ@mail.gmail.com>
	<2B571B49-1886-445D-8473-C34CC9039301@freeswitch.org>
	<AANLkTikqr10zdWni96-RUWyRrpzus+J1God_yh7JiH9A@mail.gmail.com>
Message-ID: <CCA243AB-7BC1-4A66-A80E-D671B34617F3@freeswitch.org>

I'm going to guess its missing an NPI/TON setting for the outbound call.
/b

On Jan 7, 2011, at 11:25 AM, ovvenkat wrote:

> Hi Brian,
> 
> On Fri, Jan 7, 2011 at 8:44 PM, Brian West <brian at freeswitch.org> wrote:
> You're getting no route back from your freetdm circuit.
> 
> 
> 
> When I am calling to mobile number, it works fine. 
> Its giving the problem only with fixed line numbers. 
> Can you please tell , why its so? 
> 
> 
>  
> /b

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From bansal.rajeshkr at gmail.com  Fri Jan  7 18:15:37 2011
From: bansal.rajeshkr at gmail.com (Rajesh Bansal)
Date: Fri, 7 Jan 2011 20:45:37 +0530
Subject: [Freeswitch-users] Getting Error ORA-923 in oracle query execution
Message-ID: <AANLkTi=k-cwDWxvCMne=O958yhXd8=bkr1EWtPAORxL6@mail.gmail.com>

Hi ,

I am getting error ORA -923 (from missing from statement) when i am tring to
execute a sql query from Javascript file. Here i am successfully able to
connect & execute queries with MYSQL. But in oracle connection i am getting
this error even i can successfully make a connection with oracle. with isql
& a program written in C i can connect and execute queries ok.

I am using
FreeSwitch 1.0.6
unixOdbc    2.3.0

Please tell me where is problem.

Best Regards,
Rajesh Bansal
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From cjbujold at accra.ca  Fri Jan  7 20:32:02 2011
From: cjbujold at accra.ca (Charles Bujold)
Date: Fri, 7 Jan 2011 13:32:02 -0400
Subject: [Freeswitch-users] Unable to get Freeswitch to register with
	Grandstream HT503
Message-ID: <004801cbae90$cbf44ca0$63dce5e0$@accra.ca>

Unable to get the HT-503 to register with Freeswitch.  Does anybody know
what configuration setting in the HT503 that are needed.

 

Thanks

cjb

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From Nabble at slickdeals.endjunk.com  Fri Jan  7 21:04:43 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 7 Jan 2011 10:04:43 -0800 (PST)
Subject: [Freeswitch-users] Unable to get Freeswitch to register with
	Grandstream HT503
In-Reply-To: <004801cbae90$cbf44ca0$63dce5e0$@accra.ca>
References: <004801cbae90$cbf44ca0$63dce5e0$@accra.ca>
Message-ID: <1294423483344-5900237.post@n2.nabble.com>



Charles Bujold wrote:
> Unable to get the HT-503 to register with Freeswitch.  Does anybody know
> what configuration setting in the HT503 that are needed.
I only use a Linksys PAP2v1/2 and Uniden UIP1869V phone. There isn't any
special configuration needed to make my devices registered to my FS v1.0.6
hosted on a Seagate 
http://www.seagate.com/www/en-us/products/network_storage/freeagent_dockstar
DockStar . As a matter of fact, those devices were once configured to
registered to my Asterisk PBX System and I just changed the proxy server IP
Address to as well as the username/password on my FS IP Address.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Unable-to-get-Freeswitch-to-register-with-Grandstream-HT503-tp5900205p5900237.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Fri Jan  7 21:05:41 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 7 Jan 2011 10:05:41 -0800 (PST)
Subject: [Freeswitch-users] Unable to get Freeswitch to register with
	Grandstream HT503
In-Reply-To: <004801cbae90$cbf44ca0$63dce5e0$@accra.ca>
References: <004801cbae90$cbf44ca0$63dce5e0$@accra.ca>
Message-ID: <1294423541528-5900238.post@n2.nabble.com>



Charles Bujold wrote:
> Unable to get the HT-503 to register with Freeswitch.  Does anybody know
> what configuration setting in the HT503 that are needed.
I only use a Linksys PAP2v1/2 and Uniden UIP1869V phone so I won't know
about any HT-503 devices.  For my devices, there isn't any special
configuration needed to make them registered to my FS v1.0.6 hosted on a
Seagate 
http://www.seagate.com/www/en-us/products/network_storage/freeagent_dockstar
DockStar . As a matter of fact, those devices were once configured to
registered to my Asterisk PBX System and I just changed the proxy server IP
Address to as well as the username/password on my FS IP Address.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Unable-to-get-Freeswitch-to-register-with-Grandstream-HT503-tp5900205p5900238.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Fri Jan  7 21:05:04 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 7 Jan 2011 18:05:04 +0000
Subject: [Freeswitch-users] Getting Error ORA-923 in oracle query
	execution
In-Reply-To: <AANLkTi=k-cwDWxvCMne=O958yhXd8=bkr1EWtPAORxL6@mail.gmail.com>
References: <AANLkTi=k-cwDWxvCMne=O958yhXd8=bkr1EWtPAORxL6@mail.gmail.com>
Message-ID: <AANLkTi=u9QUHh1GD3MpQTB81N5zW5xwAcWazWzuU0A6Q@mail.gmail.com>

Sounds like a SQL syntax error - can you show us the code that's
executing the statement that gives the error?

Specifically Oracle is complaining about not finding FROM where it
expects to in the SELECT.

-Steve


On 7 January 2011 15:15, Rajesh Bansal <bansal.rajeshkr at gmail.com> wrote:
> Hi ,
> I am getting error ORA -923 (from missing from statement) when i am tring to
> execute a sql query from Javascript file. Here i am successfully able to
> connect & execute queries with MYSQL. But in oracle connection i am getting
> this error even i can successfully make a connection with oracle. with isql
> & a program written in C i can connect and execute queries ok.
> I am using
> FreeSwitch 1.0.6
> unixOdbc ? ?2.3.0
> Please tell me where is problem.
> Best Regards,
> Rajesh Bansal
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From infos at madovsky.org  Fri Jan  7 21:28:00 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 7 Jan 2011 13:28:00 -0500
Subject: [Freeswitch-users] mod_conference with cluster ODBC
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705><AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com><AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com><AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com><AANLkTimutwOyeubcS=tax5EbDEniCD+Nc4iHMydQxpFb@mail.gmail.com><37761574FEE44D13BEBE4A7DE0089083@e1705><AANLkTi=VZ8JZ+PuyXoz_EYNVovY-AOuPyz9+yfKoUc3k@mail.gmail.com><337274A8CA2343259DF18D8135AF2DCE@e1705>
	<AANLkTimVaxdbwVHCTqpYC5w3q+Vi+t7-oMekrkxu4Kfd@mail.gmail.com>
Message-ID: <5B0F4D23C27A47A08D36C36B78AB981B@e1705>

I got it thanks,
but do you think it would be more interesting to reduce
bandwidth and latency between nodes and centralize the conference on one node only
by transferring the incoming user to the right node ?
  ----- Original Message ----- 
  From: Jo?o Mesquita 
  To: FreeSWITCH Users Help 
  Sent: Thursday, January 06, 2011 4:33 PM
  Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


  Ok, let me see if I can get this into your head. (giggles)


  A conference means that the audio needs to mixed in together so that all participants can talk/hear each other, right? If you implement something in C on mod_conference, you are going to essentially do the same as what an ESL app does. You _need_ to call in from one server to the other so that you can mix the audio of all the participants. The real advantage would be the management API being only one for everything and the challenge is exactly that. How to mute certain users on a conference that is spanning over 10 servers or deaf them, etc...


  A SIP "user" is easier because you don't have to bridge audio from another server necessarily. Got it?


  Regards,
  Jo?o Mesquita



  On Thu, Jan 6, 2011 at 6:25 PM, Madovsky <infos at madovsky.org> wrote:

    Rupa,

    I don't want bother anyone with this thread but why not
    to manage conference as SIP user ?
    if someone from server A call an other who is registered on server B, so
    FS do it automatically, why not with conference ? Or maybe create a param
    in mod_conference that let the choice of the admin to manage unique name in
    all cluster or not.
    like <param name="cluster_unique_name" value="true"/>
    I will try to understand the C code to hack something like this...


    ----- Original Message -----
    From: "Rupa Schomaker" <rupa at rupa.com>
    To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

    Sent: Thursday, January 06, 2011 3:01 PM
    Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


    Yes

    On Thu, Jan 6, 2011 at 1:40 PM, Madovsky <infos at madovsky.org> wrote:
    > in case of you have 8 servers you have to do it for each ?
    >
    > Thanks
    >
    > ----- Original Message -----
    > From: joy this
    > To: FreeSWITCH Users Help
    > Sent: Thursday, January 06, 2011 2:51 AM
    > Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
    > It works. Thank you everyone.
    >
    > 2011/1/5 Rupa Schomaker <rupa at rupa.com>
    >>
    >> Use the api: conference <confname> dial [{dial string
    >> options}]<endpoint_module_name>/<destination> [<callerid_number>
    >> [<callerid_name>]]
    >> To initiate the call from within conference A on server 1. Have a
    >> corresponding dialplan entry on server 2 to accept the call and add it
    >> into
    >> the conference A on server 2. You've now bridged the two conferences in
    >> the
    >> two servers.
    >
    > ________________________________
    >
    > _______________________________________________
    > FreeSWITCH-users mailing list
    > FreeSWITCH-users at lists.freeswitch.org
    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > http://www.freeswitch.org
    >
    > --
    > This message has been scanned for viruses and
    > dangerous content by MailScanner, and is
    > believed to be clean.
    > _______________________________________________
    > FreeSWITCH-users mailing list
    > FreeSWITCH-users at lists.freeswitch.org
    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > http://www.freeswitch.org
    >
    >



    --
    -Rupa

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





------------------------------------------------------------------------------


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From infos at madovsky.org  Fri Jan  7 21:38:46 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 7 Jan 2011 13:38:46 -0500
Subject: [Freeswitch-users] cepstral
Message-ID: <296FCE4397234877891847EC6F7D235D@e1705>

Is anyone know if Cepstral is alive yet ?
I bought voices and contacted support but no answer since 4 days now.

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From kahn at vestec.com  Fri Jan  7 22:15:43 2011
From: kahn at vestec.com (Kashif Kahn)
Date: Fri, 07 Jan 2011 14:15:43 -0500
Subject: [Freeswitch-users] Vestec Speech Engine: ASR 2.1 Release
Message-ID: <4D27665F.4000406@vestec.com>

Dear All,

We have launched a major upgrade to our ASR engine that offers the best 
deal around for enabling speech recognition with "command and control" 
type IVR applications. The new architecture boasts a number of 
advancements over version 1.1, including:

- Improved US English acoustic model
- DTMF recognition
- MRCP support v1 and v2
- Highly scalable architecture and Redundancy
- C++ API availability
- SRGS-XML (.grxml) grammar support
- Improved logging
- Bug Fixes

The engine can be integrated with different contact center and soft-PBX 
platforms (such as Freeswitch) using MRCP interface.

A starter kit comprising a specially priced full-function engine is 
available for $25 while a regular one channel (ie. port) license can be 
purchased for $99. Please visit Vestec webstore: http://www.vestec.com/

Regards,
-Kashif

-- 
Kashif Kahn
VP Business Development
Vestec Inc
Waterloo, ON Canada
phone: +1 519 885-7615



From mthakershi at gmail.com  Fri Jan  7 22:33:02 2011
From: mthakershi at gmail.com (Malay Thakershi)
Date: Fri, 7 Jan 2011 13:33:02 -0600
Subject: [Freeswitch-users] cepstral
In-Reply-To: <296FCE4397234877891847EC6F7D235D@e1705>
References: <296FCE4397234877891847EC6F7D235D@e1705>
Message-ID: <AANLkTinGkbeUGCGEBhBkEm-JCLDQePUhZo8q1dBHZo35@mail.gmail.com>

Few days back I had received reply on their eTicket system.

Here is the URL

http://support.cepstral.com/eticket/view.php

On Fri, Jan 7, 2011 at 12:38 PM, Madovsky <infos at madovsky.org> wrote:

>  Is anyone know if Cepstral is alive yet ?
> I bought voices and contacted support but no answer since 4 days now.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Fri Jan  7 22:39:08 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 7 Jan 2011 13:39:08 -0600
Subject: [Freeswitch-users] Vestec Speech Engine: ASR 2.1 Release
In-Reply-To: <4D27665F.4000406@vestec.com>
References: <4D27665F.4000406@vestec.com>
Message-ID: <7CB536AC-D130-464E-A516-E6935717337B@freeswitch.org>

FreeSWITCHers,

	I want to hold a contest to see who can build the neatest app using Vestec with FreeSWITCH.  The prize is a free pass to ClueCon 2011, If anyone wishes to enter please email me and Kashif.  

The only rules are it has to use FreeSWITCH with the Vestec engine.   

Thanks,
Brian

On Jan 7, 2011, at 1:15 PM, Kashif Kahn wrote:

> Dear All,
> 
> We have launched a major upgrade to our ASR engine that offers the best 
> deal around for enabling speech recognition with "command and control" 
> type IVR applications. The new architecture boasts a number of 
> advancements over version 1.1, including:
> 
> - Improved US English acoustic model
> - DTMF recognition
> - MRCP support v1 and v2
> - Highly scalable architecture and Redundancy
> - C++ API availability
> - SRGS-XML (.grxml) grammar support
> - Improved logging
> - Bug Fixes
> 
> The engine can be integrated with different contact center and soft-PBX 
> platforms (such as Freeswitch) using MRCP interface.
> 
> A starter kit comprising a specially priced full-function engine is 
> available for $25 while a regular one channel (ie. port) license can be 
> purchased for $99. Please visit Vestec webstore: http://www.vestec.com/
> 
> Regards,
> -Kashif
> 
> -- 
> Kashif Kahn
> VP Business Development
> Vestec Inc
> Waterloo, ON Canada
> phone: +1 519 885-7615
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From bernhard.suttner at winet.ch  Fri Jan  7 23:46:34 2011
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Fri, 07 Jan 2011 21:46:34 +0100
Subject: [Freeswitch-users] different mail-from/subject within voicemail
Message-ID: <20110107214634.144291f0@mail.winet.ch>

Hi,

seems like nobody knows a possible way to do that or its not possible. I wrote a small patch for mod_voicemail, which I will send over jira on Monday/Tuesday. Maybe someone will use it.

BR,
Bernhard

----- Original Message -----
From: Bernhard Suttner [mailto:bernhard.suttner at winet.ch]
To: FreeSWITCH-users at lists.freeswitch.org
Sent: Thu, 06 Jan 2011 14:07:15 +0100
Subject: [Freeswitch-users] different mail-from/subject within voicemail


> Hi,
> 
> is it somehow possible to specify the "email-from" address and subject used
> within the voicemail application within the directory (per user different)?
> 
> Best regards,
> Bernhard
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From steveayre at gmail.com  Sat Jan  8 00:00:45 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 7 Jan 2011 21:00:45 +0000
Subject: [Freeswitch-users] different mail-from/subject within voicemail
In-Reply-To: <20110107214634.144291f0@mail.winet.ch>
References: <20110107214634.144291f0@mail.winet.ch>
Message-ID: <AANLkTi=C=GygGBoUcssMp6+ZQfUZ8RO-gw9m-4DODKwG@mail.gmail.com>

I don't believe it's possible per-user, although you can create
multiple profiles.

I don't see any reason it couldn't be added as a user variable/param
though, similar to the current vm-mailto. vm-mailfrom?

-Steve



On 7 January 2011 20:46, Bernhard Suttner <bernhard.suttner at winet.ch> wrote:
> Hi,
>
> seems like nobody knows a possible way to do that or its not possible. I wrote a small patch for mod_voicemail, which I will send over jira on Monday/Tuesday. Maybe someone will use it.
>
> BR,
> Bernhard
>
> ----- Original Message -----
> From: Bernhard Suttner [mailto:bernhard.suttner at winet.ch]
> To: FreeSWITCH-users at lists.freeswitch.org
> Sent: Thu, 06 Jan 2011 14:07:15 +0100
> Subject: [Freeswitch-users] different mail-from/subject within voicemail
>
>
>> Hi,
>>
>> is it somehow possible to specify the "email-from" address and subject used
>> within the voicemail application within the directory (per user different)?
>>
>> Best regards,
>> Bernhard
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From steveayre at gmail.com  Sat Jan  8 00:08:19 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 7 Jan 2011 21:08:19 +0000
Subject: [Freeswitch-users] Incoming call from registered gateway with DID
	in To: not URI
Message-ID: <AANLkTimL-VTc0Hts3ps9h+5jxoc6ryGXVHfjPFQ2rHVG@mail.gmail.com>

Hi everyone,

I have a gateway registering to numbergroup.com, this is the configuration:

<gateway name="numbergroup">
    <param name="realm" value="sip.numbergroup-services.com"/>
    <param name="username" value="XXXXXX"/>
    <param name="password" value="XXXXXX"/>
    <param name="register" value="true"/>
    <param name="extension-in-contact" value="true"/>
    <param name="caller-id-in-from" value="true"/>
    <param name="ping" value="60"/>
    <param name="ping-max" value="10"/>
    <param name="ping-min" value="1"/>
    <param name="retry-seconds" value="5"/>
    <param name="expire-seconds" value="60"/>
</gateway>

Incoming calls from my DID to a SIP URI on the server work fine.
However I'm having problems sending it to the SIP Trunk (the
registration above).

The call arrives on the server fine, but the URI is the
username at sip.numbergroup-services.com. The DID is in the To: header

destination_number in the dialplan is the username not the DID as a
result. Does anyone know how to configure the gateway so that the
destination_number would contain the DID from the To header instead?

Here's the INVITE:

   INVITE sip:username at 81.27.101.246:5060;transport=udp;gw=numbergroup SIP/2.0
   Via: SIP/2.0/UDP 80.93.165.111;rport;branch=z9hG4bKD4m88SNemDSBg
   Max-Forwards: 65
   From: "+myphonenumber" <sip:+myphonenumber at 80.93.165.111>;tag=y9DBpB33yBHNN
   To: <sip:+mydidnumber at sip.numbergroup-services.com>
   Call-ID: 9ee8ce2d-9544-122e-c2a7-002655d1d302
   CSeq: 6881084 INVITE
   Contact: <sip:numbergroup at 80.93.165.111:5060>
   User-Agent: numbergroup.com
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Session-Expires: 900
   Min-SE: 90
   Privacy: none
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 593
   P-Charging-Vector:
icid-value=f4571a9c-1aa1-11e0-8310-5798bfd3ad89;icid-generated-at=80.93.165.110;orig-ioi=numbergroup.com
   P-Asserted-Identity: "+myphonenumber" <sip:+myphonenumber at 80.93.165.111>

   v=0
   o=numbergroup 1294416066 1294416067 IN IP4 80.93.165.111
   s=numbergroup
   c=IN IP4 80.93.165.111
   t=0 0
   m=audio 18230 RTP/AVP 8 9 98 3 18 99 100 101 102 103 104 0 105 101 13
   a=rtpmap:98 G7221/32000
   a=fmtp:98 bitrate=48000
   a=rtpmap:99 SPEEX/8000
   a=rtpmap:100 iLBC/8000
   a=fmtp:100 mode=20
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:102 G726-24/8000
   a=rtpmap:103 G726-32/8000
   a=rtpmap:104 G726-40/8000
   a=rtpmap:105 CELT/48000
   m=video 18540 RTP/AVP 106 107 34 31
   a=rtpmap:106 THEORA/90000
   a=rtpmap:107 H264/90000
   a=rtpmap:34 H263/90000
   a=rtpmap:31 H261/90000

-Steve


From david.ponzone at ipeva.fr  Sat Jan  8 00:49:00 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 7 Jan 2011 22:49:00 +0100
Subject: [Freeswitch-users] FreeSWITCH - Ploycom phone in hosted
	environment
In-Reply-To: <AANLkTimiG_GyWLO_JM+HYMM6bB0izxci1hZ=HosQwJsL@mail.gmail.com>
References: <AANLkTimiG_GyWLO_JM+HYMM6bB0izxci1hZ=HosQwJsL@mail.gmail.com>
Message-ID: <30FCC104-2CFF-4E0E-9227-4DD760700A16@ipeva.fr>

have you tried force-rport ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 07/01/2011 ? 17:05, Aloysius Lloyd a ?crit :

> Hi All,
> 
> Ploycom phone's not supporting rport. How to use Ploycom phone in the Hosted environment. Polycom phone in a NAT environment and FreeSWITCH hosted in the Datacenter with public IP.
> 
> I could not make it work. Any help is appreciated.
> 
> Thanks
> Lloyd
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From kris at livecall.com  Sat Jan  8 00:43:35 2011
From: kris at livecall.com (Kris)
Date: Fri, 7 Jan 2011 13:43:35 -0800
Subject: [Freeswitch-users] mod_conference with cluster ODBC
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705><AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com><AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com><AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com><AANLkTimutwOyeubcS=tax5EbDEniCD+Nc4iHMydQxpFb@mail.gmail.com><37761574FEE44D13BEBE4A7DE0089083@e1705><AANLkTi=VZ8JZ+PuyXoz_EYNVovY-AOuPyz9+yfKoUc3k@mail.gmail.com><337274A8CA2343259DF18D8135AF2DCE@e1705><AANLkTimVaxdbwVHCTqpYC5w3q+Vi+t7-oMekrkxu4Kfd@mail.gmail.com>
	<5B0F4D23C27A47A08D36C36B78AB981B@e1705>
Message-ID: <A7B3B85184604E1DB648F9AD0F99CD7C@stor1>

Just an idea..soon I will have to put people that are answered on multiple
servers into the same conference. I am thinking about having a table on the
central SQLServer like this: ConferenceName, ServerName. . I would lookup
the server a particular conference is on and then transfer the caller to
that server and extension that will put the caller into the appropriate
conference (dial something.. at SERVER)- I guess. I've seen the export word
that maybe the way to pass on variables to the other server such as the
ConferenceName, UserName

Then the server hosting the conference will have an extension that has the
forums profile and controls
      <action application="conference" data="${ConferenceName}@forums"/>
That way all the users are in the same conference and can be controlled
there instead of having only one link to a bunch of callers on another
server.

If you get it going, could you email the dial strings, extensions you
used.etc.I am curious..
Kris

----- Original Message ----- 
From: "Madovsky" <infos at madovsky.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, January 07, 2011 10:28 AM
Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


I got it thanks,
but do you think it would be more interesting to reduce
bandwidth and latency between nodes and centralize the conference on one 
node only
by transferring the incoming user to the right node ?
  ----- Original Message ----- 
  From: Jo?o Mesquita
  To: FreeSWITCH Users Help
  Sent: Thursday, January 06, 2011 4:33 PM
  Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


  Ok, let me see if I can get this into your head. (giggles)


  A conference means that the audio needs to mixed in together so that all 
participants can talk/hear each other, right? If you implement something in 
C on mod_conference, you are going to essentially do the same as what an ESL 
app does. You _need_ to call in from one server to the other so that you can 
mix the audio of all the participants. The real advantage would be the 
management API being only one for everything and the challenge is exactly 
that. How to mute certain users on a conference that is spanning over 10 
servers or deaf them, etc...


  A SIP "user" is easier because you don't have to bridge audio from another 
server necessarily. Got it?


  Regards,
  Jo?o Mesquita



  On Thu, Jan 6, 2011 at 6:25 PM, Madovsky <infos at madovsky.org> wrote:

    Rupa,

    I don't want bother anyone with this thread but why not
    to manage conference as SIP user ?
    if someone from server A call an other who is registered on server B, so
    FS do it automatically, why not with conference ? Or maybe create a 
param
    in mod_conference that let the choice of the admin to manage unique name 
in
    all cluster or not.
    like <param name="cluster_unique_name" value="true"/>
    I will try to understand the C code to hack something like this...


    ----- Original Message -----
    From: "Rupa Schomaker" <rupa at rupa.com>
    To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

    Sent: Thursday, January 06, 2011 3:01 PM
    Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


    Yes

    On Thu, Jan 6, 2011 at 1:40 PM, Madovsky <infos at madovsky.org> wrote:
    > in case of you have 8 servers you have to do it for each ?
    >
    > Thanks
    >
    > ----- Original Message -----
    > From: joy this
    > To: FreeSWITCH Users Help
    > Sent: Thursday, January 06, 2011 2:51 AM
    > Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
    > It works. Thank you everyone.
    >
    > 2011/1/5 Rupa Schomaker <rupa at rupa.com>
    >>
    >> Use the api: conference <confname> dial [{dial string
    >> options}]<endpoint_module_name>/<destination> [<callerid_number>
    >> [<callerid_name>]]
    >> To initiate the call from within conference A on server 1. Have a
    >> corresponding dialplan entry on server 2 to accept the call and add 
it
    >> into
    >> the conference A on server 2. You've now bridged the two conferences 
in
    >> the
    >> two servers.
    >
    > ________________________________
    >
    > _______________________________________________
    > FreeSWITCH-users mailing list
    > FreeSWITCH-users at lists.freeswitch.org
    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    > 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > http://www.freeswitch.org
    >
    > --
    > This message has been scanned for viruses and
    > dangerous content by MailScanner, and is
    > believed to be clean.
    > _______________________________________________
    > FreeSWITCH-users mailing list
    > FreeSWITCH-users at lists.freeswitch.org
    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    > 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > http://www.freeswitch.org
    >
    >



    --
    -Rupa

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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    _______________________________________________
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    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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------------------------------------------------------------------------------


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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  http://www.freeswitch.org



From lloyd.aloysius at gmail.com  Sat Jan  8 00:51:31 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 7 Jan 2011 16:51:31 -0500
Subject: [Freeswitch-users] FreeSWITCH - Ploycom phone in hosted
	environment
In-Reply-To: <30FCC104-2CFF-4E0E-9227-4DD760700A16@ipeva.fr>
References: <AANLkTimiG_GyWLO_JM+HYMM6bB0izxci1hZ=HosQwJsL@mail.gmail.com>
	<30FCC104-2CFF-4E0E-9227-4DD760700A16@ipeva.fr>
Message-ID: <AANLkTim_0wLR54D6KZBg5ioS+XpxVfJyMvUmuLWYgpc+@mail.gmail.com>

Yes but no luck.

Thanks
Lloyd


2011/1/7 David Ponzone <david.ponzone at ipeva.fr>

> have you tried force-rport ?
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 07/01/2011 ? 17:05, Aloysius Lloyd a ?crit :
>
> Hi All,
>
> Ploycom phone's not supporting rport. How to use Ploycom phone in the
> Hosted environment. Polycom phone in a NAT environment and FreeSWITCH hosted
> in the Datacenter with public IP.
>
> I could not make it work. Any help is appreciated.
>
> Thanks
> Lloyd
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Sat Jan  8 01:29:04 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 7 Jan 2011 19:29:04 -0300
Subject: [Freeswitch-users] mod_conference with cluster ODBC
In-Reply-To: <A7B3B85184604E1DB648F9AD0F99CD7C@stor1>
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705>
	<AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com>
	<AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com>
	<AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com>
	<AANLkTimutwOyeubcS=tax5EbDEniCD+Nc4iHMydQxpFb@mail.gmail.com>
	<37761574FEE44D13BEBE4A7DE0089083@e1705>
	<AANLkTi=VZ8JZ+PuyXoz_EYNVovY-AOuPyz9+yfKoUc3k@mail.gmail.com>
	<337274A8CA2343259DF18D8135AF2DCE@e1705>
	<AANLkTimVaxdbwVHCTqpYC5w3q+Vi+t7-oMekrkxu4Kfd@mail.gmail.com>
	<5B0F4D23C27A47A08D36C36B78AB981B@e1705>
	<A7B3B85184604E1DB648F9AD0F99CD7C@stor1>
Message-ID: <AANLkTi=4EM2gs+viY6gFY+qa=CGQ4dv7vMio6gj6Hi6L@mail.gmail.com>

Even tho both of your approaches are good for a "small" scale system, I
think we are missing the point here.

If you don't need to have several conferences bridged because you don't lack
machine power to hold the conference onto one server, you can use ESL to
make an INVITE and then a REPLACES. If I am not mistaken, you are able to
use the uuid_simplify command to make the replaces after the bridge is done.
Although, most people looking to have multiple conferences on multiple
servers are looking for scalability where you can have one single (or
multiple) conferences spread over several boxes that can even be
geographically spread out look like a single conference to the user and/or
systems involved.

This is the real challenge and that might be worth thinking about hacking C,
the rest is just dialplan and a bit of ESL to make the control. The first
one is way out of my league.

Jo?o Mesquita


On Fri, Jan 7, 2011 at 6:43 PM, Kris <kris at livecall.com> wrote:

> Just an idea..soon I will have to put people that are answered on multiple
> servers into the same conference. I am thinking about having a table on the
> central SQLServer like this: ConferenceName, ServerName. . I would lookup
> the server a particular conference is on and then transfer the caller to
> that server and extension that will put the caller into the appropriate
> conference (dial something.. at SERVER)- I guess. I've seen the export word
> that maybe the way to pass on variables to the other server such as the
> ConferenceName, UserName
>
> Then the server hosting the conference will have an extension that has the
> forums profile and controls
>      <action application="conference" data="${ConferenceName}@forums"/>
> That way all the users are in the same conference and can be controlled
> there instead of having only one link to a bunch of callers on another
> server.
>
> If you get it going, could you email the dial strings, extensions you
> used.etc.I am curious..
> Kris
>
> ----- Original Message -----
> From: "Madovsky" <infos at madovsky.org>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Friday, January 07, 2011 10:28 AM
> Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
>
>
> I got it thanks,
> but do you think it would be more interesting to reduce
> bandwidth and latency between nodes and centralize the conference on one
> node only
> by transferring the incoming user to the right node ?
>  ----- Original Message -----
>  From: Jo?o Mesquita
>  To: FreeSWITCH Users Help
>  Sent: Thursday, January 06, 2011 4:33 PM
>  Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
>
>
>  Ok, let me see if I can get this into your head. (giggles)
>
>
>  A conference means that the audio needs to mixed in together so that all
> participants can talk/hear each other, right? If you implement something in
> C on mod_conference, you are going to essentially do the same as what an
> ESL
> app does. You _need_ to call in from one server to the other so that you
> can
> mix the audio of all the participants. The real advantage would be the
> management API being only one for everything and the challenge is exactly
> that. How to mute certain users on a conference that is spanning over 10
> servers or deaf them, etc...
>
>
>  A SIP "user" is easier because you don't have to bridge audio from another
> server necessarily. Got it?
>
>
>  Regards,
>  Jo?o Mesquita
>
>
>
>  On Thu, Jan 6, 2011 at 6:25 PM, Madovsky <infos at madovsky.org> wrote:
>
>    Rupa,
>
>    I don't want bother anyone with this thread but why not
>    to manage conference as SIP user ?
>    if someone from server A call an other who is registered on server B, so
>    FS do it automatically, why not with conference ? Or maybe create a
> param
>    in mod_conference that let the choice of the admin to manage unique name
> in
>    all cluster or not.
>    like <param name="cluster_unique_name" value="true"/>
>    I will try to understand the C code to hack something like this...
>
>
>    ----- Original Message -----
>    From: "Rupa Schomaker" <rupa at rupa.com>
>    To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>
>    Sent: Thursday, January 06, 2011 3:01 PM
>    Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
>
>
>    Yes
>
>    On Thu, Jan 6, 2011 at 1:40 PM, Madovsky <infos at madovsky.org> wrote:
>    > in case of you have 8 servers you have to do it for each ?
>    >
>    > Thanks
>    >
>    > ----- Original Message -----
>    > From: joy this
>    > To: FreeSWITCH Users Help
>    > Sent: Thursday, January 06, 2011 2:51 AM
>    > Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
>    > It works. Thank you everyone.
>    >
>    > 2011/1/5 Rupa Schomaker <rupa at rupa.com>
>    >>
>    >> Use the api: conference <confname> dial [{dial string
>    >> options}]<endpoint_module_name>/<destination> [<callerid_number>
>    >> [<callerid_name>]]
>    >> To initiate the call from within conference A on server 1. Have a
>    >> corresponding dialplan entry on server 2 to accept the call and add
> it
>    >> into
>    >> the conference A on server 2. You've now bridged the two conferences
> in
>    >> the
>    >> two servers.
>    >
>    > ________________________________
>    >
>    > _______________________________________________
>    > FreeSWITCH-users mailing list
>    > FreeSWITCH-users at lists.freeswitch.org
>    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>    >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>    > http://www.freeswitch.org
>    >
>    > --
>    > This message has been scanned for viruses and
>    > dangerous content by MailScanner, and is
>    > believed to be clean.
>    > _______________________________________________
>    > FreeSWITCH-users mailing list
>    > FreeSWITCH-users at lists.freeswitch.org
>    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>    >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>    > http://www.freeswitch.org
>    >
>    >
>
>
>
>    --
>    -Rupa
>
>    _______________________________________________
>    FreeSWITCH-users mailing list
>    FreeSWITCH-users at lists.freeswitch.org
>    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>    UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
>    http://www.freeswitch.org
>
>
>    _______________________________________________
>    FreeSWITCH-users mailing list
>    FreeSWITCH-users at lists.freeswitch.org
>    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>    UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
>    http://www.freeswitch.org
>
>
>
>
>
>
> ------------------------------------------------------------------------------
>
>
>  _______________________________________________
>  FreeSWITCH-users mailing list
>  FreeSWITCH-users at lists.freeswitch.org
>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From chris at cloudtel.com  Sat Jan  8 01:45:21 2011
From: chris at cloudtel.com (Chris Burns)
Date: Fri, 7 Jan 2011 17:45:21 -0500
Subject: [Freeswitch-users] FreeSWITCH - Ploycom phone in hosted
	environment
In-Reply-To: <AANLkTim_0wLR54D6KZBg5ioS+XpxVfJyMvUmuLWYgpc+@mail.gmail.com>
References: <AANLkTimiG_GyWLO_JM+HYMM6bB0izxci1hZ=HosQwJsL@mail.gmail.com>
	<30FCC104-2CFF-4E0E-9227-4DD760700A16@ipeva.fr>
	<AANLkTim_0wLR54D6KZBg5ioS+XpxVfJyMvUmuLWYgpc+@mail.gmail.com>
Message-ID: <AANLkTinvcctL49YYh-yEzNXiw0RKm6_Vvhw4ff-rt2_=@mail.gmail.com>

Forcing rport in appropriate sofia profile will usually do it ... uncomment
NDLB-force-rport and restart the profile. In a certain rare situation I had
to fill out the phone's <nat/> tag dynamically when it provisioned. For
small environments you could manually assign each phone a signal port ....
yay polycom? x_X

2011/1/7 Aloysius Lloyd <lloyd.aloysius at gmail.com>

> Yes but no luck.
>
> Thanks
> Lloyd
>
>
> 2011/1/7 David Ponzone <david.ponzone at ipeva.fr>
>
> have you tried force-rport ?
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 07/01/2011 ? 17:05, Aloysius Lloyd a ?crit :
>>
>> Hi All,
>>
>> Ploycom phone's not supporting rport. How to use Ploycom phone in the
>> Hosted environment. Polycom phone in a NAT environment and FreeSWITCH hosted
>> in the Datacenter with public IP.
>>
>> I could not make it work. Any help is appreciated.
>>
>> Thanks
>> Lloyd
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Sat Jan  8 02:18:32 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 7 Jan 2011 17:18:32 -0600
Subject: [Freeswitch-users] FreeSWITCH - Ploycom phone in hosted
	environment
In-Reply-To: <AANLkTinvcctL49YYh-yEzNXiw0RKm6_Vvhw4ff-rt2_=@mail.gmail.com>
References: <AANLkTimiG_GyWLO_JM+HYMM6bB0izxci1hZ=HosQwJsL@mail.gmail.com>
	<30FCC104-2CFF-4E0E-9227-4DD760700A16@ipeva.fr>
	<AANLkTim_0wLR54D6KZBg5ioS+XpxVfJyMvUmuLWYgpc+@mail.gmail.com>
	<AANLkTinvcctL49YYh-yEzNXiw0RKm6_Vvhw4ff-rt2_=@mail.gmail.com>
Message-ID: <B93EB68C-CAD3-40C9-BB08-D988532C8913@freeswitch.org>

You guys are honestly forgetting something.

Step one.  use the VERY latests FreeSWITCH... set NDLB-force-rport to "safe" and the rest will be magical.

Step two... profit.

Step three... come to cluecon 2011.

/b


On Jan 7, 2011, at 4:45 PM, Chris Burns wrote:

> Forcing rport in appropriate sofia profile will usually do it ... uncomment NDLB-force-rport and restart the profile. In a certain rare situation I had to fill out the phone's <nat/> tag dynamically when it provisioned. For small environments you could manually assign each phone a signal port .... yay polycom? x_X
> 
> 2011/1/7 Aloysius Lloyd <lloyd.aloysius at gmail.com>
> Yes but no luck.
> 
> Thanks
> Lloyd
> 

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From steveayre at gmail.com  Sat Jan  8 02:24:17 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 7 Jan 2011 23:24:17 +0000
Subject: [Freeswitch-users] Incoming call from registered gateway with
	DID in To: not URI
In-Reply-To: <AANLkTimL-VTc0Hts3ps9h+5jxoc6ryGXVHfjPFQ2rHVG@mail.gmail.com>
References: <AANLkTimL-VTc0Hts3ps9h+5jxoc6ryGXVHfjPFQ2rHVG@mail.gmail.com>
Message-ID: <AANLkTi=kyY52e0NoULNVevNS0xkPmTyTDEp6qe_9Ls6T@mail.gmail.com>

I'm currently handling it with:

  <param name="context" value="numbergroup"/>

and:

  <context name="numbergroup">
    <extension name="numbergroup">
      <condition field="destination_number" expression="^USERNAME$">
        <action application="transfer" data="${sip_to_user} XML default"/>
      </condition>
    </extension>
  </context>

Is there a better way of handling it? Such as a param that tells Sofia
to use the To not the URI to populate destination_number?

-Steve


On 7 January 2011 21:08, Steven Ayre <steveayre at gmail.com> wrote:
> Hi everyone,
>
> I have a gateway registering to numbergroup.com, this is the configuration:
>
> <gateway name="numbergroup">
> ? ?<param name="realm" value="sip.numbergroup-services.com"/>
> ? ?<param name="username" value="XXXXXX"/>
> ? ?<param name="password" value="XXXXXX"/>
> ? ?<param name="register" value="true"/>
> ? ?<param name="extension-in-contact" value="true"/>
> ? ?<param name="caller-id-in-from" value="true"/>
> ? ?<param name="ping" value="60"/>
> ? ?<param name="ping-max" value="10"/>
> ? ?<param name="ping-min" value="1"/>
> ? ?<param name="retry-seconds" value="5"/>
> ? ?<param name="expire-seconds" value="60"/>
> </gateway>
>
> Incoming calls from my DID to a SIP URI on the server work fine.
> However I'm having problems sending it to the SIP Trunk (the
> registration above).
>
> The call arrives on the server fine, but the URI is the
> username at sip.numbergroup-services.com. The DID is in the To: header
>
> destination_number in the dialplan is the username not the DID as a
> result. Does anyone know how to configure the gateway so that the
> destination_number would contain the DID from the To header instead?
>
> Here's the INVITE:
>
> ? INVITE sip:username at 81.27.101.246:5060;transport=udp;gw=numbergroup SIP/2.0
> ? Via: SIP/2.0/UDP 80.93.165.111;rport;branch=z9hG4bKD4m88SNemDSBg
> ? Max-Forwards: 65
> ? From: "+myphonenumber" <sip:+myphonenumber at 80.93.165.111>;tag=y9DBpB33yBHNN
> ? To: <sip:+mydidnumber at sip.numbergroup-services.com>
> ? Call-ID: 9ee8ce2d-9544-122e-c2a7-002655d1d302
> ? CSeq: 6881084 INVITE
> ? Contact: <sip:numbergroup at 80.93.165.111:5060>
> ? User-Agent: numbergroup.com
> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> ? Supported: timer, precondition, path, replaces
> ? Allow-Events: talk, hold, refer
> ? Session-Expires: 900
> ? Min-SE: 90
> ? Privacy: none
> ? Content-Type: application/sdp
> ? Content-Disposition: session
> ? Content-Length: 593
> ? P-Charging-Vector:
> icid-value=f4571a9c-1aa1-11e0-8310-5798bfd3ad89;icid-generated-at=80.93.165.110;orig-ioi=numbergroup.com
> ? P-Asserted-Identity: "+myphonenumber" <sip:+myphonenumber at 80.93.165.111>
>
> ? v=0
> ? o=numbergroup 1294416066 1294416067 IN IP4 80.93.165.111
> ? s=numbergroup
> ? c=IN IP4 80.93.165.111
> ? t=0 0
> ? m=audio 18230 RTP/AVP 8 9 98 3 18 99 100 101 102 103 104 0 105 101 13
> ? a=rtpmap:98 G7221/32000
> ? a=fmtp:98 bitrate=48000
> ? a=rtpmap:99 SPEEX/8000
> ? a=rtpmap:100 iLBC/8000
> ? a=fmtp:100 mode=20
> ? a=rtpmap:101 telephone-event/8000
> ? a=fmtp:101 0-16
> ? a=rtpmap:102 G726-24/8000
> ? a=rtpmap:103 G726-32/8000
> ? a=rtpmap:104 G726-40/8000
> ? a=rtpmap:105 CELT/48000
> ? m=video 18540 RTP/AVP 106 107 34 31
> ? a=rtpmap:106 THEORA/90000
> ? a=rtpmap:107 H264/90000
> ? a=rtpmap:34 H263/90000
> ? a=rtpmap:31 H261/90000
>
> -Steve
>


From infos at madovsky.org  Sat Jan  8 02:31:45 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 7 Jan 2011 18:31:45 -0500
Subject: [Freeswitch-users] mod_conference with cluster ODBC
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705><AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com><AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com><AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com><AANLkTimutwOyeubcS=tax5EbDEniCD+Nc4iHMydQxpFb@mail.gmail.com><37761574FEE44D13BEBE4A7DE0089083@e1705><AANLkTi=VZ8JZ+PuyXoz_EYNVovY-AOuPyz9+yfKoUc3k@mail.gmail.com><337274A8CA2343259DF18D8135AF2DCE@e1705><AANLkTimVaxdbwVHCTqpYC5w3q+Vi+t7-oMekrkxu4Kfd@mail.gmail.com><5B0F4D23C27A47A08D36C36B78AB981B@e1705>
	<A7B3B85184604E1DB648F9AD0F99CD7C@stor1>
Message-ID: <5961547B8D8C429E8B6F7A938CC3E955@e1705>

it's what I thought first, but Joao is not hot for that apparently.
for now I had another idea as I don't want to spread the same
conference in several servers.

Franck

----- Original Message ----- 
From: "Kris" <kris at livecall.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, January 07, 2011 4:43 PM
Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


Just an idea..soon I will have to put people that are answered on multiple
servers into the same conference. I am thinking about having a table on the
central SQLServer like this: ConferenceName, ServerName. . I would lookup
the server a particular conference is on and then transfer the caller to
that server and extension that will put the caller into the appropriate
conference (dial something.. at SERVER)- I guess. I've seen the export word
that maybe the way to pass on variables to the other server such as the
ConferenceName, UserName

Then the server hosting the conference will have an extension that has the
forums profile and controls
      <action application="conference" data="${ConferenceName}@forums"/>
That way all the users are in the same conference and can be controlled
there instead of having only one link to a bunch of callers on another
server.

If you get it going, could you email the dial strings, extensions you
used.etc.I am curious..
Kris

----- Original Message ----- 
From: "Madovsky" <infos at madovsky.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, January 07, 2011 10:28 AM
Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


I got it thanks,
but do you think it would be more interesting to reduce
bandwidth and latency between nodes and centralize the conference on one
node only
by transferring the incoming user to the right node ?
  ----- Original Message ----- 
  From: Jo?o Mesquita
  To: FreeSWITCH Users Help
  Sent: Thursday, January 06, 2011 4:33 PM
  Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


  Ok, let me see if I can get this into your head. (giggles)


  A conference means that the audio needs to mixed in together so that all
participants can talk/hear each other, right? If you implement something in
C on mod_conference, you are going to essentially do the same as what an ESL
app does. You _need_ to call in from one server to the other so that you can
mix the audio of all the participants. The real advantage would be the
management API being only one for everything and the challenge is exactly
that. How to mute certain users on a conference that is spanning over 10
servers or deaf them, etc...


  A SIP "user" is easier because you don't have to bridge audio from another
server necessarily. Got it?


  Regards,
  Jo?o Mesquita



  On Thu, Jan 6, 2011 at 6:25 PM, Madovsky <infos at madovsky.org> wrote:

    Rupa,

    I don't want bother anyone with this thread but why not
    to manage conference as SIP user ?
    if someone from server A call an other who is registered on server B, so
    FS do it automatically, why not with conference ? Or maybe create a
param
    in mod_conference that let the choice of the admin to manage unique name
in
    all cluster or not.
    like <param name="cluster_unique_name" value="true"/>
    I will try to understand the C code to hack something like this...


    ----- Original Message -----
    From: "Rupa Schomaker" <rupa at rupa.com>
    To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

    Sent: Thursday, January 06, 2011 3:01 PM
    Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


    Yes

    On Thu, Jan 6, 2011 at 1:40 PM, Madovsky <infos at madovsky.org> wrote:
    > in case of you have 8 servers you have to do it for each ?
    >
    > Thanks
    >
    > ----- Original Message -----
    > From: joy this
    > To: FreeSWITCH Users Help
    > Sent: Thursday, January 06, 2011 2:51 AM
    > Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
    > It works. Thank you everyone.
    >
    > 2011/1/5 Rupa Schomaker <rupa at rupa.com>
    >>
    >> Use the api: conference <confname> dial [{dial string
    >> options}]<endpoint_module_name>/<destination> [<callerid_number>
    >> [<callerid_name>]]
    >> To initiate the call from within conference A on server 1. Have a
    >> corresponding dialplan entry on server 2 to accept the call and add
it
    >> into
    >> the conference A on server 2. You've now bridged the two conferences
in
    >> the
    >> two servers.
    >
    > ________________________________
    >
    > _______________________________________________
    > FreeSWITCH-users mailing list
    > FreeSWITCH-users at lists.freeswitch.org
    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > http://www.freeswitch.org
    >
    > --
    > This message has been scanned for viruses and
    > dangerous content by MailScanner, and is
    > believed to be clean.
    > _______________________________________________
    > FreeSWITCH-users mailing list
    > FreeSWITCH-users at lists.freeswitch.org
    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > http://www.freeswitch.org
    >
    >



    --
    -Rupa

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From infos at madovsky.org  Sat Jan  8 04:29:08 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 7 Jan 2011 20:29:08 -0500
Subject: [Freeswitch-users] FreeSWITCH - Ploycom phone in
	hostedenvironment
References: <AANLkTimiG_GyWLO_JM+HYMM6bB0izxci1hZ=HosQwJsL@mail.gmail.com><30FCC104-2CFF-4E0E-9227-4DD760700A16@ipeva.fr><AANLkTim_0wLR54D6KZBg5ioS+XpxVfJyMvUmuLWYgpc+@mail.gmail.com><AANLkTinvcctL49YYh-yEzNXiw0RKm6_Vvhw4ff-rt2_=@mail.gmail.com>
	<B93EB68C-CAD3-40C9-BB08-D988532C8913@freeswitch.org>
Message-ID: <5D1F1B9292A74023A294224EB7CEC04A@e1705>

> Step one.  use the VERY latests FreeSWITCH... set NDLB-force-rport to "safe" and the rest will be magical.
ok

> Step two... profit.
I hope for all who work hard on FS (but not for me now)

> Step three... come to cluecon 2011.
I would be a pleasure, but not interesting if there's no barbecue...

Step four.. I'm joking (not for Step two :(  )
  ----- Original Message ----- 
  From: Brian West 
  To: FreeSWITCH Users Help 
  Sent: Friday, January 07, 2011 6:18 PM
  Subject: Re: [Freeswitch-users] FreeSWITCH - Ploycom phone in hostedenvironment


  You guys are honestly forgetting something.


  Step one.  use the VERY latests FreeSWITCH... set NDLB-force-rport to "safe" and the rest will be magical.


  Step two... profit.


  Step three... come to cluecon 2011.


  /b




  On Jan 7, 2011, at 4:45 PM, Chris Burns wrote:


    Forcing rport in appropriate sofia profile will usually do it ... uncomment NDLB-force-rport and restart the profile. In a certain rare situation I had to fill out the phone's <nat/> tag dynamically when it provisioned. For small environments you could manually assign each phone a signal port .... yay polycom? x_X


    2011/1/7 Aloysius Lloyd <lloyd.aloysius at gmail.com>

      Yes but no luck.


      Thanks
      Lloyd






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From brian at freeswitch.org  Sat Jan  8 05:50:02 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 7 Jan 2011 20:50:02 -0600
Subject: [Freeswitch-users] Incoming call from registered gateway with
	DID in To: not URI
In-Reply-To: <AANLkTi=kyY52e0NoULNVevNS0xkPmTyTDEp6qe_9Ls6T@mail.gmail.com>
References: <AANLkTimL-VTc0Hts3ps9h+5jxoc6ryGXVHfjPFQ2rHVG@mail.gmail.com>
	<AANLkTi=kyY52e0NoULNVevNS0xkPmTyTDEp6qe_9Ls6T@mail.gmail.com>
Message-ID: <80CE5AE5-114F-47D4-8C6C-4F7F8A59D385@freeswitch.org>

update to the latest code this was fixed yesterday.  Broken the day before that.

/b

On Jan 7, 2011, at 5:24 PM, Steven Ayre wrote:

> I'm currently handling it with:
> 
>  <param name="context" value="numbergroup"/>
> 
> and:
> 
>  <context name="numbergroup">
>    <extension name="numbergroup">
>      <condition field="destination_number" expression="^USERNAME$">
>        <action application="transfer" data="${sip_to_user} XML default"/>
>      </condition>
>    </extension>
>  </context>
> 
> Is there a better way of handling it? Such as a param that tells Sofia
> to use the To not the URI to populate destination_number?
> 
> -Steve



From brian at freeswitch.org  Sat Jan  8 05:50:28 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 7 Jan 2011 20:50:28 -0600
Subject: [Freeswitch-users] Incoming call from registered gateway with
	DID in To: not URI
In-Reply-To: <AANLkTi=kyY52e0NoULNVevNS0xkPmTyTDEp6qe_9Ls6T@mail.gmail.com>
References: <AANLkTimL-VTc0Hts3ps9h+5jxoc6ryGXVHfjPFQ2rHVG@mail.gmail.com>
	<AANLkTi=kyY52e0NoULNVevNS0xkPmTyTDEp6qe_9Ls6T@mail.gmail.com>
Message-ID: <CAB3E5F4-95E2-4833-BA2C-9679BCC072C3@freeswitch.org>

Also if you set the extension to auto_to_user it'll do it also.

/b

On Jan 7, 2011, at 5:24 PM, Steven Ayre wrote:

> I'm currently handling it with:
> 
>  <param name="context" value="numbergroup"/>
> 
> and:
> 
>  <context name="numbergroup">
>    <extension name="numbergroup">
>      <condition field="destination_number" expression="^USERNAME$">
>        <action application="transfer" data="${sip_to_user} XML default"/>
>      </condition>
>    </extension>
>  </context>
> 
> Is there a better way of handling it? Such as a param that tells Sofia
> to use the To not the URI to populate destination_number?
> 
> -Steve



From u2nsam at gmail.com  Sat Jan  8 06:42:24 2011
From: u2nsam at gmail.com (Sam)
Date: Sat, 8 Jan 2011 09:12:24 +0530
Subject: [Freeswitch-users] call getting hangup after git update
In-Reply-To: <AANLkTikqr10zdWni96-RUWyRrpzus+J1God_yh7JiH9A@mail.gmail.com>
References: <AANLkTi=z5KOP0sHwdYg+=tiF38tDS_X6QY1nk7ehuuLQ@mail.gmail.com>
	<2B571B49-1886-445D-8473-C34CC9039301@freeswitch.org>
	<AANLkTikqr10zdWni96-RUWyRrpzus+J1God_yh7JiH9A@mail.gmail.com>
Message-ID: <AANLkTimptf=rLg4SFV4bzseUf9zQwqo_Ee-yh7U4_UKB@mail.gmail.com>

Hello,

Set NPI/TON to unknown and it will work !

Regards
Sam




On Fri, Jan 7, 2011 at 10:55 PM, ovvenkat <ovvenkatesan at gmail.com> wrote:

> Hi Brian,
>
> On Fri, Jan 7, 2011 at 8:44 PM, Brian West <brian at freeswitch.org> wrote:
>
>> You're getting no route back from your freetdm circuit.
>>
>>
>
> When I am calling to mobile number, it works fine.
> Its giving the problem only with fixed line numbers.
> Can you please tell , why its so?
>
>
>
>
>> /b
>>
>> On Jan 7, 2011, at 8:39 AM, ovvenkat wrote:
>>
>>
>> Hi,
>>
>> Today, I have updated freeSwitch to latest git.
>> After that, When I am trying to do outbound call,
>> Call is getting hangup. When I check the fs_cli
>> logs, its showing that,
>>
>> *mod_dptools.c:2610 Originate Failed.  Cause: NO_ROUTE_DESTINATION*
>>
>> Here is the logs for the same
>> http://pastebin.freeswitch.org/14951
>>
>> I have added STD code , before the number,
>> Still no luck. I am getting error like
>>
>> *Originate Failed.  Cause: NORMAL_UNSPECIFIED*
>>
>>
>> Here is the log after adding STD code
>>
>> http://pastebin.freeswitch.org/14952
>>
>>
>> Regards,
>> Venkat.
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
>
> If you have come to help me, you are wasting your time.
> If you have come to because your liberation is bound up in mine, we can
> work together.
>
>
> Regards
> Venkatesan OV.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Sat Jan  8 06:46:03 2011
From: u2nsam at gmail.com (Sam)
Date: Sat, 8 Jan 2011 09:16:03 +0530
Subject: [Freeswitch-users] Unable to get Freeswitch to register with
 Grandstream HT503
In-Reply-To: <004801cbae90$cbf44ca0$63dce5e0$@accra.ca>
References: <004801cbae90$cbf44ca0$63dce5e0$@accra.ca>
Message-ID: <AANLkTik_+AemFh8QnbcYPAuL3UJmV6q_kC-eELLWYOy4@mail.gmail.com>

Is it happening only with ht 503 or all the devices , do you get the trace
logs for this ?

Regards
Sam

On Fri, Jan 7, 2011 at 11:02 PM, Charles Bujold <cjbujold at accra.ca> wrote:

> Unable to get the HT-503 to register with Freeswitch.  Does anybody know
> what configuration setting in the HT503 that are needed.
>
>
>
> Thanks
>
> cjb
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bwibowo at gmail.com  Sat Jan  8 07:41:12 2011
From: bwibowo at gmail.com (Budi wibowo)
Date: Sat, 8 Jan 2011 04:41:12 +0000
Subject: [Freeswitch-users] Sip client with tunnel
Message-ID: <949141255-1294461671-cardhu_decombobulator_blackberry.rim.net-1397796539-@b25.c2.bise3.blackberry>

Hi 
I'm looking for sip client that support tunneling, because my isp seems block incoming sip connection. For sip outgoing no problem.

Thx

Budi


From steveayre at gmail.com  Sat Jan  8 12:03:44 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 8 Jan 2011 09:03:44 +0000
Subject: [Freeswitch-users] Incoming call from registered gateway with
	DID in To: not URI
In-Reply-To: <80CE5AE5-114F-47D4-8C6C-4F7F8A59D385@freeswitch.org>
References: <AANLkTimL-VTc0Hts3ps9h+5jxoc6ryGXVHfjPFQ2rHVG@mail.gmail.com>
	<AANLkTi=kyY52e0NoULNVevNS0xkPmTyTDEp6qe_9Ls6T@mail.gmail.com>
	<80CE5AE5-114F-47D4-8C6C-4F7F8A59D385@freeswitch.org>
Message-ID: <33D2FA0C-1EB2-4EC6-BA8B-F336E1F1074C@gmail.com>

Thanks Brian. :)

Steve on iPhone

On 8 Jan 2011, at 02:50, Brian West <brian at freeswitch.org> wrote:

> update to the latest code this was fixed yesterday.  Broken the day before that.
> 
> /b
> 
> On Jan 7, 2011, at 5:24 PM, Steven Ayre wrote:
> 
>> I'm currently handling it with:
>> 
>> <param name="context" value="numbergroup"/>
>> 
>> and:
>> 
>> <context name="numbergroup">
>>   <extension name="numbergroup">
>>     <condition field="destination_number" expression="^USERNAME$">
>>       <action application="transfer" data="${sip_to_user} XML default"/>
>>     </condition>
>>   </extension>
>> </context>
>> 
>> Is there a better way of handling it? Such as a param that tells Sofia
>> to use the To not the URI to populate destination_number?
>> 
>> -Steve
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From hwnorman at hotmail.com  Sat Jan  8 09:41:58 2011
From: hwnorman at hotmail.com (Norman Lam)
Date: Sat, 8 Jan 2011 14:41:58 +0800
Subject: [Freeswitch-users] Iksemel msvs compiling
In-Reply-To: <1294375438447-5898181.post@n2.nabble.com>
References: <SNT109-DS13313A1FA906527438E8CC1090@phx.gbl>	<SNT109-DS1155D401A0B9742CF63E13C10B0@phx.gbl>
	<1294375438447-5898181.post@n2.nabble.com>
Message-ID: <SNT109-DS11CB0ECE5BD0F9C0384E03C10C0@phx.gbl>

Hi Jeff

I clone the latest git and build it with your added instruction, I am
getting more error

Is it because of the new git and it didn't fix the issue



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff
Lenk
Sent: Friday, January 07, 2011 12:44 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Iksemel msvs compiling


try adding ..\..\pthreads-w32-2-7-0-release; 
to the include path and see what happens
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http://freeswitch-users.2379917.n2.nabble.com/Iksemel-msvs-compiling-tp58912
63p5898181.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

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From rafonline at hotmail.com  Sat Jan  8 16:05:14 2011
From: rafonline at hotmail.com (Rafqat .)
Date: Sat, 8 Jan 2011 13:05:14 +0000
Subject: [Freeswitch-users] server spec
Message-ID: <COL122-W4844B45D5D0452C2956202B50C0@phx.gbl>



Hi,

I am interested in purchasing a server to host my calling card application (including freeswitch itself).  The application is based on a simple lua script and which makes use of mod_lcr and mod_nibble.  I would like to support at least 500 concurrent calls and was wondering what kind of server spec should I be looking to purchase.

Any advice will be much appreciated.

Cheers

Raf
 		 	   		  
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From boris at tagnet.ru  Sat Jan  8 16:17:35 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 08 Jan 2011 18:17:35 +0500
Subject: [Freeswitch-users] user variables
Message-ID: <4D2863EF.4080604@tagnet.ru>

Hello!

     I use FreeSWITCH Version 1.0.head (git-cd13030 2011-01-06 23-17-08 
-0600) and have some troubles with user variables. So, when user is 
registered the user defined variables are present in channel variables, 
and when user isn't registered (authenticated by cidr for example) I 
can't see it variables. Is this my mistake or freeswitch bug?

-- 
Regards,
Boris




From boris at tagnet.ru  Sat Jan  8 16:29:36 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 08 Jan 2011 18:29:36 +0500
Subject: [Freeswitch-users] user variables
In-Reply-To: <4D2863EF.4080604@tagnet.ru>
References: <4D2863EF.4080604@tagnet.ru>
Message-ID: <4D2866C0.8070300@tagnet.ru>

Hello!

Sorry, this was my mistake. I forgot to reload acl.
> Hello!
>
>       I use FreeSWITCH Version 1.0.head (git-cd13030 2011-01-06 23-17-08
> -0600) and have some troubles with user variables. So, when user is
> registered the user defined variables are present in channel variables,
> and when user isn't registered (authenticated by cidr for example) I
> can't see it variables. Is this my mistake or freeswitch bug?
>


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From kris at livecall.com  Sat Jan  8 16:37:04 2011
From: kris at livecall.com (Kris)
Date: Sat, 8 Jan 2011 05:37:04 -0800
Subject: [Freeswitch-users] mod_conference with cluster ODBC
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705><AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com><AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com><AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com><AANLkTimutwOyeubcS=tax5EbDEniCD+Nc4iHMydQxpFb@mail.gmail.com><37761574FEE44D13BEBE4A7DE0089083@e1705><AANLkTi=VZ8JZ+PuyXoz_EYNVovY-AOuPyz9+yfKoUc3k@mail.gmail.com><337274A8CA2343259DF18D8135AF2DCE@e1705><AANLkTimVaxdbwVHCTqpYC5w3q+Vi+t7-oMekrkxu4Kfd@mail.gmail.com><5B0F4D23C27A47A08D36C36B78AB981B@e1705><A7B3B85184604E1DB648F9AD0F99CD7C@stor1>
	<5961547B8D8C429E8B6F7A938CC3E955@e1705>
Message-ID: <8AC5322D48564DFAA3D3D68B335C5963@stor1>

On second thought, Joao may be right. If the conference is spread accross 10 
servers, and one crashes, it keeps the other users chatting. A->B->C->D->A. 
C crashes, the conference on B has to become aware, and immediately connect 
to D. I don't know if such failover exists in mod_conference or even if 
having conferences circularly linked would cause a feedback. If they are 
circularly linked and only one fails, it should still work even without 
failover.

----- Original Message ----- 
From: "Madovsky" <infos at madovsky.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, January 07, 2011 3:31 PM
Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


it's what I thought first, but Joao is not hot for that apparently.
for now I had another idea as I don't want to spread the same
conference in several servers.

Franck

----- Original Message ----- 
From: "Kris" <kris at livecall.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, January 07, 2011 4:43 PM
Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


Just an idea..soon I will have to put people that are answered on multiple
servers into the same conference. I am thinking about having a table on the
central SQLServer like this: ConferenceName, ServerName. . I would lookup
the server a particular conference is on and then transfer the caller to
that server and extension that will put the caller into the appropriate
conference (dial something.. at SERVER)- I guess. I've seen the export word
that maybe the way to pass on variables to the other server such as the
ConferenceName, UserName

Then the server hosting the conference will have an extension that has the
forums profile and controls
      <action application="conference" data="${ConferenceName}@forums"/>
That way all the users are in the same conference and can be controlled
there instead of having only one link to a bunch of callers on another
server.

If you get it going, could you email the dial strings, extensions you
used.etc.I am curious..
Kris

----- Original Message ----- 
From: "Madovsky" <infos at madovsky.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, January 07, 2011 10:28 AM
Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


I got it thanks,
but do you think it would be more interesting to reduce
bandwidth and latency between nodes and centralize the conference on one
node only
by transferring the incoming user to the right node ?
  ----- Original Message ----- 
  From: Jo?o Mesquita
  To: FreeSWITCH Users Help
  Sent: Thursday, January 06, 2011 4:33 PM
  Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


  Ok, let me see if I can get this into your head. (giggles)


  A conference means that the audio needs to mixed in together so that all
participants can talk/hear each other, right? If you implement something in
C on mod_conference, you are going to essentially do the same as what an ESL
app does. You _need_ to call in from one server to the other so that you can
mix the audio of all the participants. The real advantage would be the
management API being only one for everything and the challenge is exactly
that. How to mute certain users on a conference that is spanning over 10
servers or deaf them, etc...


  A SIP "user" is easier because you don't have to bridge audio from another
server necessarily. Got it?


  Regards,
  Jo?o Mesquita



  On Thu, Jan 6, 2011 at 6:25 PM, Madovsky <infos at madovsky.org> wrote:

    Rupa,

    I don't want bother anyone with this thread but why not
    to manage conference as SIP user ?
    if someone from server A call an other who is registered on server B, so
    FS do it automatically, why not with conference ? Or maybe create a
param
    in mod_conference that let the choice of the admin to manage unique name
in
    all cluster or not.
    like <param name="cluster_unique_name" value="true"/>
    I will try to understand the C code to hack something like this...


    ----- Original Message -----
    From: "Rupa Schomaker" <rupa at rupa.com>
    To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

    Sent: Thursday, January 06, 2011 3:01 PM
    Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


    Yes

    On Thu, Jan 6, 2011 at 1:40 PM, Madovsky <infos at madovsky.org> wrote:
    > in case of you have 8 servers you have to do it for each ?
    >
    > Thanks
    >
    > ----- Original Message -----
    > From: joy this
    > To: FreeSWITCH Users Help
    > Sent: Thursday, January 06, 2011 2:51 AM
    > Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
    > It works. Thank you everyone.
    >
    > 2011/1/5 Rupa Schomaker <rupa at rupa.com>
    >>
    >> Use the api: conference <confname> dial [{dial string
    >> options}]<endpoint_module_name>/<destination> [<callerid_number>
    >> [<callerid_name>]]
    >> To initiate the call from within conference A on server 1. Have a
    >> corresponding dialplan entry on server 2 to accept the call and add
it
    >> into
    >> the conference A on server 2. You've now bridged the two conferences
in
    >> the
    >> two servers.
    >
    > ________________________________
    >
    > _______________________________________________
    > FreeSWITCH-users mailing list
    > FreeSWITCH-users at lists.freeswitch.org
    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > http://www.freeswitch.org
    >
    > --
    > This message has been scanned for viruses and
    > dangerous content by MailScanner, and is
    > believed to be clean.
    > _______________________________________________
    > FreeSWITCH-users mailing list
    > FreeSWITCH-users at lists.freeswitch.org
    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > http://www.freeswitch.org
    >
    >



    --
    -Rupa

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





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From steveayre at gmail.com  Sat Jan  8 17:04:31 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 8 Jan 2011 14:04:31 +0000
Subject: [Freeswitch-users] mod_conference with cluster ODBC
In-Reply-To: <8AC5322D48564DFAA3D3D68B335C5963@stor1>
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705>
	<AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com>
	<AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com>
	<AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com>
	<AANLkTimutwOyeubcS=tax5EbDEniCD+Nc4iHMydQxpFb@mail.gmail.com>
	<37761574FEE44D13BEBE4A7DE0089083@e1705>
	<AANLkTi=VZ8JZ+PuyXoz_EYNVovY-AOuPyz9+yfKoUc3k@mail.gmail.com>
	<337274A8CA2343259DF18D8135AF2DCE@e1705>
	<AANLkTimVaxdbwVHCTqpYC5w3q+Vi+t7-oMekrkxu4Kfd@mail.gmail.com>
	<5B0F4D23C27A47A08D36C36B78AB981B@e1705>
	<A7B3B85184604E1DB648F9AD0F99CD7C@stor1>
	<5961547B8D8C429E8B6F7A938CC3E955@e1705>
	<8AC5322D48564DFAA3D3D68B335C5963@stor1>
Message-ID: <94A66DE2-1401-43E9-9AF8-BC27ACE3DB57@gmail.com>

> I don't know if such failover exists in mod_conference

I think Sofia recover would handle that

>  if 
> having conferences circularly linked would cause a feedback

I think it would

Steve on iPhone

On 8 Jan 2011, at 13:37, "Kris" <kris at livecall.com> wrote:

> On second thought, Joao may be right. If the conference is spread accross 10 
> servers, and one crashes, it keeps the other users chatting. A->B->C->D->A. 
> C crashes, the conference on B has to become aware, and immediately connect 
> to D. I don't know if such failover exists in mod_conference or even if 
> having conferences circularly linked would cause a feedback. If they are 
> circularly linked and only one fails, it should still work even without 
> failover.
> 
> ----- Original Message ----- 
> From: "Madovsky" <infos at madovsky.org>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Friday, January 07, 2011 3:31 PM
> Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
> 
> 
> it's what I thought first, but Joao is not hot for that apparently.
> for now I had another idea as I don't want to spread the same
> conference in several servers.
> 
> Franck
> 
> ----- Original Message ----- 
> From: "Kris" <kris at livecall.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Friday, January 07, 2011 4:43 PM
> Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
> 
> 
> Just an idea..soon I will have to put people that are answered on multiple
> servers into the same conference. I am thinking about having a table on the
> central SQLServer like this: ConferenceName, ServerName. . I would lookup
> the server a particular conference is on and then transfer the caller to
> that server and extension that will put the caller into the appropriate
> conference (dial something.. at SERVER)- I guess. I've seen the export word
> that maybe the way to pass on variables to the other server such as the
> ConferenceName, UserName
> 
> Then the server hosting the conference will have an extension that has the
> forums profile and controls
>      <action application="conference" data="${ConferenceName}@forums"/>
> That way all the users are in the same conference and can be controlled
> there instead of having only one link to a bunch of callers on another
> server.
> 
> If you get it going, could you email the dial strings, extensions you
> used.etc.I am curious..
> Kris
> 
> ----- Original Message ----- 
> From: "Madovsky" <infos at madovsky.org>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Friday, January 07, 2011 10:28 AM
> Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
> 
> 
> I got it thanks,
> but do you think it would be more interesting to reduce
> bandwidth and latency between nodes and centralize the conference on one
> node only
> by transferring the incoming user to the right node ?
>  ----- Original Message ----- 
>  From: Jo?o Mesquita
>  To: FreeSWITCH Users Help
>  Sent: Thursday, January 06, 2011 4:33 PM
>  Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
> 
> 
>  Ok, let me see if I can get this into your head. (giggles)
> 
> 
>  A conference means that the audio needs to mixed in together so that all
> participants can talk/hear each other, right? If you implement something in
> C on mod_conference, you are going to essentially do the same as what an ESL
> app does. You _need_ to call in from one server to the other so that you can
> mix the audio of all the participants. The real advantage would be the
> management API being only one for everything and the challenge is exactly
> that. How to mute certain users on a conference that is spanning over 10
> servers or deaf them, etc...
> 
> 
>  A SIP "user" is easier because you don't have to bridge audio from another
> server necessarily. Got it?
> 
> 
>  Regards,
>  Jo?o Mesquita
> 
> 
> 
>  On Thu, Jan 6, 2011 at 6:25 PM, Madovsky <infos at madovsky.org> wrote:
> 
>    Rupa,
> 
>    I don't want bother anyone with this thread but why not
>    to manage conference as SIP user ?
>    if someone from server A call an other who is registered on server B, so
>    FS do it automatically, why not with conference ? Or maybe create a
> param
>    in mod_conference that let the choice of the admin to manage unique name
> in
>    all cluster or not.
>    like <param name="cluster_unique_name" value="true"/>
>    I will try to understand the C code to hack something like this...
> 
> 
>    ----- Original Message -----
>    From: "Rupa Schomaker" <rupa at rupa.com>
>    To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> 
>    Sent: Thursday, January 06, 2011 3:01 PM
>    Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
> 
> 
>    Yes
> 
>    On Thu, Jan 6, 2011 at 1:40 PM, Madovsky <infos at madovsky.org> wrote:
>> in case of you have 8 servers you have to do it for each ?
>> 
>> Thanks
>> 
>> ----- Original Message -----
>> From: joy this
>> To: FreeSWITCH Users Help
>> Sent: Thursday, January 06, 2011 2:51 AM
>> Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
>> It works. Thank you everyone.
>> 
>> 2011/1/5 Rupa Schomaker <rupa at rupa.com>
>>> 
>>> Use the api: conference <confname> dial [{dial string
>>> options}]<endpoint_module_name>/<destination> [<callerid_number>
>>> [<callerid_name>]]
>>> To initiate the call from within conference A on server 1. Have a
>>> corresponding dialplan entry on server 2 to accept the call and add
> it
>>> into
>>> the conference A on server 2. You've now bridged the two conferences
> in
>>> the
>>> two servers.
>> 
>> ________________________________
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> --
>> This message has been scanned for viruses and
>> dangerous content by MailScanner, and is
>> believed to be clean.
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> 
> 
>    --
>    -Rupa
> 
>    _______________________________________________
>    FreeSWITCH-users mailing list
>    FreeSWITCH-users at lists.freeswitch.org
>    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>    http://www.freeswitch.org
> 
> 
>    _______________________________________________
>    FreeSWITCH-users mailing list
>    FreeSWITCH-users at lists.freeswitch.org
>    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>    http://www.freeswitch.org
> 
> 
> 
> 
> 
> ------------------------------------------------------------------------------
> 
> 
>  _______________________________________________
>  FreeSWITCH-users mailing list
>  FreeSWITCH-users at lists.freeswitch.org
>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From cmrienzo at gmail.com  Sat Jan  8 18:28:39 2011
From: cmrienzo at gmail.com (Chris Rienzo)
Date: Sat, 8 Jan 2011 10:28:39 -0500
Subject: [Freeswitch-users] mod_conference with cluster ODBC
In-Reply-To: <8AC5322D48564DFAA3D3D68B335C5963@stor1>
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705>
	<AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com>
	<AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com>
	<AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com>
	<AANLkTimutwOyeubcS=tax5EbDEniCD+Nc4iHMydQxpFb@mail.gmail.com>
	<37761574FEE44D13BEBE4A7DE0089083@e1705>
	<AANLkTi=VZ8JZ+PuyXoz_EYNVovY-AOuPyz9+yfKoUc3k@mail.gmail.com>
	<337274A8CA2343259DF18D8135AF2DCE@e1705>
	<AANLkTimVaxdbwVHCTqpYC5w3q+Vi+t7-oMekrkxu4Kfd@mail.gmail.com>
	<5B0F4D23C27A47A08D36C36B78AB981B@e1705>
	<A7B3B85184604E1DB648F9AD0F99CD7C@stor1>
	<5961547B8D8C429E8B6F7A938CC3E955@e1705>
	<8AC5322D48564DFAA3D3D68B335C5963@stor1>
Message-ID: <E0E6652F-800E-4562-8967-4DE0461363C9@gmail.com>

I think circularly linked conferences cause echo.  Daisy chained conferences have high latency, though are simple to build.


On Jan 8, 2011, at 8:37, "Kris" <kris at livecall.com> wrote:

> On second thought, Joao may be right. If the conference is spread accross 10 
> servers, and one crashes, it keeps the other users chatting. A->B->C->D->A. 
> C crashes, the conference on B has to become aware, and immediately connect 
> to D. I don't know if such failover exists in mod_conference or even if 
> having conferences circularly linked would cause a feedback. If they are 
> circularly linked and only one fails, it should still work even without 
> failover.
> 
> ----- Original Message ----- 
> From: "Madovsky" <infos at madovsky.org>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Friday, January 07, 2011 3:31 PM
> Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
> 
> 
> it's what I thought first, but Joao is not hot for that apparently.
> for now I had another idea as I don't want to spread the same
> conference in several servers.
> 
> Franck
> 
> ----- Original Message ----- 
> From: "Kris" <kris at livecall.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Friday, January 07, 2011 4:43 PM
> Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
> 
> 
> Just an idea..soon I will have to put people that are answered on multiple
> servers into the same conference. I am thinking about having a table on the
> central SQLServer like this: ConferenceName, ServerName. . I would lookup
> the server a particular conference is on and then transfer the caller to
> that server and extension that will put the caller into the appropriate
> conference (dial something.. at SERVER)- I guess. I've seen the export word
> that maybe the way to pass on variables to the other server such as the
> ConferenceName, UserName
> 
> Then the server hosting the conference will have an extension that has the
> forums profile and controls
>      <action application="conference" data="${ConferenceName}@forums"/>
> That way all the users are in the same conference and can be controlled
> there instead of having only one link to a bunch of callers on another
> server.
> 
> If you get it going, could you email the dial strings, extensions you
> used.etc.I am curious..
> Kris
> 
> ----- Original Message ----- 
> From: "Madovsky" <infos at madovsky.org>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Friday, January 07, 2011 10:28 AM
> Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
> 
> 
> I got it thanks,
> but do you think it would be more interesting to reduce
> bandwidth and latency between nodes and centralize the conference on one
> node only
> by transferring the incoming user to the right node ?
>  ----- Original Message ----- 
>  From: Jo?o Mesquita
>  To: FreeSWITCH Users Help
>  Sent: Thursday, January 06, 2011 4:33 PM
>  Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
> 
> 
>  Ok, let me see if I can get this into your head. (giggles)
> 
> 
>  A conference means that the audio needs to mixed in together so that all
> participants can talk/hear each other, right? If you implement something in
> C on mod_conference, you are going to essentially do the same as what an ESL
> app does. You _need_ to call in from one server to the other so that you can
> mix the audio of all the participants. The real advantage would be the
> management API being only one for everything and the challenge is exactly
> that. How to mute certain users on a conference that is spanning over 10
> servers or deaf them, etc...
> 
> 
>  A SIP "user" is easier because you don't have to bridge audio from another
> server necessarily. Got it?
> 
> 
>  Regards,
>  Jo?o Mesquita
> 
> 
> 
>  On Thu, Jan 6, 2011 at 6:25 PM, Madovsky <infos at madovsky.org> wrote:
> 
>    Rupa,
> 
>    I don't want bother anyone with this thread but why not
>    to manage conference as SIP user ?
>    if someone from server A call an other who is registered on server B, so
>    FS do it automatically, why not with conference ? Or maybe create a
> param
>    in mod_conference that let the choice of the admin to manage unique name
> in
>    all cluster or not.
>    like <param name="cluster_unique_name" value="true"/>
>    I will try to understand the C code to hack something like this...
> 
> 
>    ----- Original Message -----
>    From: "Rupa Schomaker" <rupa at rupa.com>
>    To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> 
>    Sent: Thursday, January 06, 2011 3:01 PM
>    Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
> 
> 
>    Yes
> 
>    On Thu, Jan 6, 2011 at 1:40 PM, Madovsky <infos at madovsky.org> wrote:
>> in case of you have 8 servers you have to do it for each ?
>> 
>> Thanks
>> 
>> ----- Original Message -----
>> From: joy this
>> To: FreeSWITCH Users Help
>> Sent: Thursday, January 06, 2011 2:51 AM
>> Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
>> It works. Thank you everyone.
>> 
>> 2011/1/5 Rupa Schomaker <rupa at rupa.com>
>>> 
>>> Use the api: conference <confname> dial [{dial string
>>> options}]<endpoint_module_name>/<destination> [<callerid_number>
>>> [<callerid_name>]]
>>> To initiate the call from within conference A on server 1. Have a
>>> corresponding dialplan entry on server 2 to accept the call and add
> it
>>> into
>>> the conference A on server 2. You've now bridged the two conferences
> in
>>> the
>>> two servers.
>> 
>> ________________________________
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> --
>> This message has been scanned for viruses and
>> dangerous content by MailScanner, and is
>> believed to be clean.
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> 
> 
>    --
>    -Rupa
> 
>    _______________________________________________
>    FreeSWITCH-users mailing list
>    FreeSWITCH-users at lists.freeswitch.org
>    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>    http://www.freeswitch.org
> 
> 
>    _______________________________________________
>    FreeSWITCH-users mailing list
>    FreeSWITCH-users at lists.freeswitch.org
>    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>    http://www.freeswitch.org
> 
> 
> 
> 
> 
> ------------------------------------------------------------------------------
> 
> 
>  _______________________________________________
>  FreeSWITCH-users mailing list
>  FreeSWITCH-users at lists.freeswitch.org
>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From steveayre at gmail.com  Sat Jan  8 18:56:38 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 8 Jan 2011 15:56:38 +0000
Subject: [Freeswitch-users] Incoming call from registered gateway with
 DID in To: not URI
In-Reply-To: <CAB3E5F4-95E2-4833-BA2C-9679BCC072C3@freeswitch.org>
References: <AANLkTimL-VTc0Hts3ps9h+5jxoc6ryGXVHfjPFQ2rHVG@mail.gmail.com>
	<AANLkTi=kyY52e0NoULNVevNS0xkPmTyTDEp6qe_9Ls6T@mail.gmail.com>
	<CAB3E5F4-95E2-4833-BA2C-9679BCC072C3@freeswitch.org>
Message-ID: <AANLkTikHwRGhQXHb6vh8iCRcTbCvJdwWA3eWPxZm5qdu@mail.gmail.com>

<param name="extension" value="auto_to_user"/> was what did the trick
though. Thanks! :)

There's a minor issue from it where the trunk shows unregistered on
their website (they expect the Contact header to be username at ip") but
incoming calls are still sent through fine so it doesn't actually
matter at all. They can still see FS registered on their switch.

I've added the configuration to the SIP Provider Examples section of the Wiki.

One of their guys said by email they tested on Git within the last few
weeks and found calls were hanging up after 2m40s due to something
about how we handle timers... I'm going to test that next. Were there
any versions in Git recently that might have had problems?

-Steve




On 8 January 2011 02:50, Brian West <brian at freeswitch.org> wrote:
> Also if you set the extension to auto_to_user it'll do it also.
>
> /b
>
> On Jan 7, 2011, at 5:24 PM, Steven Ayre wrote:
>
>> I'm currently handling it with:
>>
>> ?<param name="context" value="numbergroup"/>
>>
>> and:
>>
>> ?<context name="numbergroup">
>> ? ?<extension name="numbergroup">
>> ? ? ?<condition field="destination_number" expression="^USERNAME$">
>> ? ? ? ?<action application="transfer" data="${sip_to_user} XML default"/>
>> ? ? ?</condition>
>> ? ?</extension>
>> ?</context>
>>
>> Is there a better way of handling it? Such as a param that tells Sofia
>> to use the To not the URI to populate destination_number?
>>
>> -Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From steveayre at gmail.com  Sat Jan  8 19:08:50 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 8 Jan 2011 16:08:50 +0000
Subject: [Freeswitch-users] Incoming call from registered gateway with
 DID in To: not URI
In-Reply-To: <AANLkTikHwRGhQXHb6vh8iCRcTbCvJdwWA3eWPxZm5qdu@mail.gmail.com>
References: <AANLkTimL-VTc0Hts3ps9h+5jxoc6ryGXVHfjPFQ2rHVG@mail.gmail.com>
	<AANLkTi=kyY52e0NoULNVevNS0xkPmTyTDEp6qe_9Ls6T@mail.gmail.com>
	<CAB3E5F4-95E2-4833-BA2C-9679BCC072C3@freeswitch.org>
	<AANLkTikHwRGhQXHb6vh8iCRcTbCvJdwWA3eWPxZm5qdu@mail.gmail.com>
Message-ID: <AANLkTinaCfYv0C9na2JaSXRNQ9GLG2LQAfXe6Yj_+Rys@mail.gmail.com>

> One of their guys said by email they tested on Git within the last few
> weeks and found calls were hanging up after 2m40s due to something
> about how we handle timers... I'm going to test that next. Were there
> any versions in Git recently that might have had problems?

There does appear to be an issue here.

Calls of 4mins duration with no problems with enable-timer=false

With enable-timer=true the call times out. The reason is they send an
INVITE and we reply 400 Bad Session Description.
http://pastebin.freeswitch.org/14957 shows those two packets.

-Steve



On 8 January 2011 15:56, Steven Ayre <steveayre at gmail.com> wrote:
> <param name="extension" value="auto_to_user"/> was what did the trick
> though. Thanks! :)
>
> There's a minor issue from it where the trunk shows unregistered on
> their website (they expect the Contact header to be username at ip") but
> incoming calls are still sent through fine so it doesn't actually
> matter at all. They can still see FS registered on their switch.
>
> I've added the configuration to the SIP Provider Examples section of the Wiki.
>
> One of their guys said by email they tested on Git within the last few
> weeks and found calls were hanging up after 2m40s due to something
> about how we handle timers... I'm going to test that next. Were there
> any versions in Git recently that might have had problems?
>
> -Steve
>
>
>
>
> On 8 January 2011 02:50, Brian West <brian at freeswitch.org> wrote:
>> Also if you set the extension to auto_to_user it'll do it also.
>>
>> /b
>>
>> On Jan 7, 2011, at 5:24 PM, Steven Ayre wrote:
>>
>>> I'm currently handling it with:
>>>
>>> ?<param name="context" value="numbergroup"/>
>>>
>>> and:
>>>
>>> ?<context name="numbergroup">
>>> ? ?<extension name="numbergroup">
>>> ? ? ?<condition field="destination_number" expression="^USERNAME$">
>>> ? ? ? ?<action application="transfer" data="${sip_to_user} XML default"/>
>>> ? ? ?</condition>
>>> ? ?</extension>
>>> ?</context>
>>>
>>> Is there a better way of handling it? Such as a param that tells Sofia
>>> to use the To not the URI to populate destination_number?
>>>
>>> -Steve
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


From steveayre at gmail.com  Sat Jan  8 19:11:10 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 8 Jan 2011 16:11:10 +0000
Subject: [Freeswitch-users] Incoming call from registered gateway with
 DID in To: not URI
In-Reply-To: <AANLkTinaCfYv0C9na2JaSXRNQ9GLG2LQAfXe6Yj_+Rys@mail.gmail.com>
References: <AANLkTimL-VTc0Hts3ps9h+5jxoc6ryGXVHfjPFQ2rHVG@mail.gmail.com>
	<AANLkTi=kyY52e0NoULNVevNS0xkPmTyTDEp6qe_9Ls6T@mail.gmail.com>
	<CAB3E5F4-95E2-4833-BA2C-9679BCC072C3@freeswitch.org>
	<AANLkTikHwRGhQXHb6vh8iCRcTbCvJdwWA3eWPxZm5qdu@mail.gmail.com>
	<AANLkTinaCfYv0C9na2JaSXRNQ9GLG2LQAfXe6Yj_+Rys@mail.gmail.com>
Message-ID: <AANLkTin+5kwn0OhjFCpVcqWsAa+C2Cx=zg3739TecwG-@mail.gmail.com>

>From the source code it looks like that means the SDP is invalid - do
you see the problem in those packet traces?

Warm Regards,
-Steve


On 8 January 2011 16:08, Steven Ayre <steveayre at gmail.com> wrote:
>> One of their guys said by email they tested on Git within the last few
>> weeks and found calls were hanging up after 2m40s due to something
>> about how we handle timers... I'm going to test that next. Were there
>> any versions in Git recently that might have had problems?
>
> There does appear to be an issue here.
>
> Calls of 4mins duration with no problems with enable-timer=false
>
> With enable-timer=true the call times out. The reason is they send an
> INVITE and we reply 400 Bad Session Description.
> http://pastebin.freeswitch.org/14957 shows those two packets.
>
> -Steve
>
>
>
> On 8 January 2011 15:56, Steven Ayre <steveayre at gmail.com> wrote:
>> <param name="extension" value="auto_to_user"/> was what did the trick
>> though. Thanks! :)
>>
>> There's a minor issue from it where the trunk shows unregistered on
>> their website (they expect the Contact header to be username at ip") but
>> incoming calls are still sent through fine so it doesn't actually
>> matter at all. They can still see FS registered on their switch.
>>
>> I've added the configuration to the SIP Provider Examples section of the Wiki.
>>
>> One of their guys said by email they tested on Git within the last few
>> weeks and found calls were hanging up after 2m40s due to something
>> about how we handle timers... I'm going to test that next. Were there
>> any versions in Git recently that might have had problems?
>>
>> -Steve
>>
>>
>>
>>
>> On 8 January 2011 02:50, Brian West <brian at freeswitch.org> wrote:
>>> Also if you set the extension to auto_to_user it'll do it also.
>>>
>>> /b
>>>
>>> On Jan 7, 2011, at 5:24 PM, Steven Ayre wrote:
>>>
>>>> I'm currently handling it with:
>>>>
>>>> ?<param name="context" value="numbergroup"/>
>>>>
>>>> and:
>>>>
>>>> ?<context name="numbergroup">
>>>> ? ?<extension name="numbergroup">
>>>> ? ? ?<condition field="destination_number" expression="^USERNAME$">
>>>> ? ? ? ?<action application="transfer" data="${sip_to_user} XML default"/>
>>>> ? ? ?</condition>
>>>> ? ?</extension>
>>>> ?</context>
>>>>
>>>> Is there a better way of handling it? Such as a param that tells Sofia
>>>> to use the To not the URI to populate destination_number?
>>>>
>>>> -Steve
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>


From steveayre at gmail.com  Sat Jan  8 19:32:25 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 8 Jan 2011 16:32:25 +0000
Subject: [Freeswitch-users] Incoming call from registered gateway with
 DID in To: not URI
In-Reply-To: <AANLkTin+5kwn0OhjFCpVcqWsAa+C2Cx=zg3739TecwG-@mail.gmail.com>
References: <AANLkTimL-VTc0Hts3ps9h+5jxoc6ryGXVHfjPFQ2rHVG@mail.gmail.com>
	<AANLkTi=kyY52e0NoULNVevNS0xkPmTyTDEp6qe_9Ls6T@mail.gmail.com>
	<CAB3E5F4-95E2-4833-BA2C-9679BCC072C3@freeswitch.org>
	<AANLkTikHwRGhQXHb6vh8iCRcTbCvJdwWA3eWPxZm5qdu@mail.gmail.com>
	<AANLkTinaCfYv0C9na2JaSXRNQ9GLG2LQAfXe6Yj_+Rys@mail.gmail.com>
	<AANLkTin+5kwn0OhjFCpVcqWsAa+C2Cx=zg3739TecwG-@mail.gmail.com>
Message-ID: <AANLkTim6SyHE5aK7wNM4HUFgG6fKCT+B8VjEJ85bintR@mail.gmail.com>

Sofia debugging shows this if it helps identify the problem at all:

nta: INVITE (6915472) going to existing leg
nta: timer shortened to 200 ms
nua: nua_stack_process_request: entering
soa_init_offer_answer(static::0x2346220) called
soa_set_remote_sdp(static::0x2346220, (nil), 0x233a90e, 521) called
nua(0x22f3260): INVITE server: error parsing SDP
nua: nua_invite_server_respond: entering
soa_clear_remote_sdp(static::0x2346220) called
tport_tsend(0x23720e0) tpn = UDP/80.93.165.111:5060
tport_resolve addrinfo = 80.93.165.111:5060
tport_by_addrinfo(0x23720e0): not found by name UDP/80.93.165.111:5060
tport_vsend(0x23720e0): 572 bytes of 572 to udp/80.93.165.111:5060
tport_vsend returned 572
nta: sent 400 Bad Session Description for INVITE (6915472)

-Steve


On 8 January 2011 16:11, Steven Ayre <steveayre at gmail.com> wrote:
> From the source code it looks like that means the SDP is invalid - do
> you see the problem in those packet traces?
>
> Warm Regards,
> -Steve
>
>
> On 8 January 2011 16:08, Steven Ayre <steveayre at gmail.com> wrote:
>>> One of their guys said by email they tested on Git within the last few
>>> weeks and found calls were hanging up after 2m40s due to something
>>> about how we handle timers... I'm going to test that next. Were there
>>> any versions in Git recently that might have had problems?
>>
>> There does appear to be an issue here.
>>
>> Calls of 4mins duration with no problems with enable-timer=false
>>
>> With enable-timer=true the call times out. The reason is they send an
>> INVITE and we reply 400 Bad Session Description.
>> http://pastebin.freeswitch.org/14957 shows those two packets.
>>
>> -Steve
>>
>>
>>
>> On 8 January 2011 15:56, Steven Ayre <steveayre at gmail.com> wrote:
>>> <param name="extension" value="auto_to_user"/> was what did the trick
>>> though. Thanks! :)
>>>
>>> There's a minor issue from it where the trunk shows unregistered on
>>> their website (they expect the Contact header to be username at ip") but
>>> incoming calls are still sent through fine so it doesn't actually
>>> matter at all. They can still see FS registered on their switch.
>>>
>>> I've added the configuration to the SIP Provider Examples section of the Wiki.
>>>
>>> One of their guys said by email they tested on Git within the last few
>>> weeks and found calls were hanging up after 2m40s due to something
>>> about how we handle timers... I'm going to test that next. Were there
>>> any versions in Git recently that might have had problems?
>>>
>>> -Steve
>>>
>>>
>>>
>>>
>>> On 8 January 2011 02:50, Brian West <brian at freeswitch.org> wrote:
>>>> Also if you set the extension to auto_to_user it'll do it also.
>>>>
>>>> /b
>>>>
>>>> On Jan 7, 2011, at 5:24 PM, Steven Ayre wrote:
>>>>
>>>>> I'm currently handling it with:
>>>>>
>>>>> ?<param name="context" value="numbergroup"/>
>>>>>
>>>>> and:
>>>>>
>>>>> ?<context name="numbergroup">
>>>>> ? ?<extension name="numbergroup">
>>>>> ? ? ?<condition field="destination_number" expression="^USERNAME$">
>>>>> ? ? ? ?<action application="transfer" data="${sip_to_user} XML default"/>
>>>>> ? ? ?</condition>
>>>>> ? ?</extension>
>>>>> ?</context>
>>>>>
>>>>> Is there a better way of handling it? Such as a param that tells Sofia
>>>>> to use the To not the URI to populate destination_number?
>>>>>
>>>>> -Steve
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>
>


From steveayre at gmail.com  Sat Jan  8 22:29:26 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 8 Jan 2011 19:29:26 +0000
Subject: [Freeswitch-users] Incoming call from registered gateway with
 DID in To: not URI
In-Reply-To: <AANLkTim6SyHE5aK7wNM4HUFgG6fKCT+B8VjEJ85bintR@mail.gmail.com>
References: <AANLkTimL-VTc0Hts3ps9h+5jxoc6ryGXVHfjPFQ2rHVG@mail.gmail.com>
	<AANLkTi=kyY52e0NoULNVevNS0xkPmTyTDEp6qe_9Ls6T@mail.gmail.com>
	<CAB3E5F4-95E2-4833-BA2C-9679BCC072C3@freeswitch.org>
	<AANLkTikHwRGhQXHb6vh8iCRcTbCvJdwWA3eWPxZm5qdu@mail.gmail.com>
	<AANLkTinaCfYv0C9na2JaSXRNQ9GLG2LQAfXe6Yj_+Rys@mail.gmail.com>
	<AANLkTin+5kwn0OhjFCpVcqWsAa+C2Cx=zg3739TecwG-@mail.gmail.com>
	<AANLkTim6SyHE5aK7wNM4HUFgG6fKCT+B8VjEJ85bintR@mail.gmail.com>
Message-ID: <AANLkTikVzO=1sEDgC5=OKMEpncfQktMBXY7a4=haJ2kC@mail.gmail.com>

Trying to reproduce this, and found it happens on every call.

The INVITE: http://pastebin.freeswitch.org/14958
The reINVITE: http://pastebin.freeswitch.org/14959
Full packet trace: http://pastebin.freeswitch.org/14960

I'm pretty sure there's a problem in the SDP they're sending in the
reINVITE. It's different from the SDP in the first invite.

The only part of the reINVITE that stands out as looking odd to me is this line:
a=rtpmap:96 /0

Would that make the Sofia SDP parser fail?

96 isn't a payload number they offer in the initial INVITE, or that we
offer in the 200 OK, so I don't know where that number shows up from.

Warm regards,
-Steve


On 8 January 2011 16:32, Steven Ayre <steveayre at gmail.com> wrote:
> Sofia debugging shows this if it helps identify the problem at all:
>
> nta: INVITE (6915472) going to existing leg
> nta: timer shortened to 200 ms
> nua: nua_stack_process_request: entering
> soa_init_offer_answer(static::0x2346220) called
> soa_set_remote_sdp(static::0x2346220, (nil), 0x233a90e, 521) called
> nua(0x22f3260): INVITE server: error parsing SDP
> nua: nua_invite_server_respond: entering
> soa_clear_remote_sdp(static::0x2346220) called
> tport_tsend(0x23720e0) tpn = UDP/80.93.165.111:5060
> tport_resolve addrinfo = 80.93.165.111:5060
> tport_by_addrinfo(0x23720e0): not found by name UDP/80.93.165.111:5060
> tport_vsend(0x23720e0): 572 bytes of 572 to udp/80.93.165.111:5060
> tport_vsend returned 572
> nta: sent 400 Bad Session Description for INVITE (6915472)
>
> -Steve
>
>
> On 8 January 2011 16:11, Steven Ayre <steveayre at gmail.com> wrote:
>> From the source code it looks like that means the SDP is invalid - do
>> you see the problem in those packet traces?
>>
>> Warm Regards,
>> -Steve
>>
>>
>> On 8 January 2011 16:08, Steven Ayre <steveayre at gmail.com> wrote:
>>>> One of their guys said by email they tested on Git within the last few
>>>> weeks and found calls were hanging up after 2m40s due to something
>>>> about how we handle timers... I'm going to test that next. Were there
>>>> any versions in Git recently that might have had problems?
>>>
>>> There does appear to be an issue here.
>>>
>>> Calls of 4mins duration with no problems with enable-timer=false
>>>
>>> With enable-timer=true the call times out. The reason is they send an
>>> INVITE and we reply 400 Bad Session Description.
>>> http://pastebin.freeswitch.org/14957 shows those two packets.
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 8 January 2011 15:56, Steven Ayre <steveayre at gmail.com> wrote:
>>>> <param name="extension" value="auto_to_user"/> was what did the trick
>>>> though. Thanks! :)
>>>>
>>>> There's a minor issue from it where the trunk shows unregistered on
>>>> their website (they expect the Contact header to be username at ip") but
>>>> incoming calls are still sent through fine so it doesn't actually
>>>> matter at all. They can still see FS registered on their switch.
>>>>
>>>> I've added the configuration to the SIP Provider Examples section of the Wiki.
>>>>
>>>> One of their guys said by email they tested on Git within the last few
>>>> weeks and found calls were hanging up after 2m40s due to something
>>>> about how we handle timers... I'm going to test that next. Were there
>>>> any versions in Git recently that might have had problems?
>>>>
>>>> -Steve
>>>>
>>>>
>>>>
>>>>
>>>> On 8 January 2011 02:50, Brian West <brian at freeswitch.org> wrote:
>>>>> Also if you set the extension to auto_to_user it'll do it also.
>>>>>
>>>>> /b
>>>>>
>>>>> On Jan 7, 2011, at 5:24 PM, Steven Ayre wrote:
>>>>>
>>>>>> I'm currently handling it with:
>>>>>>
>>>>>> ?<param name="context" value="numbergroup"/>
>>>>>>
>>>>>> and:
>>>>>>
>>>>>> ?<context name="numbergroup">
>>>>>> ? ?<extension name="numbergroup">
>>>>>> ? ? ?<condition field="destination_number" expression="^USERNAME$">
>>>>>> ? ? ? ?<action application="transfer" data="${sip_to_user} XML default"/>
>>>>>> ? ? ?</condition>
>>>>>> ? ?</extension>
>>>>>> ?</context>
>>>>>>
>>>>>> Is there a better way of handling it? Such as a param that tells Sofia
>>>>>> to use the To not the URI to populate destination_number?
>>>>>>
>>>>>> -Steve
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>
>>
>


From jeff at jefflenk.com  Sun Jan  9 04:10:04 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sat, 8 Jan 2011 17:10:04 -0800 (PST)
Subject: [Freeswitch-users] Iksemel msvs compiling
In-Reply-To: <SNT109-DS11CB0ECE5BD0F9C0384E03C10C0@phx.gbl>
References: <SNT109-DS13313A1FA906527438E8CC1090@phx.gbl>
	<SNT109-DS1155D401A0B9742CF63E13C10B0@phx.gbl>
	<1294375438447-5898181.post@n2.nabble.com>
	<SNT109-DS11CB0ECE5BD0F9C0384E03C10C0@phx.gbl>
Message-ID: <1294535404901-5903534.post@n2.nabble.com>


You had a a couple of download errors. openssl and sphinx models.

try the build again and check /libs for tarballs
openssl-1.0.0a.tar.gz
communicator_semi_6000_20080321.tar.gz
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Iksemel-msvs-compiling-tp5891263p5903534.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From covici at ccs.covici.com  Sun Jan  9 05:43:10 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sat, 08 Jan 2011 21:43:10 -0500
Subject: [Freeswitch-users] portaudio no longer working
Message-ID: <32235.1294540990@ccs.covici.com>

As of the latest git from today, port audio is no longer working.  I get
no sound and cannot even unload the module.  When I shutdown, I have to
kill fs with signal 9 and the last line in the log concerns pa.  I am
running linux gentoo x86.

Any help would be appreciated.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From boris at tagnet.ru  Sun Jan  9 10:26:50 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sun, 09 Jan 2011 12:26:50 +0500
Subject: [Freeswitch-users] Use <tab> in mod_cdr_csv
Message-ID: <4D29633A.7090206@tagnet.ru>

Hello!

     Is this possible to use special characters (<tab> for example) to 
separate fields?

-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From covici at ccs.covici.com  Sun Jan  9 17:50:54 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sun, 09 Jan 2011 09:50:54 -0500
Subject: [Freeswitch-users] portaudio no longer working
In-Reply-To: <32235.1294540990@ccs.covici.com>
References: <32235.1294540990@ccs.covici.com>
Message-ID: <17356.1294584654@ccs.covici.com>

As of the same git, dtmf detection is not working -- at least from my
did -- it does work locally.  If I go back to
70697b8835145a800f035c667f5c0f7defdc97ca then I get my dtmf detection
back,  but portaudio is still broke.

covici at ccs.covici.com wrote:

> As of the latest git from today, port audio is no longer working.  I get
> no sound and cannot even unload the module.  When I shutdown, I have to
> kill fs with signal 9 and the last line in the log concerns pa.  I am
> running linux gentoo x86.
> 
> Any help would be appreciated.
> 
> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From lloyd.aloysius at gmail.com  Sun Jan  9 19:25:34 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 9 Jan 2011 11:25:34 -0500
Subject: [Freeswitch-users] FreeSWITCH - Ploycom phone in
	hostedenvironment
In-Reply-To: <5D1F1B9292A74023A294224EB7CEC04A@e1705>
References: <AANLkTimiG_GyWLO_JM+HYMM6bB0izxci1hZ=HosQwJsL@mail.gmail.com>
	<30FCC104-2CFF-4E0E-9227-4DD760700A16@ipeva.fr>
	<AANLkTim_0wLR54D6KZBg5ioS+XpxVfJyMvUmuLWYgpc+@mail.gmail.com>
	<AANLkTinvcctL49YYh-yEzNXiw0RKm6_Vvhw4ff-rt2_=@mail.gmail.com>
	<B93EB68C-CAD3-40C9-BB08-D988532C8913@freeswitch.org>
	<5D1F1B9292A74023A294224EB7CEC04A@e1705>
Message-ID: <AANLkTineU=5j3HnXQs5AWCYEK6x1O-eh5FNaqRvP+Esn@mail.gmail.com>

Thanks Brian.

<param name="NDLB-force-rport" value="safe"/>  works and magic.

But this is the firs time using a polycom phone , configuring polycom phone
is a nightmare. Specially setting up working with DNS SRV records.

But I find the sound quality is amazing and rock solid. Compare to Aastra
and Linksys.

Yes this year definitely attending cluecon 2011.

Thanks and regards,
Lloyd


On Fri, Jan 7, 2011 at 8:29 PM, Madovsky <infos at madovsky.org> wrote:

>  > Step one.  use the VERY latests FreeSWITCH... set NDLB-force-rport to
> "safe" and the rest will be magical.
> ok
>
> > Step two... profit.
> I hope for all who work hard on FS (but not for me now)
>
> > Step three... come to cluecon 2011.
> I would be a pleasure, but not interesting if there's no barbecue...
>
> Step four.. I'm joking (not for Step two :(  )
>
> ----- Original Message -----
> *From:* Brian West <brian at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Friday, January 07, 2011 6:18 PM
> *Subject:* Re: [Freeswitch-users] FreeSWITCH - Ploycom phone in
> hostedenvironment
>
> You guys are honestly forgetting something.
>
> Step one.  use the VERY latests FreeSWITCH... set NDLB-force-rport to
> "safe" and the rest will be magical.
>
> Step two... profit.
>
> Step three... come to cluecon 2011.
>
> /b
>
>
>  On Jan 7, 2011, at 4:45 PM, Chris Burns wrote:
>
> Forcing rport in appropriate sofia profile will usually do it ... uncomment
> NDLB-force-rport and restart the profile. In a certain rare situation I had
> to fill out the phone's <nat/> tag dynamically when it provisioned. For
> small environments you could manually assign each phone a signal port ....
> yay polycom? x_X
>
> 2011/1/7 Aloysius Lloyd <lloyd.aloysius at gmail.com>
>
>> Yes but no luck.
>>
>> Thanks
>> Lloyd
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> This message has been scanned for viruses and
> dangerous content by *MailScanner* <http://www.mailscanner.info/>, and is
> believed to be clean.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Sun Jan  9 20:09:11 2011
From: brian at freeswitch.org (Brian West)
Date: Sun, 9 Jan 2011 11:09:11 -0600
Subject: [Freeswitch-users] FreeSWITCH - Ploycom phone in
	hostedenvironment
In-Reply-To: <AANLkTineU=5j3HnXQs5AWCYEK6x1O-eh5FNaqRvP+Esn@mail.gmail.com>
References: <AANLkTimiG_GyWLO_JM+HYMM6bB0izxci1hZ=HosQwJsL@mail.gmail.com>
	<30FCC104-2CFF-4E0E-9227-4DD760700A16@ipeva.fr>
	<AANLkTim_0wLR54D6KZBg5ioS+XpxVfJyMvUmuLWYgpc+@mail.gmail.com>
	<AANLkTinvcctL49YYh-yEzNXiw0RKm6_Vvhw4ff-rt2_=@mail.gmail.com>
	<B93EB68C-CAD3-40C9-BB08-D988532C8913@freeswitch.org>
	<5D1F1B9292A74023A294224EB7CEC04A@e1705>
	<AANLkTineU=5j3HnXQs5AWCYEK6x1O-eh5FNaqRvP+Esn@mail.gmail.com>
Message-ID: <8A88F903-1DBA-489D-AC51-37CF939FF0C0@freeswitch.org>

You don't have to do this step.

/b

On Jan 9, 2011, at 10:25 AM, Aloysius Lloyd wrote:

> Specially setting up working with DNS SRV records. 

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From andrew at hijacked.us  Sun Jan  9 21:47:14 2011
From: andrew at hijacked.us (Andrew Thompson)
Date: Sun, 9 Jan 2011 13:47:14 -0500
Subject: [Freeswitch-users] portaudio no longer working
In-Reply-To: <17356.1294584654@ccs.covici.com>
References: <32235.1294540990@ccs.covici.com> <17356.1294584654@ccs.covici.com>
Message-ID: <20110109184714.GG26890@hijacked.us>

On Sun, Jan 09, 2011 at 09:50:54AM -0500, covici at ccs.covici.com wrote:
> As of the same git, dtmf detection is not working -- at least from my
> did -- it does work locally.  If I go back to
> 70697b8835145a800f035c667f5c0f7defdc97ca then I get my dtmf detection
> back,  but portaudio is still broke.
> 

Portaudio hasn't changed in months, I suspect something on your system
changed and broke it rather than something in FreeSWITCH changing.
Portaudio isn't compatible with pulseaudio, for example.

Andrew


From covici at ccs.covici.com  Sun Jan  9 22:31:51 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sun, 09 Jan 2011 14:31:51 -0500
Subject: [Freeswitch-users] portaudio no longer working
In-Reply-To: <20110109184714.GG26890@hijacked.us>
References: <32235.1294540990@ccs.covici.com> <17356.1294584654@ccs.covici.com>
	<20110109184714.GG26890@hijacked.us>
Message-ID: <20447.1294601511@ccs.covici.com>

Andrew Thompson <andrew at hijacked.us> wrote:

> On Sun, Jan 09, 2011 at 09:50:54AM -0500, covici at ccs.covici.com wrote:
> > As of the same git, dtmf detection is not working -- at least from my
> > did -- it does work locally.  If I go back to
> > 70697b8835145a800f035c667f5c0f7defdc97ca then I get my dtmf detection
> > back,  but portaudio is still broke.
> > 
> 
> Portaudio hasn't changed in months, I suspect something on your system
> changed and broke it rather than something in FreeSWITCH changing.
> Portaudio isn't compatible with pulseaudio, for example.
Well, I am not using pulse audio at all, but I can use mplayer with alsa
and that works -- any other change which could break this?


-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From covici at ccs.covici.com  Sun Jan  9 22:45:20 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sun, 09 Jan 2011 14:45:20 -0500
Subject: [Freeswitch-users] portaudio no longer working
In-Reply-To: <20110109184714.GG26890@hijacked.us>
References: <32235.1294540990@ccs.covici.com> <17356.1294584654@ccs.covici.com>
	<20110109184714.GG26890@hijacked.us>
Message-ID: <12405.1294602320@ccs.covici.com>

Andrew Thompson <andrew at hijacked.us> wrote:

> On Sun, Jan 09, 2011 at 09:50:54AM -0500, covici at ccs.covici.com wrote:
> > As of the same git, dtmf detection is not working -- at least from my
> > did -- it does work locally.  If I go back to
> > 70697b8835145a800f035c667f5c0f7defdc97ca then I get my dtmf detection
> > back,  but portaudio is still broke.
> > 
> 
> Portaudio hasn't changed in months, I suspect something on your system
> changed and broke it rather than something in FreeSWITCH changing.
> Portaudio isn't compatible with pulseaudio, for example.

Also, why can't I unload portaudio, or shutdown fs without manually killing it
with signal 9?  Is this normal for portaudio?
-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From infos at madovsky.org  Sun Jan  9 23:01:17 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 9 Jan 2011 15:01:17 -0500
Subject: [Freeswitch-users] portaudio no longer working
References: <32235.1294540990@ccs.covici.com>
	<17356.1294584654@ccs.covici.com><20110109184714.GG26890@hijacked.us>
	<12405.1294602320@ccs.covici.com>
Message-ID: <3A2FD2FCAE7F467988D245E0717D5A28@e1705>

maybe try to make a fresh install and see

----- Original Message ----- 
From: <covici at ccs.covici.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Sunday, January 09, 2011 2:45 PM
Subject: Re: [Freeswitch-users] portaudio no longer working


> Andrew Thompson <andrew at hijacked.us> wrote:
>
>> On Sun, Jan 09, 2011 at 09:50:54AM -0500, covici at ccs.covici.com wrote:
>> > As of the same git, dtmf detection is not working -- at least from my
>> > did -- it does work locally.  If I go back to
>> > 70697b8835145a800f035c667f5c0f7defdc97ca then I get my dtmf detection
>> > back,  but portaudio is still broke.
>> >
>>
>> Portaudio hasn't changed in months, I suspect something on your system
>> changed and broke it rather than something in FreeSWITCH changing.
>> Portaudio isn't compatible with pulseaudio, for example.
>
> Also, why can't I unload portaudio, or shutdown fs without manually 
> killing it
> with signal 9?  Is this normal for portaudio?
> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From andrew at hijacked.us  Sun Jan  9 23:01:24 2011
From: andrew at hijacked.us (Andrew Thompson)
Date: Sun, 9 Jan 2011 15:01:24 -0500
Subject: [Freeswitch-users] portaudio no longer working
In-Reply-To: <12405.1294602320@ccs.covici.com>
References: <32235.1294540990@ccs.covici.com> <17356.1294584654@ccs.covici.com>
	<20110109184714.GG26890@hijacked.us>
	<12405.1294602320@ccs.covici.com>
Message-ID: <20110109200123.GH26890@hijacked.us>

On Sun, Jan 09, 2011 at 02:45:20PM -0500, covici at ccs.covici.com wrote:
> Andrew Thompson <andrew at hijacked.us> wrote:
> 
> > On Sun, Jan 09, 2011 at 09:50:54AM -0500, covici at ccs.covici.com wrote:
> > > As of the same git, dtmf detection is not working -- at least from my
> > > did -- it does work locally.  If I go back to
> > > 70697b8835145a800f035c667f5c0f7defdc97ca then I get my dtmf detection
> > > back,  but portaudio is still broke.
> > > 
> > 
> > Portaudio hasn't changed in months, I suspect something on your system
> > changed and broke it rather than something in FreeSWITCH changing.
> > Portaudio isn't compatible with pulseaudio, for example.
> 
> Also, why can't I unload portaudio, or shutdown fs without manually killing it
> with signal 9?  Is this normal for portaudio?

I suspect portaudio is getting blocked trying to grab a device or
something.

Andrew


From infos at madovsky.org  Sun Jan  9 23:03:52 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 9 Jan 2011 15:03:52 -0500
Subject: [Freeswitch-users] BV32 iLBC
Message-ID: <486F5962296C4CC5A5221088DA20A449@e1705>

I played with all builtin codecs today
and didn't succeed to originate a call with BV32, iLBC
and X-lite 
is it need any special settings ?

Thanks
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From matte at ahavaxthuset.se  Sun Jan  9 22:21:30 2011
From: matte at ahavaxthuset.se (Mattias Hemmingsson)
Date: Sun, 9 Jan 2011 20:21:30 +0100 (CET)
Subject: [Freeswitch-users] Two meny qestions
In-Reply-To: <15598200.261294600828264.JavaMail.root@mailserver>
Message-ID: <10669245.281294600890468.JavaMail.root@mailserver>

Hi

So first realy like the freeswitchn config use to work with trixbox but i like to have everything in config /xml files.
Bur i have stumbeld over two problems. And i think that they are realy easy to fix.

i have a nice meny that greats the caller comming in. And it works when i choose to be transferd to user 1000 in the meny by pressing one.
But when the is not online i want the user to be transferd to the users voicemail.
I have voicemail working of i call the user from a nother externsion.
But i would like it to wokr from the ivr meny as well.


And second i whould like to if you press two you should be transferd to an ring group.
I have an working ring group with extension 200.


This is my ivr meny

<include> 
<menu name="elino_ivr"
          greet-long="/opt/freeswitch/conf/elino/sounds/elino_ivr.wav"
          timeout="10000"
          inter-digit-timeout="2000"
          max-failures="1"
          max-timeouts="2"
          digit-len="1">
      <entry action="menu-exec-app" digits="1" param="bridge sofia/internal/1000%10.0.1.6"/>
      <entry action="menu-exec-app" digits="2" param="bridge sofia/internal/1001%10.0.1.6"/>
      <entry action="menu-exec-app" digits="3" param="bridge sofia/internal/1000%10.0.1.6"/>
    </menu>
</include>


i have one doman called www.elino.se set upp as well.


Regards Mattias


From brian at freeswitch.org  Mon Jan 10 00:35:00 2011
From: brian at freeswitch.org (Brian West)
Date: Sun, 9 Jan 2011 15:35:00 -0600
Subject: [Freeswitch-users] Two meny qestions
In-Reply-To: <10669245.281294600890468.JavaMail.root@mailserver>
References: <10669245.281294600890468.JavaMail.root@mailserver>
Message-ID: <9DD9A108-0CFF-4224-88C2-95CFF9E0FE76@freeswitch.org>

This is because you're calling bridge right to the users endpoint... if you were to transfer to extension 1000 or 1001 then voicemail would work exactly like you expect.

/b

On Jan 9, 2011, at 1:21 PM, Mattias Hemmingsson wrote:

> But when the is not online i want the user to be transferd to the users voicemail.
> I have voicemail working of i call the user from a nother externsion.
> But i would like it to wokr from the ivr meny as well.



From daniel-listas at gmx.net  Mon Jan 10 00:15:51 2011
From: daniel-listas at gmx.net (Daniel Bareiro)
Date: Sun, 9 Jan 2011 18:15:51 -0300
Subject: [Freeswitch-users] Prefix for outgoing calls
Message-ID: <slrniik9g2.26s.daniel-listas@defiant.freesoftware>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi all!

I'm starting to experiment with FreeSWITCH and I defined a gateway for
outgoing calls to Iptel.org on the following file:

/usr/local/freeswitch/conf/sip_profiles/external/iptel.org.xml

using the following format:

<include>
   <gateway name="iptel">
      <param name="username" value="account"/>
      <param name="realm" value="iptel.org"/>
      <param name="password" value="password"/>
      <param name="proxy" value="sip.iptel.org"/>
   </gateway>
</include>


Apparently, FreeSwitch able to register against iptel:


freeswitch at internal> sofia status
                     Name          Type                                       Data      State
=================================================================================================
                 external       profile               sip:mod_sofia at 10.1.0.45:5080      RUNNING (0)
    external::example.com       gateway                    sip:joeuser at example.com      NOREG
          external::iptel       gateway                  sip:account at sip.iptel.org      REGED
          external::ekiga       gateway                      sip:account at ekiga.net      REGED
            internal-ipv6       profile                   sip:mod_sofia@[::1]:5060      RUNNING (0)
                 internal       profile               sip:mod_sofia at 10.1.0.45:5060      RUNNING (0)
                10.1.0.45         alias                                   internal      ALIASED
=================================================================================================
3 profiles 1 alias

freeswitch at internal>


Then I tried to make any call from the predefined extensions of
FreeSwitch (1000.xml) to a number of iptel. Then calling from extension
1000 to, for example, music at iptel.org, I was able to communicate from a
softphone. The reason why the call is successfully using @iptel.org is
because it is taking the "realm" in the file iptel.org.xml?

I have been reading the FS documentation but I not found a way to use a
prefix to call a number of iptel (or some other provider) without having
to use @provider. I have no clear the syntax I should use or where
should I do this configuration.




Someone could give me a hand with this configuration?


Thanks in advance for your reply.

Regards,
Daniel

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From brian at freeswitch.org  Mon Jan 10 01:02:10 2011
From: brian at freeswitch.org (Brian West)
Date: Sun, 9 Jan 2011 16:02:10 -0600
Subject: [Freeswitch-users] Prefix for outgoing calls
In-Reply-To: <slrniik9g2.26s.daniel-listas@defiant.freesoftware>
References: <slrniik9g2.26s.daniel-listas@defiant.freesoftware>
Message-ID: <62D389FD-B699-457B-820D-DAEE5FA4FE44@freeswitch.org>


On Jan 9, 2011, at 3:15 PM, Daniel Bareiro wrote:

> Then I tried to make any call from the predefined extensions of
> FreeSwitch (1000.xml) to a number of iptel. Then calling from extension
> 1000 to, for example, music at iptel.org, I was able to communicate from a
> softphone. The reason why the call is successfully using @iptel.org is
> because it is taking the "realm" in the file iptel.org.xml?

Chances are you bypassed FS when you dialed music at iptel.org

> 
> I have been reading the FS documentation but I not found a way to use a
> prefix to call a number of iptel (or some other provider) without having
> to use @provider. I have no clear the syntax I should use or where
> should I do this configuration.


What do you mean?

  <extension name="local.example.com">
    <condition field="destination_number" expression="^5551212$">
      <action application="bridge" data="sofia/gateway/iptel.org/music"/>
    </condition>
  </extension>

That would take 5551212 from a softphone and dial music at iptel.org .. NEVER put @iptel.org at the end of that or you'll ened up calling music at iptel.org@iptel.org :P

/b

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From grsingh750 at gmail.com  Mon Jan 10 04:06:34 2011
From: grsingh750 at gmail.com (guru singh)
Date: Mon, 10 Jan 2011 06:36:34 +0530
Subject: [Freeswitch-users] Manipulate User Variables?
Message-ID: <AANLkTinkSNaw4Y6diYvmPzjD_y3E7c8G_4yekdGJ8FYh@mail.gmail.com>

Hi,

Is it possible to manipulate variables defined for users in
/directory/default/1000.xml?
Suppose I have a <variable name="flag" value="0"> for each user. Can I
dial an extension and change this value, only for the specific user
who dialed it?

Thanks
gs


From Avi at aMarcus.com  Mon Jan 10 04:26:40 2011
From: Avi at aMarcus.com (Avi Marcus)
Date: Mon, 10 Jan 2011 03:26:40 +0200
Subject: [Freeswitch-users] Manipulate User Variables?
In-Reply-To: <AANLkTinkSNaw4Y6diYvmPzjD_y3E7c8G_4yekdGJ8FYh@mail.gmail.com>
References: <AANLkTinkSNaw4Y6diYvmPzjD_y3E7c8G_4yekdGJ8FYh@mail.gmail.com>
Message-ID: <AANLkTi=wRPrRGhSifWmUhfs3YPeCsSnUn+BScn0B6oRS@mail.gmail.com>

Not that I'm aware of.
To duplicate such functionality, however, there are several options:
1) Use the db to store such variables, see:
http://wiki.freeswitch.org/wiki/Mod_db
 <http://wiki.freeswitch.org/wiki/Mod_db>2) use a complicated script that
rewrites the xml and triggers a reload xml (kind of convoluted, I
don't recommended it)
3) Use your own sql interface, e.g. using lua or mod_odbc_query from the git
contrib. However, familiarizing yourself with mod_db is probably enough.
4) Use mod_xml_curl to process what happens and run the sql query / or file
edits within that ruby/php/etc script.
I use mod_xml_curl for basically all my dialplan, if you have anything very
specific (e.g. dependent on lots of queries, or custom routing and billing)
it's the most flexible. Static XML is only so flexible.

-Avi

On Mon, Jan 10, 2011 at 3:06 AM, guru singh <grsingh750 at gmail.com> wrote:

> Hi,
>
> Is it possible to manipulate variables defined for users in
> /directory/default/1000.xml?
> Suppose I have a <variable name="flag" value="0"> for each user. Can I
> dial an extension and change this value, only for the specific user
> who dialed it?
>
> Thanks
> gs
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From max.bridgewater at gmail.com  Mon Jan 10 04:51:44 2011
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sun, 9 Jan 2011 20:51:44 -0500
Subject: [Freeswitch-users] Status of ODBC
Message-ID: <AANLkTimXPAh7Jfp_dzEQMkC32t2KY-DEDUJ65=E2MS8Z@mail.gmail.com>

Hi,

The following page seems to be outdated:
http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc. I can't find
languages/mod_spidermonkey_odbc in modules.conf. Can somebody points
me to the right direction?

Thanks,
max.


From infos at madovsky.org  Mon Jan 10 05:04:07 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 9 Jan 2011 21:04:07 -0500
Subject: [Freeswitch-users] return value in diaplan from external script
Message-ID: <DB464DBE57BD41F0A5391162350DAB72@e1705>

I'd like to get the value from a PHP or Perl function back
to the dialplan. not sure how get the value.

Thanks
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From Avi at aMarcus.com  Mon Jan 10 06:12:19 2011
From: Avi at aMarcus.com (Avi Marcus)
Date: Mon, 10 Jan 2011 05:12:19 +0200
Subject: [Freeswitch-users] return value in diaplan from external script
In-Reply-To: <DB464DBE57BD41F0A5391162350DAB72@e1705>
References: <DB464DBE57BD41F0A5391162350DAB72@e1705>
Message-ID: <AANLkTikMfoGFcaZqd5BKn6xCpt_Z4gD4dDwKnf+aqQ_T@mail.gmail.com>

You can't, or at least you can't directly. While you can call a php script,
freeswitch will not wait for the output. Here's a few options: -use lua to
do your processing. That can be run inline to set variables and the like.
(Easiest if you can get your head around the lua) -if you just need a simple
sql query, use somethibg like mod odbc query from the git contrib. -use xml
curl so you can completely control the dialplan flow -or even more if lua
isn't your thing you can create an esl php applicatio that can control the
call in realtime, rather than just generating dialplan xml. -or an iffy idea
-call the php script, have it save to sql or something, and have the xml
dialplan sleep for a moment and then retrieve it.
On Jan 10, 2011 4:08 AM, "Madovsky" <infos at madovsky.org> wrote:
> I'd like to get the value from a PHP or Perl function back
> to the dialplan. not sure how get the value.
>
> Thanks
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From infos at madovsky.org  Mon Jan 10 06:52:29 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 9 Jan 2011 22:52:29 -0500
Subject: [Freeswitch-users] return value in diaplan from external script
References: <DB464DBE57BD41F0A5391162350DAB72@e1705>
	<AANLkTikMfoGFcaZqd5BKn6xCpt_Z4gD4dDwKnf+aqQ_T@mail.gmail.com>
Message-ID: <8C9477DDB0A54D7E952539DE22F28805@e1705>

ok thanks for your info,
and Perl ? if Perl can manage session vars array,
if I set a var in it on the fly and call it from dialplan just after ?
I need only to get a little info from the DB.


  ----- Original Message ----- 
  From: Avi Marcus 
  To: FreeSWITCH Users Help 
  Sent: Sunday, January 09, 2011 10:12 PM
  Subject: Re: [Freeswitch-users] return value in diaplan from external script


  You can't, or at least you can't directly. While you can call a php script, freeswitch will not wait for the output. Here's a few options: -use lua to do your processing. That can be run inline to set variables and the like. (Easiest if you can get your head around the lua) -if you just need a simple sql query, use somethibg like mod odbc query from the git contrib. -use xml curl so you can completely control the dialplan flow -or even more if lua isn't your thing you can create an esl php applicatio that can control the call in realtime, rather than just generating dialplan xml. -or an iffy idea -call the php script, have it save to sql or something, and have the xml dialplan sleep for a moment and then retrieve it.

  On Jan 10, 2011 4:08 AM, "Madovsky" <infos at madovsky.org> wrote:
  > I'd like to get the value from a PHP or Perl function back
  > to the dialplan. not sure how get the value.
  > 
  > Thanks



------------------------------------------------------------------------------


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  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Mon Jan 10 06:53:45 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 9 Jan 2011 22:53:45 -0500
Subject: [Freeswitch-users] return value in diaplan from external script
References: <DB464DBE57BD41F0A5391162350DAB72@e1705>
	<AANLkTikMfoGFcaZqd5BKn6xCpt_Z4gD4dDwKnf+aqQ_T@mail.gmail.com>
Message-ID: <84C322EABBB54A868975E4249077BDE5@e1705>

indeed Mod_db is perfect for that

Thanks
  ----- Original Message ----- 
  From: Avi Marcus 
  To: FreeSWITCH Users Help 
  Sent: Sunday, January 09, 2011 10:12 PM
  Subject: Re: [Freeswitch-users] return value in diaplan from external script


  You can't, or at least you can't directly. While you can call a php script, freeswitch will not wait for the output. Here's a few options: -use lua to do your processing. That can be run inline to set variables and the like. (Easiest if you can get your head around the lua) -if you just need a simple sql query, use somethibg like mod odbc query from the git contrib. -use xml curl so you can completely control the dialplan flow -or even more if lua isn't your thing you can create an esl php applicatio that can control the call in realtime, rather than just generating dialplan xml. -or an iffy idea -call the php script, have it save to sql or something, and have the xml dialplan sleep for a moment and then retrieve it.

  On Jan 10, 2011 4:08 AM, "Madovsky" <infos at madovsky.org> wrote:
  > I'd like to get the value from a PHP or Perl function back
  > to the dialplan. not sure how get the value.
  > 
  > Thanks



------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From u2nsam at gmail.com  Mon Jan 10 08:33:23 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 10 Jan 2011 11:03:23 +0530
Subject: [Freeswitch-users] codec error
In-Reply-To: <2E41E910-1DD0-4F24-B613-96A3A5BE0AF6@freeswitch.org>
References: <AANLkTi=cKTUOcN--hZm2yLXY8s-pWotek+_YA1Bxw2jw@mail.gmail.com>
	<AANLkTinLm5AP9kqoeHarqrCJGjsUq6Xo8=aOnY05HNHN@mail.gmail.com>
	<AANLkTikmHcMHfc3ydk+8b2SWEKB+sR=Awj0OVri6qdrK@mail.gmail.com>
	<AANLkTi=e4sVBpb2M5HSni-1ZXViExxkxocvmnC=jBBi_@mail.gmail.com>
	<AANLkTikGM8fQzqXaFoOd=m3LaVLae6B+O7qBPVAYmzL2@mail.gmail.com>
	<AANLkTim8Bfsr7=hX=RBJJEo4TKZqUZgn5y7yUbh9d1k1@mail.gmail.com>
	<AANLkTi=KdMjptEK5_sAWZ4DcXU1zJ9DstXnA_29rEx_k@mail.gmail.com>
	<AANLkTinSByL-0TA+VthcBeiiewcHP3ZZEsR73FoMRts4@mail.gmail.com>
	<AANLkTi=X=RObn1xHxZpBR49Xpk+X3YLkRt3c=Aejo9j7@mail.gmail.com>
	<2E41E910-1DD0-4F24-B613-96A3A5BE0AF6@freeswitch.org>
Message-ID: <AANLkTi=r2sG_JAsZ+V+Bwg-0X7rD0_n4EAO69m1NM8jP@mail.gmail.com>

Which module should i load for g7222 ?

Regds
Sam

On Fri, Jan 7, 2011 at 8:43 PM, Brian West <brian at freeswitch.org> wrote:

> For one you have G7222 ... secondly you do not have mod_spandsp loaded.
>
> /b
>
> On Jan 7, 2011, at 12:58 AM, Sam wrote:
>
> > in show codec i see ;
> >
> > show codecs
> > type,name,ikey
> > codec,AMR,mod_amr
> > codec,G.711 alaw,CORE_PCM_MODULE
> > codec,G.711 ulaw,CORE_PCM_MODULE
> > codec,G.723.1 6.3k,mod_g723_1
> > codec,G.729,mod_g729
> > codec,H.261 Video (passthru),mod_h26x
> > codec,H.263 Video (passthru),mod_h26x
> > codec,H.263+ Video (passthru),mod_h26x
> > codec,H.263++ Video (passthru),mod_h26x
> > codec,H.264 Video (passthru),mod_h26x
> > codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
> > codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
> > codec,Polycom(R) G722.1/G722.1C,mod_siren
> > codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
> > codec,Speex,mod_speex
> > codec,iLBC,mod_ilbc
> >
> > i have this in vars.xml
> >
> > <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h
> ,G7221 at 16000h,G7222,PCMU,PCMA,G729"/>
> > <X-PRE-PROCESS cmd="set"
> data="outbound_codec_prefs=G722,G7222,G7221 at 32000h,G7221 at 16000h
> ,PCMU,PCMA,G729"/>
> >
> >
> >
> > Regds
> > Sam
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From marvin.n.dillon at gmail.com  Mon Jan 10 09:29:13 2011
From: marvin.n.dillon at gmail.com (Marvin Dillon)
Date: Mon, 10 Jan 2011 01:29:13 -0500
Subject: [Freeswitch-users] Unable to successfully configure icall gateway
	and route inbound DID
Message-ID: <AANLkTimahoPc=gxMD8NR+=y9jHKSfdz0Bbc_BUuLSLkp@mail.gmail.com>

Hello Team,

I am a rookie running Freeswitch 1.0.6 on Debian Lenny and need some urgent
help. I have been facing a challenge getting my icall gateways configured
and being able to route my inbound DID back to my Freeswitch platform. My
sofia status output is this right now:

 sofia status
                     Name
Type                                       Data      State
=================================================================================================
                 internal       profile
sip:mod_sofia at 192.168.1.100:5060      RUNNING (0)
                 external       profile
sip:mod_sofia at 192.168.1.100:5080      RUNNING (0)
    external::example.com       gateway
sip:joeuser at example.com <sip%3Ajoeuser at example.com>      NOREG
external::icall_international   gateway
sip:cust_mdillon at gw01-car.dal.us.icall.net<sip%3Acust_mdillon at gw01-car.dal.us.icall.net>
REGED
 external::icall_outbound       gateway
sip:cust_mdillon at sbc01-car.dal.us.icall.net<sip%3Acust_mdillon at sbc01-car.dal.us.icall.net>
FAIL_WAIT
  external::icall_inbound       gateway
sip:cust_mdillon at 72.249.14.242 <sip%3Acust_mdillon at 72.249.14.242>      REGED
      external::icall.com       gateway
sip:cust_mdillon at 72.249.14.242 <sip%3Acust_mdillon at 72.249.14.242>      REGED
           208.124.220.35         alias
internal      ALIASED
            internal-ipv6       profile
sip:mod_sofia@[::1]:5060
RUNNING (0)
=================================================================================================
3 profiles 1 alias
but I have no clue why I am getting a busy tone whenever I call my inbound
DID as the sofia output indicates my inbound gateway is registered. Can
someone please help me with this.

Thanks,
MD
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From ayhkor at gmail.com  Mon Jan 10 07:05:16 2011
From: ayhkor at gmail.com (deniro)
Date: Sun, 9 Jan 2011 23:05:16 -0500
Subject: [Freeswitch-users] conference pin
Message-ID: <AANLkTi=q5xpnW7HdGVZYpQbUNi2m=dvmFXX6TTKGvxZo@mail.gmail.com>

Hi
using conferencing software and with the phone dialing,
entering pin number it will go to a conference  identified by pin
in its default format  it is "conference at profile+pin"
in my case it will be "pin at profile+pin" since conference=pin.
how do I do that? how do I provide a pin that takes me to conference which
is
identified by pin?
thx
deniro--
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From daniel-listas at gmx.net  Mon Jan 10 03:10:19 2011
From: daniel-listas at gmx.net (Daniel Bareiro)
Date: Sun, 9 Jan 2011 21:10:19 -0300
Subject: [Freeswitch-users] Prefix for outgoing calls
Message-ID: <20110110001019.GA13793@defiant.freesoftware>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi, Brian.

> > Then I tried to make any call from the predefined extensions of
> > FreeSwitch (1000.xml) to a number of iptel. Then calling from
> > extension 1000 to, for example, music at iptel.org, I was able to
> > communicate from a softphone. The reason why the call is
> > successfully using @iptel.org is because it is taking the "realm" in
> > the file iptel.org.xml?

> Chances are you bypassed FS when you dialed music at iptel.org

Hmmm... but this softphone was registered only against FS. So, how can
it be that I can call if not through FS?

> > I have been reading the FS documentation but I not found a way to
> > use a prefix to call a number of iptel (or some other provider)
> > without having to use @provider. I have no clear the syntax I should
> > use or where should I do this configuration.

> What do you mean?

What I mean is that if, for example, I want to call the extension 112233
of iptel.org, when dialing 9112233, the call to 112233 is routed through
iptel.org.

>   <extension name="local.example.com">
>     <condition field="destination_number" expression="^5551212$">
>       <action application="bridge" data="sofia/gateway/iptel.org/music"/>
>     </condition>
>   </extension>

> That would take 5551212 from a softphone and dial music at iptel.org
> ..  NEVER put @iptel.org at the end of that or you'll ened up calling
> music at iptel.org@iptel.org :P

Anyway I think your example helped me to understand some things. When
reading the file of the extension 1000
(/usr/local/freeswitch/conf/directory/default/1000.xml), I see that it
contains something like the following:


<variable name="toll_allow" value="domestic,international,local"/>


I guess these are the contexts to which the extension can call. From
what I see, these settings are defined in the file:


/usr/local/freeswitch/conf/dialplan/default/01_example.com.xml


In this file I have commented on the block having the local.example.com
and I put the example you sent me for testing. After restarting FS, I
tried to call from the softphone registered in FS to 5551212, but I get:

Line 1: call failed.
484 Address Incomplete


What could be the problem?

Thanks for your reply.

Regards,
Daniel
- -- 
Daniel Bareiro - GNU/Linux registered user #188.598
Proudly running Debian GNU/Linux with uptime:
20:25:40 up 88 days, 20:52, 10 users,  load average: 0.01, 0.05, 0.07
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From daniel-listas at gmx.net  Mon Jan 10 03:58:28 2011
From: daniel-listas at gmx.net (Daniel Guillermo Bareiro)
Date: Mon, 10 Jan 2011 01:58:28 +0100
Subject: [Freeswitch-users] Prefix for outgoing calls
Message-ID: <20110110005828.119470@gmx.net>

Hi, Brian.

> > Then I tried to make any call from the predefined extensions of
> > FreeSwitch (1000.xml) to a number of iptel. Then calling from
> > extension 1000 to, for example, music at iptel.org, I was able to
> > communicate from a softphone. The reason why the call is
> > successfully using @iptel.org is because it is taking the "realm" in
> > the file iptel.org.xml?

> Chances are you bypassed FS when you dialed music at iptel.org

Hmmm... but this softphone was registered only against FS. So, how can
it be that I can call if not through FS?

Hmmm... but this softphone was registered only against FS. So, how can
it be that I can call if not through FS?

> > I have been reading the FS documentation but I not found a way to
> > use a prefix to call a number of iptel (or some other provider)
> > without having to use @provider. I have no clear the syntax I should
> > use or where should I do this configuration.

> What do you mean?

What I mean is that if, for example, I want to call the extension 112233
of iptel.org, when dialing 9112233, the call to 112233 is routed through
iptel.org.

>   <extension name="local.example.com">
>     <condition field="destination_number" expression="^5551212$">
>       <action application="bridge" data="sofia/gateway/iptel.org/music"/>
>     </condition>
>   </extension>

> That would take 5551212 from a softphone and dial music at iptel.org
> ..  NEVER put @iptel.org at the end of that or you'll ened up calling
> music at iptel.org@iptel.org :P

Anyway I think your example helped me to understand some things. When
reading the file of the extension 1000
(/usr/local/freeswitch/conf/directory/default/1000.xml), I see that it
contains something like the following:

<variable name="toll_allow" value="domestic,international,local"/>


I guess these are the contexts to which the extension can call. From
what I see, these settings are defined in the file:


/usr/local/freeswitch/conf/dialplan/default/01_example.com.xml


In this file I have commented on the block having the local.example.com
and I put the example you sent me for testing. After restarting FS, I
tried to call from the softphone registered in FS to 5551212, but I get:

Line 1: call failed.
484 Address Incomplete


What could be the problem?

Thanks for your reply.

Regards,
Daniel
-- 
GMX DSL Doppel-Flat ab 19,99 Euro/mtl.! Jetzt mit 
gratis Handy-Flat! http://portal.gmx.net/de/go/dsl


From hadyn_whx at hotmail.com  Mon Jan 10 06:33:55 2011
From: hadyn_whx at hotmail.com (Alex Wang)
Date: Sun, 9 Jan 2011 22:33:55 -0500
Subject: [Freeswitch-users] auto-nat is not working
Message-ID: <BLU0-SMTP1648BEF319A9AA8367FF99D8D0E0@phx.gbl>

Hi

I am very new about the freeswitch and just finished setup a new
freeswitch and config a test sip account from my voip provider.

in the fs_cli,
originate sofia/gateway/gw1/xxxxxx(my other number)  &echo()
I got the phone ring but no echo.
originate sofia/user/1000 &echo() works fine.

nat_map status shows;
freeswitch at internal> nat_map status
Nat Type: UNKNOWN, ExtIP: 

0 total.

freeswitch at internal> sofia status profile internal
=================================================================================================
Name                    internal
Domain Name             N/A
Auto-NAT                false
DBName                  sofia_reg_internal

Just wonder how to fix the NAT issue on my freeswitch.

The router is WRT54GL with Tomato 1.28 on it, which seems support upnp...

Thanks a lot

Alex




From hwnorman at hotmail.com  Mon Jan 10 08:23:03 2011
From: hwnorman at hotmail.com (Norman Lam)
Date: Mon, 10 Jan 2011 13:23:03 +0800
Subject: [Freeswitch-users] Iksemel msvs compiling
In-Reply-To: <1294535404901-5903534.post@n2.nabble.com>
References: <SNT109-DS13313A1FA906527438E8CC1090@phx.gbl>	<SNT109-DS1155D401A0B9742CF63E13C10B0@phx.gbl>	<1294375438447-5898181.post@n2.nabble.com>	<SNT109-DS11CB0ECE5BD0F9C0384E03C10C0@phx.gbl>
	<1294535404901-5903534.post@n2.nabble.com>
Message-ID: <SNT109-DS14B90C48B33126784F294AC10E0@phx.gbl>

Hi Jeff 

I am still getting the error on the iksemel build 
:
:
28>------ Build started: Project: iksemel, Configuration: Debug Win32 ------
26>eng_all.c
28>Compiling...
28>dom.c
28>..\..\iksemel\src\dom.c(152) : error C2065: 'ENOENT' : undeclared
identifier
28>filter.c
26>cms_smime.c
28>..\..\iksemel\src\filter.c(59) : error C2059: syntax error : 'type'
28>..\..\iksemel\src\filter.c(64) : error C2059: syntax error : 'type'
28>..\..\iksemel\src\filter.c(67) : error C2059: syntax error : 'type'
28>..\..\iksemel\src\filter.c(70) : error C2059: syntax error : 'type'
28>..\..\iksemel\src\filter.c(73) : error C2059: syntax error : 'type'
28>..\..\iksemel\src\filter.c(76) : error C2059: syntax error : 'type'
28>..\..\iksemel\src\filter.c(79) : error C2059: syntax error : 'type'
28>iks.c
28>ikstack.c
28>io-posix.c
26>cms_sd.c
28>jabber.c
28>md5.c
26>cms_lib.c
28>sax.c
28>sha.c
28>stream.c
26>Generating Code...
28>..\..\iksemel\src\stream.c(19) : fatal error C1083: Cannot open include
file: 'gnutls/gnutls.h': No such file or directory
28>utility.c
26>Compiling...
26>cms_io.c
28>base64.c
28>Generating Code...
26>cms_ess.c
28>Build log was saved at
"file://c:\FS_GIT2\libs\win32\iksemel\Debug\BuildLog.htm"
28>iksemel - 9 error(s), 0 warning(s)
:
:
125>------ Build started: Project: mod_dingaling, Configuration: Debug Win32
------
124>Linking...
125>Compiling...
125>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler
125>mod_dingaling.c
124>   Creating library Win32\Debug/mod_commands.2008.lib and object
Win32\Debug/mod_commands.2008.exp
124>Embedding manifest...
124>Build log was saved at
"file://c:\FS_GIT2\src\mod\applications\mod_commands\Win32\Debug\BuildLog.ht
m"
124>mod_commands - 0 error(s), 1 warning(s)
126>------ Build started: Project: mod_ilbc, Configuration: Debug Win32
------
126>Compiling...
126>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler
126>mod_ilbc.c
125>Linking...
125>LINK : fatal error LNK1181: cannot open input file
'..\..\..\..\libs\win32\iksemel\debug\iksemel.lib'
125>Build log was saved at
"file://c:\FS_GIT2\src\mod\endpoints\mod_dingaling\Win32\Debug\BuildLog.htm"
125>mod_dingaling - 1 error(s), 1 warning(s)
:
:
Please advise

Norman Lam 

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff
Lenk
Sent: Sunday, January 09, 2011 9:10 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Iksemel msvs compiling


You had a a couple of download errors. openssl and sphinx models.

try the build again and check /libs for tarballs
openssl-1.0.0a.tar.gz
communicator_semi_6000_20080321.tar.gz
-- 
View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/Iksemel-msvs-compiling-tp58912
63p5903534.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From u2nsam at gmail.com  Mon Jan 10 10:09:47 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 10 Jan 2011 12:39:47 +0530
Subject: [Freeswitch-users] conference pin
In-Reply-To: <AANLkTi=q5xpnW7HdGVZYpQbUNi2m=dvmFXX6TTKGvxZo@mail.gmail.com>
References: <AANLkTi=q5xpnW7HdGVZYpQbUNi2m=dvmFXX6TTKGvxZo@mail.gmail.com>
Message-ID: <AANLkTin3F-TXJp_Mv-8vwGbkLmAwbi9mE2bobiB2uyQx@mail.gmail.com>

use channels variables in freeswitch.

http://wiki.freeswitch.org/wiki/Channel_Variables#Channel_Variable_Manipulation

Regds
Sam



On Mon, Jan 10, 2011 at 9:35 AM, deniro <ayhkor at gmail.com> wrote:

> Hi
> using conferencing software and with the phone dialing,
> entering pin number it will go to a conference  identified by pin
> in its default format  it is "conference at profile+pin"
> in my case it will be "pin at profile+pin" since conference=pin.
> how do I do that? how do I provide a pin that takes me to conference which
> is
> identified by pin?
> thx
> deniro--
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Mon Jan 10 10:36:01 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 10 Jan 2011 07:36:01 +0000
Subject: [Freeswitch-users] conference pin
In-Reply-To: <AANLkTi=q5xpnW7HdGVZYpQbUNi2m=dvmFXX6TTKGvxZo@mail.gmail.com>
References: <AANLkTi=q5xpnW7HdGVZYpQbUNi2m=dvmFXX6TTKGvxZo@mail.gmail.com>
Message-ID: <F7A3A716-9B0F-4B2E-B2A0-F1CD19371D1B@gmail.com>

You can collect the pin via ivr prompts and store it in a channel variable using play_and_g

Steve on iPhone

On 10 Jan 2011, at 04:05, deniro <ayhkor at gmail.com> wrote:

> Hi
> using conferencing software and with the phone dialing, 
> entering pin number it will go to a conference  identified by pin
> in its default format  it is "conference at profile+pin"
> in my case it will be "pin at profile+pin" since conference=pin.
> how do I do that? how do I provide a pin that takes me to conference which is 
> identified by pin?
> thx
> deniro--
>  
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From steveayre at gmail.com  Mon Jan 10 10:39:43 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 10 Jan 2011 07:39:43 +0000
Subject: [Freeswitch-users] conference pin
In-Reply-To: <AANLkTi=q5xpnW7HdGVZYpQbUNi2m=dvmFXX6TTKGvxZo@mail.gmail.com>
References: <AANLkTi=q5xpnW7HdGVZYpQbUNi2m=dvmFXX6TTKGvxZo@mail.gmail.com>
Message-ID: <AANLkTikVC=ScXK6J6Zrdu4+HTUvd3GO84Kmg_xngiFhe@mail.gmail.com>

You can collect the pin via ivr prompts and store it in a channel
variable using play_and_get_digits
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits

Some IVR prompts can be downloaded from
http://files.freeswitch.org/freeswitch-sounds-en-us-callie-8000-1.0.14.tar.gz
(en/us/callie/ivr), or you can download your own.

-Steve



On 10 January 2011 04:05, deniro <ayhkor at gmail.com> wrote:
> Hi
> using conferencing software and with the phone dialing,
> entering pin number it will go to a conference? identified by pin
> in its default format? it is "conference at profile+pin"
> in my case it will be "pin at profile+pin" since conference=pin.
> how do I do that? how do I provide a pin that takes me to conference which
> is
> identified by pin?
> thx
> deniro--
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From david.ponzone at ipeva.fr  Mon Jan 10 11:33:28 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 10 Jan 2011 09:33:28 +0100
Subject: [Freeswitch-users] codec error
In-Reply-To: <AANLkTi=r2sG_JAsZ+V+Bwg-0X7rD0_n4EAO69m1NM8jP@mail.gmail.com>
References: <AANLkTi=cKTUOcN--hZm2yLXY8s-pWotek+_YA1Bxw2jw@mail.gmail.com>
	<AANLkTinLm5AP9kqoeHarqrCJGjsUq6Xo8=aOnY05HNHN@mail.gmail.com>
	<AANLkTikmHcMHfc3ydk+8b2SWEKB+sR=Awj0OVri6qdrK@mail.gmail.com>
	<AANLkTi=e4sVBpb2M5HSni-1ZXViExxkxocvmnC=jBBi_@mail.gmail.com>
	<AANLkTikGM8fQzqXaFoOd=m3LaVLae6B+O7qBPVAYmzL2@mail.gmail.com>
	<AANLkTim8Bfsr7=hX=RBJJEo4TKZqUZgn5y7yUbh9d1k1@mail.gmail.com>
	<AANLkTi=KdMjptEK5_sAWZ4DcXU1zJ9DstXnA_29rEx_k@mail.gmail.com>
	<AANLkTinSByL-0TA+VthcBeiiewcHP3ZZEsR73FoMRts4@mail.gmail.com>
	<AANLkTi=X=RObn1xHxZpBR49Xpk+X3YLkRt3c=Aejo9j7@mail.gmail.com>
	<2E41E910-1DD0-4F24-B613-96A3A5BE0AF6@freeswitch.org>
	<AANLkTi=r2sG_JAsZ+V+Bwg-0X7rD0_n4EAO69m1NM8jP@mail.gmail.com>
Message-ID: <7F832A02-FDAE-4648-8766-C95FDDB434A5@ipeva.fr>

Sam,

Brian pointed out 2 mistakes:

> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G7222,PCMU,PCMA,G729"/>

Here, you have G7222. It is G722.

And for G722, you need mod_spandsp, as he said in his last reply.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 10/01/2011 ? 06:33, Sam a ?crit :

> Which module should i load for g7222 ?
> 
> Regds
> Sam
> 
> On Fri, Jan 7, 2011 at 8:43 PM, Brian West <brian at freeswitch.org> wrote:
> For one you have G7222 ... secondly you do not have mod_spandsp loaded.
> 
> /b
> 
> On Jan 7, 2011, at 12:58 AM, Sam wrote:
> 
> > in show codec i see ;
> >
> > show codecs
> > type,name,ikey
> > codec,AMR,mod_amr
> > codec,G.711 alaw,CORE_PCM_MODULE
> > codec,G.711 ulaw,CORE_PCM_MODULE
> > codec,G.723.1 6.3k,mod_g723_1
> > codec,G.729,mod_g729
> > codec,H.261 Video (passthru),mod_h26x
> > codec,H.263 Video (passthru),mod_h26x
> > codec,H.263+ Video (passthru),mod_h26x
> > codec,H.263++ Video (passthru),mod_h26x
> > codec,H.264 Video (passthru),mod_h26x
> > codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
> > codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
> > codec,Polycom(R) G722.1/G722.1C,mod_siren
> > codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
> > codec,Speex,mod_speex
> > codec,iLBC,mod_ilbc
> >
> > i have this in vars.xml
> >
> > <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G7222,PCMU,PCMA,G729"/>
> > <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,G7222,G7221 at 32000h,G7221 at 16000h,PCMU,PCMA,G729"/>
> >
> >
> >
> > Regds
> > Sam
> >
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From u2nsam at gmail.com  Mon Jan 10 12:28:47 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 10 Jan 2011 14:58:47 +0530
Subject: [Freeswitch-users] codec error
In-Reply-To: <7F832A02-FDAE-4648-8766-C95FDDB434A5@ipeva.fr>
References: <AANLkTi=cKTUOcN--hZm2yLXY8s-pWotek+_YA1Bxw2jw@mail.gmail.com>
	<AANLkTinLm5AP9kqoeHarqrCJGjsUq6Xo8=aOnY05HNHN@mail.gmail.com>
	<AANLkTikmHcMHfc3ydk+8b2SWEKB+sR=Awj0OVri6qdrK@mail.gmail.com>
	<AANLkTi=e4sVBpb2M5HSni-1ZXViExxkxocvmnC=jBBi_@mail.gmail.com>
	<AANLkTikGM8fQzqXaFoOd=m3LaVLae6B+O7qBPVAYmzL2@mail.gmail.com>
	<AANLkTim8Bfsr7=hX=RBJJEo4TKZqUZgn5y7yUbh9d1k1@mail.gmail.com>
	<AANLkTi=KdMjptEK5_sAWZ4DcXU1zJ9DstXnA_29rEx_k@mail.gmail.com>
	<AANLkTinSByL-0TA+VthcBeiiewcHP3ZZEsR73FoMRts4@mail.gmail.com>
	<AANLkTi=X=RObn1xHxZpBR49Xpk+X3YLkRt3c=Aejo9j7@mail.gmail.com>
	<2E41E910-1DD0-4F24-B613-96A3A5BE0AF6@freeswitch.org>
	<AANLkTi=r2sG_JAsZ+V+Bwg-0X7rD0_n4EAO69m1NM8jP@mail.gmail.com>
	<7F832A02-FDAE-4648-8766-C95FDDB434A5@ipeva.fr>
Message-ID: <AANLkTinTnwQxhK4pDpdGcejkrJ8qXg3OfRvJndAYkrwL@mail.gmail.com>

I have now ,

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7222,G7221 at 32000h
,G7221 at 16000h,G722,PCMU,PCMA,G729"/>

The G722 started working, but i wanted to know for G722.2 which module
works?

Regds
Sam


On Mon, Jan 10, 2011 at 2:03 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Sam,
>
> Brian pointed out 2 mistakes:
>
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h
> ,G7221 at 16000h,G7222,PCMU,PCMA,G729"/>
>
>
> Here, you have G7222. It is G722.
>
> And for G722, you need mod_spandsp, as he said in his last reply.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 10/01/2011 ? 06:33, Sam a ?crit :
>
> Which module should i load for g7222 ?
>
> Regds
> Sam
>
> On Fri, Jan 7, 2011 at 8:43 PM, Brian West <brian at freeswitch.org> wrote:
>
>> For one you have G7222 ... secondly you do not have mod_spandsp loaded.
>>
>> /b
>>
>> On Jan 7, 2011, at 12:58 AM, Sam wrote:
>>
>> > in show codec i see ;
>> >
>> > show codecs
>> > type,name,ikey
>> > codec,AMR,mod_amr
>> > codec,G.711 alaw,CORE_PCM_MODULE
>> > codec,G.711 ulaw,CORE_PCM_MODULE
>> > codec,G.723.1 6.3k,mod_g723_1
>> > codec,G.729,mod_g729
>> > codec,H.261 Video (passthru),mod_h26x
>> > codec,H.263 Video (passthru),mod_h26x
>> > codec,H.263+ Video (passthru),mod_h26x
>> > codec,H.263++ Video (passthru),mod_h26x
>> > codec,H.264 Video (passthru),mod_h26x
>> > codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
>> > codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
>> > codec,Polycom(R) G722.1/G722.1C,mod_siren
>> > codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
>> > codec,Speex,mod_speex
>> > codec,iLBC,mod_ilbc
>> >
>> > i have this in vars.xml
>> >
>> > <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h
>> ,G7221 at 16000h,G7222,PCMU,PCMA,G729"/>
>> > <X-PRE-PROCESS cmd="set"
>> data="outbound_codec_prefs=G722,G7222,G7221 at 32000h,G7221 at 16000h
>> ,PCMU,PCMA,G729"/>
>> >
>> >
>> >
>> > Regds
>> > Sam
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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>
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From tayeb.meftah at gmail.com  Mon Jan 10 12:29:49 2011
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Mon, 10 Jan 2011 10:29:49 +0100
Subject: [Freeswitch-users] Status of ODBC
In-Reply-To: <AANLkTimXPAh7Jfp_dzEQMkC32t2KY-DEDUJ65=E2MS8Z@mail.gmail.com>
References: <AANLkTimXPAh7Jfp_dzEQMkC32t2KY-DEDUJ65=E2MS8Z@mail.gmail.com>
Message-ID: <4D2AD18D.905@gmail.com>

the spydermonkey ODBC is not module but is a submodule of spydermonkey 
module
you just need to have UnixODBC developmant headers to by able to compile 
it, the configure script will auto manage this
after, uncommant it in spydermonkey.conf.xml
thanks
Le 10/01/2011 02:51, Max Bridgewater a ?crit :
> Hi,
>
> The following page seems to be outdated:
> http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc. I can't find
> languages/mod_spidermonkey_odbc in modules.conf. Can somebody points
> me to the right direction?
>
> Thanks,
> max.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313



From tayeb.meftah at gmail.com  Mon Jan 10 12:32:12 2011
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Mon, 10 Jan 2011 10:32:12 +0100
Subject: [Freeswitch-users] BV32 iLBC
In-Reply-To: <486F5962296C4CC5A5221088DA20A449@e1705>
References: <486F5962296C4CC5A5221088DA20A449@e1705>
Message-ID: <4D2AD21C.8060309@gmail.com>

if default config you don't need anything except of your codecs in 
vars.xml and your codec modules loaded
load mod_bv
load mod_ilbc (loaded by default)
i just tried it.
Le 09/01/2011 21:03, Madovsky a ?crit :
> I played with all builtin codecs today
> and didn't succeed to originate a call with BV32, iLBC
> and X-lite
> is it need any special settings ?
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313

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From david.ponzone at ipeva.fr  Mon Jan 10 12:39:14 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 10 Jan 2011 10:39:14 +0100
Subject: [Freeswitch-users] codec error
In-Reply-To: <AANLkTinTnwQxhK4pDpdGcejkrJ8qXg3OfRvJndAYkrwL@mail.gmail.com>
References: <AANLkTi=cKTUOcN--hZm2yLXY8s-pWotek+_YA1Bxw2jw@mail.gmail.com>
	<AANLkTinLm5AP9kqoeHarqrCJGjsUq6Xo8=aOnY05HNHN@mail.gmail.com>
	<AANLkTikmHcMHfc3ydk+8b2SWEKB+sR=Awj0OVri6qdrK@mail.gmail.com>
	<AANLkTi=e4sVBpb2M5HSni-1ZXViExxkxocvmnC=jBBi_@mail.gmail.com>
	<AANLkTikGM8fQzqXaFoOd=m3LaVLae6B+O7qBPVAYmzL2@mail.gmail.com>
	<AANLkTim8Bfsr7=hX=RBJJEo4TKZqUZgn5y7yUbh9d1k1@mail.gmail.com>
	<AANLkTi=KdMjptEK5_sAWZ4DcXU1zJ9DstXnA_29rEx_k@mail.gmail.com>
	<AANLkTinSByL-0TA+VthcBeiiewcHP3ZZEsR73FoMRts4@mail.gmail.com>
	<AANLkTi=X=RObn1xHxZpBR49Xpk+X3YLkRt3c=Aejo9j7@mail.gmail.com>
	<2E41E910-1DD0-4F24-B613-96A3A5BE0AF6@freeswitch.org>
	<AANLkTi=r2sG_JAsZ+V+Bwg-0X7rD0_n4EAO69m1NM8jP@mail.gmail.com>
	<7F832A02-FDAE-4648-8766-C95FDDB434A5@ipeva.fr>
	<AANLkTinTnwQxhK4pDpdGcejkrJ8qXg3OfRvJndAYkrwL@mail.gmail.com>
Message-ID: <67626821-60D6-48B7-9860-487FA33C7D2F@ipeva.fr>

G722.1 is in mod_siren

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 10/01/2011 ? 10:28, Sam a ?crit :

> I have now ,
> 
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7222,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,G729"/>
> 
> The G722 started working, but i wanted to know for G722.2 which module works?
> 
> Regds
> Sam
> 
> 
> On Mon, Jan 10, 2011 at 2:03 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
> Sam,
> 
> Brian pointed out 2 mistakes:
> 
>> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G7222,PCMU,PCMA,G729"/>
> 
> Here, you have G7222. It is G722.
> 
> And for G722, you need mod_spandsp, as he said in his last reply.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 10/01/2011 ? 06:33, Sam a ?crit :
> 
>> Which module should i load for g7222 ?
>> 
>> Regds
>> Sam
>> 
>> On Fri, Jan 7, 2011 at 8:43 PM, Brian West <brian at freeswitch.org> wrote:
>> For one you have G7222 ... secondly you do not have mod_spandsp loaded.
>> 
>> /b
>> 
>> On Jan 7, 2011, at 12:58 AM, Sam wrote:
>> 
>> > in show codec i see ;
>> >
>> > show codecs
>> > type,name,ikey
>> > codec,AMR,mod_amr
>> > codec,G.711 alaw,CORE_PCM_MODULE
>> > codec,G.711 ulaw,CORE_PCM_MODULE
>> > codec,G.723.1 6.3k,mod_g723_1
>> > codec,G.729,mod_g729
>> > codec,H.261 Video (passthru),mod_h26x
>> > codec,H.263 Video (passthru),mod_h26x
>> > codec,H.263+ Video (passthru),mod_h26x
>> > codec,H.263++ Video (passthru),mod_h26x
>> > codec,H.264 Video (passthru),mod_h26x
>> > codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
>> > codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
>> > codec,Polycom(R) G722.1/G722.1C,mod_siren
>> > codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
>> > codec,Speex,mod_speex
>> > codec,iLBC,mod_ilbc
>> >
>> > i have this in vars.xml
>> >
>> > <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G7222,PCMU,PCMA,G729"/>
>> > <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,G7222,G7221 at 32000h,G7221 at 16000h,PCMU,PCMA,G729"/>
>> >
>> >
>> >
>> > Regds
>> > Sam
>> >
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From u2nsam at gmail.com  Mon Jan 10 12:45:17 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 10 Jan 2011 15:15:17 +0530
Subject: [Freeswitch-users] conversion to mp3
Message-ID: <AANLkTinnMcrMi5KWuvCnx7dTqmtOEuoq_1uPFJ_3Hfx9@mail.gmail.com>

Hello,

How to i convert the recording on the fly to mp3 ?

<param name="auto-record"
value="$${recordings_dir}/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>


Regards
Sam
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From grsingh750 at gmail.com  Mon Jan 10 12:47:30 2011
From: grsingh750 at gmail.com (guru singh)
Date: Mon, 10 Jan 2011 15:17:30 +0530
Subject: [Freeswitch-users] Manipulate User Variables?
In-Reply-To: <AANLkTi=wRPrRGhSifWmUhfs3YPeCsSnUn+BScn0B6oRS@mail.gmail.com>
References: <AANLkTinkSNaw4Y6diYvmPzjD_y3E7c8G_4yekdGJ8FYh@mail.gmail.com>
	<AANLkTi=wRPrRGhSifWmUhfs3YPeCsSnUn+BScn0B6oRS@mail.gmail.com>
Message-ID: <AANLkTim+cSSkAUDGPuT3rmQ8HEWuci9HHweChPxn6cYz@mail.gmail.com>

Avi, Thanks for the reply,

I'd also thought of mod_db, sounds like the best, least convoluted
way to do it.
Another thing I tried was using set_global to initially set the var,
and then test it using global_getvar, this works as expected for users
who've dialed an extension to set the var, but for users who haven't
the var remains UNDEF and my test fails, is it possible to check if a
var is (some value| UNDEF)?

thanks

On Mon, Jan 10, 2011 at 6:56 AM, Avi Marcus <Avi at amarcus.com> wrote:
> Not that I'm aware of.
> To duplicate such functionality, however, there are several options:
> 1) Use the db to store such variables,
> see:?http://wiki.freeswitch.org/wiki/Mod_db
> 2) use a complicated script that rewrites the xml and triggers a reload xml
> (kind of convoluted, I don't?recommended?it)
> 3) Use your own sql interface, e.g. using lua or mod_odbc_query from the git
> contrib. However, familiarizing yourself with mod_db is probably enough.
> 4) Use mod_xml_curl to process what happens and run the sql query / or file
> edits within that ruby/php/etc script.
> I use mod_xml_curl for basically all my dialplan, if you have anything very
> specific (e.g. dependent on lots of queries, or custom routing and billing)
> it's the most flexible. Static XML is only so flexible.
> -Avi
>
> On Mon, Jan 10, 2011 at 3:06 AM, guru singh <grsingh750 at gmail.com> wrote:
>>
>> Hi,
>>
>> Is it possible to manipulate variables defined for users in
>> /directory/default/1000.xml?
>> Suppose I have a <variable name="flag" value="0"> for each user. Can I
>> dial an extension and change this value, only for the specific user
>> who dialed it?
>>
>> Thanks
>> gs
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From Avi at aMarcus.com  Mon Jan 10 12:52:02 2011
From: Avi at aMarcus.com (Avi Marcus)
Date: Mon, 10 Jan 2011 11:52:02 +0200
Subject: [Freeswitch-users] conversion to mp3
In-Reply-To: <AANLkTinnMcrMi5KWuvCnx7dTqmtOEuoq_1uPFJ_3Hfx9@mail.gmail.com>
References: <AANLkTinnMcrMi5KWuvCnx7dTqmtOEuoq_1uPFJ_3Hfx9@mail.gmail.com>
Message-ID: <AANLkTim4+NFRUbE87G9DagaK42B0UD+uxh5_3xkD0W6X@mail.gmail.com>

Set a hangup hook to run a script to do that for you.
-Avi
On Jan 10, 2011 11:49 AM, "Sam" <u2nsam at gmail.com> wrote:
> Hello,
>
> How to i convert the recording on the fly to mp3 ?
>
> <param name="auto-record"
>
value="$${recordings_dir}/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>
>
> Regards
> Sam
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From bernhard.suttner at winet.ch  Mon Jan 10 12:55:30 2011
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Mon, 10 Jan 2011 10:55:30 +0100
Subject: [Freeswitch-users] different mail-from/subject within voicemail
In-Reply-To: <AANLkTi=C=GygGBoUcssMp6+ZQfUZ8RO-gw9m-4DODKwG@mail.gmail.com>
References: <20110107214634.144291f0@mail.winet.ch>
	<AANLkTi=C=GygGBoUcssMp6+ZQfUZ8RO-gw9m-4DODKwG@mail.gmail.com>
Message-ID: <88026121-a6e2-4276-8a14-fd036d90e711@winet.ch>

Tested patch is attached at ticket FS-2972 within jira.

BR,
Bernhard

-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Steven Ayre
Gesendet: Freitag, 7. Januar 2011 22:01
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] different mail-from/subject within voicemail

I don't believe it's possible per-user, although you can create
multiple profiles.

I don't see any reason it couldn't be added as a user variable/param
though, similar to the current vm-mailto. vm-mailfrom?

-Steve



On 7 January 2011 20:46, Bernhard Suttner <bernhard.suttner at winet.ch> wrote:
> Hi,
>
> seems like nobody knows a possible way to do that or its not possible. I wrote a small patch for mod_voicemail, which I will send over jira on Monday/Tuesday. Maybe someone will use it.
>
> BR,
> Bernhard
>
> ----- Original Message -----
> From: Bernhard Suttner [mailto:bernhard.suttner at winet.ch]
> To: FreeSWITCH-users at lists.freeswitch.org
> Sent: Thu, 06 Jan 2011 14:07:15 +0100
> Subject: [Freeswitch-users] different mail-from/subject within voicemail
>
>
>> Hi,
>>
>> is it somehow possible to specify the "email-from" address and subject used
>> within the voicemail application within the directory (per user different)?
>>
>> Best regards,
>> Bernhard
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From a.afzali2003 at gmail.com  Mon Jan 10 13:04:47 2011
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Mon, 10 Jan 2011 13:34:47 +0330
Subject: [Freeswitch-users] conversion to mp3
In-Reply-To: <AANLkTim4+NFRUbE87G9DagaK42B0UD+uxh5_3xkD0W6X@mail.gmail.com>
References: <AANLkTinnMcrMi5KWuvCnx7dTqmtOEuoq_1uPFJ_3Hfx9@mail.gmail.com>
	<AANLkTim4+NFRUbE87G9DagaK42B0UD+uxh5_3xkD0W6X@mail.gmail.com>
Message-ID: <AANLkTi=ycWifr5Hi2KPQmzd2BZccr_-i_hdCp+W4p=4J@mail.gmail.com>

I use LAME .

-- afshin

On Mon, Jan 10, 2011 at 1:22 PM, Avi Marcus <Avi at amarcus.com> wrote:

> Set a hangup hook to run a script to do that for you.
> -Avi
> On Jan 10, 2011 11:49 AM, "Sam" <u2nsam at gmail.com> wrote:
> > Hello,
> >
> > How to i convert the recording on the fly to mp3 ?
> >
> > <param name="auto-record"
> >
> value="$${recordings_dir}/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
> >
> >
> > Regards
> > Sam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ovvenkatesan at gmail.com  Mon Jan 10 13:14:20 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Mon, 10 Jan 2011 15:44:20 +0530
Subject: [Freeswitch-users] Vestec Speech Engine: ASR 2.1 Release
In-Reply-To: <7CB536AC-D130-464E-A516-E6935717337B@freeswitch.org>
References: <4D27665F.4000406@vestec.com>
	<7CB536AC-D130-464E-A516-E6935717337B@freeswitch.org>
Message-ID: <AANLkTi=myoT__u6=6c-xzw=PRTe9YqcrS33cr6jg37c6@mail.gmail.com>

Will it support , Indian languages ;)

On Sat, Jan 8, 2011 at 1:09 AM, Brian West <brian at freeswitch.org> wrote:

> FreeSWITCHers,
>
>        I want to hold a contest to see who can build the neatest app using
> Vestec with FreeSWITCH.  The prize is a free pass to ClueCon 2011, If anyone
> wishes to enter please email me and Kashif.
>
> The only rules are it has to use FreeSWITCH with the Vestec engine.
>
> Thanks,
> Brian
>
> On Jan 7, 2011, at 1:15 PM, Kashif Kahn wrote:
>
> > Dear All,
> >
> > We have launched a major upgrade to our ASR engine that offers the best
> > deal around for enabling speech recognition with "command and control"
> > type IVR applications. The new architecture boasts a number of
> > advancements over version 1.1, including:
> >
> > - Improved US English acoustic model
> > - DTMF recognition
> > - MRCP support v1 and v2
> > - Highly scalable architecture and Redundancy
> > - C++ API availability
> > - SRGS-XML (.grxml) grammar support
> > - Improved logging
> > - Bug Fixes
> >
> > The engine can be integrated with different contact center and soft-PBX
> > platforms (such as Freeswitch) using MRCP interface.
> >
> > A starter kit comprising a specially priced full-function engine is
> > available for $25 while a regular one channel (ie. port) license can be
> > purchased for $99. Please visit Vestec webstore: http://www.vestec.com/
> >
> > Regards,
> > -Kashif
> >
> > --
> > Kashif Kahn
> > VP Business Development
> > Vestec Inc
> > Waterloo, ON Canada
> > phone: +1 519 885-7615
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
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From u2nsam at gmail.com  Mon Jan 10 13:16:36 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 10 Jan 2011 15:46:36 +0530
Subject: [Freeswitch-users] conversion to mp3
In-Reply-To: <AANLkTi=ycWifr5Hi2KPQmzd2BZccr_-i_hdCp+W4p=4J@mail.gmail.com>
References: <AANLkTinnMcrMi5KWuvCnx7dTqmtOEuoq_1uPFJ_3Hfx9@mail.gmail.com>
	<AANLkTim4+NFRUbE87G9DagaK42B0UD+uxh5_3xkD0W6X@mail.gmail.com>
	<AANLkTi=ycWifr5Hi2KPQmzd2BZccr_-i_hdCp+W4p=4J@mail.gmail.com>
Message-ID: <AANLkTi=+=8Xsb9YsJvUt_sQ0064CrVqauzDQ6SXRLbZ-@mail.gmail.com>

I used to use lame on asterisk .. how to use it in dialplan on FS ?

Regds
Sam

On Mon, Jan 10, 2011 at 3:34 PM, afshin afzali <a.afzali2003 at gmail.com>wrote:

> I use LAME .
>
> -- afshin
>
> On Mon, Jan 10, 2011 at 1:22 PM, Avi Marcus <Avi at amarcus.com> wrote:
>
>> Set a hangup hook to run a script to do that for you.
>> -Avi
>> On Jan 10, 2011 11:49 AM, "Sam" <u2nsam at gmail.com> wrote:
>> > Hello,
>> >
>> > How to i convert the recording on the fly to mp3 ?
>> >
>> > <param name="auto-record"
>> >
>> value="$${recordings_dir}/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>> >
>> >
>> > Regards
>> > Sam
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Mon Jan 10 13:22:58 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 10 Jan 2011 15:52:58 +0530
Subject: [Freeswitch-users] conversion to mp3
In-Reply-To: <AANLkTi=+=8Xsb9YsJvUt_sQ0064CrVqauzDQ6SXRLbZ-@mail.gmail.com>
References: <AANLkTinnMcrMi5KWuvCnx7dTqmtOEuoq_1uPFJ_3Hfx9@mail.gmail.com>
	<AANLkTim4+NFRUbE87G9DagaK42B0UD+uxh5_3xkD0W6X@mail.gmail.com>
	<AANLkTi=ycWifr5Hi2KPQmzd2BZccr_-i_hdCp+W4p=4J@mail.gmail.com>
	<AANLkTi=+=8Xsb9YsJvUt_sQ0064CrVqauzDQ6SXRLbZ-@mail.gmail.com>
Message-ID: <AANLkTi=3ELd47nCX=3CW9BXVfM2TKoU_2eto=ko19g56@mail.gmail.com>

Is it possible to issue linux commands with the dialplan ?

Regds
Sam

On Mon, Jan 10, 2011 at 3:46 PM, Sam <u2nsam at gmail.com> wrote:

> I used to use lame on asterisk .. how to use it in dialplan on FS ?
>
> Regds
> Sam
>
>
> On Mon, Jan 10, 2011 at 3:34 PM, afshin afzali <a.afzali2003 at gmail.com>wrote:
>
>> I use LAME .
>>
>> -- afshin
>>
>> On Mon, Jan 10, 2011 at 1:22 PM, Avi Marcus <Avi at amarcus.com> wrote:
>>
>>>  Set a hangup hook to run a script to do that for you.
>>> -Avi
>>> On Jan 10, 2011 11:49 AM, "Sam" <u2nsam at gmail.com> wrote:
>>> > Hello,
>>> >
>>> > How to i convert the recording on the fly to mp3 ?
>>> >
>>> > <param name="auto-record"
>>> >
>>> value="$${recordings_dir}/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>>> >
>>> >
>>> > Regards
>>> > Sam
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From bernhard.suttner at winet.ch  Mon Jan 10 13:27:33 2011
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Mon, 10 Jan 2011 11:27:33 +0100
Subject: [Freeswitch-users] conversion to mp3
In-Reply-To: <AANLkTi=3ELd47nCX=3CW9BXVfM2TKoU_2eto=ko19g56@mail.gmail.com>
References: <AANLkTinnMcrMi5KWuvCnx7dTqmtOEuoq_1uPFJ_3Hfx9@mail.gmail.com>
	<AANLkTim4+NFRUbE87G9DagaK42B0UD+uxh5_3xkD0W6X@mail.gmail.com>
	<AANLkTi=ycWifr5Hi2KPQmzd2BZccr_-i_hdCp+W4p=4J@mail.gmail.com>
	<AANLkTi=+=8Xsb9YsJvUt_sQ0064CrVqauzDQ6SXRLbZ-@mail.gmail.com>
	<AANLkTi=3ELd47nCX=3CW9BXVfM2TKoU_2eto=ko19g56@mail.gmail.com>
Message-ID: <add8e665-966f-4644-8e77-222b68433d24@winet.ch>

Check the command system
 
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Sam
Gesendet: Montag, 10. Januar 2011 11:23
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] conversion to mp3
 
Is it possible to issue linux commands with the dialplan ?

Regds
Sam
On Mon, Jan 10, 2011 at 3:46 PM, Sam <u2nsam at gmail.com> wrote:
I used to use lame on asterisk .. how to use it in dialplan on FS ?

Regds
Sam
 
On Mon, Jan 10, 2011 at 3:34 PM, afshin afzali <a.afzali2003 at gmail.com> wrote:
I use LAME .

-- afshin
On Mon, Jan 10, 2011 at 1:22 PM, Avi Marcus <Avi at amarcus.com> wrote:
Set a hangup hook to run a script to do that for you.
-Avi
On Jan 10, 2011 11:49 AM, "Sam" <u2nsam at gmail.com> wrote:
> Hello,
> 
> How to i convert the recording on the fly to mp3 ?
> 
> <param name="auto-record"
> value="$${recordings_dir}/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
> 
> 
> Regards
> Sam
 
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
 
 
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From steveayre at gmail.com  Mon Jan 10 14:38:48 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 10 Jan 2011 11:38:48 +0000
Subject: [Freeswitch-users] codec error
In-Reply-To: <67626821-60D6-48B7-9860-487FA33C7D2F@ipeva.fr>
References: <AANLkTi=cKTUOcN--hZm2yLXY8s-pWotek+_YA1Bxw2jw@mail.gmail.com>
	<AANLkTinLm5AP9kqoeHarqrCJGjsUq6Xo8=aOnY05HNHN@mail.gmail.com>
	<AANLkTikmHcMHfc3ydk+8b2SWEKB+sR=Awj0OVri6qdrK@mail.gmail.com>
	<AANLkTi=e4sVBpb2M5HSni-1ZXViExxkxocvmnC=jBBi_@mail.gmail.com>
	<AANLkTikGM8fQzqXaFoOd=m3LaVLae6B+O7qBPVAYmzL2@mail.gmail.com>
	<AANLkTim8Bfsr7=hX=RBJJEo4TKZqUZgn5y7yUbh9d1k1@mail.gmail.com>
	<AANLkTi=KdMjptEK5_sAWZ4DcXU1zJ9DstXnA_29rEx_k@mail.gmail.com>
	<AANLkTinSByL-0TA+VthcBeiiewcHP3ZZEsR73FoMRts4@mail.gmail.com>
	<AANLkTi=X=RObn1xHxZpBR49Xpk+X3YLkRt3c=Aejo9j7@mail.gmail.com>
	<2E41E910-1DD0-4F24-B613-96A3A5BE0AF6@freeswitch.org>
	<AANLkTi=r2sG_JAsZ+V+Bwg-0X7rD0_n4EAO69m1NM8jP@mail.gmail.com>
	<7F832A02-FDAE-4648-8766-C95FDDB434A5@ipeva.fr>
	<AANLkTinTnwQxhK4pDpdGcejkrJ8qXg3OfRvJndAYkrwL@mail.gmail.com>
	<67626821-60D6-48B7-9860-487FA33C7D2F@ipeva.fr>
Message-ID: <AANLkTinUFJyhEAhLy6z8j_gqC09d5xQoA-52HFOMBqO4@mail.gmail.com>

David,
G.722 G.722.1 and G.722.2 codecs are 3 different codecs.

Sam,
G.722 - mod_spandsp
G.722.1 - mod_siren
G.722.2 (aka AMR-WB) - AFAIK there is no FS implementation, because it
requires patent licenses

Regards,
-Steve



On 10 January 2011 09:39, David Ponzone <david.ponzone at ipeva.fr> wrote:
> G722.1 is in mod_siren
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 10/01/2011 ? 10:28, Sam a ?crit :
>
> I have now ,
>
> <X-PRE-PROCESS cmd="set"
> data="global_codec_prefs=G7222,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,G729"/>
>
> The G722 started working, but i wanted to know for G722.2 which module
> works?
>
> Regds
> Sam
>
>
> On Mon, Jan 10, 2011 at 2:03 PM, David Ponzone <david.ponzone at ipeva.fr>
> wrote:
>>
>> Sam,
>> Brian pointed out 2 mistakes:
>>
>> <X-PRE-PROCESS cmd="set"
>> data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G7222,PCMU,PCMA,G729"/>
>>
>> Here, you have G7222. It is G722.
>> And for G722, you need mod_spandsp, as he said in his last reply.
>> David Ponzone ?Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel: ? ? ?01 74 03 18 97
>> gsm: ? 06 66 98 76 34
>> Service Client?IPeva
>> tel: ? ? ?0811 46 26 26
>> www.ipeva.fr? -? ?www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>> Le 10/01/2011 ? 06:33, Sam a ?crit :
>>
>> Which module should i load for g7222 ?
>>
>> Regds
>> Sam
>>
>> On Fri, Jan 7, 2011 at 8:43 PM, Brian West <brian at freeswitch.org> wrote:
>>>
>>> For one you have G7222 ... secondly you do not have mod_spandsp loaded.
>>>
>>> /b
>>>
>>> On Jan 7, 2011, at 12:58 AM, Sam wrote:
>>>
>>> > in show codec i see ;
>>> >
>>> > show codecs
>>> > type,name,ikey
>>> > codec,AMR,mod_amr
>>> > codec,G.711 alaw,CORE_PCM_MODULE
>>> > codec,G.711 ulaw,CORE_PCM_MODULE
>>> > codec,G.723.1 6.3k,mod_g723_1
>>> > codec,G.729,mod_g729
>>> > codec,H.261 Video (passthru),mod_h26x
>>> > codec,H.263 Video (passthru),mod_h26x
>>> > codec,H.263+ Video (passthru),mod_h26x
>>> > codec,H.263++ Video (passthru),mod_h26x
>>> > codec,H.264 Video (passthru),mod_h26x
>>> > codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
>>> > codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
>>> > codec,Polycom(R) G722.1/G722.1C,mod_siren
>>> > codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
>>> > codec,Speex,mod_speex
>>> > codec,iLBC,mod_ilbc
>>> >
>>> > i have this in vars.xml
>>> >
>>> > <X-PRE-PROCESS cmd="set"
>>> > data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G7222,PCMU,PCMA,G729"/>
>>> > <X-PRE-PROCESS cmd="set"
>>> > data="outbound_codec_prefs=G722,G7222,G7221 at 32000h,G7221 at 16000h,PCMU,PCMA,G729"/>
>>> >
>>> >
>>> >
>>> > Regds
>>> > Sam
>>> >
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steveayre at gmail.com  Mon Jan 10 14:41:19 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 10 Jan 2011 11:41:19 +0000
Subject: [Freeswitch-users] codec error
In-Reply-To: <AANLkTinUFJyhEAhLy6z8j_gqC09d5xQoA-52HFOMBqO4@mail.gmail.com>
References: <AANLkTi=cKTUOcN--hZm2yLXY8s-pWotek+_YA1Bxw2jw@mail.gmail.com>
	<AANLkTinLm5AP9kqoeHarqrCJGjsUq6Xo8=aOnY05HNHN@mail.gmail.com>
	<AANLkTikmHcMHfc3ydk+8b2SWEKB+sR=Awj0OVri6qdrK@mail.gmail.com>
	<AANLkTi=e4sVBpb2M5HSni-1ZXViExxkxocvmnC=jBBi_@mail.gmail.com>
	<AANLkTikGM8fQzqXaFoOd=m3LaVLae6B+O7qBPVAYmzL2@mail.gmail.com>
	<AANLkTim8Bfsr7=hX=RBJJEo4TKZqUZgn5y7yUbh9d1k1@mail.gmail.com>
	<AANLkTi=KdMjptEK5_sAWZ4DcXU1zJ9DstXnA_29rEx_k@mail.gmail.com>
	<AANLkTinSByL-0TA+VthcBeiiewcHP3ZZEsR73FoMRts4@mail.gmail.com>
	<AANLkTi=X=RObn1xHxZpBR49Xpk+X3YLkRt3c=Aejo9j7@mail.gmail.com>
	<2E41E910-1DD0-4F24-B613-96A3A5BE0AF6@freeswitch.org>
	<AANLkTi=r2sG_JAsZ+V+Bwg-0X7rD0_n4EAO69m1NM8jP@mail.gmail.com>
	<7F832A02-FDAE-4648-8766-C95FDDB434A5@ipeva.fr>
	<AANLkTinTnwQxhK4pDpdGcejkrJ8qXg3OfRvJndAYkrwL@mail.gmail.com>
	<67626821-60D6-48B7-9860-487FA33C7D2F@ipeva.fr>
	<AANLkTinUFJyhEAhLy6z8j_gqC09d5xQoA-52HFOMBqO4@mail.gmail.com>
Message-ID: <AANLkTimZBDZ2_gWQSHx9V+t=gF-8YDZz3dcpLdV-kgwf@mail.gmail.com>

> G.722.2 (aka AMR-WB) - AFAIK there is no FS implementation, because it
> requires patent licenses

Correction: there is mod_amrwb

However, because of the patent licences it is passthrough-only so it
may not suit your needs.

Regards
-Steve



On 10 January 2011 11:38, Steven Ayre <steveayre at gmail.com> wrote:
> David,
> G.722 G.722.1 and G.722.2 codecs are 3 different codecs.
>
> Sam,
> G.722 - mod_spandsp
> G.722.1 - mod_siren
> G.722.2 (aka AMR-WB) - AFAIK there is no FS implementation, because it
> requires patent licenses
>
> Regards,
> -Steve
>
>
>
> On 10 January 2011 09:39, David Ponzone <david.ponzone at ipeva.fr> wrote:
>> G722.1 is in mod_siren
>> David Ponzone ?Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel: ? ? ?01 74 03 18 97
>> gsm: ? 06 66 98 76 34
>> Service Client?IPeva
>> tel: ? ? ?0811 46 26 26
>> www.ipeva.fr? -? ?www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>> Le 10/01/2011 ? 10:28, Sam a ?crit :
>>
>> I have now ,
>>
>> <X-PRE-PROCESS cmd="set"
>> data="global_codec_prefs=G7222,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,G729"/>
>>
>> The G722 started working, but i wanted to know for G722.2 which module
>> works?
>>
>> Regds
>> Sam
>>
>>
>> On Mon, Jan 10, 2011 at 2:03 PM, David Ponzone <david.ponzone at ipeva.fr>
>> wrote:
>>>
>>> Sam,
>>> Brian pointed out 2 mistakes:
>>>
>>> <X-PRE-PROCESS cmd="set"
>>> data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G7222,PCMU,PCMA,G729"/>
>>>
>>> Here, you have G7222. It is G722.
>>> And for G722, you need mod_spandsp, as he said in his last reply.
>>> David Ponzone ?Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel: ? ? ?01 74 03 18 97
>>> gsm: ? 06 66 98 76 34
>>> Service Client?IPeva
>>> tel: ? ? ?0811 46 26 26
>>> www.ipeva.fr? -? ?www.ipeva-studio.com
>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>
>>>
>>>
>>> Le 10/01/2011 ? 06:33, Sam a ?crit :
>>>
>>> Which module should i load for g7222 ?
>>>
>>> Regds
>>> Sam
>>>
>>> On Fri, Jan 7, 2011 at 8:43 PM, Brian West <brian at freeswitch.org> wrote:
>>>>
>>>> For one you have G7222 ... secondly you do not have mod_spandsp loaded.
>>>>
>>>> /b
>>>>
>>>> On Jan 7, 2011, at 12:58 AM, Sam wrote:
>>>>
>>>> > in show codec i see ;
>>>> >
>>>> > show codecs
>>>> > type,name,ikey
>>>> > codec,AMR,mod_amr
>>>> > codec,G.711 alaw,CORE_PCM_MODULE
>>>> > codec,G.711 ulaw,CORE_PCM_MODULE
>>>> > codec,G.723.1 6.3k,mod_g723_1
>>>> > codec,G.729,mod_g729
>>>> > codec,H.261 Video (passthru),mod_h26x
>>>> > codec,H.263 Video (passthru),mod_h26x
>>>> > codec,H.263+ Video (passthru),mod_h26x
>>>> > codec,H.263++ Video (passthru),mod_h26x
>>>> > codec,H.264 Video (passthru),mod_h26x
>>>> > codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
>>>> > codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
>>>> > codec,Polycom(R) G722.1/G722.1C,mod_siren
>>>> > codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
>>>> > codec,Speex,mod_speex
>>>> > codec,iLBC,mod_ilbc
>>>> >
>>>> > i have this in vars.xml
>>>> >
>>>> > <X-PRE-PROCESS cmd="set"
>>>> > data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G7222,PCMU,PCMA,G729"/>
>>>> > <X-PRE-PROCESS cmd="set"
>>>> > data="outbound_codec_prefs=G722,G7222,G7221 at 32000h,G7221 at 16000h,PCMU,PCMA,G729"/>
>>>> >
>>>> >
>>>> >
>>>> > Regds
>>>> > Sam
>>>> >
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>


From david.ponzone at ipeva.fr  Mon Jan 10 15:03:31 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 10 Jan 2011 13:03:31 +0100
Subject: [Freeswitch-users] codec error
In-Reply-To: <AANLkTinUFJyhEAhLy6z8j_gqC09d5xQoA-52HFOMBqO4@mail.gmail.com>
References: <AANLkTi=cKTUOcN--hZm2yLXY8s-pWotek+_YA1Bxw2jw@mail.gmail.com>
	<AANLkTinLm5AP9kqoeHarqrCJGjsUq6Xo8=aOnY05HNHN@mail.gmail.com>
	<AANLkTikmHcMHfc3ydk+8b2SWEKB+sR=Awj0OVri6qdrK@mail.gmail.com>
	<AANLkTi=e4sVBpb2M5HSni-1ZXViExxkxocvmnC=jBBi_@mail.gmail.com>
	<AANLkTikGM8fQzqXaFoOd=m3LaVLae6B+O7qBPVAYmzL2@mail.gmail.com>
	<AANLkTim8Bfsr7=hX=RBJJEo4TKZqUZgn5y7yUbh9d1k1@mail.gmail.com>
	<AANLkTi=KdMjptEK5_sAWZ4DcXU1zJ9DstXnA_29rEx_k@mail.gmail.com>
	<AANLkTinSByL-0TA+VthcBeiiewcHP3ZZEsR73FoMRts4@mail.gmail.com>
	<AANLkTi=X=RObn1xHxZpBR49Xpk+X3YLkRt3c=Aejo9j7@mail.gmail.com>
	<2E41E910-1DD0-4F24-B613-96A3A5BE0AF6@freeswitch.org>
	<AANLkTi=r2sG_JAsZ+V+Bwg-0X7rD0_n4EAO69m1NM8jP@mail.gmail.com>
	<7F832A02-FDAE-4648-8766-C95FDDB434A5@ipeva.fr>
	<AANLkTinTnwQxhK4pDpdGcejkrJ8qXg3OfRvJndAYkrwL@mail.gmail.com>
	<67626821-60D6-48B7-9860-487FA33C7D2F@ipeva.fr>
	<AANLkTinUFJyhEAhLy6z8j_gqC09d5xQoA-52HFOMBqO4@mail.gmail.com>
Message-ID: <D7597764-F1E2-4955-8808-E4C6D3ABE9E2@ipeva.fr>

Sure :)

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 10/01/2011 ? 12:38, Steven Ayre a ?crit :

> David,
> G.722 G.722.1 and G.722.2 codecs are 3 different codecs.
> 
> Sam,
> G.722 - mod_spandsp
> G.722.1 - mod_siren
> G.722.2 (aka AMR-WB) - AFAIK there is no FS implementation, because it
> requires patent licenses
> 
> Regards,
> -Steve
> 
> 
> 
> On 10 January 2011 09:39, David Ponzone <david.ponzone at ipeva.fr> wrote:
>> G722.1 is in mod_siren
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> 
>> 
>> 
>> Le 10/01/2011 ? 10:28, Sam a ?crit :
>> 
>> I have now ,
>> 
>> <X-PRE-PROCESS cmd="set"
>> data="global_codec_prefs=G7222,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,G729"/>
>> 
>> The G722 started working, but i wanted to know for G722.2 which module
>> works?
>> 
>> Regds
>> Sam
>> 
>> 
>> On Mon, Jan 10, 2011 at 2:03 PM, David Ponzone <david.ponzone at ipeva.fr>
>> wrote:
>>> 
>>> Sam,
>>> Brian pointed out 2 mistakes:
>>> 
>>> <X-PRE-PROCESS cmd="set"
>>> data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G7222,PCMU,PCMA,G729"/>
>>> 
>>> Here, you have G7222. It is G722.
>>> And for G722, you need mod_spandsp, as he said in his last reply.
>>> David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>> 
>>> 
>>> 
>>> Le 10/01/2011 ? 06:33, Sam a ?crit :
>>> 
>>> Which module should i load for g7222 ?
>>> 
>>> Regds
>>> Sam
>>> 
>>> On Fri, Jan 7, 2011 at 8:43 PM, Brian West <brian at freeswitch.org> wrote:
>>>> 
>>>> For one you have G7222 ... secondly you do not have mod_spandsp loaded.
>>>> 
>>>> /b
>>>> 
>>>> On Jan 7, 2011, at 12:58 AM, Sam wrote:
>>>> 
>>>>> in show codec i see ;
>>>>> 
>>>>> show codecs
>>>>> type,name,ikey
>>>>> codec,AMR,mod_amr
>>>>> codec,G.711 alaw,CORE_PCM_MODULE
>>>>> codec,G.711 ulaw,CORE_PCM_MODULE
>>>>> codec,G.723.1 6.3k,mod_g723_1
>>>>> codec,G.729,mod_g729
>>>>> codec,H.261 Video (passthru),mod_h26x
>>>>> codec,H.263 Video (passthru),mod_h26x
>>>>> codec,H.263+ Video (passthru),mod_h26x
>>>>> codec,H.263++ Video (passthru),mod_h26x
>>>>> codec,H.264 Video (passthru),mod_h26x
>>>>> codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
>>>>> codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
>>>>> codec,Polycom(R) G722.1/G722.1C,mod_siren
>>>>> codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
>>>>> codec,Speex,mod_speex
>>>>> codec,iLBC,mod_ilbc
>>>>> 
>>>>> i have this in vars.xml
>>>>> 
>>>>> <X-PRE-PROCESS cmd="set"
>>>>> data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G7222,PCMU,PCMA,G729"/>
>>>>> <X-PRE-PROCESS cmd="set"
>>>>> data="outbound_codec_prefs=G722,G7222,G7221 at 32000h,G7221 at 16000h,PCMU,PCMA,G729"/>
>>>>> 
>>>>> 
>>>>> 
>>>>> Regds
>>>>> Sam
>>>>> 
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From u2nsam at gmail.com  Mon Jan 10 16:07:14 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 10 Jan 2011 18:37:14 +0530
Subject: [Freeswitch-users] codec error
In-Reply-To: <AANLkTimZBDZ2_gWQSHx9V+t=gF-8YDZz3dcpLdV-kgwf@mail.gmail.com>
References: <AANLkTi=cKTUOcN--hZm2yLXY8s-pWotek+_YA1Bxw2jw@mail.gmail.com>
	<AANLkTinLm5AP9kqoeHarqrCJGjsUq6Xo8=aOnY05HNHN@mail.gmail.com>
	<AANLkTikmHcMHfc3ydk+8b2SWEKB+sR=Awj0OVri6qdrK@mail.gmail.com>
	<AANLkTi=e4sVBpb2M5HSni-1ZXViExxkxocvmnC=jBBi_@mail.gmail.com>
	<AANLkTikGM8fQzqXaFoOd=m3LaVLae6B+O7qBPVAYmzL2@mail.gmail.com>
	<AANLkTim8Bfsr7=hX=RBJJEo4TKZqUZgn5y7yUbh9d1k1@mail.gmail.com>
	<AANLkTi=KdMjptEK5_sAWZ4DcXU1zJ9DstXnA_29rEx_k@mail.gmail.com>
	<AANLkTinSByL-0TA+VthcBeiiewcHP3ZZEsR73FoMRts4@mail.gmail.com>
	<AANLkTi=X=RObn1xHxZpBR49Xpk+X3YLkRt3c=Aejo9j7@mail.gmail.com>
	<2E41E910-1DD0-4F24-B613-96A3A5BE0AF6@freeswitch.org>
	<AANLkTi=r2sG_JAsZ+V+Bwg-0X7rD0_n4EAO69m1NM8jP@mail.gmail.com>
	<7F832A02-FDAE-4648-8766-C95FDDB434A5@ipeva.fr>
	<AANLkTinTnwQxhK4pDpdGcejkrJ8qXg3OfRvJndAYkrwL@mail.gmail.com>
	<67626821-60D6-48B7-9860-487FA33C7D2F@ipeva.fr>
	<AANLkTinUFJyhEAhLy6z8j_gqC09d5xQoA-52HFOMBqO4@mail.gmail.com>
	<AANLkTimZBDZ2_gWQSHx9V+t=gF-8YDZz3dcpLdV-kgwf@mail.gmail.com>
Message-ID: <AANLkTimydDsHkWYiaWddr2VVh9bBX=UwpzY38Bzgkscm@mail.gmail.com>

thanks Steve !

Regds
Sam

On Mon, Jan 10, 2011 at 5:11 PM, Steven Ayre <steveayre at gmail.com> wrote:

> > G.722.2 (aka AMR-WB) - AFAIK there is no FS implementation, because it
> > requires patent licenses
>
> Correction: there is mod_amrwb
>
> However, because of the patent licences it is passthrough-only so it
> may not suit your needs.
>
> Regards
> -Steve
>
>
>
> On 10 January 2011 11:38, Steven Ayre <steveayre at gmail.com> wrote:
> > David,
> > G.722 G.722.1 and G.722.2 codecs are 3 different codecs.
> >
> > Sam,
> > G.722 - mod_spandsp
> > G.722.1 - mod_siren
> > G.722.2 (aka AMR-WB) - AFAIK there is no FS implementation, because it
> > requires patent licenses
> >
> > Regards,
> > -Steve
> >
> >
> >
> > On 10 January 2011 09:39, David Ponzone <david.ponzone at ipeva.fr> wrote:
> >> G722.1 is in mod_siren
> >> David Ponzone  Direction Technique
> >> email: david.ponzone at ipeva.fr
> >> tel:      01 74 03 18 97
> >> gsm:   06 66 98 76 34
> >> Service Client IPeva
> >> tel:      0811 46 26 26
> >> www.ipeva.fr  -   www.ipeva-studio.com
> >> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> >> l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion
> >> non autoris?e est interdite. Tout message ?lectronique est susceptible
> >> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il
> >> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> >> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> >>
> >>
> >>
> >> Le 10/01/2011 ? 10:28, Sam a ?crit :
> >>
> >> I have now ,
> >>
> >> <X-PRE-PROCESS cmd="set"
> >> data="global_codec_prefs=G7222,G7221 at 32000h,G7221 at 16000h
> ,G722,PCMU,PCMA,G729"/>
> >>
> >> The G722 started working, but i wanted to know for G722.2 which module
> >> works?
> >>
> >> Regds
> >> Sam
> >>
> >>
> >> On Mon, Jan 10, 2011 at 2:03 PM, David Ponzone <david.ponzone at ipeva.fr>
> >> wrote:
> >>>
> >>> Sam,
> >>> Brian pointed out 2 mistakes:
> >>>
> >>> <X-PRE-PROCESS cmd="set"
> >>> data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h
> ,G7222,PCMU,PCMA,G729"/>
> >>>
> >>> Here, you have G7222. It is G722.
> >>> And for G722, you need mod_spandsp, as he said in his last reply.
> >>> David Ponzone  Direction Technique
> >>> email: david.ponzone at ipeva.fr
> >>> tel:      01 74 03 18 97
> >>> gsm:   06 66 98 76 34
> >>> Service Client IPeva
> >>> tel:      0811 46 26 26
> >>> www.ipeva.fr  -   www.ipeva-studio.com
> >>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> >>> l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion
> >>> non autoris?e est interdite. Tout message ?lectronique est susceptible
> >>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il
> >>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de
> ce
> >>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> >>>
> >>>
> >>>
> >>> Le 10/01/2011 ? 06:33, Sam a ?crit :
> >>>
> >>> Which module should i load for g7222 ?
> >>>
> >>> Regds
> >>> Sam
> >>>
> >>> On Fri, Jan 7, 2011 at 8:43 PM, Brian West <brian at freeswitch.org>
> wrote:
> >>>>
> >>>> For one you have G7222 ... secondly you do not have mod_spandsp
> loaded.
> >>>>
> >>>> /b
> >>>>
> >>>> On Jan 7, 2011, at 12:58 AM, Sam wrote:
> >>>>
> >>>> > in show codec i see ;
> >>>> >
> >>>> > show codecs
> >>>> > type,name,ikey
> >>>> > codec,AMR,mod_amr
> >>>> > codec,G.711 alaw,CORE_PCM_MODULE
> >>>> > codec,G.711 ulaw,CORE_PCM_MODULE
> >>>> > codec,G.723.1 6.3k,mod_g723_1
> >>>> > codec,G.729,mod_g729
> >>>> > codec,H.261 Video (passthru),mod_h26x
> >>>> > codec,H.263 Video (passthru),mod_h26x
> >>>> > codec,H.263+ Video (passthru),mod_h26x
> >>>> > codec,H.263++ Video (passthru),mod_h26x
> >>>> > codec,H.264 Video (passthru),mod_h26x
> >>>> > codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
> >>>> > codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
> >>>> > codec,Polycom(R) G722.1/G722.1C,mod_siren
> >>>> > codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
> >>>> > codec,Speex,mod_speex
> >>>> > codec,iLBC,mod_ilbc
> >>>> >
> >>>> > i have this in vars.xml
> >>>> >
> >>>> > <X-PRE-PROCESS cmd="set"
> >>>> > data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h
> ,G7222,PCMU,PCMA,G729"/>
> >>>> > <X-PRE-PROCESS cmd="set"
> >>>> > data="outbound_codec_prefs=G722,G7222,G7221 at 32000h,G7221 at 16000h
> ,PCMU,PCMA,G729"/>
> >>>> >
> >>>> >
> >>>> >
> >>>> > Regds
> >>>> > Sam
> >>>> >
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From grsingh750 at gmail.com  Mon Jan 10 16:52:01 2011
From: grsingh750 at gmail.com (guru singh)
Date: Mon, 10 Jan 2011 19:22:01 +0530
Subject: [Freeswitch-users] Manipulate User Variables?
In-Reply-To: <AANLkTim+cSSkAUDGPuT3rmQ8HEWuci9HHweChPxn6cYz@mail.gmail.com>
References: <AANLkTinkSNaw4Y6diYvmPzjD_y3E7c8G_4yekdGJ8FYh@mail.gmail.com>
	<AANLkTi=wRPrRGhSifWmUhfs3YPeCsSnUn+BScn0B6oRS@mail.gmail.com>
	<AANLkTim+cSSkAUDGPuT3rmQ8HEWuci9HHweChPxn6cYz@mail.gmail.com>
Message-ID: <AANLkTimrs4-CiP_xZ9BSam0Et2v8-0cL5tU9ot=MqFP5@mail.gmail.com>

Got it with help on IRC, UNDEF is just a match to empty string, simple regex.

Thanks

On Mon, Jan 10, 2011 at 3:17 PM, guru singh <grsingh750 at gmail.com> wrote:
> Avi, Thanks for the reply,
>
> I'd also thought of mod_db, sounds like the best, least convoluted
> way to do it.
> Another thing I tried was using set_global to initially set the var,
> and then test it using global_getvar, this works as expected for users
> who've dialed an extension to set the var, but for users who haven't
> the var remains UNDEF and my test fails, is it possible to check if a
> var is (some value| UNDEF)?
>
> thanks
>
> On Mon, Jan 10, 2011 at 6:56 AM, Avi Marcus <Avi at amarcus.com> wrote:
>> Not that I'm aware of.
>> To duplicate such functionality, however, there are several options:
>> 1) Use the db to store such variables,
>> see:?http://wiki.freeswitch.org/wiki/Mod_db
>> 2) use a complicated script that rewrites the xml and triggers a reload xml
>> (kind of convoluted, I don't?recommended?it)
>> 3) Use your own sql interface, e.g. using lua or mod_odbc_query from the git
>> contrib. However, familiarizing yourself with mod_db is probably enough.
>> 4) Use mod_xml_curl to process what happens and run the sql query / or file
>> edits within that ruby/php/etc script.
>> I use mod_xml_curl for basically all my dialplan, if you have anything very
>> specific (e.g. dependent on lots of queries, or custom routing and billing)
>> it's the most flexible. Static XML is only so flexible.
>> -Avi
>>
>> On Mon, Jan 10, 2011 at 3:06 AM, guru singh <grsingh750 at gmail.com> wrote:
>>>
>>> Hi,
>>>
>>> Is it possible to manipulate variables defined for users in
>>> /directory/default/1000.xml?
>>> Suppose I have a <variable name="flag" value="0"> for each user. Can I
>>> dial an extension and change this value, only for the specific user
>>> who dialed it?
>>>
>>> Thanks
>>> gs
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>


From kahn at vestec.com  Mon Jan 10 17:24:34 2011
From: kahn at vestec.com (Kashif Kahn)
Date: Mon, 10 Jan 2011 09:24:34 -0500
Subject: [Freeswitch-users] Vestec Speech Engine: ASR 2.1 Release
In-Reply-To: <AANLkTi=myoT__u6=6c-xzw=PRTe9YqcrS33cr6jg37c6@mail.gmail.com>
References: <4D27665F.4000406@vestec.com>	<7CB536AC-D130-464E-A516-E6935717337B@freeswitch.org>
	<AANLkTi=myoT__u6=6c-xzw=PRTe9YqcrS33cr6jg37c6@mail.gmail.com>
Message-ID: <4D2B16A2.3070805@vestec.com>

Support of Indian languages is very much part of Vestec ASR road-map.
-Kashif

On 1/10/2011 5:14 AM, ovvenkat wrote:
> Will it support , Indian languages ;)
>
> On Sat, Jan 8, 2011 at 1:09 AM, Brian West <brian at freeswitch.org 
> <mailto:brian at freeswitch.org>> wrote:
>
>     FreeSWITCHers,
>
>            I want to hold a contest to see who can build the neatest
>     app using Vestec with FreeSWITCH.  The prize is a free pass to
>     ClueCon 2011, If anyone wishes to enter please email me and Kashif.
>
>     The only rules are it has to use FreeSWITCH with the Vestec engine.
>
>     Thanks,
>     Brian
>
>     On Jan 7, 2011, at 1:15 PM, Kashif Kahn wrote:
>
>     > Dear All,
>     >
>     > We have launched a major upgrade to our ASR engine that offers
>     the best
>     > deal around for enabling speech recognition with "command and
>     control"
>     > type IVR applications. The new architecture boasts a number of
>     > advancements over version 1.1, including:
>     >
>     > - Improved US English acoustic model
>     > - DTMF recognition
>     > - MRCP support v1 and v2
>     > - Highly scalable architecture and Redundancy
>     > - C++ API availability
>     > - SRGS-XML (.grxml) grammar support
>     > - Improved logging
>     > - Bug Fixes
>     >
>     > The engine can be integrated with different contact center and
>     soft-PBX
>     > platforms (such as Freeswitch) using MRCP interface.
>     >
>     > A starter kit comprising a specially priced full-function engine is
>     > available for $25 while a regular one channel (ie. port) license
>     can be
>     > purchased for $99. Please visit Vestec webstore:
>     http://www.vestec.com/
>     >
>     > Regards,
>     > -Kashif
>     >
>     > --
>     > Kashif Kahn
>     > VP Business Development
>     > Vestec Inc
>     > Waterloo, ON Canada
>     > phone: +1 519 885-7615
>     >
>     >
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
>
> If you have come to help me, you are wasting your time.
> If you have come to because your liberation is bound up in mine, we 
> can work together.
>
>
> Regards
> Venkatesan OV.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From w8hdkim at gmail.com  Mon Jan 10 17:35:31 2011
From: w8hdkim at gmail.com (Kim Culhan)
Date: Mon, 10 Jan 2011 09:35:31 -0500
Subject: [Freeswitch-users] IVR menu not observing timeout value
Message-ID: <AANLkTinw+KeRM-ecF6viSh7MD+aogXW-54B-wnMRbqy2@mail.gmail.com>

FreeSWITCH Version 1.0.head (git-4e95227 2010-12-26 09-09-14 -0600)

Not possible to send a 4-digit string to the IVR, the recv string appears to
be evaluated
before the timeout value is reached:

2011-01-10 09:09:42.237843 [DEBUG] switch_rtp.c:3037 RTP RECV DTMF 6:2400
2011-01-10 09:09:42.237843 [DEBUG] switch_ivr_play_say.c:1573 done playing
file
2011-01-10 09:09:42.357031 [DEBUG] switch_ivr_menu.c:343 waiting for 3/4
digits t/o 2000
2011-01-10 09:09:43.378101 [DEBUG] switch_ivr_menu.c:390 digits '6'
2011-01-10 09:09:43.378101 [DEBUG] switch_ivr_menu.c:484 action regex [6]
[/^(1[01][0-9])$/] [0]
2011-01-10 09:09:43.378101 [DEBUG] switch_ivr_menu.c:484 action regex [6]
[/^(6[34578][01][0-9])$/] [0]
2011-01-10 09:09:43.378101 [DEBUG] switch_ivr_menu.c:574 IVR menu
'ivr_kimnet' caught invalid input '6'

Any help is greatly appreciated

-kim
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From brian at freeswitch.org  Mon Jan 10 18:14:22 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Jan 2011 09:14:22 -0600
Subject: [Freeswitch-users] Unable to successfully configure icall
	gateway and route inbound DID
In-Reply-To: <AANLkTimahoPc=gxMD8NR+=y9jHKSfdz0Bbc_BUuLSLkp@mail.gmail.com>
References: <AANLkTimahoPc=gxMD8NR+=y9jHKSfdz0Bbc_BUuLSLkp@mail.gmail.com>
Message-ID: <67907271-E9E5-4608-9D3F-41D2D4102BF1@freeswitch.org>

Um I don't think you register to the outbound server.


/b

On Jan 10, 2011, at 12:29 AM, Marvin Dillon wrote:

> Hello Team,
>  
> I am a rookie running Freeswitch 1.0.6 on Debian Lenny and need some urgent help. I have been facing a challenge getting my icall gateways configured and being able to route my inbound DID back to my Freeswitch platform. My sofia status output is this right now:
>  
>  sofia status
>                      Name          Type                                       Data      State
> =================================================================================================
>                  internal       profile           sip:mod_sofia at 192.168.1.100:5060      RUNNING (0)
>                  external       profile           sip:mod_sofia at 192.168.1.100:5080      RUNNING (0)
>     external::example.com       gateway                    sip:joeuser at example.com      NOREG
> external::icall_international   gateway   sip:cust_mdillon at gw01-car.dal.us.icall.net    REGED
>  external::icall_outbound       gateway   sip:cust_mdillon at sbc01-car.dal.us.icall.net   FAIL_WAIT
>   external::icall_inbound       gateway             sip:cust_mdillon at 72.249.14.242      REGED
>       external::icall.com       gateway             sip:cust_mdillon at 72.249.14.242      REGED
>            208.124.220.35         alias                                   internal      ALIASED
>             internal-ipv6       profile                   sip:mod_sofia@[::1]:5060      RUNNING (0)
> =================================================================================================
> 3 profiles 1 alias
> but I have no clue why I am getting a busy tone whenever I call my inbound DID as the sofia output indicates my inbound gateway is registered. Can someone please help me with this.
>  
> Thanks,
> MD
>  

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From jeff at jefflenk.com  Mon Jan 10 18:27:16 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 10 Jan 2011 07:27:16 -0800 (PST)
Subject: [Freeswitch-users] Iksemel msvs compiling
In-Reply-To: <SNT109-DS14B90C48B33126784F294AC10E0@phx.gbl>
References: <SNT109-DS13313A1FA906527438E8CC1090@phx.gbl>
	<SNT109-DS1155D401A0B9742CF63E13C10B0@phx.gbl>
	<1294375438447-5898181.post@n2.nabble.com>
	<SNT109-DS11CB0ECE5BD0F9C0384E03C10C0@phx.gbl>
	<1294535404901-5903534.post@n2.nabble.com>
	<SNT109-DS14B90C48B33126784F294AC10E0@phx.gbl>
Message-ID: <1294673236806-5907287.post@n2.nabble.com>


make sure you put the include path for ;..\..\pthreads-w32-2-7-0-release;
at the end of the list. There seems to be some include file conflicts but
this seems to take care of it.

If you are succesfull with all of this please take the time to update the
wiki with more detail.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Iksemel-msvs-compiling-tp5891263p5907287.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From bansal.rajeshkr at gmail.com  Mon Jan 10 11:09:12 2011
From: bansal.rajeshkr at gmail.com (Rajesh Bansal)
Date: Mon, 10 Jan 2011 13:39:12 +0530
Subject: [Freeswitch-users] Getting Error ORA-923 in oracle query
	execution
In-Reply-To: <AANLkTi=u9QUHh1GD3MpQTB81N5zW5xwAcWazWzuU0A6Q@mail.gmail.com>
References: <AANLkTi=k-cwDWxvCMne=O958yhXd8=bkr1EWtPAORxL6@mail.gmail.com>
	<AANLkTi=u9QUHh1GD3MpQTB81N5zW5xwAcWazWzuU0A6Q@mail.gmail.com>
Message-ID: <AANLkTin+T3yML3iZpVg0qgUFV1Vxjn0AYO9sAsNrOGHZ@mail.gmail.com>

Hi Steve,

I am using following sample code.

use("ODBC");

var dsn     = "rajesh_dsn";
var db_user = "avc";
var db_pass = "avc";
var db      = new ODBC(dsn, db_user, db_pass);
var sql = "select sysdate  from dual";

console_log("info","\nbefore connect with DB\n");

if(db.connect())
{
if(session.ready())
 session.answer();
console_log("info","\n\nafter connect with DB\n");

if (db.exec(sql))
        session.hangup(); //might want to say something nice instead.
else
        console_log("info","\n\nconnect with DB\n");
while (db.nextRow())
{
  row = db.getData();
  console_log("info", "UserName: " );
}}
else

console_log("info","\n\nunable to connect with DB\n");

Best Regards,
Rajesh Bansal



On Fri, Jan 7, 2011 at 11:35 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Sounds like a SQL syntax error - can you show us the code that's
> executing the statement that gives the error?
>
> Specifically Oracle is complaining about not finding FROM where it
> expects to in the SELECT.
>
> -Steve
>
>
> On 7 January 2011 15:15, Rajesh Bansal <bansal.rajeshkr at gmail.com> wrote:
> > Hi ,
> > I am getting error ORA -923 (from missing from statement) when i am tring
> to
> > execute a sql query from Javascript file. Here i am successfully able to
> > connect & execute queries with MYSQL. But in oracle connection i am
> getting
> > this error even i can successfully make a connection with oracle. with
> isql
> > & a program written in C i can connect and execute queries ok.
> > I am using
> > FreeSwitch 1.0.6
> > unixOdbc    2.3.0
> > Please tell me where is problem.
> > Best Regards,
> > Rajesh Bansal
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bansal.rajeshkr at gmail.com  Mon Jan 10 14:39:40 2011
From: bansal.rajeshkr at gmail.com (Rajesh Bansal)
Date: Mon, 10 Jan 2011 17:09:40 +0530
Subject: [Freeswitch-users] Getting Error ORA-923 in oracle query
	execution
In-Reply-To: <AANLkTin+T3yML3iZpVg0qgUFV1Vxjn0AYO9sAsNrOGHZ@mail.gmail.com>
References: <AANLkTi=k-cwDWxvCMne=O958yhXd8=bkr1EWtPAORxL6@mail.gmail.com>
	<AANLkTi=u9QUHh1GD3MpQTB81N5zW5xwAcWazWzuU0A6Q@mail.gmail.com>
	<AANLkTin+T3yML3iZpVg0qgUFV1Vxjn0AYO9sAsNrOGHZ@mail.gmail.com>
Message-ID: <AANLkTin2mx0K=NOx4xrM06AQX5vRV7Hu0poDGsR6AUAr@mail.gmail.com>

Hi All,

I have find reason of problem with UnixOdbc (with Oracle). When we are using
freeswitch with ODBC support before executing a sql query it run a test
query "Select 1" which works at MYSQL but didn't work at ORACLE. This change
should be in switch_odbc.c file. Please someone raise it as a minor bug so
that it can be fixed in next releases.

Best Regards,
Rajesh Bansal



On Mon, Jan 10, 2011 at 1:39 PM, Rajesh Bansal <bansal.rajeshkr at gmail.com>wrote:

> Hi Steve,
>
> I am using following sample code.
>
> use("ODBC");
>
> var dsn     = "rajesh_dsn";
> var db_user = "avc";
> var db_pass = "avc";
> var db      = new ODBC(dsn, db_user, db_pass);
> var sql = "select sysdate  from dual";
>
> console_log("info","\nbefore connect with DB\n");
>
> if(db.connect())
> {
> if(session.ready())
>  session.answer();
> console_log("info","\n\nafter connect with DB\n");
>
> if (db.exec(sql))
>         session.hangup(); //might want to say something nice instead.
> else
>         console_log("info","\n\nconnect with DB\n");
> while (db.nextRow())
> {
>   row = db.getData();
>   console_log("info", "UserName: " );
> }}
> else
>
> console_log("info","\n\nunable to connect with DB\n");
>
> Best Regards,
> Rajesh Bansal
>
>
>
>
> On Fri, Jan 7, 2011 at 11:35 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Sounds like a SQL syntax error - can you show us the code that's
>> executing the statement that gives the error?
>>
>> Specifically Oracle is complaining about not finding FROM where it
>> expects to in the SELECT.
>>
>> -Steve
>>
>>
>> On 7 January 2011 15:15, Rajesh Bansal <bansal.rajeshkr at gmail.com> wrote:
>> > Hi ,
>> > I am getting error ORA -923 (from missing from statement) when i am
>> tring to
>> > execute a sql query from Javascript file. Here i am successfully able to
>> > connect & execute queries with MYSQL. But in oracle connection i am
>> getting
>> > this error even i can successfully make a connection with oracle. with
>> isql
>> > & a program written in C i can connect and execute queries ok.
>> > I am using
>> > FreeSwitch 1.0.6
>> > unixOdbc    2.3.0
>> > Please tell me where is problem.
>> > Best Regards,
>> > Rajesh Bansal
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From pivanet at gmail.com  Mon Jan 10 12:39:23 2011
From: pivanet at gmail.com (Leonid K)
Date: Mon, 10 Jan 2011 11:39:23 +0200
Subject: [Freeswitch-users] Asterisk & FreeSwitch in recording view
Message-ID: <AANLkTinZT1WiNnae4oMKd+5YdEUO9xq4_cJr6R+4siiK@mail.gmail.com>

Hi,

currently we're looking for voip recording solution - it must record all
incoming/outgoing/internal/conference calls within the company. later on we
going to develop applications that let us fing/analyze recordings. the main
problem as I see at the moment is huge count of calls that is going through
switch.

what do u think about using Asterisk or/and FreeSwitch for this task? thanks
in advance!

-- 
Sincerely,
Leonid Kryvoruchko

Mobile   +38 093 7609175
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From hadyn_whx at hotmail.com  Mon Jan 10 17:13:03 2011
From: hadyn_whx at hotmail.com (Alex Wang)
Date: Mon, 10 Jan 2011 09:13:03 -0500
Subject: [Freeswitch-users] Any one use FreeCanadianCalls with FreeSwitch?
Message-ID: <BLU0-SMTP84FDB44AA90E9823D8E3B18D0E0@phx.gbl>

Hi 

Any one is using FreeCanadianCalls with FreeSwitch? Would you mind to
share the xml setting?

Thanks

Alex




From cjbujold at accra.ca  Mon Jan 10 19:20:29 2011
From: cjbujold at accra.ca (Charles Bujold)
Date: Mon, 10 Jan 2011 12:20:29 -0400
Subject: [Freeswitch-users] Unable to get Freeswitch to register with
 Grandstream HT503
In-Reply-To: <AANLkTik_+AemFh8QnbcYPAuL3UJmV6q_kC-eELLWYOy4@mail.gmail.com>
References: <004801cbae90$cbf44ca0$63dce5e0$@accra.ca>
	<AANLkTik_+AemFh8QnbcYPAuL3UJmV6q_kC-eELLWYOy4@mail.gmail.com>
Message-ID: <011001cbb0e2$4c156880$e4403980$@accra.ca>

Thanks for the help I finally got it to register, I did not expect the FXO
to register as an extension I was trying to register it as a gateway.  Now I
need to figure out how to set up the dialplan to route calls to the FXO
extension so I can make outgoing calls.  If you have a suggestion It would
be most appreciated.

 

Thanks 

 

\Newbie CJB

.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sam
Sent: January-07-11 11:46 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Unable to get Freeswitch to register with
Grandstream HT503

 

Is it happening only with ht 503 or all the devices , do you get the trace
logs for this ?

Regards
Sam

On Fri, Jan 7, 2011 at 11:02 PM, Charles Bujold <cjbujold at accra.ca> wrote:

Unable to get the HT-503 to register with Freeswitch.  Does anybody know
what configuration setting in the HT503 that are needed.

 

Thanks

cjb


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From w8hdkim at gmail.com  Mon Jan 10 19:15:22 2011
From: w8hdkim at gmail.com (Kim Culhan)
Date: Mon, 10 Jan 2011 11:15:22 -0500
Subject: [Freeswitch-users] IVR menu not observing timeout value
Message-ID: <AANLkTikc_YS=vCZLAh0mMZ0Eu9qP_vyCFq0oKSRORou7@mail.gmail.com>

On Mon, January 10, 2011 9:35 am, Kim Culhan wrote:
> FreeSWITCH Version 1.0.head (git-4e95227 2010-12-26 09-09-14 -0600)
Correction on the FS version, it should read:

FreeSWITCH Version 1.0.head (git-3003489 2011-01-09 14-42-42 -0500)

thanks
-kim
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From rupa at rupa.com  Mon Jan 10 20:13:16 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 10 Jan 2011 11:13:16 -0600
Subject: [Freeswitch-users] auto-nat is not working
In-Reply-To: <BLU0-SMTP1648BEF319A9AA8367FF99D8D0E0@phx.gbl>
References: <BLU0-SMTP1648BEF319A9AA8367FF99D8D0E0@phx.gbl>
Message-ID: <AANLkTikO6j50jp6hC7keHnnU13C=ND1vRH0DxvtBjWm4@mail.gmail.com>

1) Ensure you have UPNP turned on in tomato.  Double check.

2) Ensure debug logs go to your log *file* (since you want to capture
info at startup and what you need will scroll off the screen or be too
early to get in fs_cli)

3) start freeswitch

4) check the log file or msgs originating from switch_nat.c (nice that
logs show the file, eh?)

5) pastebin the info in pastebin.freeswitch.org (or just the whole log
from a freeswitch start, that might be better so you don't
accidentally remove something needed)

6) Profit. :)   Well, maybe not but I can look at 'em.

On Sun, Jan 9, 2011 at 9:33 PM, Alex Wang <hadyn_whx at hotmail.com> wrote:
> Hi
>
> I am very new about the freeswitch and just finished setup a new
> freeswitch and config a test sip account from my voip provider.
>
> in the fs_cli,
> originate sofia/gateway/gw1/xxxxxx(my other number) ?&echo()
> I got the phone ring but no echo.
> originate sofia/user/1000 &echo() works fine.
>
> nat_map status shows;
> freeswitch at internal> nat_map status
> Nat Type: UNKNOWN, ExtIP:
>
> 0 total.
>
> freeswitch at internal> sofia status profile internal
> =================================================================================================
> Name ? ? ? ? ? ? ? ? ? ?internal
> Domain Name ? ? ? ? ? ? N/A
> Auto-NAT ? ? ? ? ? ? ? ?false
> DBName ? ? ? ? ? ? ? ? ?sofia_reg_internal
>
> Just wonder how to fix the NAT issue on my freeswitch.
>
> The router is WRT54GL with Tomato 1.28 on it, which seems support upnp...
>
> Thanks a lot
>
> Alex
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa


From rupa at rupa.com  Mon Jan 10 20:16:17 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 10 Jan 2011 11:16:17 -0600
Subject: [Freeswitch-users] Asterisk & FreeSwitch in recording view
In-Reply-To: <AANLkTinZT1WiNnae4oMKd+5YdEUO9xq4_cJr6R+4siiK@mail.gmail.com>
References: <AANLkTinZT1WiNnae4oMKd+5YdEUO9xq4_cJr6R+4siiK@mail.gmail.com>
Message-ID: <AANLkTinvVqYuZKT6zHAQPW+QdqMoSaabB1WG6DLQOocO@mail.gmail.com>

I think the standard answer to that is www.orecx.com.

On Mon, Jan 10, 2011 at 3:39 AM, Leonid K <pivanet at gmail.com> wrote:
> Hi,
> currently we're looking for voip recording solution - it must record all
> incoming/outgoing/internal/conference calls within the company. later on we
> going to develop applications that let us fing/analyze recordings. the main
> problem as I see at the moment is huge count of calls that is going through
> switch.
>
> what do u think about using Asterisk or/and FreeSwitch for this task? thanks
> in advance!
> --
> Sincerely,
> Leonid Kryvoruchko
>
> Mobile?? +38 093 7609175
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From rupa at rupa.com  Mon Jan 10 20:18:37 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 10 Jan 2011 11:18:37 -0600
Subject: [Freeswitch-users] Getting Error ORA-923 in oracle query
	execution
In-Reply-To: <AANLkTin2mx0K=NOx4xrM06AQX5vRV7Hu0poDGsR6AUAr@mail.gmail.com>
References: <AANLkTi=k-cwDWxvCMne=O958yhXd8=bkr1EWtPAORxL6@mail.gmail.com>
	<AANLkTi=u9QUHh1GD3MpQTB81N5zW5xwAcWazWzuU0A6Q@mail.gmail.com>
	<AANLkTin+T3yML3iZpVg0qgUFV1Vxjn0AYO9sAsNrOGHZ@mail.gmail.com>
	<AANLkTin2mx0K=NOx4xrM06AQX5vRV7Hu0poDGsR6AUAr@mail.gmail.com>
Message-ID: <AANLkTi=Vbgyh4Yw0VuCXymOrbNpu8aFMUrk9vfR0UsQr@mail.gmail.com>

Someone should raise it as a minor bug with Oracle. :(  They are the
only modern db that I know of that requires a hack of "FROM dual" on
the SQL.

I suppose either a) switch_odbc can auto detect oracle (not easy, it
would have to carry state in the dbh I guess) or b) a setting (again,
probably on the dbh) that you can put in FS's config to tell FS not to
do the select 1.

Either one is... umm.. yuck. :(

On Mon, Jan 10, 2011 at 5:39 AM, Rajesh Bansal
<bansal.rajeshkr at gmail.com> wrote:
> Hi All,
> I have find reason of problem with UnixOdbc (with Oracle). When we are using
> freeswitch with ODBC support before executing a sql query it run a test
> query "Select 1" which works at MYSQL but didn't work at ORACLE. This change
> should be in switch_odbc.c file. Please someone raise it as a minor bug so
> that it can be fixed in next releases.
> Best Regards,
> Rajesh Bansal

>
>
> On Mon, Jan 10, 2011 at 1:39 PM, Rajesh Bansal <bansal.rajeshkr at gmail.com>
> wrote:
>>
>> Hi Steve,
>> I am using following sample code.
>> use("ODBC");
>> var dsn ? ? = "rajesh_dsn";
>> var db_user = "avc";
>> var db_pass = "avc";
>> var db ? ? ?= new ODBC(dsn, db_user, db_pass);
>> var sql = "select sysdate ?from dual";
>> console_log("info","\nbefore connect with DB\n");
>> if(db.connect())
>> {
>> if(session.ready())
>> ?session.answer();
>> console_log("info","\n\nafter connect with DB\n");
>> if (db.exec(sql))
>> ?? ? ? ?session.hangup(); //might want to say something nice instead.
>> else
>> ?? ? ? ?console_log("info","\n\nconnect with DB\n");
>> while (db.nextRow())
>> {
>> ??row = db.getData();
>> ??console_log("info", "UserName: " );
>> }}
>> else
>> console_log("info","\n\nunable to connect with DB\n");
>> Best Regards,
>> Rajesh Bansal
>>
>>
>>
>> On Fri, Jan 7, 2011 at 11:35 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>>
>>> Sounds like a SQL syntax error - can you show us the code that's
>>> executing the statement that gives the error?
>>>
>>> Specifically Oracle is complaining about not finding FROM where it
>>> expects to in the SELECT.
>>>
>>> -Steve
>>>
>>>
>>> On 7 January 2011 15:15, Rajesh Bansal <bansal.rajeshkr at gmail.com> wrote:
>>> > Hi ,
>>> > I am getting error ORA -923 (from missing from statement) when i am
>>> > tring to
>>> > execute a sql query from Javascript file. Here i am successfully able
>>> > to
>>> > connect & execute queries with MYSQL. But in oracle connection i am
>>> > getting
>>> > this error even i can successfully make a connection with oracle. with
>>> > isql
>>> > & a program written in C i can connect and execute queries ok.
>>> > I am using
>>> > FreeSwitch 1.0.6
>>> > unixOdbc ? ?2.3.0
>>> > Please tell me where is problem.
>>> > Best Regards,
>>> > Rajesh Bansal
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From steveayre at gmail.com  Mon Jan 10 20:19:04 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 10 Jan 2011 17:19:04 +0000
Subject: [Freeswitch-users] Asterisk & FreeSwitch in recording view
In-Reply-To: <AANLkTinZT1WiNnae4oMKd+5YdEUO9xq4_cJr6R+4siiK@mail.gmail.com>
References: <AANLkTinZT1WiNnae4oMKd+5YdEUO9xq4_cJr6R+4siiK@mail.gmail.com>
Message-ID: <AANLkTimR=W4mOfc_XpPqL5koak3qqf8XCKcgq_FtyiHG@mail.gmail.com>

FreeSWITCH does have the ability to record calls. You can control when
to do so from the dialplan. I assume Asterisk has the ability too.

Either way you'll have to direct the calls through FreeSWITCH/Asterisk
in order to get the media so you can record them. Personally I have
found FreeSWITCH can handle a greater number of calls more reliably,
so I'd say FreeSWITCH is the better choice.

-Steve



On 10 January 2011 09:39, Leonid K <pivanet at gmail.com> wrote:
> Hi,
> currently we're looking for voip recording solution - it must record all
> incoming/outgoing/internal/conference calls within the company. later on we
> going to develop applications that let us fing/analyze recordings. the main
> problem as I see at the moment is huge count of calls that is going through
> switch.
>
> what do u think about using Asterisk or/and FreeSwitch for this task? thanks
> in advance!
> --
> Sincerely,
> Leonid Kryvoruchko
>
> Mobile?? +38 093 7609175
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steveayre at gmail.com  Mon Jan 10 20:24:08 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 10 Jan 2011 17:24:08 +0000
Subject: [Freeswitch-users] Getting Error ORA-923 in oracle query
	execution
In-Reply-To: <AANLkTi=Vbgyh4Yw0VuCXymOrbNpu8aFMUrk9vfR0UsQr@mail.gmail.com>
References: <AANLkTi=k-cwDWxvCMne=O958yhXd8=bkr1EWtPAORxL6@mail.gmail.com>
	<AANLkTi=u9QUHh1GD3MpQTB81N5zW5xwAcWazWzuU0A6Q@mail.gmail.com>
	<AANLkTin+T3yML3iZpVg0qgUFV1Vxjn0AYO9sAsNrOGHZ@mail.gmail.com>
	<AANLkTin2mx0K=NOx4xrM06AQX5vRV7Hu0poDGsR6AUAr@mail.gmail.com>
	<AANLkTi=Vbgyh4Yw0VuCXymOrbNpu8aFMUrk9vfR0UsQr@mail.gmail.com>
Message-ID: <AANLkTimdbke-Kj4B09eaXhaOJzphB3rwa7zi9zSxJ21s@mail.gmail.com>

We already have a similar autodetect for firebird there though, so
probably no more yuck than it already is.

-Steve



On 10 January 2011 17:18, Rupa Schomaker <rupa at rupa.com> wrote:
> Someone should raise it as a minor bug with Oracle. :( ?They are the
> only modern db that I know of that requires a hack of "FROM dual" on
> the SQL.
>
> I suppose either a) switch_odbc can auto detect oracle (not easy, it
> would have to carry state in the dbh I guess) or b) a setting (again,
> probably on the dbh) that you can put in FS's config to tell FS not to
> do the select 1.
>
> Either one is... umm.. yuck. :(
>
> On Mon, Jan 10, 2011 at 5:39 AM, Rajesh Bansal
> <bansal.rajeshkr at gmail.com> wrote:
>> Hi All,
>> I have find reason of problem with UnixOdbc (with Oracle). When we are using
>> freeswitch with ODBC support before executing a sql query it run a test
>> query "Select 1" which works at MYSQL but didn't work at ORACLE. This change
>> should be in switch_odbc.c file. Please someone raise it as a minor bug so
>> that it can be fixed in next releases.
>> Best Regards,
>> Rajesh Bansal
>
>>
>>
>> On Mon, Jan 10, 2011 at 1:39 PM, Rajesh Bansal <bansal.rajeshkr at gmail.com>
>> wrote:
>>>
>>> Hi Steve,
>>> I am using following sample code.
>>> use("ODBC");
>>> var dsn ? ? = "rajesh_dsn";
>>> var db_user = "avc";
>>> var db_pass = "avc";
>>> var db ? ? ?= new ODBC(dsn, db_user, db_pass);
>>> var sql = "select sysdate ?from dual";
>>> console_log("info","\nbefore connect with DB\n");
>>> if(db.connect())
>>> {
>>> if(session.ready())
>>> ?session.answer();
>>> console_log("info","\n\nafter connect with DB\n");
>>> if (db.exec(sql))
>>> ?? ? ? ?session.hangup(); //might want to say something nice instead.
>>> else
>>> ?? ? ? ?console_log("info","\n\nconnect with DB\n");
>>> while (db.nextRow())
>>> {
>>> ??row = db.getData();
>>> ??console_log("info", "UserName: " );
>>> }}
>>> else
>>> console_log("info","\n\nunable to connect with DB\n");
>>> Best Regards,
>>> Rajesh Bansal
>>>
>>>
>>>
>>> On Fri, Jan 7, 2011 at 11:35 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>>>
>>>> Sounds like a SQL syntax error - can you show us the code that's
>>>> executing the statement that gives the error?
>>>>
>>>> Specifically Oracle is complaining about not finding FROM where it
>>>> expects to in the SELECT.
>>>>
>>>> -Steve
>>>>
>>>>
>>>> On 7 January 2011 15:15, Rajesh Bansal <bansal.rajeshkr at gmail.com> wrote:
>>>> > Hi ,
>>>> > I am getting error ORA -923 (from missing from statement) when i am
>>>> > tring to
>>>> > execute a sql query from Javascript file. Here i am successfully able
>>>> > to
>>>> > connect & execute queries with MYSQL. But in oracle connection i am
>>>> > getting
>>>> > this error even i can successfully make a connection with oracle. with
>>>> > isql
>>>> > & a program written in C i can connect and execute queries ok.
>>>> > I am using
>>>> > FreeSwitch 1.0.6
>>>> > unixOdbc ? ?2.3.0
>>>> > Please tell me where is problem.
>>>> > Best Regards,
>>>> > Rajesh Bansal
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >
>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From rupa at rupa.com  Mon Jan 10 20:25:29 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 10 Jan 2011 11:25:29 -0600
Subject: [Freeswitch-users] Asterisk & FreeSwitch in recording view
In-Reply-To: <AANLkTinvVqYuZKT6zHAQPW+QdqMoSaabB1WG6DLQOocO@mail.gmail.com>
References: <AANLkTinZT1WiNnae4oMKd+5YdEUO9xq4_cJr6R+4siiK@mail.gmail.com>
	<AANLkTinvVqYuZKT6zHAQPW+QdqMoSaabB1WG6DLQOocO@mail.gmail.com>
Message-ID: <AANLkTi=gPxWQgunBT604zT13gDe4oU8xNzWF3Xj_1aJ+@mail.gmail.com>

Let me clarify. :)

If you are ok with routing all your traffic through FreeSWITCH (or *),
then FreeSWITCH can definitely do what you want.  I use it that way
(record all external calls) without issue.  If your volume is high,
you'll want to record as PMCU WAV to a ramdisk and then have a cron
job that converts completed calls to mp3 or speex or whatever.  I have
scripts that do this well including preserving the metadata that FS
can write to the WAV files for CID info or whatever else you ask it to
put in there.  (just using sox or lame will loose that metadata from
the WAV file since they don't bother to preserve it).

That being said.  orecx is designed to record your VOIP traffic
without having to do anything put ensure all VOIP traffic ends up on
the network segment orecx is attached to.  This "transparent"
recording is definitely "the way to go" if you want to separate your
phone infrastructure from your recording infrastructure.

On Mon, Jan 10, 2011 at 11:16 AM, Rupa Schomaker <rupa at rupa.com> wrote:
> I think the standard answer to that is www.orecx.com.
>
> On Mon, Jan 10, 2011 at 3:39 AM, Leonid K <pivanet at gmail.com> wrote:
>> Hi,
>> currently we're looking for voip recording solution - it must record all
>> incoming/outgoing/internal/conference calls within the company. later on we
>> going to develop applications that let us fing/analyze recordings. the main
>> problem as I see at the moment is huge count of calls that is going through
>> switch.
>>
>> what do u think about using Asterisk or/and FreeSwitch for this task? thanks
>> in advance!
>> --
>> Sincerely,
>> Leonid Kryvoruchko
>>
>> Mobile?? +38 093 7609175
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> -Rupa
>



-- 
-Rupa


From steveayre at gmail.com  Mon Jan 10 20:33:25 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 10 Jan 2011 17:33:25 +0000
Subject: [Freeswitch-users] Getting Error ORA-923 in oracle query
	execution
In-Reply-To: <AANLkTimdbke-Kj4B09eaXhaOJzphB3rwa7zi9zSxJ21s@mail.gmail.com>
References: <AANLkTi=k-cwDWxvCMne=O958yhXd8=bkr1EWtPAORxL6@mail.gmail.com>
	<AANLkTi=u9QUHh1GD3MpQTB81N5zW5xwAcWazWzuU0A6Q@mail.gmail.com>
	<AANLkTin+T3yML3iZpVg0qgUFV1Vxjn0AYO9sAsNrOGHZ@mail.gmail.com>
	<AANLkTin2mx0K=NOx4xrM06AQX5vRV7Hu0poDGsR6AUAr@mail.gmail.com>
	<AANLkTi=Vbgyh4Yw0VuCXymOrbNpu8aFMUrk9vfR0UsQr@mail.gmail.com>
	<AANLkTimdbke-Kj4B09eaXhaOJzphB3rwa7zi9zSxJ21s@mail.gmail.com>
Message-ID: <AANLkTikT+i4BASNm041=qxX0+UXdT+3fug5=s9Uz78yF@mail.gmail.com>

Quick patch attached, it's not actually tested and we'd need to check
what the odbc driver name really looked like.

-Steve

On 10 January 2011 17:24, Steven Ayre <steveayre at gmail.com> wrote:
> We already have a similar autodetect for firebird there though, so
> probably no more yuck than it already is.
>
> -Steve
>
>
>
> On 10 January 2011 17:18, Rupa Schomaker <rupa at rupa.com> wrote:
>> Someone should raise it as a minor bug with Oracle. :( ?They are the
>> only modern db that I know of that requires a hack of "FROM dual" on
>> the SQL.
>>
>> I suppose either a) switch_odbc can auto detect oracle (not easy, it
>> would have to carry state in the dbh I guess) or b) a setting (again,
>> probably on the dbh) that you can put in FS's config to tell FS not to
>> do the select 1.
>>
>> Either one is... umm.. yuck. :(
>>
>> On Mon, Jan 10, 2011 at 5:39 AM, Rajesh Bansal
>> <bansal.rajeshkr at gmail.com> wrote:
>>> Hi All,
>>> I have find reason of problem with UnixOdbc (with Oracle). When we are using
>>> freeswitch with ODBC support before executing a sql query it run a test
>>> query "Select 1" which works at MYSQL but didn't work at ORACLE. This change
>>> should be in switch_odbc.c file. Please someone raise it as a minor bug so
>>> that it can be fixed in next releases.
>>> Best Regards,
>>> Rajesh Bansal
>>
>>>
>>>
>>> On Mon, Jan 10, 2011 at 1:39 PM, Rajesh Bansal <bansal.rajeshkr at gmail.com>
>>> wrote:
>>>>
>>>> Hi Steve,
>>>> I am using following sample code.
>>>> use("ODBC");
>>>> var dsn ? ? = "rajesh_dsn";
>>>> var db_user = "avc";
>>>> var db_pass = "avc";
>>>> var db ? ? ?= new ODBC(dsn, db_user, db_pass);
>>>> var sql = "select sysdate ?from dual";
>>>> console_log("info","\nbefore connect with DB\n");
>>>> if(db.connect())
>>>> {
>>>> if(session.ready())
>>>> ?session.answer();
>>>> console_log("info","\n\nafter connect with DB\n");
>>>> if (db.exec(sql))
>>>> ?? ? ? ?session.hangup(); //might want to say something nice instead.
>>>> else
>>>> ?? ? ? ?console_log("info","\n\nconnect with DB\n");
>>>> while (db.nextRow())
>>>> {
>>>> ??row = db.getData();
>>>> ??console_log("info", "UserName: " );
>>>> }}
>>>> else
>>>> console_log("info","\n\nunable to connect with DB\n");
>>>> Best Regards,
>>>> Rajesh Bansal
>>>>
>>>>
>>>>
>>>> On Fri, Jan 7, 2011 at 11:35 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>>>>
>>>>> Sounds like a SQL syntax error - can you show us the code that's
>>>>> executing the statement that gives the error?
>>>>>
>>>>> Specifically Oracle is complaining about not finding FROM where it
>>>>> expects to in the SELECT.
>>>>>
>>>>> -Steve
>>>>>
>>>>>
>>>>> On 7 January 2011 15:15, Rajesh Bansal <bansal.rajeshkr at gmail.com> wrote:
>>>>> > Hi ,
>>>>> > I am getting error ORA -923 (from missing from statement) when i am
>>>>> > tring to
>>>>> > execute a sql query from Javascript file. Here i am successfully able
>>>>> > to
>>>>> > connect & execute queries with MYSQL. But in oracle connection i am
>>>>> > getting
>>>>> > this error even i can successfully make a connection with oracle. with
>>>>> > isql
>>>>> > & a program written in C i can connect and execute queries ok.
>>>>> > I am using
>>>>> > FreeSwitch 1.0.6
>>>>> > unixOdbc ? ?2.3.0
>>>>> > Please tell me where is problem.
>>>>> > Best Regards,
>>>>> > Rajesh Bansal
>>>>> >
>>>>> >
>>>>> > _______________________________________________
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >
>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>> >
>>>>> >
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
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From boris at tagnet.ru  Mon Jan 10 20:43:09 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Mon, 10 Jan 2011 22:43:09 +0500
Subject: [Freeswitch-users] mod_lcr and extra_vars
Message-ID: <4D2B452D.8060004@tagnet.ru>

Hello!

     I need to set extra vars with mod_lcr. I did as wiki recomended:
1) created sql column
2) modified sql quer
3) added <param name="export_fields" value="lcr_gw_extra_vars"/> to profile
So, may lcr output looks nice:
| Digit Match | Carrier     | Rate     | Codec | CID Regexp          | 
Limit | 
Dialstring                                                                                                                                                               
|
  | 734353      | tagnet.ru   | 0.00000  |       |                     
|       | 
[lcr_carrier=tagnet.ru,lcr_rate=0.00000,lcr_gw_extra_vars=,v_tagnet_ats_dstport=50000]sofia/epbx/73435350101 at X.X.X.X:5060

Unfortunatelly I can't see (and can't use) v_tagnet_ats_dstport variable 
in my cdr records (even with b-leg the variable isn't present) until I 
set import=v_tagnet_ats_dstport variable. But I thinked mod_lcr should 
do this?
Something wrong with my configuration?


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From andy at fabulous4.co.uk  Mon Jan 10 21:28:08 2011
From: andy at fabulous4.co.uk (Andy Ayers)
Date: Mon, 10 Jan 2011 18:28:08 -0000
Subject: [Freeswitch-users] Problems with recorded messages cutting off at
	60 seconds involving remote SDP
Message-ID: <056401cbb0f4$22948a60$67bd9f20$@fabulous4.co.uk>

Hi,

 

I have a weird problem where on certain calls when someone tries to leave a
recorded message the call hangs up exactly 60 seconds after the start of the
recording. The recordFile is sending the stream to icecast using a shout:
address in MP3 format.

 

I've done a line by line compare of a successful call and one where the
recording fails and the issue seems to be with the remote SDP. 

 

RECORDING SUCCEEDS WHEN SDP IS..

 

v=0

o=root 1060 1060 IN IP4 77.240.60.33

s=session

c=IN IP4 77.240.60.33

t=0 0

m=audio 13850 RTP/AVP 8 0 3 97 7 110 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:97 iLBC/8000

a=fmtp:97 mode=30

a=rtpmap:7 LPC/8000

a=rtpmap:110 speex/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

 

RECORDING CUTS OFF AFTER 60 SECS WHEN SDP IS..

 

v=0

o=voip 302847118 302847118 IN IP4 77.240.60.4

s=voip

c=IN IP4 77.240.60.4

t=0 0

m=audio 13622 RTP/AVP 8 0 3 97 7 110 5 10 18 112 111 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:97 iLBC/8000

a=fmtp:97 mode=30

a=rtpmap:7 LPC/8000

a=rtpmap:110 speex/8000

a=rtpmap:5 DVI4/8000

a=rtpmap:10 L16/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:112 AAL2-G726-32/8000

a=rtpmap:111 G726-32/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

 

Can anyone tell me the significance of the o=voip, s=voip as opposed to
o=root,s=session? Is this likely to be the cause of the problem and any idea
how I fix it?

 

Many thanks as always

Andy Ayers

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From hadyn_whx at hotmail.com  Mon Jan 10 21:55:38 2011
From: hadyn_whx at hotmail.com (Alex Wang)
Date: Mon, 10 Jan 2011 13:55:38 -0500
Subject: [Freeswitch-users] auto-nat is not working
In-Reply-To: <AANLkTikO6j50jp6hC7keHnnU13C=ND1vRH0DxvtBjWm4@mail.gmail.com>
References: <BLU0-SMTP1648BEF319A9AA8367FF99D8D0E0@phx.gbl>
	<AANLkTikO6j50jp6hC7keHnnU13C=ND1vRH0DxvtBjWm4@mail.gmail.com>
Message-ID: <BLU0-SMTP11508FD72A79AE674F3BB378D0E0@phx.gbl>

Thanks Rupa

Tomato 1.28 enable uPNP. Sometime works sometime not. I am doing more
test.
I also see outside connection trying to register extension after I turn
on the uPNP, I know the uPNP also map the port out. How to disable that?

Thanks

Alex

On Mon, 10 Jan 2011 11:13:16 -0600
Rupa Schomaker <rupa at rupa.com> wrote:

> 1) Ensure you have UPNP turned on in tomato.  Double check.
> 
> 2) Ensure debug logs go to your log *file* (since you want to capture
> info at startup and what you need will scroll off the screen or be too
> early to get in fs_cli)
> 
> 3) start freeswitch
> 
> 4) check the log file or msgs originating from switch_nat.c (nice that
> logs show the file, eh?)
> 
> 5) pastebin the info in pastebin.freeswitch.org (or just the whole log
> from a freeswitch start, that might be better so you don't
> accidentally remove something needed)
> 
> 6) Profit. :)   Well, maybe not but I can look at 'em.
> 
> On Sun, Jan 9, 2011 at 9:33 PM, Alex Wang <hadyn_whx at hotmail.com> wrote:
> > Hi
> >
> > I am very new about the freeswitch and just finished setup a new
> > freeswitch and config a test sip account from my voip provider.
> >
> > in the fs_cli,
> > originate sofia/gateway/gw1/xxxxxx(my other number) ?&echo()
> > I got the phone ring but no echo.
> > originate sofia/user/1000 &echo() works fine.
> >
> > nat_map status shows;
> > freeswitch at internal> nat_map status
> > Nat Type: UNKNOWN, ExtIP:
> >
> > 0 total.
> >
> > freeswitch at internal> sofia status profile internal
> > =================================================================================================
> > Name ? ? ? ? ? ? ? ? ? ?internal
> > Domain Name ? ? ? ? ? ? N/A
> > Auto-NAT ? ? ? ? ? ? ? ?false
> > DBName ? ? ? ? ? ? ? ? ?sofia_reg_internal
> >
> > Just wonder how to fix the NAT issue on my freeswitch.
> >
> > The router is WRT54GL with Tomato 1.28 on it, which seems support upnp...
> >
> > Thanks a lot
> >
> > Alex
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> 
> -- 
> -Rupa
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From matte at ahavaxthuset.se  Mon Jan 10 21:23:45 2011
From: matte at ahavaxthuset.se (Mattias Hemmingsson)
Date: Mon, 10 Jan 2011 19:23:45 +0100 (CET)
Subject: [Freeswitch-users] Two meny qestions
In-Reply-To: <9DD9A108-0CFF-4224-88C2-95CFF9E0FE76@freeswitch.org>
Message-ID: <25504058.721294683825512.JavaMail.root@mailserver>

Hi


Still having problems transfering the call to the right extension.
I have one domain in my directory called www.elino.se and in there all my users all.
So i test to set upp the meny to transfer the call to my extension  like this

<entry action="menu-exec-app" digits="1" param="transfer 1000 at www.elino.se"/>

but it dont work.

i have also test to transfer with 


<entry action="menu-exec-app" digits="1" param="transfer 1000 XML default"/>

but the i only get an godbye.


what im i doing wrong ?

// Matte


----- Ursprungligt meddelande -----
Fr?n: "Brian West" <brian at freeswitch.org>
Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Skickat: s?ndag, 9 jan 2011 22:35:00
?mne: Re: [Freeswitch-users] Two meny qestions

This is because you're calling bridge right to the users endpoint... if you were to transfer to extension 1000 or 1001 then voicemail would work exactly like you expect.

/b

On Jan 9, 2011, at 1:21 PM, Mattias Hemmingsson wrote:

> But when the is not online i want the user to be transferd to the users voicemail.
> I have voicemail working of i call the user from a nother externsion.
> But i would like it to wokr from the ivr meny as well.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From hesser4900 at gmail.com  Mon Jan 10 21:42:36 2011
From: hesser4900 at gmail.com (Holger Esser)
Date: Mon, 10 Jan 2011 12:42:36 -0600
Subject: [Freeswitch-users] playing files through a remote URL
Message-ID: <AANLkTikYdXfOVYF6=vUsj8gB7kHUuwOddrYy_dFg6ByG@mail.gmail.com>

Hello,

I am trying to write a simple ivr script that answers the call, plays a
greeting and collects a few digits.
Is it possible to fetch the audio through a remote URL (i.e.
http://messageServer/en.us/greeting.wav<http://messageserver/en.us/greeting.wav>
)?

Or does it have to reside on the local file system or a mapped drive?

Kind regards,

Holger
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From brian at freeswitch.org  Mon Jan 10 22:24:27 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Jan 2011 13:24:27 -0600
Subject: [Freeswitch-users] auto-nat is not working
In-Reply-To: <BLU0-SMTP11508FD72A79AE674F3BB378D0E0@phx.gbl>
References: <BLU0-SMTP1648BEF319A9AA8367FF99D8D0E0@phx.gbl>
	<AANLkTikO6j50jp6hC7keHnnU13C=ND1vRH0DxvtBjWm4@mail.gmail.com>
	<BLU0-SMTP11508FD72A79AE674F3BB378D0E0@phx.gbl>
Message-ID: <606869C1-13C3-4C5A-8671-A3BA18A2C65F@freeswitch.org>

use nat-pmp upnp is unreliable sometimes.

/b

On Jan 10, 2011, at 12:55 PM, Alex Wang wrote:

> Thanks Rupa
> 
> Tomato 1.28 enable uPNP. Sometime works sometime not. I am doing more
> test.
> I also see outside connection trying to register extension after I turn
> on the uPNP, I know the uPNP also map the port out. How to disable that?
> 
> Thanks
> 
> Alex



From hesser4900 at gmail.com  Mon Jan 10 22:09:37 2011
From: hesser4900 at gmail.com (Holger Esser)
Date: Mon, 10 Jan 2011 13:09:37 -0600
Subject: [Freeswitch-users] playing files through a remote URL
In-Reply-To: <AANLkTikYdXfOVYF6=vUsj8gB7kHUuwOddrYy_dFg6ByG@mail.gmail.com>
References: <AANLkTikYdXfOVYF6=vUsj8gB7kHUuwOddrYy_dFg6ByG@mail.gmail.com>
Message-ID: <AANLkTik-d6Wb2dyStNSF-rVKCvaHzXq4pWKsYRKA9sf7@mail.gmail.com>

Hello,

I am trying to write a simple ivr script that answers the call, plays a
greeting and collects a few digits.
Is it possible to fetch the audio through a remote URL (i.e.
http://messageServer/en.us/greeting.wav<http://messageserver/en.us/greeting.wav>
)?

Or does it have to reside on the local file system or a mapped drive?

Kind regards,

Holger
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From hadyn_whx at hotmail.com  Mon Jan 10 23:21:19 2011
From: hadyn_whx at hotmail.com (Alex Wang)
Date: Mon, 10 Jan 2011 15:21:19 -0500
Subject: [Freeswitch-users] auto-nat is not working
In-Reply-To: <606869C1-13C3-4C5A-8671-A3BA18A2C65F@freeswitch.org>
References: <BLU0-SMTP11508FD72A79AE674F3BB378D0E0@phx.gbl>
	<606869C1-13C3-4C5A-8671-A3BA18A2C65F@freeswitch.org>
Message-ID: <BLU0-SMTP716CAC65E87813D51D132C8D0E0@phx.gbl>

Thanks Brian

Alex

On Mon, 10 Jan 2011 13:24:27 -0600
Brian West <brian at freeswitch.org> wrote:

> use nat-pmp upnp is unreliable sometimes.
> 
> /b
> 
> On Jan 10, 2011, at 12:55 PM, Alex Wang wrote:
> 
> > Thanks Rupa
> > 
> > Tomato 1.28 enable uPNP. Sometime works sometime not. I am doing more
> > test.
> > I also see outside connection trying to register extension after I turn
> > on the uPNP, I know the uPNP also map the port out. How to disable that?
> > 
> > Thanks
> > 
> > Alex
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From darren at aleph-com.net  Mon Jan 10 23:26:35 2011
From: darren at aleph-com.net (Darren Wiebe)
Date: Mon, 10 Jan 2011 13:26:35 -0700
Subject: [Freeswitch-users] Link2voip
Message-ID: <4D2B6B7B.4090102@aleph-com.net>

Good Afternoon,

I'm trying to get my freeswitch box talking to Link2voip.  Does anybody 
have sample XML files for them?

-- 
Darren Wiebe
Aleph Communications
--------------------
Phone: 1-877-702-2900
Fax:   1-866-274-4506
Email: darren at aleph-com.net



From lloyd.aloysius at gmail.com  Mon Jan 10 23:39:51 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 10 Jan 2011 15:39:51 -0500
Subject: [Freeswitch-users] auto-nat is not working
In-Reply-To: <BLU0-SMTP716CAC65E87813D51D132C8D0E0@phx.gbl>
References: <BLU0-SMTP11508FD72A79AE674F3BB378D0E0@phx.gbl>
	<606869C1-13C3-4C5A-8671-A3BA18A2C65F@freeswitch.org>
	<BLU0-SMTP716CAC65E87813D51D132C8D0E0@phx.gbl>
Message-ID: <AANLkTino1pcsnJtBwSyVUJD-sKXesvzwQd-zKYP2b5c8@mail.gmail.com>

Alex,

There are some bugs in Tomato Firmware. Sometimes work and some
times suddenly it strip some sip messages. Sometime no passwords. I find
very difficult on this.

Switch to some other firmware ... DD-WRT. your problem go away and you do
not need to do any changes on the router firmware configuration and
FreeSWITCH.

I have this problem for a longtime finally figure out with the debug. Then I
use the DD-WRT for Linksys WRT54GL the voip firmware version problem go
away.

Hope this helps.

Thanks
Lloyd


On Mon, Jan 10, 2011 at 3:21 PM, Alex Wang <hadyn_whx at hotmail.com> wrote:

> Thanks Brian
>
> Alex
>
> On Mon, 10 Jan 2011 13:24:27 -0600
> Brian West <brian at freeswitch.org> wrote:
>
> > use nat-pmp upnp is unreliable sometimes.
> >
> > /b
> >
> > On Jan 10, 2011, at 12:55 PM, Alex Wang wrote:
> >
> > > Thanks Rupa
> > >
> > > Tomato 1.28 enable uPNP. Sometime works sometime not. I am doing more
> > > test.
> > > I also see outside connection trying to register extension after I turn
> > > on the uPNP, I know the uPNP also map the port out. How to disable
> that?
> > >
> > > Thanks
> > >
> > > Alex
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Tue Jan 11 00:24:27 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Jan 2011 15:24:27 -0600
Subject: [Freeswitch-users] auto-nat is not working
In-Reply-To: <AANLkTino1pcsnJtBwSyVUJD-sKXesvzwQd-zKYP2b5c8@mail.gmail.com>
References: <BLU0-SMTP11508FD72A79AE674F3BB378D0E0@phx.gbl>
	<606869C1-13C3-4C5A-8671-A3BA18A2C65F@freeswitch.org>
	<BLU0-SMTP716CAC65E87813D51D132C8D0E0@phx.gbl>
	<AANLkTino1pcsnJtBwSyVUJD-sKXesvzwQd-zKYP2b5c8@mail.gmail.com>
Message-ID: <ACB9C3B1-9D5B-4F4B-B0A8-414F7034B95D@freeswitch.org>

This sentence makes NO sense... passwords aren't set in plain text on the wire for SIP auth... so what exactly are you talking about?  I for one have not had issues with Tomato.

/b

On Jan 10, 2011, at 2:39 PM, Aloysius Lloyd wrote:

> There are some bugs in Tomato Firmware. Sometimes work and some times suddenly it strip some sip messages. Sometime no passwords. I find very difficult on this.

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From brian at freeswitch.org  Tue Jan 11 00:24:51 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Jan 2011 15:24:51 -0600
Subject: [Freeswitch-users] Link2voip
In-Reply-To: <4D2B6B7B.4090102@aleph-com.net>
References: <4D2B6B7B.4090102@aleph-com.net>
Message-ID: <9F467B41-E6CE-4A64-A228-52896E428CD4@freeswitch.org>

can you put up a sip trace or something so we can help guide you?

/b

On Jan 10, 2011, at 2:26 PM, Darren Wiebe wrote:

> Good Afternoon,
> 
> I'm trying to get my freeswitch box talking to Link2voip.  Does anybody 
> have sample XML files for them?
> 
> -- 
> Darren Wiebe
> Aleph Communications
> --------------------
> Phone: 1-877-702-2900
> Fax:   1-866-274-4506
> Email: darren at aleph-com.net
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From hadyn_whx at hotmail.com  Tue Jan 11 00:37:42 2011
From: hadyn_whx at hotmail.com (Alex Wang)
Date: Mon, 10 Jan 2011 16:37:42 -0500
Subject: [Freeswitch-users] auto-nat is not working
In-Reply-To: <AANLkTino1pcsnJtBwSyVUJD-sKXesvzwQd-zKYP2b5c8@mail.gmail.com>
References: <BLU0-SMTP716CAC65E87813D51D132C8D0E0@phx.gbl>
	<AANLkTino1pcsnJtBwSyVUJD-sKXesvzwQd-zKYP2b5c8@mail.gmail.com>
Message-ID: <BLU0-SMTP2135DFD02F20F00E38AC0158D0E0@phx.gbl>

I had bad experience about dd-wrt, not stable on my router, always cause
the problem or hang up over night. I will try the nat-pmp on tomato.
Switch back to dd-wrt will be my last option.

BTW, how to disable extension register from external after I turn the
upnp up? All the extensions are in my LAN and I do see some ip trying to
register the extension from outside.

Thanks

Alex

On Mon, 10 Jan 2011 15:39:51 -0500
Aloysius Lloyd <lloyd.aloysius at gmail.com> wrote:

> Alex,
> 
> There are some bugs in Tomato Firmware. Sometimes work and some
> times suddenly it strip some sip messages. Sometime no passwords. I find
> very difficult on this.
> 
> Switch to some other firmware ... DD-WRT. your problem go away and you do
> not need to do any changes on the router firmware configuration and
> FreeSWITCH.
> 
> I have this problem for a longtime finally figure out with the debug. Then I
> use the DD-WRT for Linksys WRT54GL the voip firmware version problem go
> away.
> 
> Hope this helps.
> 
> Thanks
> Lloyd
> 
> 
> On Mon, Jan 10, 2011 at 3:21 PM, Alex Wang <hadyn_whx at hotmail.com> wrote:
> 
> > Thanks Brian
> >
> > Alex
> >
> > On Mon, 10 Jan 2011 13:24:27 -0600
> > Brian West <brian at freeswitch.org> wrote:
> >
> > > use nat-pmp upnp is unreliable sometimes.
> > >
> > > /b
> > >
> > > On Jan 10, 2011, at 12:55 PM, Alex Wang wrote:
> > >
> > > > Thanks Rupa
> > > >
> > > > Tomato 1.28 enable uPNP. Sometime works sometime not. I am doing more
> > > > test.
> > > > I also see outside connection trying to register extension after I turn
> > > > on the uPNP, I know the uPNP also map the port out. How to disable
> > that?
> > > >
> > > > Thanks
> > > >
> > > > Alex
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >





From hadyn_whx at hotmail.com  Tue Jan 11 00:40:50 2011
From: hadyn_whx at hotmail.com (Alex Wang)
Date: Mon, 10 Jan 2011 16:40:50 -0500
Subject: [Freeswitch-users] auto-nat is not working
In-Reply-To: <ACB9C3B1-9D5B-4F4B-B0A8-414F7034B95D@freeswitch.org>
References: <AANLkTino1pcsnJtBwSyVUJD-sKXesvzwQd-zKYP2b5c8@mail.gmail.com>
	<ACB9C3B1-9D5B-4F4B-B0A8-414F7034B95D@freeswitch.org>
Message-ID: <BLU0-SMTP755CD2410AA0BD8BB859CE8D0E0@phx.gbl>

Hi Brian

I do have some issue this morning relate to the NAT. But it hasn't back
since I reboot the unit so I am not sure if the bug is true or not. I
will update if I had. (hope not, I don't like dd-wrt)

Alex

On Mon, 10 Jan 2011 15:24:27 -0600
Brian West <brian at freeswitch.org> wrote:

> This sentence makes NO sense... passwords aren't set in plain text on the wire for SIP auth... so what exactly are you talking about?  I for one have not had issues with Tomato.
> 
> /b
> 
> On Jan 10, 2011, at 2:39 PM, Aloysius Lloyd wrote:
> 
> > There are some bugs in Tomato Firmware. Sometimes work and some times suddenly it strip some sip messages. Sometime no passwords. I find very difficult on this.
> 





From msc at freeswitch.org  Tue Jan 11 00:52:25 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Jan 2011 13:52:25 -0800
Subject: [Freeswitch-users] Link2voip
In-Reply-To: <9F467B41-E6CE-4A64-A228-52896E428CD4@freeswitch.org>
References: <4D2B6B7B.4090102@aleph-com.net>
	<9F467B41-E6CE-4A64-A228-52896E428CD4@freeswitch.org>
Message-ID: <AANLkTinwz46psUH_KzuEBDqQuXxXGD4QOwrphOA83t3O@mail.gmail.com>

Or just give us your credentials and we'll "test it thoroughly" for you. :)
-MC

On Mon, Jan 10, 2011 at 1:24 PM, Brian West <brian at freeswitch.org> wrote:

> can you put up a sip trace or something so we can help guide you?
>
> /b
>
> On Jan 10, 2011, at 2:26 PM, Darren Wiebe wrote:
>
> > Good Afternoon,
> >
> > I'm trying to get my freeswitch box talking to Link2voip.  Does anybody
> > have sample XML files for them?
> >
> > --
> > Darren Wiebe
> > Aleph Communications
> > --------------------
> > Phone: 1-877-702-2900
> > Fax:   1-866-274-4506
> > Email: darren at aleph-com.net
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lloyd.aloysius at gmail.com  Tue Jan 11 00:46:28 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 10 Jan 2011 16:46:28 -0500
Subject: [Freeswitch-users] auto-nat is not working
In-Reply-To: <BLU0-SMTP2135DFD02F20F00E38AC0158D0E0@phx.gbl>
References: <BLU0-SMTP716CAC65E87813D51D132C8D0E0@phx.gbl>
	<AANLkTino1pcsnJtBwSyVUJD-sKXesvzwQd-zKYP2b5c8@mail.gmail.com>
	<BLU0-SMTP2135DFD02F20F00E38AC0158D0E0@phx.gbl>
Message-ID: <AANLkTi=k3cDaubpL20zgFDNDB_WT=gH2YRvYbDka+ba1@mail.gmail.com>

Brian,

Yes that correct. Sometimes Tomato suddenly send wrong messages to the sip
server ... that causes authentication fails. We suspect the password strip
off some characters. This is happen suddenly and after some time it will
work back normal.

I try with 1.25 , 1.27 and 1.28  ..... no luck then I switch to dd-wrt all
problems go away. I am using
Firmware: DD-WRT v24-sp2 (08/07/10) voip

what is the Tomato firmware u r using?

Thanks
Lloyd


On Mon, Jan 10, 2011 at 4:37 PM, Alex Wang <hadyn_whx at hotmail.com> wrote:

> I had bad experience about dd-wrt, not stable on my router, always cause
> the problem or hang up over night. I will try the nat-pmp on tomato.
> Switch back to dd-wrt will be my last option.
>
> BTW, how to disable extension register from external after I turn the
> upnp up? All the extensions are in my LAN and I do see some ip trying to
> register the extension from outside.
>
> Thanks
>
> Alex
>
> On Mon, 10 Jan 2011 15:39:51 -0500
> Aloysius Lloyd <lloyd.aloysius at gmail.com> wrote:
>
> > Alex,
> >
> > There are some bugs in Tomato Firmware. Sometimes work and some
> > times suddenly it strip some sip messages. Sometime no passwords. I find
> > very difficult on this.
> >
> > Switch to some other firmware ... DD-WRT. your problem go away and you do
> > not need to do any changes on the router firmware configuration and
> > FreeSWITCH.
> >
> > I have this problem for a longtime finally figure out with the debug.
> Then I
> > use the DD-WRT for Linksys WRT54GL the voip firmware version problem go
> > away.
> >
> > Hope this helps.
> >
> > Thanks
> > Lloyd
> >
> >
> > On Mon, Jan 10, 2011 at 3:21 PM, Alex Wang <hadyn_whx at hotmail.com>
> wrote:
> >
> > > Thanks Brian
> > >
> > > Alex
> > >
> > > On Mon, 10 Jan 2011 13:24:27 -0600
> > > Brian West <brian at freeswitch.org> wrote:
> > >
> > > > use nat-pmp upnp is unreliable sometimes.
> > > >
> > > > /b
> > > >
> > > > On Jan 10, 2011, at 12:55 PM, Alex Wang wrote:
> > > >
> > > > > Thanks Rupa
> > > > >
> > > > > Tomato 1.28 enable uPNP. Sometime works sometime not. I am doing
> more
> > > > > test.
> > > > > I also see outside connection trying to register extension after I
> turn
> > > > > on the uPNP, I know the uPNP also map the port out. How to disable
> > > that?
> > > > >
> > > > > Thanks
> > > > >
> > > > > Alex
> > > >
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > http://www.freeswitch.org
> > >
> > >
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lloyd.aloysius at gmail.com  Tue Jan 11 01:01:53 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 10 Jan 2011 17:01:53 -0500
Subject: [Freeswitch-users] auto-nat is not working
In-Reply-To: <AANLkTi=k3cDaubpL20zgFDNDB_WT=gH2YRvYbDka+ba1@mail.gmail.com>
References: <BLU0-SMTP716CAC65E87813D51D132C8D0E0@phx.gbl>
	<AANLkTino1pcsnJtBwSyVUJD-sKXesvzwQd-zKYP2b5c8@mail.gmail.com>
	<BLU0-SMTP2135DFD02F20F00E38AC0158D0E0@phx.gbl>
	<AANLkTi=k3cDaubpL20zgFDNDB_WT=gH2YRvYbDka+ba1@mail.gmail.com>
Message-ID: <AANLkTi=4DJ0152Hg3YsUPAJhc8T9jrmGm+1nTPTr_TFw@mail.gmail.com>

Here is the thread I post before ...

http://lists.freeswitch.org/pipermail/freeswitch-users/2010-April/055696.html

All sort of problem from Tomato.

Also I can share you the experience.

1. If you unplug the power and restart the router . Phones never connect the
server.
2. If you reboot from web interface . Then all phones start to work.
3. Etc .....

Here is another link
http://www.dslreports.com/forum/r23423987-Equipment-Tomato-with-VOIP-warning

Google : Tomato + SIP issues

I am using dd-wrt for last four months never have any problem.FreeSWITCH
default configuration working all the time.

That is my experience. Hope this may help you.

Thanks and regards,
Loyd


On Mon, Jan 10, 2011 at 4:46 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Brian,
>
> Yes that correct. Sometimes Tomato suddenly send wrong messages to the sip
> server ... that causes authentication fails. We suspect the password strip
> off some characters. This is happen suddenly and after some time it will
> work back normal.
>
> I try with 1.25 , 1.27 and 1.28  ..... no luck then I switch to dd-wrt all
> problems go away. I am using
> Firmware: DD-WRT v24-sp2 (08/07/10) voip
>
> what is the Tomato firmware u r using?
>
> Thanks
> Lloyd
>
>
> On Mon, Jan 10, 2011 at 4:37 PM, Alex Wang <hadyn_whx at hotmail.com> wrote:
>
>> I had bad experience about dd-wrt, not stable on my router, always cause
>> the problem or hang up over night. I will try the nat-pmp on tomato.
>> Switch back to dd-wrt will be my last option.
>>
>> BTW, how to disable extension register from external after I turn the
>> upnp up? All the extensions are in my LAN and I do see some ip trying to
>> register the extension from outside.
>>
>> Thanks
>>
>> Alex
>>
>> On Mon, 10 Jan 2011 15:39:51 -0500
>> Aloysius Lloyd <lloyd.aloysius at gmail.com> wrote:
>>
>> > Alex,
>> >
>> > There are some bugs in Tomato Firmware. Sometimes work and some
>> > times suddenly it strip some sip messages. Sometime no passwords. I find
>> > very difficult on this.
>> >
>> > Switch to some other firmware ... DD-WRT. your problem go away and you
>> do
>> > not need to do any changes on the router firmware configuration and
>> > FreeSWITCH.
>> >
>> > I have this problem for a longtime finally figure out with the debug.
>> Then I
>> > use the DD-WRT for Linksys WRT54GL the voip firmware version problem go
>> > away.
>> >
>> > Hope this helps.
>> >
>> > Thanks
>> > Lloyd
>> >
>> >
>> > On Mon, Jan 10, 2011 at 3:21 PM, Alex Wang <hadyn_whx at hotmail.com>
>> wrote:
>> >
>> > > Thanks Brian
>> > >
>> > > Alex
>> > >
>> > > On Mon, 10 Jan 2011 13:24:27 -0600
>> > > Brian West <brian at freeswitch.org> wrote:
>> > >
>> > > > use nat-pmp upnp is unreliable sometimes.
>> > > >
>> > > > /b
>> > > >
>> > > > On Jan 10, 2011, at 12:55 PM, Alex Wang wrote:
>> > > >
>> > > > > Thanks Rupa
>> > > > >
>> > > > > Tomato 1.28 enable uPNP. Sometime works sometime not. I am doing
>> more
>> > > > > test.
>> > > > > I also see outside connection trying to register extension after I
>> turn
>> > > > > on the uPNP, I know the uPNP also map the port out. How to disable
>> > > that?
>> > > > >
>> > > > > Thanks
>> > > > >
>> > > > > Alex
>> > > >
>> > > >
>> > > > _______________________________________________
>> > > > FreeSWITCH-users mailing list
>> > > > FreeSWITCH-users at lists.freeswitch.org
>> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > > http://www.freeswitch.org
>> > >
>> > >
>> > >
>> > >
>> > > _______________________________________________
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> > >
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From brian at freeswitch.org  Tue Jan 11 01:14:00 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Jan 2011 16:14:00 -0600
Subject: [Freeswitch-users] auto-nat is not working
In-Reply-To: <AANLkTi=k3cDaubpL20zgFDNDB_WT=gH2YRvYbDka+ba1@mail.gmail.com>
References: <BLU0-SMTP716CAC65E87813D51D132C8D0E0@phx.gbl>
	<AANLkTino1pcsnJtBwSyVUJD-sKXesvzwQd-zKYP2b5c8@mail.gmail.com>
	<BLU0-SMTP2135DFD02F20F00E38AC0158D0E0@phx.gbl>
	<AANLkTi=k3cDaubpL20zgFDNDB_WT=gH2YRvYbDka+ba1@mail.gmail.com>
Message-ID: <2DD31FE2-FBE4-4060-BCE2-6A2DA6B8161E@freeswitch.org>

Odd since Tomato doesn't have a SIP alg built in... 

/b

On Jan 10, 2011, at 3:46 PM, Aloysius Lloyd wrote:

> Brian,
> 
> Yes that correct. Sometimes Tomato suddenly send wrong messages to the sip server ... that causes authentication fails. We suspect the password strip off some characters. This is happen suddenly and after some time it will work back normal.
> 
> I try with 1.25 , 1.27 and 1.28  ..... no luck then I switch to dd-wrt all problems go away. I am using 
> Firmware: DD-WRT v24-sp2 (08/07/10) voip
> 
> what is the Tomato firmware u r using?
> 
> Thanks
> Lloyd
> 

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From brian at freeswitch.org  Tue Jan 11 01:15:58 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Jan 2011 16:15:58 -0600
Subject: [Freeswitch-users] auto-nat is not working
In-Reply-To: <AANLkTi=4DJ0152Hg3YsUPAJhc8T9jrmGm+1nTPTr_TFw@mail.gmail.com>
References: <BLU0-SMTP716CAC65E87813D51D132C8D0E0@phx.gbl>
	<AANLkTino1pcsnJtBwSyVUJD-sKXesvzwQd-zKYP2b5c8@mail.gmail.com>
	<BLU0-SMTP2135DFD02F20F00E38AC0158D0E0@phx.gbl>
	<AANLkTi=k3cDaubpL20zgFDNDB_WT=gH2YRvYbDka+ba1@mail.gmail.com>
	<AANLkTi=4DJ0152Hg3YsUPAJhc8T9jrmGm+1nTPTr_TFw@mail.gmail.com>
Message-ID: <E15D3DE6-0C48-4329-B8B3-264A5BD47015@freeswitch.org>

I have never had these issue with Tomato... but reading that thread is somewhat comical.

SIP Nat friendly? HA!

/b

On Jan 10, 2011, at 4:01 PM, Aloysius Lloyd wrote:

> 
> Here is another link http://www.dslreports.com/forum/r23423987-Equipment-Tomato-with-VOIP-warning
> 
> Google : Tomato + SIP issues
> 
> I am using dd-wrt for last four months never have any problem.FreeSWITCH default configuration working all the time.
> 
> That is my experience. Hope this may help you.
> 
> Thanks and regards,
> Loyd

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From lloyd.aloysius at gmail.com  Tue Jan 11 01:31:03 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 10 Jan 2011 17:31:03 -0500
Subject: [Freeswitch-users] auto-nat is not working
In-Reply-To: <E15D3DE6-0C48-4329-B8B3-264A5BD47015@freeswitch.org>
References: <BLU0-SMTP716CAC65E87813D51D132C8D0E0@phx.gbl>
	<AANLkTino1pcsnJtBwSyVUJD-sKXesvzwQd-zKYP2b5c8@mail.gmail.com>
	<BLU0-SMTP2135DFD02F20F00E38AC0158D0E0@phx.gbl>
	<AANLkTi=k3cDaubpL20zgFDNDB_WT=gH2YRvYbDka+ba1@mail.gmail.com>
	<AANLkTi=4DJ0152Hg3YsUPAJhc8T9jrmGm+1nTPTr_TFw@mail.gmail.com>
	<E15D3DE6-0C48-4329-B8B3-264A5BD47015@freeswitch.org>
Message-ID: <AANLkTikJchZfWjM7frmuyFJ9D06jbZoYK5F3ee1--MeJ@mail.gmail.com>

What is your Tomato firmware version ? Let me give a Try on the version.

Thanks
Lloyd


On Mon, Jan 10, 2011 at 5:15 PM, Brian West <brian at freeswitch.org> wrote:

> I have never had these issue with Tomato... but reading that thread is
> somewhat comical.
>
> SIP Nat friendly? HA!
>
> /b
>
> On Jan 10, 2011, at 4:01 PM, Aloysius Lloyd wrote:
>
>
> Here is another link
> http://www.dslreports.com/forum/r23423987-Equipment-Tomato-with-VOIP-warning
>
> Google : Tomato + SIP issues
>
> I am using dd-wrt for last four months never have any problem.FreeSWITCH
> default configuration working all the time.
>
> That is my experience. Hope this may help you.
>
> Thanks and regards,
> Loyd
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Tue Jan 11 01:42:45 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Jan 2011 16:42:45 -0600
Subject: [Freeswitch-users] auto-nat is not working
In-Reply-To: <AANLkTikJchZfWjM7frmuyFJ9D06jbZoYK5F3ee1--MeJ@mail.gmail.com>
References: <BLU0-SMTP716CAC65E87813D51D132C8D0E0@phx.gbl>
	<AANLkTino1pcsnJtBwSyVUJD-sKXesvzwQd-zKYP2b5c8@mail.gmail.com>
	<BLU0-SMTP2135DFD02F20F00E38AC0158D0E0@phx.gbl>
	<AANLkTi=k3cDaubpL20zgFDNDB_WT=gH2YRvYbDka+ba1@mail.gmail.com>
	<AANLkTi=4DJ0152Hg3YsUPAJhc8T9jrmGm+1nTPTr_TFw@mail.gmail.com>
	<E15D3DE6-0C48-4329-B8B3-264A5BD47015@freeswitch.org>
	<AANLkTikJchZfWjM7frmuyFJ9D06jbZoYK5F3ee1--MeJ@mail.gmail.com>
Message-ID: <3A279A2B-5C56-4F95-AE7A-3E9A5F45C930@freeswitch.org>

1.25 was the last time I updated.  Used it extensively when writing all that crap to deal with nat in freeswitch.

/b

On Jan 10, 2011, at 4:31 PM, Aloysius Lloyd wrote:

> What is your Tomato firmware version ? Let me give a Try on the version.
> 
> Thanks
> Lloyd
> 
> 
> On Mon, Jan 10, 2011 at 5:15 PM, Brian West <brian at freeswitch.org> wrote:
> I have never had these issue with Tomato... but reading that thread is somewhat comical.
> 
> SIP Nat friendly? HA!
> 
> /b
> 

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From rafonline at hotmail.com  Tue Jan 11 01:46:05 2011
From: rafonline at hotmail.com (Rafqat .)
Date: Mon, 10 Jan 2011 22:46:05 +0000
Subject: [Freeswitch-users] bind_digit_action
Message-ID: <COL122-W20D6C00E627218901B0691B50E0@phx.gbl>




Hi,

Can someone please help here.? 

I am trying to use bind_digit_action to hangup leg B only, if leg A presses ##, but it doesn't seem to be working.? The INFO message is displayed but when I press ##, nothing happens.

Any help will be much appreciated.



??? <extension name="call-me-service">
??? ??? <condition field="destination_number" expression="^9999$">
??? ??? ??? <action application="answer"/>
??????????? .........
??? ??? ??? <action application="set" data="bridge_pre_execute_bleg_app=execute_extension"/>
??? ??? ??? <action application="set" data="bridge_pre_execute_bleg_data=START_LISTENING XML private"/>
??? ??? ??? .......
??? ??? </condition>
??? </extension>

??? 
?? <extension name="setup bind_digit_action_listening">??????????????? 
??? <condition field="destination_number" expression="^START_LISTENING$">
????? <action application="log" data="INFO Configuring bind_digit_action to listen..."/>
????? <action application="bind_digit_action" data="start,##,exec:hangup,unknown"/>
??? ? <action application="digit_action_set_realm" data="start"/>
??? </condition>???????????????????????????????????? 
? </extension>


Cheers

Raf? 
 		 	   		  

From msc at freeswitch.org  Tue Jan 11 02:03:16 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Jan 2011 23:03:16 +0000
Subject: [Freeswitch-users] auto-nat is not working
In-Reply-To: <3A279A2B-5C56-4F95-AE7A-3E9A5F45C930@freeswitch.org>
References: <BLU0-SMTP716CAC65E87813D51D132C8D0E0@phx.gbl>
	<AANLkTino1pcsnJtBwSyVUJD-sKXesvzwQd-zKYP2b5c8@mail.gmail.com>
	<BLU0-SMTP2135DFD02F20F00E38AC0158D0E0@phx.gbl>
	<AANLkTi=k3cDaubpL20zgFDNDB_WT=gH2YRvYbDka+ba1@mail.gmail.com>
	<AANLkTi=4DJ0152Hg3YsUPAJhc8T9jrmGm+1nTPTr_TFw@mail.gmail.com>
	<E15D3DE6-0C48-4329-B8B3-264A5BD47015@freeswitch.org>
	<AANLkTikJchZfWjM7frmuyFJ9D06jbZoYK5F3ee1--MeJ@mail.gmail.com>
	<3A279A2B-5C56-4F95-AE7A-3E9A5F45C930@freeswitch.org>
Message-ID: <AANLkTinPiuT4m2-NqTX8SiPxV=6p7X1ev=Na9vSeKEwF@mail.gmail.com>

FWIW,

I have never had any issues with Tomato and SIP. It "just worked" for me.
-MC

On Mon, Jan 10, 2011 at 10:42 PM, Brian West <brian at freeswitch.org> wrote:

> 1.25 was the last time I updated.  Used it extensively when writing all
> that crap to deal with nat in freeswitch.
>
> /b
>
> On Jan 10, 2011, at 4:31 PM, Aloysius Lloyd wrote:
>
> What is your Tomato firmware version ? Let me give a Try on the version.
>
> Thanks
> Lloyd
>
>
> On Mon, Jan 10, 2011 at 5:15 PM, Brian West <brian at freeswitch.org> wrote:
>
>> I have never had these issue with Tomato... but reading that thread is
>> somewhat comical.
>>
>> SIP Nat friendly? HA!
>>
>> /b
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Jan 11 02:04:52 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Jan 2011 23:04:52 +0000
Subject: [Freeswitch-users] bind_digit_action
In-Reply-To: <COL122-W20D6C00E627218901B0691B50E0@phx.gbl>
References: <COL122-W20D6C00E627218901B0691B50E0@phx.gbl>
Message-ID: <AANLkTim-+iLNZAhNaWU0npAfL2Sn++8mCyqwANXXbVax@mail.gmail.com>

Just to confirm: you want Leg A to press ## to hangup leg B? If so then you
need to set the bind_digit_action on Leg A. The way you have it now is that
Leg B would need to dial ##.

-MC

On Mon, Jan 10, 2011 at 10:46 PM, Rafqat . <rafonline at hotmail.com> wrote:

>
>
>
> Hi,
>
> Can someone please help here.
>
> I am trying to use bind_digit_action to hangup leg B only, if leg A presses
> ##, but it doesn't seem to be working.  The INFO message is displayed but
> when I press ##, nothing happens.
>
> Any help will be much appreciated.
>
>
>
>     <extension name="call-me-service">
>         <condition field="destination_number" expression="^9999$">
>             <action application="answer"/>
>             .........
>             <action application="set"
> data="bridge_pre_execute_bleg_app=execute_extension"/>
>             <action application="set"
> data="bridge_pre_execute_bleg_data=START_LISTENING XML private"/>
>             .......
>         </condition>
>     </extension>
>
>
>    <extension name="setup bind_digit_action_listening">
>     <condition field="destination_number" expression="^START_LISTENING$">
>       <action application="log" data="INFO Configuring bind_digit_action to
> listen..."/>
>       <action application="bind_digit_action"
> data="start,##,exec:hangup,unknown"/>
>       <action application="digit_action_set_realm" data="start"/>
>     </condition>
>   </extension>
>
>
> Cheers
>
> Raf
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From zacw at safisystems.com  Tue Jan 11 01:03:47 2011
From: zacw at safisystems.com (Zac Wolfe)
Date: Mon, 10 Jan 2011 14:03:47 -0800
Subject: [Freeswitch-users] New FreeSWITCH IVR coming, but need HELP!
Message-ID: <AANLkTi=L+EFi50Q4Mb-=X2HW=SsA_TG8tBcg9k6Dgmth@mail.gmail.com>

Hi guys,

First some good news:  we're finally close to releasing our free IVR
Development platform SafiServer/SafiWorkshop (www.safisystems.com) with
FreeSWITCH support! It's happening much later than we originally anticipated
as we've been unexpectedly busy with contracting opportunities but I think
it will be worth the wait.  Currently everything is working fine with one
minor exception:  if the user-created script (we call them Saflets) doesn't
explicitly hang up the call, the call will remain parked until the caller
hangs up.  Some details:

In Asterisk we invoke our server-side scripting applications via the
extensions.conf using the following syntax:

exten = 1111,1,Agi(agi://
192.168.0.10:3573/safletEngine.agi?saflet=project1/callflow1)

Here '192.168.0.10' is the IP address of the SafiServer and
project1/callflow1 identifies the Saflet to be executed.  Asterisk creates a
socket connection to the SafiServer and, once the socket is disconnected,
the call proceeds to the next entry in the dialplan (typically 'hangup').

For FreeSWITCH, the process is slightly different.  Currently, rather than
create a separate socket connection for each incoming call, we're invoking
an event that informs the SafiServer that there is a new incoming call.  The
event contains the contextual information including the Saflet name.  For
example:

<extension name="9194">
      <condition field="destination_number" expression="9194$">
        <action inline="true" application="set" data="saficall=true"/>
        <action application="event"
data="Event-Subclass=saficall::incoming,Event-Name=CUSTOM,saflet_project=test,saflet=flow1,new_saficall=true"/>

       <!-- <action application="set" data="park_timeout=10"/> -->
        <action application="park" />
      </condition>
    </extension>

So once the event is fired, the call is parked to prevent further execution
within the dialplan.  From there on, SafiServer is controlling the call via
Inbound Mod event socket.

So this works perfectly, except that if the invoked Saflet doesn't
explicitly hang-up the call it will remain parked until the caller hangs
up.  My question is, is there a better way to do this?  Is there some better
alternative to park in this case?  Ideally I'd like to initiate a 'session'
of some kind when the SafiServer is "controlling" the call and then exit
that session as soon as the Saflet is complete, at which point the call
would continue on to the next entry in the dialplan.  I understand I could
use Outbound sockets to achieve this but, as I mentioned, I'd like to avoid
the overhead of a separate socket connection for each incoming call.

I actually have a mod_saficall.c app that does basically the the same thing
as I described in the dialplan entry.  Perhaps there's something more I
could do in code that would allow me to be notified when the session is
complete.  Here's the relevant code I have so far:

    switch_channel_t *channel = NULL;
    switch_event_t *event;
    const char *safiCallFlag = NULL;
    channel = switch_core_session_get_channel(session);

    safiCallFlag = switch_channel_get_variable(channel, "saficall");

    if (!safiCallFlag)
        switch_channel_set_variable(channel, "saficall", "true");


    if (switch_event_create_subclass(&event, SWITCH_EVENT_CUSTOM,
"saficall::incoming") == SWITCH_STATUS_SUCCESS) {

        switch_event_add_header_string(event, SWITCH_STACK_BOTTOM,
"new_saficall", safiCallFlag ? "false" : "true");

        switch_channel_event_set_data(channel, event);
        switch_event_fire(&event);
        switch_ivr_park(session, NULL);
    }

Any ideas you might have on this are welcome.

Thanks,
Zac Wolfe
Safi Systems LLC
www.safisystems.com
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From darren at aleph-com.net  Tue Jan 11 02:30:11 2011
From: darren at aleph-com.net (Darren Wiebe)
Date: Mon, 10 Jan 2011 16:30:11 -0700
Subject: [Freeswitch-users] Link2voip
In-Reply-To: <AANLkTinwz46psUH_KzuEBDqQuXxXGD4QOwrphOA83t3O@mail.gmail.com>
References: <4D2B6B7B.4090102@aleph-com.net>	<9F467B41-E6CE-4A64-A228-52896E428CD4@freeswitch.org>
	<AANLkTinwz46psUH_KzuEBDqQuXxXGD4QOwrphOA83t3O@mail.gmail.com>
Message-ID: <4D2B9683.2000405@aleph-com.net>

Yeah, I bet.  :)  The outgoing call problem was my fault, I had an 
incorrect piece of dialplan.  Here's the trace on an incoming call.  I'm 
trying to get it to come to a particular DID in the public context 
instead of this.

2011-01-10 16:22:36.430652 [INFO] mod_dialplan_xml.c:331 Processing CLID 
NAME
<7806283672>->sipuser in context public

What am I missing?

Here's the relevant provider entry from the external sip profile

<include>
<gateway name="link2voip1">
<param name="username" value="sipuser"/>
<param name="password" value="password"/>
<param name="proxy" value="sip.ca1.link2voip.com"/>
<param name="register" value="true"/>
<param name="register-transport" value="udp"/>
<param name="retry-seconds" value="30"/>
</gateway>
</include>

Sip Trace:

    From: "CLID NAME" <sip:7806283672 at CUSTOMERIP>;tag=as01e3f5de
    Call-ID: 0472a7de73e00fc578b12f6679b6dd9d at CUSTOMERIP
    To: <sip:14034883602 at sip.ca2.link2voip.com>;tag=m0Ur3NvDZma5H
    CSeq: 103 ACK
    Content-Length: 0

    ------------------------------------------------------------------------
recv 942 bytes from udp/[66.51.110.210]:5060 at 23:22:36.415027:
    ------------------------------------------------------------------------
    INVITE 
sip:gw+link2voip2 at 192.168.35.1:5080;transport=udp;gw=link2voip2 SIP/2.
0
    Record-Route: <sip:66.51.110.210;lr;ftag=as368d01dd>
    Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0
    Via: SIP/2.0/UDP CUSTOMERIP:5060;branch=z9hG4bK0bdd0021;rport=5060
    From: "CLID NAME" <sip:7806283672 at CUSTOMERIP>;tag=as368d01dd
    To: <sip:14034883602 at sip.ca2.link2voip.com>
    Contact: <sip:7806283672 at CUSTOMERIP>
    Call-ID: 2306e7456006e56f62f3e67673965635 at CUSTOMERIP
    CSeq: 103 INVITE
    User-Agent: Asterisk PBX
    Max-Forwards: 69
    Date: Mon, 10 Jan 2011 23:22:36 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Content-Type: application/sdp
    Content-Length: 235

    v=0
    o=root 12790 12791 IN IP4 CUSTOMERIP
    s=session
    c=IN IP4 66.51.110.210
    t=0 0
    m=audio 14648 RTP/AVP 0 96
    a=rtpmap:0 PCMU/8000
    a=rtpmap:96 telephone-event/8000
    a=fmtp:96 0-16
    a=silenceSupp:off - - - -
    a=nortpproxy:yes
    ------------------------------------------------------------------------
send 503 bytes to udp/[66.51.110.210]:5060 at 23:22:36.415027:
    ------------------------------------------------------------------------
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0
    Via: SIP/2.0/UDP CUSTOMERIP:5060;branch=z9hG4bK0bdd0021;rport=5060
    Record-Route: <sip:66.51.110.210;lr;ftag=as368d01dd>
    From: "CLID NAME" <sip:7806283672 at CUSTOMERIP>;tag=as368d01dd
    To: <sip:14034883602 at sip.ca2.link2voip.com>
    Call-ID: 2306e7456006e56f62f3e67673965635 at CUSTOMERIP
    CSeq: 103 INVITE
    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f 2010-12-29 
13-15-14 -06
00
    Content-Length: 0

    ------------------------------------------------------------------------
2011-01-10 16:22:36.415027 [NOTICE] switch_channel.c:784 New Channel 
sofia/exter
nal/7806283672 at CUSTOMERIP [c58ce64a-c99e-4a20-94a9-680e488f8fab]
2011-01-10 16:22:36.430652 [INFO] mod_dialplan_xml.c:331 Processing CLID 
NAME
<7806283672>->sipuser in context public
2011-01-10 16:22:36.430652 [NOTICE] switch_core_state_machine.c:189 
sofia/extern
al/7806283672 at CUSTOMERIP has executed the last dialplan instruction, hangin
g up.
2011-01-10 16:22:36.430652 [NOTICE] switch_core_state_machine.c:191 
Hangup sofia
/external/7806283672 at CUSTOMERIP [CS_EXECUTE] [NORMAL_CLEARING]
send 818 bytes to udp/[66.51.110.210]:5060 at 23:22:36.430652:
    ------------------------------------------------------------------------
    SIP/2.0 480 Temporarily Unavailable
    Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0
    Via: SIP/2.0/UDP CUSTOMERIP:5060;branch=z9hG4bK0bdd0021;rport=5060
    From: "CLID NAME" <sip:7806283672 at CUSTOMERIP>;tag=as368d01dd
    To: <sip:14034883602 at sip.ca2.link2voip.com>;tag=N9mH5gDHvX0QD
    Call-ID: 2306e7456006e56f62f3e67673965635 at CUSTOMERIP
    CSeq: 103 INVITE
    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f 2010-12-29 
13-15-14 -06
00
    Accept: application/sdp
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, RE
FER, NOTIFY
    Supported: timer, precondition, path, replaces
    Allow-Events: talk, hold, refer
    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
    Content-Length: 0
    Remote-Party-ID: "sipuser" 
<sip:sipuser at 192.168.35.1>;party=calling;privacy=o
ff;screen=no

    ------------------------------------------------------------------------
2011-01-10 16:22:36.430652 [NOTICE] switch_core_session.c:1273 Session 
53 (sofia
/external/7806283672 at CUSTOMERIP) Ended
2011-01-10 16:22:36.430652 [NOTICE] switch_core_session.c:1275 Close 
Channel sof
ia/external/7806283672 at CUSTOMERIP [CS_DESTROY]
recv 367 bytes from udp/[66.51.110.210]:5060 at 23:22:36.696268:
    ------------------------------------------------------------------------
    ACK sip:gw+link2voip2 at 192.168.35.1:5080;transport=udp;gw=link2voip2 
SIP/2.0
    Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0
    From: "CLID NAME" <sip:7806283672 at CUSTOMERIP>;tag=as368d01dd
    Call-ID: 2306e7456006e56f62f3e67673965635 at CUSTOMERIP
    To: <sip:14034883602 at sip.ca2.link2voip.com>;tag=N9mH5gDHvX0QD
    CSeq: 103 ACK
    Content-Length: 0

    ------------------------------------------------------------------------
recv 1230 bytes from udp/[66.51.110.210]:5060 at 23:22:37.789983:
    ------------------------------------------------------------------------
    INVITE 
sip:gw+link2voip2 at 192.168.35.1:5080;transport=udp;gw=link2voip2 SIP/2.
0
    Record-Route: <sip:66.51.110.210;lr;ftag=ZNcB4yyH3SD3e>
    Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0
    Via: SIP/2.0/UDP 
66.51.127.163:5080;rport=5080;branch=z9hG4bK0D66gH7m2614F
    Max-Forwards: 66
    From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=ZNcB4yyH3SD3e
    To: <sip:14034883602 at 66.51.110.210>
    Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad
    CSeq: 7014814 INVITE
    Contact: <sip:ciscosip at 66.51.127.163:5080>
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, 
NOTIFY,
REFER, UPDATE, REGISTER, INFO
    Supported: timer, precondition, path, replaces
    Allow-Events: talk, refer
    Content-Type: application/sdp
    Content-Disposition: session
    Content-Length: 323
    Remote-Party-ID: "CLID NAME" 
<sip:7806283672 at 66.51.127.163>;screen=yes;pri
vacy=off

    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7458343035650957072 
338127983379325658 IN IP4
66.51.127.163
    s=SIP Call
    c=IN IP4 66.51.110.210
    t=0 0
    m=audio 14650 RTP/AVP 0 18 101 13
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=rtpmap:13 CN/8000
    a=ptime:20
    a=nortpproxy:yes
    ------------------------------------------------------------------------
send 494 bytes to udp/[66.51.110.210]:5060 at 23:22:37.789983:
    ------------------------------------------------------------------------
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0
    Via: SIP/2.0/UDP 
66.51.127.163:5080;rport=5080;branch=z9hG4bK0D66gH7m2614F
    Record-Route: <sip:66.51.110.210;lr;ftag=ZNcB4yyH3SD3e>
    From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=ZNcB4yyH3SD3e
    To: <sip:14034883602 at 66.51.110.210>
    Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad
    CSeq: 7014814 INVITE
    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f 2010-12-29 
13-15-14 -06
00
    Content-Length: 0

    ------------------------------------------------------------------------
2011-01-10 16:22:37.789983 [NOTICE] switch_channel.c:784 New Channel 
sofia/exter
nal/7806283672 at 66.51.127.163 [86b78ecd-469f-4a1c-9fe5-692a5941ff37]
2011-01-10 16:22:37.805608 [INFO] mod_dialplan_xml.c:331 Processing CLID 
NAME
<7806283672>->sipuser in context public
2011-01-10 16:22:37.805608 [NOTICE] switch_core_state_machine.c:189 
sofia/extern
al/7806283672 at 66.51.127.163 has executed the last dialplan instruction, 
hanging
up.
2011-01-10 16:22:37.805608 [NOTICE] switch_core_state_machine.c:191 
Hangup sofia
/external/7806283672 at 66.51.127.163 [CS_EXECUTE] [NORMAL_CLEARING]
send 806 bytes to udp/[66.51.110.210]:5060 at 23:22:37.805608:
    ------------------------------------------------------------------------
    SIP/2.0 480 Temporarily Unavailable
    Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0
    Via: SIP/2.0/UDP 
66.51.127.163:5080;rport=5080;branch=z9hG4bK0D66gH7m2614F
    From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=ZNcB4yyH3SD3e
    To: <sip:14034883602 at 66.51.110.210>;tag=pjea7BymS6paS
    Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad
    CSeq: 7014814 INVITE
    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f 2010-12-29 
13-15-14 -06
00
    Accept: application/sdp
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, RE
FER, NOTIFY
    Supported: timer, precondition, path, replaces
    Allow-Events: talk, hold, refer
    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
    Content-Length: 0
    Remote-Party-ID: "sipuser" 
<sip:sipuser at 192.168.35.1>;party=calling;privacy=o
ff;screen=no

    ------------------------------------------------------------------------
2011-01-10 16:22:37.805608 [NOTICE] switch_core_session.c:1273 Session 
54 (sofia
/external/7806283672 at 66.51.127.163) Ended
2011-01-10 16:22:37.805608 [NOTICE] switch_core_session.c:1275 Close 
Channel sof
ia/external/7806283672 at 66.51.127.163 [CS_DESTROY]
recv 352 bytes from udp/[66.51.110.210]:5060 at 23:22:37.914979:
    ------------------------------------------------------------------------
    ACK sip:gw+link2voip2 at 192.168.35.1:5080;transport=udp;gw=link2voip2 
SIP/2.0
    Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0
    From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=ZNcB4yyH3SD3e
    Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad
    To: <sip:14034883602 at 66.51.110.210>;tag=pjea7BymS6paS
    CSeq: 7014814 ACK
    Content-Length: 0

    ------------------------------------------------------------------------
recv 1230 bytes from udp/[66.51.127.173]:5060 at 23:22:37.977477:
    ------------------------------------------------------------------------
    INVITE 
sip:gw+link2voip1 at 192.168.35.1:5080;transport=udp;gw=link2voip1 SIP/2.
0
    Record-Route: <sip:66.51.127.173;lr;ftag=17yv7m0rXBt8N>
    Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0
    Via: SIP/2.0/UDP 
66.51.127.163:5080;rport=5080;branch=z9hG4bK2Zrrm78UvreaQ
    Max-Forwards: 66
    From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=17yv7m0rXBt8N
    To: <sip:14034883602 at 66.51.127.173>
    Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad
    CSeq: 7014814 INVITE
    Contact: <sip:ciscosip at 66.51.127.163:5080>
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, 
NOTIFY,
REFER, UPDATE, REGISTER, INFO
    Supported: timer, precondition, path, replaces
    Allow-Events: talk, refer
    Content-Type: application/sdp
    Content-Disposition: session
    Content-Length: 323
    Remote-Party-ID: "CLID NAME" 
<sip:7806283672 at 66.51.127.163>;screen=yes;pri
vacy=off

    v=0
    o=CiscoSystemsSIP-GW-UserAgent 4919904105816548778 
787843793424096957 IN IP4
66.51.127.163
    s=SIP Call
    c=IN IP4 66.51.127.173
    t=0 0
    m=audio 15488 RTP/AVP 0 18 101 13
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=rtpmap:13 CN/8000
    a=ptime:20
    a=nortpproxy:yes
    ------------------------------------------------------------------------
send 494 bytes to udp/[66.51.127.173]:5060 at 23:22:37.977477:
    ------------------------------------------------------------------------
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0
    Via: SIP/2.0/UDP 
66.51.127.163:5080;rport=5080;branch=z9hG4bK2Zrrm78UvreaQ
    Record-Route: <sip:66.51.127.173;lr;ftag=17yv7m0rXBt8N>
    From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=17yv7m0rXBt8N
    To: <sip:14034883602 at 66.51.127.173>
    Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad
    CSeq: 7014814 INVITE
    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f 2010-12-29 
13-15-14 -06
00
    Content-Length: 0

    ------------------------------------------------------------------------
2011-01-10 16:22:37.977477 [NOTICE] switch_channel.c:784 New Channel 
sofia/exter
nal/7806283672 at 66.51.127.163 [86c939d1-4de0-4a46-9203-518e0d6f7bc5]
2011-01-10 16:22:37.977477 [INFO] mod_dialplan_xml.c:331 Processing CLID 
NAME
<7806283672>->sipuser in context public
2011-01-10 16:22:37.977477 [NOTICE] switch_core_state_machine.c:189 
sofia/extern
al/7806283672 at 66.51.127.163 has executed the last dialplan instruction, 
hanging
up.
2011-01-10 16:22:37.977477 [NOTICE] switch_core_state_machine.c:191 
Hangup sofia
/external/7806283672 at 66.51.127.163 [CS_EXECUTE] [NORMAL_CLEARING]
send 806 bytes to udp/[66.51.127.173]:5060 at 23:22:37.977477:
    ------------------------------------------------------------------------
    SIP/2.0 480 Temporarily Unavailable
    Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0
    Via: SIP/2.0/UDP 
66.51.127.163:5080;rport=5080;branch=z9hG4bK2Zrrm78UvreaQ
    From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=17yv7m0rXBt8N
    To: <sip:14034883602 at 66.51.127.173>;tag=QU7286erpFDXm
    Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad
    CSeq: 7014814 INVITE
    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f 2010-12-29 
13-15-14 -06
00
    Accept: application/sdp
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, RE
FER, NOTIFY
    Supported: timer, precondition, path, replaces
    Allow-Events: talk, hold, refer
    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
    Content-Length: 0
    Remote-Party-ID: "sipuser" 
<sip:sipuser at 192.168.35.1>;party=calling;privacy=o
ff;screen=no

    ------------------------------------------------------------------------
2011-01-10 16:22:37.977477 [NOTICE] switch_core_session.c:1273 Session 
55 (sofia
/external/7806283672 at 66.51.127.163) Ended
2011-01-10 16:22:37.977477 [NOTICE] switch_core_session.c:1275 Close 
Channel sof
ia/external/7806283672 at 66.51.127.163 [CS_DESTROY]
recv 352 bytes from udp/[66.51.127.173]:5060 at 23:22:38.071224:
    ------------------------------------------------------------------------
    ACK sip:gw+link2voip1 at 192.168.35.1:5080;transport=udp;gw=link2voip1 
SIP/2.0
    Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0
    From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=17yv7m0rXBt8N
    Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad
    To: <sip:14034883602 at 66.51.127.173>;tag=QU7286erpFDXm
    CSeq: 7014814 ACK
    Content-Length: 0

    ------------------------------------------------------------------------


On 10/01/2011 2:52 PM, Michael Collins wrote:
> Or just give us your credentials and we'll "test it thoroughly" for 
> you. :)
> -MC
>
> On Mon, Jan 10, 2011 at 1:24 PM, Brian West <brian at freeswitch.org 
> <mailto:brian at freeswitch.org>> wrote:
>
>     can you put up a sip trace or something so we can help guide you?
>
>     /b
>
>     On Jan 10, 2011, at 2:26 PM, Darren Wiebe wrote:
>
>     > Good Afternoon,
>     >
>     > I'm trying to get my freeswitch box talking to Link2voip.  Does
>     anybody
>     > have sample XML files for them?
>     >
>     > --
>     > Darren Wiebe
>     > Aleph Communications
>     > --------------------
>     > Phone: 1-877-702-2900
>     > Fax:   1-866-274-4506
>     > Email: darren at aleph-com.net <mailto:darren at aleph-com.net>
>     >
>     >
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Darren Wiebe
Aleph Communications
--------------------
Phone: 1-877-702-2900
Fax:   1-866-274-4506
Email: darren at aleph-com.net

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From dujinfang at gmail.com  Tue Jan 11 03:19:30 2011
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 11 Jan 2011 08:19:30 +0800
Subject: [Freeswitch-users] playing files through a remote URL
In-Reply-To: <AANLkTik-d6Wb2dyStNSF-rVKCvaHzXq4pWKsYRKA9sf7@mail.gmail.com>
References: <AANLkTikYdXfOVYF6=vUsj8gB7kHUuwOddrYy_dFg6ByG@mail.gmail.com>
	<AANLkTik-d6Wb2dyStNSF-rVKCvaHzXq4pWKsYRKA9sf7@mail.gmail.com>
Message-ID: <AANLkTimCERQqj5aT1B2_QJZ1+MKJ31YLJQ4fqJFOg=5h@mail.gmail.com>

you can plan mp3 in mod_shout, or maybe play from a MRCP server with
mod_unimrcp ?

On Tue, Jan 11, 2011 at 3:09 AM, Holger Esser <hesser4900 at gmail.com> wrote:
> Hello,
> I am trying to write a simple ivr script that answers the call, plays a
> greeting and collects a few digits.
> Is it possible to fetch the audio through a remote URL
> (i.e.?http://messageServer/en.us/greeting.wav)?
> Or does it have to reside on the local file system or a mapped drive?
> Kind regards,
> Holger
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From joaocarlosleme at gmail.com  Tue Jan 11 03:39:59 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Mon, 10 Jan 2011 16:39:59 -0800
Subject: [Freeswitch-users] Remote LogIn to Freeswitch?
Message-ID: <AANLkTinREKsePLFxXHhJobO0VkNBe0zt89araJJdBEnz@mail.gmail.com>

Hi There,
What do I have to do to be able to LogIn to Freeswitch from Home (server is
located at office) starting from the basic/original configuration?
I'm using X-Lite. I've been able to LogIn replacing the internal IP by the
external IP from the Office but the sound is not working so I wanted to know
what are the configuration changes that have to be done to allow it. Do I
have to create a different profile? I want be able to do the same just as if
I was at the Office.
Thanks,
John
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From Nabble at slickdeals.endjunk.com  Tue Jan 11 03:46:54 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 10 Jan 2011 16:46:54 -0800 (PST)
Subject: [Freeswitch-users] Remote LogIn to Freeswitch?
In-Reply-To: <AANLkTinREKsePLFxXHhJobO0VkNBe0zt89araJJdBEnz@mail.gmail.com>
References: <AANLkTinREKsePLFxXHhJobO0VkNBe0zt89araJJdBEnz@mail.gmail.com>
Message-ID: <1294706814105-5909221.post@n2.nabble.com>



Joao Leme wrote:
> What do I have to do to be able to LogIn to Freeswitch from Home (server
> is
> located at office) starting from the basic/original configuration?
You can either use a telnet or ssh to perform a remote login into your FS.
The later one is preferable because it uses a secured shell.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Remote-LogIn-to-Freeswitch-tp5909220p5909221.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From lfurrea at gmail.com  Tue Jan 11 04:09:57 2011
From: lfurrea at gmail.com (Luis F Urrea)
Date: Mon, 10 Jan 2011 19:09:57 -0600
Subject: [Freeswitch-users] FS 502- Bad gateway to Patton SN4960
Message-ID: <AANLkTinGL3Rf9Y0B-bauch1aY=--MYj6rDq5-NdSc36m@mail.gmail.com>

Hi guys,

I have a call coming through a Patton GW in the same LAN as FS, call comes
in and FS responds with 502 to GW.

Here's a console + siptrace log

http://pastebin.freeswitch.org/14978

I appreciate your input and advice on where to direct my troubleshooting
efforts!

TIA
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From msc at freeswitch.org  Tue Jan 11 04:12:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Jan 2011 01:12:36 +0000
Subject: [Freeswitch-users] Link2voip
In-Reply-To: <4D2B9683.2000405@aleph-com.net>
References: <4D2B6B7B.4090102@aleph-com.net>
	<9F467B41-E6CE-4A64-A228-52896E428CD4@freeswitch.org>
	<AANLkTinwz46psUH_KzuEBDqQuXxXGD4QOwrphOA83t3O@mail.gmail.com>
	<4D2B9683.2000405@aleph-com.net>
Message-ID: <AANLkTin4348ir9zSAd=5PaqLjvzCexEATFeviHRsQA0j@mail.gmail.com>

How many different DIDs do you have for this user? Just one? If so can you
not map the user to a specific DID? In any case, throw a "info" app in your
public dialplan and call the DID. You'll see there's all sorts of variables
you can use for routing if you need to.

-MC

On Mon, Jan 10, 2011 at 11:30 PM, Darren Wiebe <darren at aleph-com.net> wrote:

>  Yeah, I bet.  :)  The outgoing call problem was my fault, I had an
> incorrect piece of dialplan.  Here's the trace on an incoming call.  I'm
> trying to get it to come to a particular DID in the public context instead
> of this.
>
> 2011-01-10 16:22:36.430652 [INFO] mod_dialplan_xml.c:331 Processing CLID
> NAME
>  <7806283672>->sipuser in context public
>
> What am I missing?
>
> Here's the relevant provider entry from the external sip profile
>
> <include>
>   <gateway name="link2voip1">
>   <param name="username" value="sipuser"/>
>   <param name="password" value="password"/>
>   <param name="proxy" value="sip.ca1.link2voip.com"/>
>   <param name="register" value="true"/>
>   <param name="register-transport" value="udp"/>
>   <param name="retry-seconds" value="30"/>
>   </gateway>
> </include>
>
> Sip Trace:
>
>    From: "CLID NAME" <sip:7806283672 at CUSTOMERIP>;tag=as01e3f5de
>    Call-ID: 0472a7de73e00fc578b12f6679b6dd9d at CUSTOMERIP
>    To: <sip:14034883602 at sip.ca2.link2voip.com>;tag=m0Ur3NvDZma5H
>    CSeq: 103 ACK
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 942 bytes from udp/[66.51.110.210]:5060 at 23:22:36.415027:
>    ------------------------------------------------------------------------
>    INVITE sip:gw+link2voip2 at 192.168.35.1:5080;transport=udp;gw=link2voip2SIP/2.
> 0
>    Record-Route: <sip:66.51.110.210;lr;ftag=as368d01dd>
>    Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0
>    Via: SIP/2.0/UDP CUSTOMERIP:5060;branch=z9hG4bK0bdd0021;rport=5060
>    From: "CLID NAME" <sip:7806283672 at CUSTOMERIP>;tag=as368d01dd
>    To: <sip:14034883602 at sip.ca2.link2voip.com>
>    Contact: <sip:7806283672 at CUSTOMERIP>
>    Call-ID: 2306e7456006e56f62f3e67673965635 at CUSTOMERIP
>    CSeq: 103 INVITE
>    User-Agent: Asterisk PBX
>    Max-Forwards: 69
>    Date: Mon, 10 Jan 2011 23:22:36 GMT
>    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>    Content-Type: application/sdp
>    Content-Length: 235
>
>    v=0
>    o=root 12790 12791 IN IP4 CUSTOMERIP
>    s=session
>    c=IN IP4 66.51.110.210
>    t=0 0
>    m=audio 14648 RTP/AVP 0 96
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:96 telephone-event/8000
>    a=fmtp:96 0-16
>    a=silenceSupp:off - - - -
>    a=nortpproxy:yes
>    ------------------------------------------------------------------------
> send 503 bytes to udp/[66.51.110.210]:5060 at 23:22:36.415027:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0
>    Via: SIP/2.0/UDP CUSTOMERIP:5060;branch=z9hG4bK0bdd0021;rport=5060
>    Record-Route: <sip:66.51.110.210;lr;ftag=as368d01dd>
>    From: "CLID NAME" <sip:7806283672 at CUSTOMERIP>;tag=as368d01dd
>    To: <sip:14034883602 at sip.ca2.link2voip.com>
>    Call-ID: 2306e7456006e56f62f3e67673965635 at CUSTOMERIP
>    CSeq: 103 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f 2010-12-29
> 13-15-14 -06
> 00
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> 2011-01-10 16:22:36.415027 [NOTICE] switch_channel.c:784 New Channel
> sofia/exter
> nal/7806283672 at CUSTOMERIP [c58ce64a-c99e-4a20-94a9-680e488f8fab]
> 2011-01-10 16:22:36.430652 [INFO] mod_dialplan_xml.c:331 Processing CLID
> NAME
>  <7806283672>->sipuser in context public
> 2011-01-10 16:22:36.430652 [NOTICE] switch_core_state_machine.c:189
> sofia/extern
> al/7806283672 at CUSTOMERIP has executed the last dialplan instruction,
> hangin
> g up.
> 2011-01-10 16:22:36.430652 [NOTICE] switch_core_state_machine.c:191 Hangup
> sofia
> /external/7806283672 at CUSTOMERIP [CS_EXECUTE] [NORMAL_CLEARING]
> send 818 bytes to udp/[66.51.110.210]:5060 at 23:22:36.430652:
>    ------------------------------------------------------------------------
>    SIP/2.0 480 Temporarily Unavailable
>    Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0
>    Via: SIP/2.0/UDP CUSTOMERIP:5060;branch=z9hG4bK0bdd0021;rport=5060
>    From: "CLID NAME" <sip:7806283672 at CUSTOMERIP>;tag=as368d01dd
>    To: <sip:14034883602 at sip.ca2.link2voip.com>;tag=N9mH5gDHvX0QD
>    Call-ID: 2306e7456006e56f62f3e67673965635 at CUSTOMERIP
>    CSeq: 103 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f 2010-12-29
> 13-15-14 -06
> 00
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, RE
> FER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, refer
>    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>    Content-Length: 0
>    Remote-Party-ID: "sipuser" <sip:sipuser at 192.168.35.1>
> ;party=calling;privacy=o
> ff;screen=no
>
>    ------------------------------------------------------------------------
> 2011-01-10 16:22:36.430652 [NOTICE] switch_core_session.c:1273 Session 53
> (sofia
> /external/7806283672 at CUSTOMERIP) Ended
> 2011-01-10 16:22:36.430652 [NOTICE] switch_core_session.c:1275 Close
> Channel sof
> ia/external/7806283672 at CUSTOMERIP [CS_DESTROY]
> recv 367 bytes from udp/[66.51.110.210]:5060 at 23:22:36.696268:
>    ------------------------------------------------------------------------
>    ACK sip:gw+link2voip2 at 192.168.35.1:5080;transport=udp;gw=link2voip2SIP/2.0
>    Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0
>    From: "CLID NAME" <sip:7806283672 at CUSTOMERIP>;tag=as368d01dd
>    Call-ID: 2306e7456006e56f62f3e67673965635 at CUSTOMERIP
>    To: <sip:14034883602 at sip.ca2.link2voip.com>;tag=N9mH5gDHvX0QD
>    CSeq: 103 ACK
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 1230 bytes from udp/[66.51.110.210]:5060 at 23:22:37.789983:
>    ------------------------------------------------------------------------
>    INVITE sip:gw+link2voip2 at 192.168.35.1:5080;transport=udp;gw=link2voip2SIP/2.
> 0
>    Record-Route: <sip:66.51.110.210;lr;ftag=ZNcB4yyH3SD3e>
>    Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0
>    Via: SIP/2.0/UDP 66.51.127.163:5080
> ;rport=5080;branch=z9hG4bK0D66gH7m2614F
>    Max-Forwards: 66
>    From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=ZNcB4yyH3SD3e
>    To: <sip:14034883602 at 66.51.110.210>
>    Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad
>    CSeq: 7014814 INVITE
>    Contact: <sip:ciscosip at 66.51.127.163:5080>
>    User-Agent: Cisco-SIPGateway/IOS-12.x
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY,
> REFER, UPDATE, REGISTER, INFO
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, refer
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 323
>    Remote-Party-ID: "CLID NAME" <sip:7806283672 at 66.51.127.163>
> ;screen=yes;pri
> vacy=off
>
>    v=0
>    o=CiscoSystemsSIP-GW-UserAgent 7458343035650957072 338127983379325658 IN
> IP4
> 66.51.127.163
>    s=SIP Call
>    c=IN IP4 66.51.110.210
>    t=0 0
>    m=audio 14650 RTP/AVP 0 18 101 13
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:18 G729/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=rtpmap:13 CN/8000
>    a=ptime:20
>    a=nortpproxy:yes
>    ------------------------------------------------------------------------
> send 494 bytes to udp/[66.51.110.210]:5060 at 23:22:37.789983:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0
>    Via: SIP/2.0/UDP 66.51.127.163:5080
> ;rport=5080;branch=z9hG4bK0D66gH7m2614F
>    Record-Route: <sip:66.51.110.210;lr;ftag=ZNcB4yyH3SD3e>
>    From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=ZNcB4yyH3SD3e
>    To: <sip:14034883602 at 66.51.110.210>
>    Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad
>    CSeq: 7014814 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f 2010-12-29
> 13-15-14 -06
> 00
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> 2011-01-10 16:22:37.789983 [NOTICE] switch_channel.c:784 New Channel
> sofia/exter
> nal/7806283672 at 66.51.127.163 [86b78ecd-469f-4a1c-9fe5-692a5941ff37]
> 2011-01-10 16:22:37.805608 [INFO] mod_dialplan_xml.c:331 Processing CLID
> NAME
>  <7806283672>->sipuser in context public
> 2011-01-10 16:22:37.805608 [NOTICE] switch_core_state_machine.c:189
> sofia/extern
> al/7806283672 at 66.51.127.163 has executed the last dialplan instruction,
> hanging
> up.
> 2011-01-10 16:22:37.805608 [NOTICE] switch_core_state_machine.c:191 Hangup
> sofia
> /external/7806283672 at 66.51.127.163 [CS_EXECUTE] [NORMAL_CLEARING]
> send 806 bytes to udp/[66.51.110.210]:5060 at 23:22:37.805608:
>    ------------------------------------------------------------------------
>    SIP/2.0 480 Temporarily Unavailable
>    Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0
>    Via: SIP/2.0/UDP 66.51.127.163:5080
> ;rport=5080;branch=z9hG4bK0D66gH7m2614F
>    From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=ZNcB4yyH3SD3e
>    To: <sip:14034883602 at 66.51.110.210>;tag=pjea7BymS6paS
>    Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad
>    CSeq: 7014814 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f 2010-12-29
> 13-15-14 -06
> 00
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, RE
> FER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, refer
>    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>    Content-Length: 0
>    Remote-Party-ID: "sipuser" <sip:sipuser at 192.168.35.1>
> ;party=calling;privacy=o
> ff;screen=no
>
>    ------------------------------------------------------------------------
> 2011-01-10 16:22:37.805608 [NOTICE] switch_core_session.c:1273 Session 54
> (sofia
> /external/7806283672 at 66.51.127.163) Ended
> 2011-01-10 16:22:37.805608 [NOTICE] switch_core_session.c:1275 Close
> Channel sof
> ia/external/7806283672 at 66.51.127.163 [CS_DESTROY]
> recv 352 bytes from udp/[66.51.110.210]:5060 at 23:22:37.914979:
>    ------------------------------------------------------------------------
>    ACK sip:gw+link2voip2 at 192.168.35.1:5080;transport=udp;gw=link2voip2SIP/2.0
>    Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0
>    From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=ZNcB4yyH3SD3e
>    Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad
>    To: <sip:14034883602 at 66.51.110.210>;tag=pjea7BymS6paS
>    CSeq: 7014814 ACK
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 1230 bytes from udp/[66.51.127.173]:5060 at 23:22:37.977477:
>    ------------------------------------------------------------------------
>    INVITE sip:gw+link2voip1 at 192.168.35.1:5080;transport=udp;gw=link2voip1SIP/2.
> 0
>    Record-Route: <sip:66.51.127.173;lr;ftag=17yv7m0rXBt8N>
>    Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0
>    Via: SIP/2.0/UDP 66.51.127.163:5080
> ;rport=5080;branch=z9hG4bK2Zrrm78UvreaQ
>    Max-Forwards: 66
>    From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=17yv7m0rXBt8N
>    To: <sip:14034883602 at 66.51.127.173>
>    Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad
>    CSeq: 7014814 INVITE
>    Contact: <sip:ciscosip at 66.51.127.163:5080>
>    User-Agent: Cisco-SIPGateway/IOS-12.x
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY,
> REFER, UPDATE, REGISTER, INFO
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, refer
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 323
>    Remote-Party-ID: "CLID NAME" <sip:7806283672 at 66.51.127.163>
> ;screen=yes;pri
> vacy=off
>
>    v=0
>    o=CiscoSystemsSIP-GW-UserAgent 4919904105816548778 787843793424096957 IN
> IP4
> 66.51.127.163
>    s=SIP Call
>    c=IN IP4 66.51.127.173
>    t=0 0
>    m=audio 15488 RTP/AVP 0 18 101 13
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:18 G729/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=rtpmap:13 CN/8000
>    a=ptime:20
>    a=nortpproxy:yes
>    ------------------------------------------------------------------------
> send 494 bytes to udp/[66.51.127.173]:5060 at 23:22:37.977477:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0
>    Via: SIP/2.0/UDP 66.51.127.163:5080
> ;rport=5080;branch=z9hG4bK2Zrrm78UvreaQ
>    Record-Route: <sip:66.51.127.173;lr;ftag=17yv7m0rXBt8N>
>    From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=17yv7m0rXBt8N
>    To: <sip:14034883602 at 66.51.127.173>
>    Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad
>    CSeq: 7014814 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f 2010-12-29
> 13-15-14 -06
> 00
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> 2011-01-10 16:22:37.977477 [NOTICE] switch_channel.c:784 New Channel
> sofia/exter
> nal/7806283672 at 66.51.127.163 [86c939d1-4de0-4a46-9203-518e0d6f7bc5]
> 2011-01-10 16:22:37.977477 [INFO] mod_dialplan_xml.c:331 Processing CLID
> NAME
>  <7806283672>->sipuser in context public
> 2011-01-10 16:22:37.977477 [NOTICE] switch_core_state_machine.c:189
> sofia/extern
> al/7806283672 at 66.51.127.163 has executed the last dialplan instruction,
> hanging
> up.
> 2011-01-10 16:22:37.977477 [NOTICE] switch_core_state_machine.c:191 Hangup
> sofia
> /external/7806283672 at 66.51.127.163 [CS_EXECUTE] [NORMAL_CLEARING]
> send 806 bytes to udp/[66.51.127.173]:5060 at 23:22:37.977477:
>    ------------------------------------------------------------------------
>    SIP/2.0 480 Temporarily Unavailable
>    Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0
>    Via: SIP/2.0/UDP 66.51.127.163:5080
> ;rport=5080;branch=z9hG4bK2Zrrm78UvreaQ
>    From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=17yv7m0rXBt8N
>    To: <sip:14034883602 at 66.51.127.173>;tag=QU7286erpFDXm
>    Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad
>    CSeq: 7014814 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f 2010-12-29
> 13-15-14 -06
> 00
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, RE
> FER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, refer
>    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>    Content-Length: 0
>    Remote-Party-ID: "sipuser" <sip:sipuser at 192.168.35.1>
> ;party=calling;privacy=o
> ff;screen=no
>
>    ------------------------------------------------------------------------
> 2011-01-10 16:22:37.977477 [NOTICE] switch_core_session.c:1273 Session 55
> (sofia
> /external/7806283672 at 66.51.127.163) Ended
> 2011-01-10 16:22:37.977477 [NOTICE] switch_core_session.c:1275 Close
> Channel sof
> ia/external/7806283672 at 66.51.127.163 [CS_DESTROY]
> recv 352 bytes from udp/[66.51.127.173]:5060 at 23:22:38.071224:
>    ------------------------------------------------------------------------
>    ACK sip:gw+link2voip1 at 192.168.35.1:5080;transport=udp;gw=link2voip1SIP/2.0
>    Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0
>    From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=17yv7m0rXBt8N
>    Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad
>    To: <sip:14034883602 at 66.51.127.173>;tag=QU7286erpFDXm
>    CSeq: 7014814 ACK
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
>
>
>
> On 10/01/2011 2:52 PM, Michael Collins wrote:
>
> Or just give us your credentials and we'll "test it thoroughly" for you. :)
>
> -MC
>
> On Mon, Jan 10, 2011 at 1:24 PM, Brian West <brian at freeswitch.org> wrote:
>
>> can you put up a sip trace or something so we can help guide you?
>>
>> /b
>>
>> On Jan 10, 2011, at 2:26 PM, Darren Wiebe wrote:
>>
>> > Good Afternoon,
>> >
>> > I'm trying to get my freeswitch box talking to Link2voip.  Does anybody
>> > have sample XML files for them?
>> >
>> > --
>> > Darren Wiebe
>> > Aleph Communications
>> > --------------------
>> > Phone: 1-877-702-2900
>> > Fax:   1-866-274-4506
>> > Email: darren at aleph-com.net
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> Darren Wiebe
> Aleph Communications
> --------------------
> Phone: 1-877-702-2900
> Fax:   1-866-274-4506
> Email: darren at aleph-com.net
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ayhkor at gmail.com  Tue Jan 11 02:40:59 2011
From: ayhkor at gmail.com (deniro)
Date: Mon, 10 Jan 2011 18:40:59 -0500
Subject: [Freeswitch-users] gateway context
Message-ID: <AANLkTikDHenNrxvDb-rVjxWkw+rs8n9Dyu0ehuqD23uG@mail.gmail.com>

Hi,
This is in regards to how to tell my new sip profile or gateway definition
to use different context within new gateway definition.

 In conf/sip_profile/external.xml  I see it tells to use public context
    <settings>
    <param name="context" value="public"/>

I define a new sip_profile or gateway and within that I try to tell to use a
different context like
     conf/sip_profile/external/incoming.xml
     <gateway>
    ...........
    <param name="context" value="incoming"/>
     <gateway/>

 (i put that "incoming" context under dialplan/public/xxx.xml file)
The problem, it still it reads public context from external.xml (checked log
files)
How do I make my gateway/sip profile use a  different context  than the
public context in external.xml file.


thx
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From ayhkor at gmail.com  Tue Jan 11 03:09:20 2011
From: ayhkor at gmail.com (deniro)
Date: Mon, 10 Jan 2011 19:09:20 -0500
Subject: [Freeswitch-users] conference pin
In-Reply-To: <AANLkTin3F-TXJp_Mv-8vwGbkLmAwbi9mE2bobiB2uyQx@mail.gmail.com>
References: <AANLkTi=q5xpnW7HdGVZYpQbUNi2m=dvmFXX6TTKGvxZo@mail.gmail.com>
	<AANLkTin3F-TXJp_Mv-8vwGbkLmAwbi9mE2bobiB2uyQx@mail.gmail.com>
Message-ID: <AANLkTimSRbCLU_+F3TfR7eK9NXTx=wiVUnVJ-7-yxNqv@mail.gmail.com>

Hi
any expamples of how to store asked pin (phone call) by channel var and
re-use it in conferencing to go to that conference?
thx


On Mon, Jan 10, 2011 at 2:09 AM, Sam <u2nsam at gmail.com> wrote:

> use channels variables in freeswitch.
>
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#Channel_Variable_Manipulation
>
> Regds
> Sam
>
>
>
>   On Mon, Jan 10, 2011 at 9:35 AM, deniro <ayhkor at gmail.com> wrote:
>
>>   Hi
>> using conferencing software and with the phone dialing,
>> entering pin number it will go to a conference  identified by pin
>> in its default format  it is "conference at profile+pin"
>> in my case it will be "pin at profile+pin" since conference=pin.
>> how do I do that? how do I provide a pin that takes me to conference which
>> is
>> identified by pin?
>> thx
>> deniro--
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From darren at aleph-com.net  Tue Jan 11 04:42:38 2011
From: darren at aleph-com.net (Darren Wiebe)
Date: Mon, 10 Jan 2011 18:42:38 -0700
Subject: [Freeswitch-users] Link2voip
In-Reply-To: <AANLkTin4348ir9zSAd=5PaqLjvzCexEATFeviHRsQA0j@mail.gmail.com>
References: <4D2B6B7B.4090102@aleph-com.net>	<9F467B41-E6CE-4A64-A228-52896E428CD4@freeswitch.org>	<AANLkTinwz46psUH_KzuEBDqQuXxXGD4QOwrphOA83t3O@mail.gmail.com>	<4D2B9683.2000405@aleph-com.net>
	<AANLkTin4348ir9zSAd=5PaqLjvzCexEATFeviHRsQA0j@mail.gmail.com>
Message-ID: <4D2BB58E.4010002@aleph-com.net>

this user we've only got 6.  Thanks, that's a good idea.  I'll try it.

Darren Wiebe

On 11-01-10 06:12 PM, Michael Collins wrote:
> How many different DIDs do you have for this user? Just one? If so can 
> you not map the user to a specific DID? In any case, throw a "info" 
> app in your public dialplan and call the DID. You'll see there's all 
> sorts of variables you can use for routing if you need to.
>
> -MC
>
> On Mon, Jan 10, 2011 at 11:30 PM, Darren Wiebe <darren at aleph-com.net 
> <mailto:darren at aleph-com.net>> wrote:
>
>     Yeah, I bet.  :)  The outgoing call problem was my fault, I had an
>     incorrect piece of dialplan.  Here's the trace on an incoming
>     call.  I'm trying to get it to come to a particular DID in the
>     public context instead of this.
>
>     2011-01-10 16:22:36.430652 [INFO] mod_dialplan_xml.c:331
>     Processing CLID NAME
>     <7806283672>->sipuser in context public
>
>     What am I missing?
>
>     Here's the relevant provider entry from the external sip profile
>
>     <include>
>     <gateway name="link2voip1">
>     <param name="username" value="sipuser"/>
>     <param name="password" value="password"/>
>     <param name="proxy" value="sip.ca1.link2voip.com
>     <http://sip.ca1.link2voip.com>"/>
>     <param name="register" value="true"/>
>     <param name="register-transport" value="udp"/>
>     <param name="retry-seconds" value="30"/>
>     </gateway>
>     </include>
>
>     Sip Trace:
>
>        From: "CLID NAME" <sip:7806283672 at CUSTOMERIP>;tag=as01e3f5de
>        Call-ID: 0472a7de73e00fc578b12f6679b6dd9d at CUSTOMERIP
>        To: <sip:14034883602 at sip.ca2.link2voip.com>;tag=m0Ur3NvDZma5H
>        CSeq: 103 ACK
>        Content-Length: 0
>
>       
>     ------------------------------------------------------------------------
>     recv 942 bytes from udp/[66.51.110.210]:5060 at 23:22:36.415027:
>       
>     ------------------------------------------------------------------------
>        INVITE
>     sip:gw+link2voip2 at 192.168.35.1:5080;transport=udp;gw=link2voip2 SIP/2.
>     0
>        Record-Route: <sip:66.51.110.210;lr;ftag=as368d01dd>
>        Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0
>        Via: SIP/2.0/UDP CUSTOMERIP:5060;branch=z9hG4bK0bdd0021;rport=5060
>        From: "CLID NAME" <sip:7806283672 at CUSTOMERIP>;tag=as368d01dd
>        To: <sip:14034883602 at sip.ca2.link2voip.com>
>        Contact: <sip:7806283672 at CUSTOMERIP>
>        Call-ID: 2306e7456006e56f62f3e67673965635 at CUSTOMERIP
>        CSeq: 103 INVITE
>        User-Agent: Asterisk PBX
>        Max-Forwards: 69
>        Date: Mon, 10 Jan 2011 23:22:36 GMT
>        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>        Content-Type: application/sdp
>        Content-Length: 235
>
>        v=0
>        o=root 12790 12791 IN IP4 CUSTOMERIP
>        s=session
>        c=IN IP4 66.51.110.210
>        t=0 0
>        m=audio 14648 RTP/AVP 0 96
>        a=rtpmap:0 PCMU/8000
>        a=rtpmap:96 telephone-event/8000
>        a=fmtp:96 0-16
>        a=silenceSupp:off - - - -
>        a=nortpproxy:yes
>       
>     ------------------------------------------------------------------------
>     send 503 bytes to udp/[66.51.110.210]:5060 at 23:22:36.415027:
>       
>     ------------------------------------------------------------------------
>        SIP/2.0 100 Trying
>        Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0
>        Via: SIP/2.0/UDP CUSTOMERIP:5060;branch=z9hG4bK0bdd0021;rport=5060
>        Record-Route: <sip:66.51.110.210;lr;ftag=as368d01dd>
>        From: "CLID NAME" <sip:7806283672 at CUSTOMERIP>;tag=as368d01dd
>        To: <sip:14034883602 at sip.ca2.link2voip.com>
>        Call-ID: 2306e7456006e56f62f3e67673965635 at CUSTOMERIP
>        CSeq: 103 INVITE
>        User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f
>     2010-12-29 13-15-14 -06
>     00
>        Content-Length: 0
>
>       
>     ------------------------------------------------------------------------
>     2011-01-10 16:22:36.415027 [NOTICE] switch_channel.c:784 New
>     Channel sofia/exter
>     nal/7806283672 at CUSTOMERIP [c58ce64a-c99e-4a20-94a9-680e488f8fab]
>     2011-01-10 16:22:36.430652 [INFO] mod_dialplan_xml.c:331
>     Processing CLID NAME
>     <7806283672>->sipuser in context public
>     2011-01-10 16:22:36.430652 [NOTICE]
>     switch_core_state_machine.c:189 sofia/extern
>     al/7806283672 at CUSTOMERIP has executed the last dialplan
>     instruction, hangin
>     g up.
>     2011-01-10 16:22:36.430652 [NOTICE]
>     switch_core_state_machine.c:191 Hangup sofia
>     /external/7806283672 at CUSTOMERIP [CS_EXECUTE] [NORMAL_CLEARING]
>     send 818 bytes to udp/[66.51.110.210]:5060 at 23:22:36.430652:
>       
>     ------------------------------------------------------------------------
>        SIP/2.0 480 Temporarily Unavailable
>        Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0
>        Via: SIP/2.0/UDP CUSTOMERIP:5060;branch=z9hG4bK0bdd0021;rport=5060
>        From: "CLID NAME" <sip:7806283672 at CUSTOMERIP>;tag=as368d01dd
>        To: <sip:14034883602 at sip.ca2.link2voip.com>;tag=N9mH5gDHvX0QD
>        Call-ID: 2306e7456006e56f62f3e67673965635 at CUSTOMERIP
>        CSeq: 103 INVITE
>        User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f
>     2010-12-29 13-15-14 -06
>     00
>        Accept: application/sdp
>        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
>     INFO, REGISTER, RE
>     FER, NOTIFY
>        Supported: timer, precondition, path, replaces
>        Allow-Events: talk, hold, refer
>        Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>        Content-Length: 0
>        Remote-Party-ID: "sipuser"
>     <sip:sipuser at 192.168.35.1>;party=calling;privacy=o
>     ff;screen=no
>
>       
>     ------------------------------------------------------------------------
>     2011-01-10 16:22:36.430652 [NOTICE] switch_core_session.c:1273
>     Session 53 (sofia
>     /external/7806283672 at CUSTOMERIP) Ended
>     2011-01-10 16:22:36.430652 [NOTICE] switch_core_session.c:1275
>     Close Channel sof
>     ia/external/7806283672 at CUSTOMERIP [CS_DESTROY]
>     recv 367 bytes from udp/[66.51.110.210]:5060 at 23:22:36.696268:
>       
>     ------------------------------------------------------------------------
>        ACK
>     sip:gw+link2voip2 at 192.168.35.1:5080;transport=udp;gw=link2voip2
>     SIP/2.0
>        Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0
>        From: "CLID NAME" <sip:7806283672 at CUSTOMERIP>;tag=as368d01dd
>        Call-ID: 2306e7456006e56f62f3e67673965635 at CUSTOMERIP
>        To: <sip:14034883602 at sip.ca2.link2voip.com>;tag=N9mH5gDHvX0QD
>        CSeq: 103 ACK
>        Content-Length: 0
>
>       
>     ------------------------------------------------------------------------
>     recv 1230 bytes from udp/[66.51.110.210]:5060 at 23:22:37.789983:
>       
>     ------------------------------------------------------------------------
>        INVITE
>     sip:gw+link2voip2 at 192.168.35.1:5080;transport=udp;gw=link2voip2 SIP/2.
>     0
>        Record-Route: <sip:66.51.110.210;lr;ftag=ZNcB4yyH3SD3e>
>        Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0
>        Via: SIP/2.0/UDP
>     66.51.127.163:5080;rport=5080;branch=z9hG4bK0D66gH7m2614F
>        Max-Forwards: 66
>        From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=ZNcB4yyH3SD3e
>        To: <sip:14034883602 at 66.51.110.210>
>        Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad
>        CSeq: 7014814 INVITE
>        Contact: <sip:ciscosip at 66.51.127.163:5080>
>        User-Agent: Cisco-SIPGateway/IOS-12.x
>        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,
>     SUBSCRIBE, NOTIFY,
>     REFER, UPDATE, REGISTER, INFO
>        Supported: timer, precondition, path, replaces
>        Allow-Events: talk, refer
>        Content-Type: application/sdp
>        Content-Disposition: session
>        Content-Length: 323
>        Remote-Party-ID: "CLID NAME"
>     <sip:7806283672 at 66.51.127.163>;screen=yes;pri
>     vacy=off
>
>        v=0
>        o=CiscoSystemsSIP-GW-UserAgent 7458343035650957072
>     338127983379325658 IN IP4
>     66.51.127.163
>        s=SIP Call
>        c=IN IP4 66.51.110.210
>        t=0 0
>        m=audio 14650 RTP/AVP 0 18 101 13
>        a=rtpmap:0 PCMU/8000
>        a=rtpmap:18 G729/8000
>        a=rtpmap:101 telephone-event/8000
>        a=fmtp:101 0-16
>        a=rtpmap:13 CN/8000
>        a=ptime:20
>        a=nortpproxy:yes
>       
>     ------------------------------------------------------------------------
>     send 494 bytes to udp/[66.51.110.210]:5060 at 23:22:37.789983:
>       
>     ------------------------------------------------------------------------
>        SIP/2.0 100 Trying
>        Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0
>        Via: SIP/2.0/UDP
>     66.51.127.163:5080;rport=5080;branch=z9hG4bK0D66gH7m2614F
>        Record-Route: <sip:66.51.110.210;lr;ftag=ZNcB4yyH3SD3e>
>        From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=ZNcB4yyH3SD3e
>        To: <sip:14034883602 at 66.51.110.210>
>        Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad
>        CSeq: 7014814 INVITE
>        User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f
>     2010-12-29 13-15-14 -06
>     00
>        Content-Length: 0
>
>       
>     ------------------------------------------------------------------------
>     2011-01-10 16:22:37.789983 [NOTICE] switch_channel.c:784 New
>     Channel sofia/exter
>     nal/7806283672 at 66.51.127.163 <mailto:nal/7806283672 at 66.51.127.163>
>     [86b78ecd-469f-4a1c-9fe5-692a5941ff37]
>     2011-01-10 16:22:37.805608 [INFO] mod_dialplan_xml.c:331
>     Processing CLID NAME
>     <7806283672>->sipuser in context public
>     2011-01-10 16:22:37.805608 [NOTICE]
>     switch_core_state_machine.c:189 sofia/extern
>     al/7806283672 at 66.51.127.163 <mailto:al/7806283672 at 66.51.127.163>
>     has executed the last dialplan instruction, hanging
>     up.
>     2011-01-10 16:22:37.805608 [NOTICE]
>     switch_core_state_machine.c:191 Hangup sofia
>     /external/7806283672 at 66.51.127.163
>     <mailto:7806283672 at 66.51.127.163> [CS_EXECUTE] [NORMAL_CLEARING]
>     send 806 bytes to udp/[66.51.110.210]:5060 at 23:22:37.805608:
>       
>     ------------------------------------------------------------------------
>        SIP/2.0 480 Temporarily Unavailable
>        Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0
>        Via: SIP/2.0/UDP
>     66.51.127.163:5080;rport=5080;branch=z9hG4bK0D66gH7m2614F
>        From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=ZNcB4yyH3SD3e
>        To: <sip:14034883602 at 66.51.110.210>;tag=pjea7BymS6paS
>        Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad
>        CSeq: 7014814 INVITE
>        User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f
>     2010-12-29 13-15-14 -06
>     00
>        Accept: application/sdp
>        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
>     INFO, REGISTER, RE
>     FER, NOTIFY
>        Supported: timer, precondition, path, replaces
>        Allow-Events: talk, hold, refer
>        Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>        Content-Length: 0
>        Remote-Party-ID: "sipuser"
>     <sip:sipuser at 192.168.35.1>;party=calling;privacy=o
>     ff;screen=no
>
>       
>     ------------------------------------------------------------------------
>     2011-01-10 16:22:37.805608 [NOTICE] switch_core_session.c:1273
>     Session 54 (sofia
>     /external/7806283672 at 66.51.127.163
>     <mailto:7806283672 at 66.51.127.163>) Ended
>     2011-01-10 16:22:37.805608 [NOTICE] switch_core_session.c:1275
>     Close Channel sof
>     ia/external/7806283672 at 66.51.127.163
>     <mailto:ia/external/7806283672 at 66.51.127.163> [CS_DESTROY]
>     recv 352 bytes from udp/[66.51.110.210]:5060 at 23:22:37.914979:
>       
>     ------------------------------------------------------------------------
>        ACK
>     sip:gw+link2voip2 at 192.168.35.1:5080;transport=udp;gw=link2voip2
>     SIP/2.0
>        Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0
>        From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=ZNcB4yyH3SD3e
>        Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad
>        To: <sip:14034883602 at 66.51.110.210>;tag=pjea7BymS6paS
>        CSeq: 7014814 ACK
>        Content-Length: 0
>
>       
>     ------------------------------------------------------------------------
>     recv 1230 bytes from udp/[66.51.127.173]:5060 at 23:22:37.977477:
>       
>     ------------------------------------------------------------------------
>        INVITE
>     sip:gw+link2voip1 at 192.168.35.1:5080;transport=udp;gw=link2voip1 SIP/2.
>     0
>        Record-Route: <sip:66.51.127.173;lr;ftag=17yv7m0rXBt8N>
>        Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0
>        Via: SIP/2.0/UDP
>     66.51.127.163:5080;rport=5080;branch=z9hG4bK2Zrrm78UvreaQ
>        Max-Forwards: 66
>        From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=17yv7m0rXBt8N
>        To: <sip:14034883602 at 66.51.127.173>
>        Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad
>        CSeq: 7014814 INVITE
>        Contact: <sip:ciscosip at 66.51.127.163:5080>
>        User-Agent: Cisco-SIPGateway/IOS-12.x
>        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,
>     SUBSCRIBE, NOTIFY,
>     REFER, UPDATE, REGISTER, INFO
>        Supported: timer, precondition, path, replaces
>        Allow-Events: talk, refer
>        Content-Type: application/sdp
>        Content-Disposition: session
>        Content-Length: 323
>        Remote-Party-ID: "CLID NAME"
>     <sip:7806283672 at 66.51.127.163>;screen=yes;pri
>     vacy=off
>
>        v=0
>        o=CiscoSystemsSIP-GW-UserAgent 4919904105816548778
>     787843793424096957 IN IP4
>     66.51.127.163
>        s=SIP Call
>        c=IN IP4 66.51.127.173
>        t=0 0
>        m=audio 15488 RTP/AVP 0 18 101 13
>        a=rtpmap:0 PCMU/8000
>        a=rtpmap:18 G729/8000
>        a=rtpmap:101 telephone-event/8000
>        a=fmtp:101 0-16
>        a=rtpmap:13 CN/8000
>        a=ptime:20
>        a=nortpproxy:yes
>       
>     ------------------------------------------------------------------------
>     send 494 bytes to udp/[66.51.127.173]:5060 at 23:22:37.977477:
>       
>     ------------------------------------------------------------------------
>        SIP/2.0 100 Trying
>        Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0
>        Via: SIP/2.0/UDP
>     66.51.127.163:5080;rport=5080;branch=z9hG4bK2Zrrm78UvreaQ
>        Record-Route: <sip:66.51.127.173;lr;ftag=17yv7m0rXBt8N>
>        From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=17yv7m0rXBt8N
>        To: <sip:14034883602 at 66.51.127.173>
>        Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad
>        CSeq: 7014814 INVITE
>        User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f
>     2010-12-29 13-15-14 -06
>     00
>        Content-Length: 0
>
>       
>     ------------------------------------------------------------------------
>     2011-01-10 16:22:37.977477 [NOTICE] switch_channel.c:784 New
>     Channel sofia/exter
>     nal/7806283672 at 66.51.127.163 <mailto:nal/7806283672 at 66.51.127.163>
>     [86c939d1-4de0-4a46-9203-518e0d6f7bc5]
>     2011-01-10 16:22:37.977477 [INFO] mod_dialplan_xml.c:331
>     Processing CLID NAME
>     <7806283672>->sipuser in context public
>     2011-01-10 16:22:37.977477 [NOTICE]
>     switch_core_state_machine.c:189 sofia/extern
>     al/7806283672 at 66.51.127.163 <mailto:al/7806283672 at 66.51.127.163>
>     has executed the last dialplan instruction, hanging
>     up.
>     2011-01-10 16:22:37.977477 [NOTICE]
>     switch_core_state_machine.c:191 Hangup sofia
>     /external/7806283672 at 66.51.127.163
>     <mailto:7806283672 at 66.51.127.163> [CS_EXECUTE] [NORMAL_CLEARING]
>     send 806 bytes to udp/[66.51.127.173]:5060 at 23:22:37.977477:
>       
>     ------------------------------------------------------------------------
>        SIP/2.0 480 Temporarily Unavailable
>        Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0
>        Via: SIP/2.0/UDP
>     66.51.127.163:5080;rport=5080;branch=z9hG4bK2Zrrm78UvreaQ
>        From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=17yv7m0rXBt8N
>        To: <sip:14034883602 at 66.51.127.173>;tag=QU7286erpFDXm
>        Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad
>        CSeq: 7014814 INVITE
>        User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f
>     2010-12-29 13-15-14 -06
>     00
>        Accept: application/sdp
>        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
>     INFO, REGISTER, RE
>     FER, NOTIFY
>        Supported: timer, precondition, path, replaces
>        Allow-Events: talk, hold, refer
>        Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>        Content-Length: 0
>        Remote-Party-ID: "sipuser"
>     <sip:sipuser at 192.168.35.1>;party=calling;privacy=o
>     ff;screen=no
>
>       
>     ------------------------------------------------------------------------
>     2011-01-10 16:22:37.977477 [NOTICE] switch_core_session.c:1273
>     Session 55 (sofia
>     /external/7806283672 at 66.51.127.163
>     <mailto:7806283672 at 66.51.127.163>) Ended
>     2011-01-10 16:22:37.977477 [NOTICE] switch_core_session.c:1275
>     Close Channel sof
>     ia/external/7806283672 at 66.51.127.163
>     <mailto:ia/external/7806283672 at 66.51.127.163> [CS_DESTROY]
>     recv 352 bytes from udp/[66.51.127.173]:5060 at 23:22:38.071224:
>       
>     ------------------------------------------------------------------------
>        ACK
>     sip:gw+link2voip1 at 192.168.35.1:5080;transport=udp;gw=link2voip1
>     SIP/2.0
>        Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0
>        From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=17yv7m0rXBt8N
>        Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad
>        To: <sip:14034883602 at 66.51.127.173>;tag=QU7286erpFDXm
>        CSeq: 7014814 ACK
>        Content-Length: 0
>
>       
>     ------------------------------------------------------------------------
>
>
>
>
>     On 10/01/2011 2:52 PM, Michael Collins wrote:
>>     Or just give us your credentials and we'll "test it thoroughly"
>>     for you. :)
>>     -MC
>>
>>     On Mon, Jan 10, 2011 at 1:24 PM, Brian West <brian at freeswitch.org
>>     <mailto:brian at freeswitch.org>> wrote:
>>
>>         can you put up a sip trace or something so we can help guide you?
>>
>>         /b
>>
>>         On Jan 10, 2011, at 2:26 PM, Darren Wiebe wrote:
>>
>>         > Good Afternoon,
>>         >
>>         > I'm trying to get my freeswitch box talking to Link2voip.
>>          Does anybody
>>         > have sample XML files for them?
>>         >
>>         > --
>>         > Darren Wiebe
>>         > Aleph Communications
>>         > --------------------
>>         > Phone: 1-877-702-2900
>>         > Fax:   1-866-274-4506
>>         > Email: darren at aleph-com.net <mailto:darren at aleph-com.net>
>>         >
>>         >
>>         > _______________________________________________
>>         > FreeSWITCH-users mailing list
>>         > FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         >
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         > http://www.freeswitch.org
>>
>>
>>         _______________________________________________
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     -- 
>     Darren Wiebe
>     Aleph Communications
>     --------------------
>     Phone: 1-877-702-2900
>     Fax:   1-866-274-4506
>     Email:darren at aleph-com.net  <mailto:darren at aleph-com.net>
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Tue Jan 11 05:14:15 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Jan 2011 20:14:15 -0600
Subject: [Freeswitch-users] gateway context
In-Reply-To: <AANLkTikDHenNrxvDb-rVjxWkw+rs8n9Dyu0ehuqD23uG@mail.gmail.com>
References: <AANLkTikDHenNrxvDb-rVjxWkw+rs8n9Dyu0ehuqD23uG@mail.gmail.com>
Message-ID: <79CED570-0B25-4761-A51D-F4B521DDEB8F@freeswitch.org>

Chances are your provider doesn't call the contact you registered with otherwise it would work.

Can you show us the sip dialog of the register and an inbound invite?

/b

On Jan 10, 2011, at 5:40 PM, deniro wrote:

> How do I make my gateway/sip profile use a  different context  than the public context in external.xml file.
>  
>  



From Nabble at slickdeals.endjunk.com  Tue Jan 11 05:50:42 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 10 Jan 2011 18:50:42 -0800 (PST)
Subject: [Freeswitch-users] Link2voip
In-Reply-To: <4D2B6B7B.4090102@aleph-com.net>
References: <4D2B6B7B.4090102@aleph-com.net>
Message-ID: <1294714242894-5909450.post@n2.nabble.com>



Darren Wiebe wrote:
> I'm trying to get my freeswitch box talking to Link2voip.  Does anybody 
> have sample XML files for them?
Have you read the FS wiki on Provider Configuration Examples for 
http://wiki.freeswitch.org/wiki/Provider_Configuration:_Link2VoIP Link2VoIP
?

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Link2voip-tp5908479p5909450.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From darren at aleph-com.net  Tue Jan 11 05:49:23 2011
From: darren at aleph-com.net (Darren Wiebe)
Date: Mon, 10 Jan 2011 19:49:23 -0700
Subject: [Freeswitch-users] Link2voip
In-Reply-To: <AANLkTin4348ir9zSAd=5PaqLjvzCexEATFeviHRsQA0j@mail.gmail.com>
References: <4D2B6B7B.4090102@aleph-com.net>	<9F467B41-E6CE-4A64-A228-52896E428CD4@freeswitch.org>	<AANLkTinwz46psUH_KzuEBDqQuXxXGD4QOwrphOA83t3O@mail.gmail.com>	<4D2B9683.2000405@aleph-com.net>
	<AANLkTin4348ir9zSAd=5PaqLjvzCexEATFeviHRsQA0j@mail.gmail.com>
Message-ID: <4D2BC533.8070002@aleph-com.net>

Thanks for your help. I've updated the wiki page at 
http://wiki.freeswitch.org/wiki/Provider_Configuration:_Link2VoIP#Link2VoIP

Darren Wiebe

On 10/01/2011 6:12 PM, Michael Collins wrote:
> How many different DIDs do you have for this user? Just one? If so can 
> you not map the user to a specific DID? In any case, throw a "info" 
> app in your public dialplan and call the DID. You'll see there's all 
> sorts of variables you can use for routing if you need to.
>
> -MC
>
> On Mon, Jan 10, 2011 at 11:30 PM, Darren Wiebe <darren at aleph-com.net 
> <mailto:darren at aleph-com.net>> wrote:
>
>     Yeah, I bet.  :)  The outgoing call problem was my fault, I had an
>     incorrect piece of dialplan.  Here's the trace on an incoming
>     call.  I'm trying to get it to come to a particular DID in the
>     public context instead of this.
>
>     2011-01-10 16:22:36.430652 [INFO] mod_dialplan_xml.c:331
>     Processing CLID NAME
>     <7806283672>->sipuser in context public
>
>     What am I missing?
>
>     Here's the relevant provider entry from the external sip profile
>
>     <include>
>     <gateway name="link2voip1">
>     <param name="username" value="sipuser"/>
>     <param name="password" value="password"/>
>     <param name="proxy" value="sip.ca1.link2voip.com
>     <http://sip.ca1.link2voip.com>"/>
>     <param name="register" value="true"/>
>     <param name="register-transport" value="udp"/>
>     <param name="retry-seconds" value="30"/>
>     </gateway>
>     </include>
>
>     Sip Trace:
>
>        From: "CLID NAME" <sip:7806283672 at CUSTOMERIP>;tag=as01e3f5de
>        Call-ID: 0472a7de73e00fc578b12f6679b6dd9d at CUSTOMERIP
>        To: <sip:14034883602 at sip.ca2.link2voip.com>;tag=m0Ur3NvDZma5H
>        CSeq: 103 ACK
>        Content-Length: 0
>
>       
>     ------------------------------------------------------------------------
>     recv 942 bytes from udp/[66.51.110.210]:5060 at 23:22:36.415027:
>       
>     ------------------------------------------------------------------------
>        INVITE
>     sip:gw+link2voip2 at 192.168.35.1:5080;transport=udp;gw=link2voip2 SIP/2.
>     0
>        Record-Route: <sip:66.51.110.210;lr;ftag=as368d01dd>
>        Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0
>        Via: SIP/2.0/UDP CUSTOMERIP:5060;branch=z9hG4bK0bdd0021;rport=5060
>        From: "CLID NAME" <sip:7806283672 at CUSTOMERIP>;tag=as368d01dd
>        To: <sip:14034883602 at sip.ca2.link2voip.com>
>        Contact: <sip:7806283672 at CUSTOMERIP>
>        Call-ID: 2306e7456006e56f62f3e67673965635 at CUSTOMERIP
>        CSeq: 103 INVITE
>        User-Agent: Asterisk PBX
>        Max-Forwards: 69
>        Date: Mon, 10 Jan 2011 23:22:36 GMT
>        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>        Content-Type: application/sdp
>        Content-Length: 235
>
>        v=0
>        o=root 12790 12791 IN IP4 CUSTOMERIP
>        s=session
>        c=IN IP4 66.51.110.210
>        t=0 0
>        m=audio 14648 RTP/AVP 0 96
>        a=rtpmap:0 PCMU/8000
>        a=rtpmap:96 telephone-event/8000
>        a=fmtp:96 0-16
>        a=silenceSupp:off - - - -
>        a=nortpproxy:yes
>       
>     ------------------------------------------------------------------------
>     send 503 bytes to udp/[66.51.110.210]:5060 at 23:22:36.415027:
>       
>     ------------------------------------------------------------------------
>        SIP/2.0 100 Trying
>        Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0
>        Via: SIP/2.0/UDP CUSTOMERIP:5060;branch=z9hG4bK0bdd0021;rport=5060
>        Record-Route: <sip:66.51.110.210;lr;ftag=as368d01dd>
>        From: "CLID NAME" <sip:7806283672 at CUSTOMERIP>;tag=as368d01dd
>        To: <sip:14034883602 at sip.ca2.link2voip.com>
>        Call-ID: 2306e7456006e56f62f3e67673965635 at CUSTOMERIP
>        CSeq: 103 INVITE
>        User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f
>     2010-12-29 13-15-14 -06
>     00
>        Content-Length: 0
>
>       
>     ------------------------------------------------------------------------
>     2011-01-10 16:22:36.415027 [NOTICE] switch_channel.c:784 New
>     Channel sofia/exter
>     nal/7806283672 at CUSTOMERIP [c58ce64a-c99e-4a20-94a9-680e488f8fab]
>     2011-01-10 16:22:36.430652 [INFO] mod_dialplan_xml.c:331
>     Processing CLID NAME
>     <7806283672>->sipuser in context public
>     2011-01-10 16:22:36.430652 [NOTICE]
>     switch_core_state_machine.c:189 sofia/extern
>     al/7806283672 at CUSTOMERIP has executed the last dialplan
>     instruction, hangin
>     g up.
>     2011-01-10 16:22:36.430652 [NOTICE]
>     switch_core_state_machine.c:191 Hangup sofia
>     /external/7806283672 at CUSTOMERIP [CS_EXECUTE] [NORMAL_CLEARING]
>     send 818 bytes to udp/[66.51.110.210]:5060 at 23:22:36.430652:
>       
>     ------------------------------------------------------------------------
>        SIP/2.0 480 Temporarily Unavailable
>        Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0
>        Via: SIP/2.0/UDP CUSTOMERIP:5060;branch=z9hG4bK0bdd0021;rport=5060
>        From: "CLID NAME" <sip:7806283672 at CUSTOMERIP>;tag=as368d01dd
>        To: <sip:14034883602 at sip.ca2.link2voip.com>;tag=N9mH5gDHvX0QD
>        Call-ID: 2306e7456006e56f62f3e67673965635 at CUSTOMERIP
>        CSeq: 103 INVITE
>        User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f
>     2010-12-29 13-15-14 -06
>     00
>        Accept: application/sdp
>        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
>     INFO, REGISTER, RE
>     FER, NOTIFY
>        Supported: timer, precondition, path, replaces
>        Allow-Events: talk, hold, refer
>        Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>        Content-Length: 0
>        Remote-Party-ID: "sipuser"
>     <sip:sipuser at 192.168.35.1>;party=calling;privacy=o
>     ff;screen=no
>
>       
>     ------------------------------------------------------------------------
>     2011-01-10 16:22:36.430652 [NOTICE] switch_core_session.c:1273
>     Session 53 (sofia
>     /external/7806283672 at CUSTOMERIP) Ended
>     2011-01-10 16:22:36.430652 [NOTICE] switch_core_session.c:1275
>     Close Channel sof
>     ia/external/7806283672 at CUSTOMERIP [CS_DESTROY]
>     recv 367 bytes from udp/[66.51.110.210]:5060 at 23:22:36.696268:
>       
>     ------------------------------------------------------------------------
>        ACK
>     sip:gw+link2voip2 at 192.168.35.1:5080;transport=udp;gw=link2voip2
>     SIP/2.0
>        Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0
>        From: "CLID NAME" <sip:7806283672 at CUSTOMERIP>;tag=as368d01dd
>        Call-ID: 2306e7456006e56f62f3e67673965635 at CUSTOMERIP
>        To: <sip:14034883602 at sip.ca2.link2voip.com>;tag=N9mH5gDHvX0QD
>        CSeq: 103 ACK
>        Content-Length: 0
>
>       
>     ------------------------------------------------------------------------
>     recv 1230 bytes from udp/[66.51.110.210]:5060 at 23:22:37.789983:
>       
>     ------------------------------------------------------------------------
>        INVITE
>     sip:gw+link2voip2 at 192.168.35.1:5080;transport=udp;gw=link2voip2 SIP/2.
>     0
>        Record-Route: <sip:66.51.110.210;lr;ftag=ZNcB4yyH3SD3e>
>        Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0
>        Via: SIP/2.0/UDP
>     66.51.127.163:5080;rport=5080;branch=z9hG4bK0D66gH7m2614F
>        Max-Forwards: 66
>        From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=ZNcB4yyH3SD3e
>        To: <sip:14034883602 at 66.51.110.210>
>        Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad
>        CSeq: 7014814 INVITE
>        Contact: <sip:ciscosip at 66.51.127.163:5080>
>        User-Agent: Cisco-SIPGateway/IOS-12.x
>        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,
>     SUBSCRIBE, NOTIFY,
>     REFER, UPDATE, REGISTER, INFO
>        Supported: timer, precondition, path, replaces
>        Allow-Events: talk, refer
>        Content-Type: application/sdp
>        Content-Disposition: session
>        Content-Length: 323
>        Remote-Party-ID: "CLID NAME"
>     <sip:7806283672 at 66.51.127.163>;screen=yes;pri
>     vacy=off
>
>        v=0
>        o=CiscoSystemsSIP-GW-UserAgent 7458343035650957072
>     338127983379325658 IN IP4
>     66.51.127.163
>        s=SIP Call
>        c=IN IP4 66.51.110.210
>        t=0 0
>        m=audio 14650 RTP/AVP 0 18 101 13
>        a=rtpmap:0 PCMU/8000
>        a=rtpmap:18 G729/8000
>        a=rtpmap:101 telephone-event/8000
>        a=fmtp:101 0-16
>        a=rtpmap:13 CN/8000
>        a=ptime:20
>        a=nortpproxy:yes
>       
>     ------------------------------------------------------------------------
>     send 494 bytes to udp/[66.51.110.210]:5060 at 23:22:37.789983:
>       
>     ------------------------------------------------------------------------
>        SIP/2.0 100 Trying
>        Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0
>        Via: SIP/2.0/UDP
>     66.51.127.163:5080;rport=5080;branch=z9hG4bK0D66gH7m2614F
>        Record-Route: <sip:66.51.110.210;lr;ftag=ZNcB4yyH3SD3e>
>        From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=ZNcB4yyH3SD3e
>        To: <sip:14034883602 at 66.51.110.210>
>        Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad
>        CSeq: 7014814 INVITE
>        User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f
>     2010-12-29 13-15-14 -06
>     00
>        Content-Length: 0
>
>       
>     ------------------------------------------------------------------------
>     2011-01-10 16:22:37.789983 [NOTICE] switch_channel.c:784 New
>     Channel sofia/exter
>     nal/7806283672 at 66.51.127.163 <mailto:nal/7806283672 at 66.51.127.163>
>     [86b78ecd-469f-4a1c-9fe5-692a5941ff37]
>     2011-01-10 16:22:37.805608 [INFO] mod_dialplan_xml.c:331
>     Processing CLID NAME
>     <7806283672>->sipuser in context public
>     2011-01-10 16:22:37.805608 [NOTICE]
>     switch_core_state_machine.c:189 sofia/extern
>     al/7806283672 at 66.51.127.163 <mailto:al/7806283672 at 66.51.127.163>
>     has executed the last dialplan instruction, hanging
>     up.
>     2011-01-10 16:22:37.805608 [NOTICE]
>     switch_core_state_machine.c:191 Hangup sofia
>     /external/7806283672 at 66.51.127.163
>     <mailto:7806283672 at 66.51.127.163> [CS_EXECUTE] [NORMAL_CLEARING]
>     send 806 bytes to udp/[66.51.110.210]:5060 at 23:22:37.805608:
>       
>     ------------------------------------------------------------------------
>        SIP/2.0 480 Temporarily Unavailable
>        Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0
>        Via: SIP/2.0/UDP
>     66.51.127.163:5080;rport=5080;branch=z9hG4bK0D66gH7m2614F
>        From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=ZNcB4yyH3SD3e
>        To: <sip:14034883602 at 66.51.110.210>;tag=pjea7BymS6paS
>        Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad
>        CSeq: 7014814 INVITE
>        User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f
>     2010-12-29 13-15-14 -06
>     00
>        Accept: application/sdp
>        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
>     INFO, REGISTER, RE
>     FER, NOTIFY
>        Supported: timer, precondition, path, replaces
>        Allow-Events: talk, hold, refer
>        Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>        Content-Length: 0
>        Remote-Party-ID: "sipuser"
>     <sip:sipuser at 192.168.35.1>;party=calling;privacy=o
>     ff;screen=no
>
>       
>     ------------------------------------------------------------------------
>     2011-01-10 16:22:37.805608 [NOTICE] switch_core_session.c:1273
>     Session 54 (sofia
>     /external/7806283672 at 66.51.127.163
>     <mailto:7806283672 at 66.51.127.163>) Ended
>     2011-01-10 16:22:37.805608 [NOTICE] switch_core_session.c:1275
>     Close Channel sof
>     ia/external/7806283672 at 66.51.127.163
>     <mailto:ia/external/7806283672 at 66.51.127.163> [CS_DESTROY]
>     recv 352 bytes from udp/[66.51.110.210]:5060 at 23:22:37.914979:
>       
>     ------------------------------------------------------------------------
>        ACK
>     sip:gw+link2voip2 at 192.168.35.1:5080;transport=udp;gw=link2voip2
>     SIP/2.0
>        Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0
>        From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=ZNcB4yyH3SD3e
>        Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad
>        To: <sip:14034883602 at 66.51.110.210>;tag=pjea7BymS6paS
>        CSeq: 7014814 ACK
>        Content-Length: 0
>
>       
>     ------------------------------------------------------------------------
>     recv 1230 bytes from udp/[66.51.127.173]:5060 at 23:22:37.977477:
>       
>     ------------------------------------------------------------------------
>        INVITE
>     sip:gw+link2voip1 at 192.168.35.1:5080;transport=udp;gw=link2voip1 SIP/2.
>     0
>        Record-Route: <sip:66.51.127.173;lr;ftag=17yv7m0rXBt8N>
>        Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0
>        Via: SIP/2.0/UDP
>     66.51.127.163:5080;rport=5080;branch=z9hG4bK2Zrrm78UvreaQ
>        Max-Forwards: 66
>        From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=17yv7m0rXBt8N
>        To: <sip:14034883602 at 66.51.127.173>
>        Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad
>        CSeq: 7014814 INVITE
>        Contact: <sip:ciscosip at 66.51.127.163:5080>
>        User-Agent: Cisco-SIPGateway/IOS-12.x
>        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,
>     SUBSCRIBE, NOTIFY,
>     REFER, UPDATE, REGISTER, INFO
>        Supported: timer, precondition, path, replaces
>        Allow-Events: talk, refer
>        Content-Type: application/sdp
>        Content-Disposition: session
>        Content-Length: 323
>        Remote-Party-ID: "CLID NAME"
>     <sip:7806283672 at 66.51.127.163>;screen=yes;pri
>     vacy=off
>
>        v=0
>        o=CiscoSystemsSIP-GW-UserAgent 4919904105816548778
>     787843793424096957 IN IP4
>     66.51.127.163
>        s=SIP Call
>        c=IN IP4 66.51.127.173
>        t=0 0
>        m=audio 15488 RTP/AVP 0 18 101 13
>        a=rtpmap:0 PCMU/8000
>        a=rtpmap:18 G729/8000
>        a=rtpmap:101 telephone-event/8000
>        a=fmtp:101 0-16
>        a=rtpmap:13 CN/8000
>        a=ptime:20
>        a=nortpproxy:yes
>       
>     ------------------------------------------------------------------------
>     send 494 bytes to udp/[66.51.127.173]:5060 at 23:22:37.977477:
>       
>     ------------------------------------------------------------------------
>        SIP/2.0 100 Trying
>        Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0
>        Via: SIP/2.0/UDP
>     66.51.127.163:5080;rport=5080;branch=z9hG4bK2Zrrm78UvreaQ
>        Record-Route: <sip:66.51.127.173;lr;ftag=17yv7m0rXBt8N>
>        From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=17yv7m0rXBt8N
>        To: <sip:14034883602 at 66.51.127.173>
>        Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad
>        CSeq: 7014814 INVITE
>        User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f
>     2010-12-29 13-15-14 -06
>     00
>        Content-Length: 0
>
>       
>     ------------------------------------------------------------------------
>     2011-01-10 16:22:37.977477 [NOTICE] switch_channel.c:784 New
>     Channel sofia/exter
>     nal/7806283672 at 66.51.127.163 <mailto:nal/7806283672 at 66.51.127.163>
>     [86c939d1-4de0-4a46-9203-518e0d6f7bc5]
>     2011-01-10 16:22:37.977477 [INFO] mod_dialplan_xml.c:331
>     Processing CLID NAME
>     <7806283672>->sipuser in context public
>     2011-01-10 16:22:37.977477 [NOTICE]
>     switch_core_state_machine.c:189 sofia/extern
>     al/7806283672 at 66.51.127.163 <mailto:al/7806283672 at 66.51.127.163>
>     has executed the last dialplan instruction, hanging
>     up.
>     2011-01-10 16:22:37.977477 [NOTICE]
>     switch_core_state_machine.c:191 Hangup sofia
>     /external/7806283672 at 66.51.127.163
>     <mailto:7806283672 at 66.51.127.163> [CS_EXECUTE] [NORMAL_CLEARING]
>     send 806 bytes to udp/[66.51.127.173]:5060 at 23:22:37.977477:
>       
>     ------------------------------------------------------------------------
>        SIP/2.0 480 Temporarily Unavailable
>        Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0
>        Via: SIP/2.0/UDP
>     66.51.127.163:5080;rport=5080;branch=z9hG4bK2Zrrm78UvreaQ
>        From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=17yv7m0rXBt8N
>        To: <sip:14034883602 at 66.51.127.173>;tag=QU7286erpFDXm
>        Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad
>        CSeq: 7014814 INVITE
>        User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f
>     2010-12-29 13-15-14 -06
>     00
>        Accept: application/sdp
>        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
>     INFO, REGISTER, RE
>     FER, NOTIFY
>        Supported: timer, precondition, path, replaces
>        Allow-Events: talk, hold, refer
>        Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>        Content-Length: 0
>        Remote-Party-ID: "sipuser"
>     <sip:sipuser at 192.168.35.1>;party=calling;privacy=o
>     ff;screen=no
>
>       
>     ------------------------------------------------------------------------
>     2011-01-10 16:22:37.977477 [NOTICE] switch_core_session.c:1273
>     Session 55 (sofia
>     /external/7806283672 at 66.51.127.163
>     <mailto:7806283672 at 66.51.127.163>) Ended
>     2011-01-10 16:22:37.977477 [NOTICE] switch_core_session.c:1275
>     Close Channel sof
>     ia/external/7806283672 at 66.51.127.163
>     <mailto:ia/external/7806283672 at 66.51.127.163> [CS_DESTROY]
>     recv 352 bytes from udp/[66.51.127.173]:5060 at 23:22:38.071224:
>       
>     ------------------------------------------------------------------------
>        ACK
>     sip:gw+link2voip1 at 192.168.35.1:5080;transport=udp;gw=link2voip1
>     SIP/2.0
>        Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0
>        From: "CLID NAME" <sip:7806283672 at 66.51.127.163>;tag=17yv7m0rXBt8N
>        Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad
>        To: <sip:14034883602 at 66.51.127.173>;tag=QU7286erpFDXm
>        CSeq: 7014814 ACK
>        Content-Length: 0
>
>       
>     ------------------------------------------------------------------------
>
>
>
>
>     On 10/01/2011 2:52 PM, Michael Collins wrote:
>>     Or just give us your credentials and we'll "test it thoroughly"
>>     for you. :)
>>     -MC
>>
>>     On Mon, Jan 10, 2011 at 1:24 PM, Brian West <brian at freeswitch.org
>>     <mailto:brian at freeswitch.org>> wrote:
>>
>>         can you put up a sip trace or something so we can help guide you?
>>
>>         /b
>>
>>         On Jan 10, 2011, at 2:26 PM, Darren Wiebe wrote:
>>
>>         > Good Afternoon,
>>         >
>>         > I'm trying to get my freeswitch box talking to Link2voip.
>>          Does anybody
>>         > have sample XML files for them?
>>         >
>>         > --
>>         > Darren Wiebe
>>         > Aleph Communications
>>         > --------------------
>>         > Phone: 1-877-702-2900
>>         > Fax:   1-866-274-4506
>>         > Email: darren at aleph-com.net <mailto:darren at aleph-com.net>
>>         >
>>
>
>     -- 
>     Darren Wiebe
>     Aleph Communications
>     --------------------
>     Phone: 1-877-702-2900
>     Fax:   1-866-274-4506
>     Email:darren at aleph-com.net  <mailto:darren at aleph-com.net>
>
>


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From darren at aleph-com.net  Tue Jan 11 05:54:10 2011
From: darren at aleph-com.net (Darren Wiebe)
Date: Mon, 10 Jan 2011 19:54:10 -0700
Subject: [Freeswitch-users] Link2voip
In-Reply-To: <1294714242894-5909450.post@n2.nabble.com>
References: <4D2B6B7B.4090102@aleph-com.net>
	<1294714242894-5909450.post@n2.nabble.com>
Message-ID: <4D2BC652.6090802@aleph-com.net>

On 10/01/2011 7:50 PM, mazilo wrote:
>
> Darren Wiebe wrote:
>> I'm trying to get my freeswitch box talking to Link2voip.  Does anybody
>> have sample XML files for them?
> Have you read the FS wiki on Provider Configuration Examples for
> http://wiki.freeswitch.org/wiki/Provider_Configuration:_Link2VoIP Link2VoIP
> ?
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.

I actually wrote that page after I got it working. :)--

Darren Wiebe
Aleph Communications
--------------------
Phone: 1-877-702-2900
Fax:   1-866-274-4506
Email: darren at aleph-com.net



From cmrienzo at gmail.com  Tue Jan 11 06:12:54 2011
From: cmrienzo at gmail.com (Chris Rienzo)
Date: Mon, 10 Jan 2011 22:12:54 -0500
Subject: [Freeswitch-users] playing files through a remote URL
In-Reply-To: <AANLkTik-d6Wb2dyStNSF-rVKCvaHzXq4pWKsYRKA9sf7@mail.gmail.com>
References: <AANLkTikYdXfOVYF6=vUsj8gB7kHUuwOddrYy_dFg6ByG@mail.gmail.com>
	<AANLkTik-d6Wb2dyStNSF-rVKCvaHzXq4pWKsYRKA9sf7@mail.gmail.com>
Message-ID: <93166BD9-5C05-4E41-A017-CBED60057429@gmail.com>

Maybe you could write a script to download the file.  There are plenty of command line tools to do http get (curl, wget).



On Jan 10, 2011, at 14:09, Holger Esser <hesser4900 at gmail.com> wrote:

> Hello,
> 
> I am trying to write a simple ivr script that answers the call, plays a greeting and collects a few digits. 
> Is it possible to fetch the audio through a remote URL (i.e. http://messageServer/en.us/greeting.wav)?
> 
> Or does it have to reside on the local file system or a mapped drive?
> 
> Kind regards,
> 
> Holger
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From hadyn_whx at hotmail.com  Tue Jan 11 07:17:01 2011
From: hadyn_whx at hotmail.com (Alex Wang)
Date: Mon, 10 Jan 2011 23:17:01 -0500
Subject: [Freeswitch-users] How to disable 5060-5080 with auto-nat at upnp?
Message-ID: <BLU0-SMTP97E64484D5F1A2D4F0FEFB8D0F0@phx.gbl>

Hi All

Just don't want to open these port to the public, all my extensions are
in my LAN. If I disable upnp, the outside sip register is not working, I
mean no sound.
How to disable the auto-nat publish 5060-5080 on the internet?


Thanks

Alex




From u2nsam at gmail.com  Tue Jan 11 07:56:12 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 11 Jan 2011 10:26:12 +0530
Subject: [Freeswitch-users] conference pin
In-Reply-To: <AANLkTimSRbCLU_+F3TfR7eK9NXTx=wiVUnVJ-7-yxNqv@mail.gmail.com>
References: <AANLkTi=q5xpnW7HdGVZYpQbUNi2m=dvmFXX6TTKGvxZo@mail.gmail.com>
	<AANLkTin3F-TXJp_Mv-8vwGbkLmAwbi9mE2bobiB2uyQx@mail.gmail.com>
	<AANLkTimSRbCLU_+F3TfR7eK9NXTx=wiVUnVJ-7-yxNqv@mail.gmail.com>
Message-ID: <AANLkTinKHit3YREk4StHM4d1hrxWzrHwhfGNsLObvPSb@mail.gmail.com>

In features.xml you will see a dialplan you can use it and modify as per
your requirement.

<extension name="att_xfer">
     <condition field="destination_number" expression="^att_xfer$">
       <action application="read" data="3 4
'tone_stream://%(5000,0,350,440)' digits 30000 #"/>
       <action application="set" data="origination_cancel_key=#"/>
       <action application="att_xfer" data="user/${digits}@$${domain}"/>
     </condition>
    </extension>

here ${digits} refers to the dtmf you punch in and you can create such
variable where ever you require.

Regds
Sam



On Tue, Jan 11, 2011 at 5:39 AM, deniro <ayhkor at gmail.com> wrote:

> Hi
> any expamples of how to store asked pin (phone call) by channel var and
> re-use it in conferencing to go to that conference?
> thx
>
>
> On Mon, Jan 10, 2011 at 2:09 AM, Sam <u2nsam at gmail.com> wrote:
>
>> use channels variables in freeswitch.
>>
>>
>> http://wiki.freeswitch.org/wiki/Channel_Variables#Channel_Variable_Manipulation
>>
>> Regds
>> Sam
>>
>>
>>
>>   On Mon, Jan 10, 2011 at 9:35 AM, deniro <ayhkor at gmail.com> wrote:
>>
>>>   Hi
>>> using conferencing software and with the phone dialing,
>>> entering pin number it will go to a conference  identified by pin
>>> in its default format  it is "conference at profile+pin"
>>> in my case it will be "pin at profile+pin" since conference=pin.
>>> how do I do that? how do I provide a pin that takes me to conference
>>> which is
>>> identified by pin?
>>> thx
>>> deniro--
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Jan 11 08:03:21 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Jan 2011 05:03:21 +0000
Subject: [Freeswitch-users] conference pin
In-Reply-To: <AANLkTimSRbCLU_+F3TfR7eK9NXTx=wiVUnVJ-7-yxNqv@mail.gmail.com>
References: <AANLkTi=q5xpnW7HdGVZYpQbUNi2m=dvmFXX6TTKGvxZo@mail.gmail.com>
	<AANLkTin3F-TXJp_Mv-8vwGbkLmAwbi9mE2bobiB2uyQx@mail.gmail.com>
	<AANLkTimSRbCLU_+F3TfR7eK9NXTx=wiVUnVJ-7-yxNqv@mail.gmail.com>
Message-ID: <AANLkTinRMuz2zuspEvp9ZaDugpA5G6GGm0n2HKuMTaiF@mail.gmail.com>

Did the example I gave you not work? (http://pastebin.freeswitch.org/14979)
-MC

On Tue, Jan 11, 2011 at 12:09 AM, deniro <ayhkor at gmail.com> wrote:

> Hi
> any expamples of how to store asked pin (phone call) by channel var and
> re-use it in conferencing to go to that conference?
> thx
>
>
> On Mon, Jan 10, 2011 at 2:09 AM, Sam <u2nsam at gmail.com> wrote:
>
>> use channels variables in freeswitch.
>>
>>
>> http://wiki.freeswitch.org/wiki/Channel_Variables#Channel_Variable_Manipulation
>>
>> Regds
>> Sam
>>
>>
>>
>>   On Mon, Jan 10, 2011 at 9:35 AM, deniro <ayhkor at gmail.com> wrote:
>>
>>>   Hi
>>> using conferencing software and with the phone dialing,
>>> entering pin number it will go to a conference  identified by pin
>>> in its default format  it is "conference at profile+pin"
>>> in my case it will be "pin at profile+pin" since conference=pin.
>>> how do I do that? how do I provide a pin that takes me to conference
>>> which is
>>> identified by pin?
>>> thx
>>> deniro--
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From curriegrad2004 at gmail.com  Tue Jan 11 08:09:34 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 10 Jan 2011 21:09:34 -0800
Subject: [Freeswitch-users] Any one use FreeCanadianCalls with
	FreeSwitch?
In-Reply-To: <BLU0-SMTP84FDB44AA90E9823D8E3B18D0E0@phx.gbl>
References: <BLU0-SMTP84FDB44AA90E9823D8E3B18D0E0@phx.gbl>
Message-ID: <AANLkTin+smtuNsHmjD3pOtaUu030bxxxwJpzn7Rtc2g_@mail.gmail.com>

Just register it as a regular gateway. Shouldn't be too hard to do. As
for incoming calls, use the destination_number variable instead of
${sip_to_user} in the dialplan. Hope this clears things up for you.

On Mon, Jan 10, 2011 at 6:13 AM, Alex Wang <hadyn_whx at hotmail.com> wrote:
> Hi
>
> Any one is using FreeCanadianCalls with FreeSwitch? Would you mind to
> share the xml setting?
>
> Thanks
>
> Alex
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From u2nsam at gmail.com  Tue Jan 11 08:17:13 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 11 Jan 2011 10:47:13 +0530
Subject: [Freeswitch-users] conference pin
In-Reply-To: <AANLkTinRMuz2zuspEvp9ZaDugpA5G6GGm0n2HKuMTaiF@mail.gmail.com>
References: <AANLkTi=q5xpnW7HdGVZYpQbUNi2m=dvmFXX6TTKGvxZo@mail.gmail.com>
	<AANLkTin3F-TXJp_Mv-8vwGbkLmAwbi9mE2bobiB2uyQx@mail.gmail.com>
	<AANLkTimSRbCLU_+F3TfR7eK9NXTx=wiVUnVJ-7-yxNqv@mail.gmail.com>
	<AANLkTinRMuz2zuspEvp9ZaDugpA5G6GGm0n2HKuMTaiF@mail.gmail.com>
Message-ID: <AANLkTi=fR-yLnagCicFXi9U_o6V8CLXeF7Wz01kJ6EDu@mail.gmail.com>

Kool example MC !

Regds
Sam

On Tue, Jan 11, 2011 at 10:33 AM, Michael Collins <msc at freeswitch.org>wrote:

> Did the example I gave you not work? (http://pastebin.freeswitch.org/14979
> )
> -MC
>
>
> On Tue, Jan 11, 2011 at 12:09 AM, deniro <ayhkor at gmail.com> wrote:
>
>> Hi
>> any expamples of how to store asked pin (phone call) by channel var and
>> re-use it in conferencing to go to that conference?
>> thx
>>
>>
>> On Mon, Jan 10, 2011 at 2:09 AM, Sam <u2nsam at gmail.com> wrote:
>>
>>> use channels variables in freeswitch.
>>>
>>>
>>> http://wiki.freeswitch.org/wiki/Channel_Variables#Channel_Variable_Manipulation
>>>
>>> Regds
>>> Sam
>>>
>>>
>>>
>>>   On Mon, Jan 10, 2011 at 9:35 AM, deniro <ayhkor at gmail.com> wrote:
>>>
>>>>   Hi
>>>> using conferencing software and with the phone dialing,
>>>> entering pin number it will go to a conference  identified by pin
>>>> in its default format  it is "conference at profile+pin"
>>>> in my case it will be "pin at profile+pin" since conference=pin.
>>>> how do I do that? how do I provide a pin that takes me to conference
>>>> which is
>>>> identified by pin?
>>>> thx
>>>> deniro--
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Jan 11 08:39:34 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Jan 2011 05:39:34 +0000
Subject: [Freeswitch-users] Two meny qestions
In-Reply-To: <25504058.721294683825512.JavaMail.root@mailserver>
References: <9DD9A108-0CFF-4224-88C2-95CFF9E0FE76@freeswitch.org>
	<25504058.721294683825512.JavaMail.root@mailserver>
Message-ID: <AANLkTikd_A3JCYO3GC0m8KTwWCorQvP+USvO27SBk5Wv@mail.gmail.com>

get a console log and put it in pastebin. That should help us to see what is
going on. Look at this page if you need more information on getting debug
information:

http://wiki.freeswitch.org/wiki/Reporting_Bugs

-MC

On Mon, Jan 10, 2011 at 6:23 PM, Mattias Hemmingsson
<matte at ahavaxthuset.se>wrote:

> Hi
>
>
> Still having problems transfering the call to the right extension.
> I have one domain in my directory called www.elino.se and in there all my
> users all.
> So i test to set upp the meny to transfer the call to my extension  like
> this
>
> <entry action="menu-exec-app" digits="1" param="transfer 1000 at www.elino.se
> "/>
>
> but it dont work.
>
> i have also test to transfer with
>
>
> <entry action="menu-exec-app" digits="1" param="transfer 1000 XML
> default"/>
>
> but the i only get an godbye.
>
>
> what im i doing wrong ?
>
> // Matte
>
>
> ----- Ursprungligt meddelande -----
> Fr?n: "Brian West" <brian at freeswitch.org>
> Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Skickat: s?ndag, 9 jan 2011 22:35:00
> ?mne: Re: [Freeswitch-users] Two meny qestions
>
> This is because you're calling bridge right to the users endpoint... if you
> were to transfer to extension 1000 or 1001 then voicemail would work exactly
> like you expect.
>
> /b
>
> On Jan 9, 2011, at 1:21 PM, Mattias Hemmingsson wrote:
>
> > But when the is not online i want the user to be transferd to the users
> voicemail.
> > I have voicemail working of i call the user from a nother externsion.
> > But i would like it to wokr from the ivr meny as well.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Jan 11 09:20:34 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Jan 2011 06:20:34 +0000
Subject: [Freeswitch-users] IVR menu not observing timeout value
In-Reply-To: <AANLkTikc_YS=vCZLAh0mMZ0Eu9qP_vyCFq0oKSRORou7@mail.gmail.com>
References: <AANLkTikc_YS=vCZLAh0mMZ0Eu9qP_vyCFq0oKSRORou7@mail.gmail.com>
Message-ID: <AANLkTik8r9-utKqEUB+UvkchYNE9TUaL7sfqnGOLVHmb@mail.gmail.com>

I was not able to reproduce this using the demo IVR. Can you try it with the
demo IVR and let us know what you get?
-MC

On Mon, Jan 10, 2011 at 4:15 PM, Kim Culhan <w8hdkim at gmail.com> wrote:

> On Mon, January 10, 2011 9:35 am, Kim Culhan wrote:
> > FreeSWITCH Version 1.0.head (git-4e95227 2010-12-26 09-09-14 -0600)
> Correction on the FS version, it should read:
>
> FreeSWITCH Version 1.0.head (git-3003489 2011-01-09 14-42-42 -0500)
>
> thanks
> -kim
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rafonline at hotmail.com  Tue Jan 11 10:57:26 2011
From: rafonline at hotmail.com (Rafqat .)
Date: Tue, 11 Jan 2011 07:57:26 +0000
Subject: [Freeswitch-users] bind_digit_action
In-Reply-To: <AANLkTim-+iLNZAhNaWU0npAfL2Sn++8mCyqwANXXbVax@mail.gmail.com>
References: <COL122-W20D6C00E627218901B0691B50E0@phx.gbl>,
	<AANLkTim-+iLNZAhNaWU0npAfL2Sn++8mCyqwANXXbVax@mail.gmail.com>
Message-ID: <COL122-W297FC85979AA72D5DBC40B50F0@phx.gbl>



Hi,

I moved the bind_digit_action so my dialplan looks something like this:

<extension name="call-me-service">
  <condition field="destination_number" expression="^9999$">
    <action application="answer"/>
    .........
    <action application="set" data="bridge_pre_execute_bleg_app=execute_extension"/>
    <action application="set" data="bridge_pre_execute_bleg_data=START_LISTENING XML private"/>
    <action application="bind_digit_action" data="start,##,exec:hangup,unknown"/>
    .......
  </condition>
</extension>

<extension name="setup bind_digit_action_listening">
  <condition field="destination_number" expression="^START_LISTENING$">
    <action application="log" data="INFO Configuring bind_digit_action to listen..."/>
    <action application="digit_action_set_realm" data="start"/>
  </condition>
</extension>

Unfortunalety, now when leg A presses ##, it hangs up leg A. I would like to make it hang up leg B only.

I am sure it is my lack of understanding of bind_digit_action.

Apreciate all the help I can get.

Cheers

Raf


________________________________
> Date: Mon, 10 Jan 2011 23:04:52 +0000
> From: msc at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] bind_digit_action
>
> Just to confirm: you want Leg A to press ## to hangup leg B? If so then
> you need to set the bind_digit_action on Leg A. The way you have it now
> is that Leg B would need to dial ##.
>
> -MC
>
> On Mon, Jan 10, 2011 at 10:46 PM, Rafqat .
> > wrote:
>
>
>
> Hi,
>
> Can someone please help here.
>
> I am trying to use bind_digit_action to hangup leg B only, if leg A
> presses ##, but it doesn't seem to be working. The INFO message is
> displayed but when I press ##, nothing happens.
>
> Any help will be much appreciated.
>
>
>
>
>
>
> .........
> > data="bridge_pre_execute_bleg_app=execute_extension"/>
> > data="bridge_pre_execute_bleg_data=START_LISTENING XML private"/>
> .......
>
>
>
>
>
>
>
> > data="start,##,exec:hangup,unknown"/>
>
>
>
>
>
> Cheers
>
> Raf
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________ FreeSWITCH-users
> mailing list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  

From pivanet at gmail.com  Tue Jan 11 10:59:15 2011
From: pivanet at gmail.com (Leonid K)
Date: Tue, 11 Jan 2011 09:59:15 +0200
Subject: [Freeswitch-users] Asterisk & FreeSwitch in recording view
In-Reply-To: <AANLkTi=gPxWQgunBT604zT13gDe4oU8xNzWF3Xj_1aJ+@mail.gmail.com>
References: <AANLkTinZT1WiNnae4oMKd+5YdEUO9xq4_cJr6R+4siiK@mail.gmail.com>
	<AANLkTinvVqYuZKT6zHAQPW+QdqMoSaabB1WG6DLQOocO@mail.gmail.com>
	<AANLkTi=gPxWQgunBT604zT13gDe4oU8xNzWF3Xj_1aJ+@mail.gmail.com>
Message-ID: <AANLkTinB6zn9e0ti7NB8y7hWQtuQzWaBjznpUo0+m3aF@mail.gmail.com>

a few more questions:
1) can I rely on orecx in it's openSource view (looks like it hasn't been
updated since 2009) - OR it's better to get their commercial solution?
2)what about limitations of the count of concurent calls that FreeSwitch can
record? what if i neet it to be huge - like 18.000 calls !!!?


On Mon, Jan 10, 2011 at 7:25 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> Let me clarify. :)
>
> If you are ok with routing all your traffic through FreeSWITCH (or *),
> then FreeSWITCH can definitely do what you want.  I use it that way
> (record all external calls) without issue.  If your volume is high,
> you'll want to record as PMCU WAV to a ramdisk and then have a cron
> job that converts completed calls to mp3 or speex or whatever.  I have
> scripts that do this well including preserving the metadata that FS
> can write to the WAV files for CID info or whatever else you ask it to
> put in there.  (just using sox or lame will loose that metadata from
> the WAV file since they don't bother to preserve it).
>
> That being said.  orecx is designed to record your VOIP traffic
> without having to do anything put ensure all VOIP traffic ends up on
> the network segment orecx is attached to.  This "transparent"
> recording is definitely "the way to go" if you want to separate your
> phone infrastructure from your recording infrastructure.
>
> On Mon, Jan 10, 2011 at 11:16 AM, Rupa Schomaker <rupa at rupa.com> wrote:
> > I think the standard answer to that is www.orecx.com.
> >
> > On Mon, Jan 10, 2011 at 3:39 AM, Leonid K <pivanet at gmail.com> wrote:
> >> Hi,
> >> currently we're looking for voip recording solution - it must record all
> >> incoming/outgoing/internal/conference calls within the company. later on
> we
> >> going to develop applications that let us fing/analyze recordings. the
> main
> >> problem as I see at the moment is huge count of calls that is going
> through
> >> switch.
> >>
> >> what do u think about using Asterisk or/and FreeSwitch for this task?
> thanks
> >> in advance!
> >> --
> >> Sincerely,
> >> Leonid Kryvoruchko
> >>
> >> Mobile   +38 093 7609175
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
> >
> > --
> > -Rupa
> >
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,
Leonid Kryvoruchko

Mobile   +38 093 7609175
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From infos at madovsky.org  Tue Jan 11 11:12:25 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 11 Jan 2011 03:12:25 -0500
Subject: [Freeswitch-users] cepstral problem
Message-ID: <6DCDF4F8008043CDA4BD7F00005611F4@e1705>

I have one sentence like this in my dialplan
<action application="speak" data="cepstral|david|want pizza"/>

so when I start FS from a console directly in foreground the voice works.
if I start FS from the init script in background, the voice becomes silence
and nothing is running after this in the dialplan.
if I want to shutdown manually FS, it makes a segmentation fault with core dump.

Any idea ?

Thanks
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From infos at madovsky.org  Tue Jan 11 11:33:45 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 11 Jan 2011 03:33:45 -0500
Subject: [Freeswitch-users] cepstral problem
References: <6DCDF4F8008043CDA4BD7F00005611F4@e1705>
Message-ID: <693A2C5B2175445CA6FC02821FACFE80@e1705>

I'm sure to have set mod_cepstral in modules.conf.xml
but in the log I can see

2011-01-11 03:32:14.999468 [ERR] switch_core_speech.c:61 Invalid speech module [cepstral]!
2011-01-11 03:32:14.999468 [ERR] switch_ivr_play_say.c:2266 Invalid TTS module!

I use Cepstral 5.1

Thanks
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, January 11, 2011 3:12 AM
  Subject: [Freeswitch-users] cepstral problem


  I have one sentence like this in my dialplan
  <action application="speak" data="cepstral|david|want pizza"/>

  so when I start FS from a console directly in foreground the voice works.
  if I start FS from the init script in background, the voice becomes silence
  and nothing is running after this in the dialplan.
  if I want to shutdown manually FS, it makes a segmentation fault with core dump.

  Any idea ?

  Thanks


------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From steveayre at gmail.com  Tue Jan 11 12:32:11 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 11 Jan 2011 09:32:11 +0000
Subject: [Freeswitch-users] cepstral problem
In-Reply-To: <693A2C5B2175445CA6FC02821FACFE80@e1705>
References: <6DCDF4F8008043CDA4BD7F00005611F4@e1705>
	<693A2C5B2175445CA6FC02821FACFE80@e1705>
Message-ID: <AANLkTimkDFymovrMg7mX3ZDJT+ubQH=7mz0QZga1Acja@mail.gmail.com>

Check your log from startup - did mod_cepstral fail to load?

It could be that you didn't compile it, it can't find a shared library
or there's a problem in the config file. It'll show you in the log.

'show modules mod_cepstral' will show you whether it's loaded from cli.

-Steve

On 11 January 2011 08:33, Madovsky <infos at madovsky.org> wrote:
> I'm sure to have set mod_cepstral in modules.conf.xml
> but in the log I can see
>
> 2011-01-11 03:32:14.999468 [ERR] switch_core_speech.c:61 Invalid speech
> module [cepstral]!
> 2011-01-11 03:32:14.999468 [ERR] switch_ivr_play_say.c:2266 Invalid TTS
> module!
> I use Cepstral 5.1
>
> Thanks
>
> ----- Original Message -----
> From: Madovsky
> To: freeswitch-users at lists.freeswitch.org
> Sent: Tuesday, January 11, 2011 3:12 AM
> Subject: [Freeswitch-users] cepstral problem
> I have one sentence like this in my dialplan
> <action application="speak" data="cepstral|david|want pizza"/>
>
> so when I start FS from a console directly in foreground the voice works.
> if I start FS from the init script in background, the voice becomes silence
> and nothing is running after this in the dialplan.
> if I want to shutdown manually FS, it makes a segmentation fault with core
> dump.
>
> Any idea ?
>
> Thanks
>
> ________________________________
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steveayre at gmail.com  Tue Jan 11 12:32:30 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 11 Jan 2011 09:32:30 +0000
Subject: [Freeswitch-users] cepstral problem
In-Reply-To: <AANLkTimkDFymovrMg7mX3ZDJT+ubQH=7mz0QZga1Acja@mail.gmail.com>
References: <6DCDF4F8008043CDA4BD7F00005611F4@e1705>
	<693A2C5B2175445CA6FC02821FACFE80@e1705>
	<AANLkTimkDFymovrMg7mX3ZDJT+ubQH=7mz0QZga1Acja@mail.gmail.com>
Message-ID: <AANLkTinCqzaXDybb62sD4jq=YaocjPnF6OVvx4oG4UAe@mail.gmail.com>

If in doubt, try 'load mod_cepstral' from the cli

-Steve


On 11 January 2011 09:32, Steven Ayre <steveayre at gmail.com> wrote:
> Check your log from startup - did mod_cepstral fail to load?
>
> It could be that you didn't compile it, it can't find a shared library
> or there's a problem in the config file. It'll show you in the log.
>
> 'show modules mod_cepstral' will show you whether it's loaded from cli.
>
> -Steve
>
> On 11 January 2011 08:33, Madovsky <infos at madovsky.org> wrote:
>> I'm sure to have set mod_cepstral in modules.conf.xml
>> but in the log I can see
>>
>> 2011-01-11 03:32:14.999468 [ERR] switch_core_speech.c:61 Invalid speech
>> module [cepstral]!
>> 2011-01-11 03:32:14.999468 [ERR] switch_ivr_play_say.c:2266 Invalid TTS
>> module!
>> I use Cepstral 5.1
>>
>> Thanks
>>
>> ----- Original Message -----
>> From: Madovsky
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Tuesday, January 11, 2011 3:12 AM
>> Subject: [Freeswitch-users] cepstral problem
>> I have one sentence like this in my dialplan
>> <action application="speak" data="cepstral|david|want pizza"/>
>>
>> so when I start FS from a console directly in foreground the voice works.
>> if I start FS from the init script in background, the voice becomes silence
>> and nothing is running after this in the dialplan.
>> if I want to shutdown manually FS, it makes a segmentation fault with core
>> dump.
>>
>> Any idea ?
>>
>> Thanks
>>
>> ________________________________
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>


From steveayre at gmail.com  Tue Jan 11 12:33:50 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 11 Jan 2011 09:33:50 +0000
Subject: [Freeswitch-users] Asterisk & FreeSwitch in recording view
In-Reply-To: <AANLkTinB6zn9e0ti7NB8y7hWQtuQzWaBjznpUo0+m3aF@mail.gmail.com>
References: <AANLkTinZT1WiNnae4oMKd+5YdEUO9xq4_cJr6R+4siiK@mail.gmail.com>
	<AANLkTinvVqYuZKT6zHAQPW+QdqMoSaabB1WG6DLQOocO@mail.gmail.com>
	<AANLkTi=gPxWQgunBT604zT13gDe4oU8xNzWF3Xj_1aJ+@mail.gmail.com>
	<AANLkTinB6zn9e0ti7NB8y7hWQtuQzWaBjznpUo0+m3aF@mail.gmail.com>
Message-ID: <AANLkTikqJi+6OMFJw49kfRert6dmFCf5DLi5Y3YzepnS@mail.gmail.com>

> 2)what about limitations of the count of concurent calls that FreeSwitch can
> record? what if i neet it to be huge - like 18.000 calls !!!?

For that many, you'd want to send calls through a cluster of FS boxes.
Something like orecx would be much better.

-Steve


On 11 January 2011 07:59, Leonid K <pivanet at gmail.com> wrote:
> a few more questions:
> 1) can I rely on orecx?in it's openSource view (looks like it hasn't been
> updated since 2009) - OR it's better to get their commercial solution?
> 2)what about limitations of the count of concurent calls that FreeSwitch can
> record? what if i neet it to be huge - like 18.000 calls !!!?
>
> On Mon, Jan 10, 2011 at 7:25 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>> Let me clarify. :)
>>
>> If you are ok with routing all your traffic through FreeSWITCH (or *),
>> then FreeSWITCH can definitely do what you want. ?I use it that way
>> (record all external calls) without issue. ?If your volume is high,
>> you'll want to record as PMCU WAV to a ramdisk and then have a cron
>> job that converts completed calls to mp3 or speex or whatever. ?I have
>> scripts that do this well including preserving the metadata that FS
>> can write to the WAV files for CID info or whatever else you ask it to
>> put in there. ?(just using sox or lame will loose that metadata from
>> the WAV file since they don't bother to preserve it).
>>
>> That being said. ?orecx is designed to record your VOIP traffic
>> without having to do anything put ensure all VOIP traffic ends up on
>> the network segment orecx is attached to. ?This "transparent"
>> recording is definitely "the way to go" if you want to separate your
>> phone infrastructure from your recording infrastructure.
>>
>> On Mon, Jan 10, 2011 at 11:16 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>> > I think the standard answer to that is www.orecx.com.
>> >
>> > On Mon, Jan 10, 2011 at 3:39 AM, Leonid K <pivanet at gmail.com> wrote:
>> >> Hi,
>> >> currently we're looking for voip recording solution - it must record
>> >> all
>> >> incoming/outgoing/internal/conference calls within the company. later
>> >> on we
>> >> going to develop applications that let us fing/analyze recordings. the
>> >> main
>> >> problem as I see at the moment is huge count of calls that is going
>> >> through
>> >> switch.
>> >>
>> >> what do u think about using Asterisk or/and FreeSwitch for this task?
>> >> thanks
>> >> in advance!
>> >> --
>> >> Sincerely,
>> >> Leonid Kryvoruchko
>> >>
>> >> Mobile?? +38 093 7609175
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> >
>> >
>> > --
>> > -Rupa
>> >
>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Sincerely,
> Leonid Kryvoruchko
>
> Mobile?? +38 093 7609175
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steveayre at gmail.com  Tue Jan 11 12:35:47 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 11 Jan 2011 09:35:47 +0000
Subject: [Freeswitch-users] Remote LogIn to Freeswitch?
In-Reply-To: <1294706814105-5909221.post@n2.nabble.com>
References: <AANLkTinREKsePLFxXHhJobO0VkNBe0zt89araJJdBEnz@mail.gmail.com>
	<1294706814105-5909221.post@n2.nabble.com>
Message-ID: <AANLkTink7yKq5_Wj8nssWgK=y4h9kxpPLOMnrpC8qCWN@mail.gmail.com>

Not what he was asking.

On 11 January 2011 00:46, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
>
> Joao Leme wrote:
>> What do I have to do to be able to LogIn to Freeswitch from Home (server
>> is
>> located at office) starting from the basic/original configuration?
> You can either use a telnet or ssh to perform a remote login into your FS.
> The later one is preferable because it uses a secured shell.
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Remote-LogIn-to-Freeswitch-tp5909220p5909221.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From steveayre at gmail.com  Tue Jan 11 12:37:35 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 11 Jan 2011 09:37:35 +0000
Subject: [Freeswitch-users] Remote LogIn to Freeswitch?
In-Reply-To: <AANLkTinREKsePLFxXHhJobO0VkNBe0zt89araJJdBEnz@mail.gmail.com>
References: <AANLkTinREKsePLFxXHhJobO0VkNBe0zt89araJJdBEnz@mail.gmail.com>
Message-ID: <AANLkTinQiG5T6SrGW-Vr_kqiqSC8Mj9ZeMwnw4X+pyK1@mail.gmail.com>

If you're able to dial in but you're getting no sound, it's probably
NAT stopping the audio get through.

There's quite a bit of information on NAT on the Wiki that might be of use.

http://wiki.freeswitch.org/wiki/NAT

-Steve


On 11 January 2011 00:39, Joao Leme <joaocarlosleme at gmail.com> wrote:
> Hi There,
> What do I have to do to be able to LogIn to Freeswitch from Home (server is
> located at office) starting from the basic/original configuration?
> I'm using X-Lite. I've been able to LogIn replacing the internal IP by the
> external IP from the Office but the sound is not working so I wanted to know
> what are the configuration changes that have to be done to allow it. Do I
> have to create a different profile? I want be able to do the same just as if
> I was at the Office.
> Thanks,
> John
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steveayre at gmail.com  Tue Jan 11 12:51:37 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 11 Jan 2011 09:51:37 +0000
Subject: [Freeswitch-users] Sofia fails to detect offline gateway
Message-ID: <AANLkTim3PeS3fELJPQ21N38c9v5p8J0wGNkBrUvc9R=t@mail.gmail.com>

Hi Guys,

A couple of our gateways have been shut down for hardware upgrades.
However, the FreeSWITCH servers didn't spot the gateways go offline so
they're still trying to send calls to them.

I see this in the logs:

2011-01-11 09:35:05.396610 [WARNING] sofia.c:3981 Ping failed gw4 with
code 503 - count 1/0/10, state UP
2011-01-11 09:35:05.396610 [WARNING] sofia.c:3981 Ping failed gw3 with
code 503 - count 1/0/10, state UP
2011-01-11 09:35:13.408674 [WARNING] sofia.c:3981 Ping failed gw4 with
code 503 - count 1/0/10, state UP
2011-01-11 09:35:13.408674 [WARNING] sofia.c:3981 Ping failed gw3 with
code 503 - count 1/0/10, state UP
2011-01-11 09:35:21.420884 [WARNING] sofia.c:3981 Ping failed gw4 with
code 503 - count 1/0/10, state UP
2011-01-11 09:35:21.420884 [WARNING] sofia.c:3981 Ping failed gw3 with
code 503 - count 1/0/10, state UP
2011-01-11 09:35:29.458114 [WARNING] sofia.c:3981 Ping failed gw4 with
code 503 - count 1/0/10, state UP
2011-01-11 09:35:29.458114 [WARNING] sofia.c:3981 Ping failed gw3 with
code 503 - count 1/0/10, state UP
2011-01-11 09:35:37.472424 [WARNING] sofia.c:3981 Ping failed gw4 with
code 503 - count 1/0/10, state UP
2011-01-11 09:35:37.472424 [WARNING] sofia.c:3981 Ping failed gw3 with
code 503 - count 1/0/10, state UP
2011-01-11 09:35:46.475934 [WARNING] sofia.c:3981 Ping failed gw4 with
code 503 - count 1/0/10, state UP
2011-01-11 09:35:46.475934 [WARNING] sofia.c:3981 Ping failed gw3 with
code 503 - count 1/0/10, state UP
2011-01-11 09:35:55.489967 [WARNING] sofia.c:3981 Ping failed gw4 with
code 503 - count 1/0/10, state UP
2011-01-11 09:35:55.489967 [WARNING] sofia.c:3981 Ping failed gw3 with
code 503 - count 1/0/10, state UP
... you get the idea.

I've done some testing and it happens both when FS is running and the
gateway goes offline (code 408 then) and when FS starts up with the
gateway already offline (code 503 as above).

Both IP addresses are entirely offline. I had expected these would go
to state DOWN but they haven't, I just get the above repeating.

The gateway definitions look like this:

<gateway name="gw3">
  <param name="realm" value="a.b.c.d"/>
  <param name="username" value="nouser"/>
  <param name="password" value="nopass"/>
  <param name="register" value="false"/>
  <param name="caller-id-in-from" value="true"/>
  <param name="ping" value="5"/>
  <param name="ping-max" value="10"/>
  <param name="ping-min" value="1"/>
  <param name="retry-seconds" value="5"/>
  <param name="expire-seconds" value="60"/>
</gateway>


Is there a mistake in my configuration?

-Steve


From pivanet at gmail.com  Tue Jan 11 13:11:58 2011
From: pivanet at gmail.com (Leonid K)
Date: Tue, 11 Jan 2011 12:11:58 +0200
Subject: [Freeswitch-users] Asterisk & FreeSwitch in recording view
In-Reply-To: <AANLkTikqJi+6OMFJw49kfRert6dmFCf5DLi5Y3YzepnS@mail.gmail.com>
References: <AANLkTinZT1WiNnae4oMKd+5YdEUO9xq4_cJr6R+4siiK@mail.gmail.com>
	<AANLkTinvVqYuZKT6zHAQPW+QdqMoSaabB1WG6DLQOocO@mail.gmail.com>
	<AANLkTi=gPxWQgunBT604zT13gDe4oU8xNzWF3Xj_1aJ+@mail.gmail.com>
	<AANLkTinB6zn9e0ti7NB8y7hWQtuQzWaBjznpUo0+m3aF@mail.gmail.com>
	<AANLkTikqJi+6OMFJw49kfRert6dmFCf5DLi5Y3YzepnS@mail.gmail.com>
Message-ID: <AANLkTikL=EMSBi57utb0JLOb3TJm5-wJE1rJve3HyAQi@mail.gmail.com>

thx Steve. I'm going to look at Oreka... is there any specific info that I
have to know before "merging" FreeSwitch with it? a saw that it has some
patch for Asterisk... so i'd need to write smth for FreeSwitch?

On Tue, Jan 11, 2011 at 11:33 AM, Steven Ayre <steveayre at gmail.com> wrote:

> > 2)what about limitations of the count of concurent calls that FreeSwitch
> can
> > record? what if i neet it to be huge - like 18.000 calls !!!?
>
> For that many, you'd want to send calls through a cluster of FS boxes.
> Something like orecx would be much better.
>
> -Steve
>
>
> On 11 January 2011 07:59, Leonid K <pivanet at gmail.com> wrote:
> > a few more questions:
> > 1) can I rely on orecx in it's openSource view (looks like it hasn't been
> > updated since 2009) - OR it's better to get their commercial solution?
> > 2)what about limitations of the count of concurent calls that FreeSwitch
> can
> > record? what if i neet it to be huge - like 18.000 calls !!!?
> >
> > On Mon, Jan 10, 2011 at 7:25 PM, Rupa Schomaker <rupa at rupa.com> wrote:
> >>
> >> Let me clarify. :)
> >>
> >> If you are ok with routing all your traffic through FreeSWITCH (or *),
> >> then FreeSWITCH can definitely do what you want.  I use it that way
> >> (record all external calls) without issue.  If your volume is high,
> >> you'll want to record as PMCU WAV to a ramdisk and then have a cron
> >> job that converts completed calls to mp3 or speex or whatever.  I have
> >> scripts that do this well including preserving the metadata that FS
> >> can write to the WAV files for CID info or whatever else you ask it to
> >> put in there.  (just using sox or lame will loose that metadata from
> >> the WAV file since they don't bother to preserve it).
> >>
> >> That being said.  orecx is designed to record your VOIP traffic
> >> without having to do anything put ensure all VOIP traffic ends up on
> >> the network segment orecx is attached to.  This "transparent"
> >> recording is definitely "the way to go" if you want to separate your
> >> phone infrastructure from your recording infrastructure.
> >>
> >> On Mon, Jan 10, 2011 at 11:16 AM, Rupa Schomaker <rupa at rupa.com> wrote:
> >> > I think the standard answer to that is www.orecx.com.
> >> >
> >> > On Mon, Jan 10, 2011 at 3:39 AM, Leonid K <pivanet at gmail.com> wrote:
> >> >> Hi,
> >> >> currently we're looking for voip recording solution - it must record
> >> >> all
> >> >> incoming/outgoing/internal/conference calls within the company. later
> >> >> on we
> >> >> going to develop applications that let us fing/analyze recordings.
> the
> >> >> main
> >> >> problem as I see at the moment is huge count of calls that is going
> >> >> through
> >> >> switch.
> >> >>
> >> >> what do u think about using Asterisk or/and FreeSwitch for this task?
> >> >> thanks
> >> >> in advance!
> >> >> --
> >> >> Sincerely,
> >> >> Leonid Kryvoruchko
> >> >>
> >> >> Mobile   +38 093 7609175
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >>
> >> >
> >> >
> >> >
> >> > --
> >> > -Rupa
> >> >
> >>
> >>
> >>
> >> --
> >> -Rupa
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Sincerely,
> > Leonid Kryvoruchko
> >
> > Mobile   +38 093 7609175
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,
Leonid Kryvoruchko

Mobile   +38 093 7609175
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From steveayre at gmail.com  Tue Jan 11 13:34:46 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 11 Jan 2011 10:34:46 +0000
Subject: [Freeswitch-users] Asterisk & FreeSwitch in recording view
In-Reply-To: <AANLkTikL=EMSBi57utb0JLOb3TJm5-wJE1rJve3HyAQi@mail.gmail.com>
References: <AANLkTinZT1WiNnae4oMKd+5YdEUO9xq4_cJr6R+4siiK@mail.gmail.com>
	<AANLkTinvVqYuZKT6zHAQPW+QdqMoSaabB1WG6DLQOocO@mail.gmail.com>
	<AANLkTi=gPxWQgunBT604zT13gDe4oU8xNzWF3Xj_1aJ+@mail.gmail.com>
	<AANLkTinB6zn9e0ti7NB8y7hWQtuQzWaBjznpUo0+m3aF@mail.gmail.com>
	<AANLkTikqJi+6OMFJw49kfRert6dmFCf5DLi5Y3YzepnS@mail.gmail.com>
	<AANLkTikL=EMSBi57utb0JLOb3TJm5-wJE1rJve3HyAQi@mail.gmail.com>
Message-ID: <AANLkTim+1+RnKgRi5aWfU9NjXfs2h=P+6HnfiPBhKEtm@mail.gmail.com>

Personally I'm not familiar with Oreka. However you shouldn't need
anything from Asterisk/FreeSWITCH, just put it on the same network
segment.

http://oreka.sourceforge.net/oreka-user-manual.html#gettingvoiptraffic

-Steve

On 11 January 2011 10:11, Leonid K <pivanet at gmail.com> wrote:
> thx Steve. I'm going to look at Oreka... is there any specific info that I
> have to know before "merging" FreeSwitch with it? a saw that it has some
> patch for Asterisk... so i'd need to write smth for FreeSwitch?
>
> On Tue, Jan 11, 2011 at 11:33 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> > 2)what about limitations of the count of concurent calls that FreeSwitch
>> > can
>> > record? what if i neet it to be huge - like 18.000 calls !!!?
>>
>> For that many, you'd want to send calls through a cluster of FS boxes.
>> Something like orecx would be much better.
>>
>> -Steve
>>
>>
>> On 11 January 2011 07:59, Leonid K <pivanet at gmail.com> wrote:
>> > a few more questions:
>> > 1) can I rely on orecx?in it's openSource view (looks like it hasn't
>> > been
>> > updated since 2009) - OR it's better to get their commercial solution?
>> > 2)what about limitations of the count of concurent calls that FreeSwitch
>> > can
>> > record? what if i neet it to be huge - like 18.000 calls !!!?
>> >
>> > On Mon, Jan 10, 2011 at 7:25 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>> >>
>> >> Let me clarify. :)
>> >>
>> >> If you are ok with routing all your traffic through FreeSWITCH (or *),
>> >> then FreeSWITCH can definitely do what you want. ?I use it that way
>> >> (record all external calls) without issue. ?If your volume is high,
>> >> you'll want to record as PMCU WAV to a ramdisk and then have a cron
>> >> job that converts completed calls to mp3 or speex or whatever. ?I have
>> >> scripts that do this well including preserving the metadata that FS
>> >> can write to the WAV files for CID info or whatever else you ask it to
>> >> put in there. ?(just using sox or lame will loose that metadata from
>> >> the WAV file since they don't bother to preserve it).
>> >>
>> >> That being said. ?orecx is designed to record your VOIP traffic
>> >> without having to do anything put ensure all VOIP traffic ends up on
>> >> the network segment orecx is attached to. ?This "transparent"
>> >> recording is definitely "the way to go" if you want to separate your
>> >> phone infrastructure from your recording infrastructure.
>> >>
>> >> On Mon, Jan 10, 2011 at 11:16 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>> >> > I think the standard answer to that is www.orecx.com.
>> >> >
>> >> > On Mon, Jan 10, 2011 at 3:39 AM, Leonid K <pivanet at gmail.com> wrote:
>> >> >> Hi,
>> >> >> currently we're looking for voip recording solution - it must record
>> >> >> all
>> >> >> incoming/outgoing/internal/conference calls within the company.
>> >> >> later
>> >> >> on we
>> >> >> going to develop applications that let us fing/analyze recordings.
>> >> >> the
>> >> >> main
>> >> >> problem as I see at the moment is huge count of calls that is going
>> >> >> through
>> >> >> switch.
>> >> >>
>> >> >> what do u think about using Asterisk or/and FreeSwitch for this
>> >> >> task?
>> >> >> thanks
>> >> >> in advance!
>> >> >> --
>> >> >> Sincerely,
>> >> >> Leonid Kryvoruchko
>> >> >>
>> >> >> Mobile?? +38 093 7609175
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >>
>> >> >>
>> >> >
>> >> >
>> >> >
>> >> > --
>> >> > -Rupa
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> -Rupa
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Sincerely,
>> > Leonid Kryvoruchko
>> >
>> > Mobile?? +38 093 7609175
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Sincerely,
> Leonid Kryvoruchko
>
> Mobile?? +38 093 7609175
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From u2nsam at gmail.com  Tue Jan 11 13:44:38 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 11 Jan 2011 16:14:38 +0530
Subject: [Freeswitch-users] console
Message-ID: <AANLkTi=_ZNGF1BKsWh6CP8ExrmMYYqwzEx8C5O9OXCZ8@mail.gmail.com>

A query,

I have 2 FS running on one server on 2 different ips,
so when i do fs_cli going to respective bins , i see console of only the
first server.

Is there any way to get the console of both the FS on the same server .
I tried changing the port of event socket to 8022 but it donot works.
<param name="listen-port" value="8022"/>

Is there some method to start the console of both the instances.

Regds
Sam
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From steveayre at gmail.com  Tue Jan 11 13:50:41 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 11 Jan 2011 10:50:41 +0000
Subject: [Freeswitch-users] console
In-Reply-To: <AANLkTi=_ZNGF1BKsWh6CP8ExrmMYYqwzEx8C5O9OXCZ8@mail.gmail.com>
References: <AANLkTi=_ZNGF1BKsWh6CP8ExrmMYYqwzEx8C5O9OXCZ8@mail.gmail.com>
Message-ID: <AANLkTi=0ZCQYgpR7T4LF11oPvGn75WeinYLPYF-BQXJn@mail.gmail.com>

    <param name="listen-ip" value="127.0.0.1"/>
    <param name="listen-port" value="8021"/>

You can bind both to port 8021 on their individual IPs, or different
ports on the same IP.

A listen IP of 0.0.0.0 will mean any IP.

-Steve

On 11 January 2011 10:44, Sam <u2nsam at gmail.com> wrote:
> A query,
>
> I have 2 FS running on one server on 2 different ips,
> so when i do fs_cli going to respective bins , i see console of only the
> first server.
>
> Is there any way to get the console of both the FS on the same server .
> I tried changing the port of event socket to 8022 but it donot works.
> <param name="listen-port" value="8022"/>
>
> Is there some method to start the console of both the instances.
>
> Regds
> Sam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From pivanet at gmail.com  Tue Jan 11 14:36:10 2011
From: pivanet at gmail.com (Leonid K)
Date: Tue, 11 Jan 2011 13:36:10 +0200
Subject: [Freeswitch-users] Asterisk & FreeSwitch in recording view
In-Reply-To: <AANLkTim+1+RnKgRi5aWfU9NjXfs2h=P+6HnfiPBhKEtm@mail.gmail.com>
References: <AANLkTinZT1WiNnae4oMKd+5YdEUO9xq4_cJr6R+4siiK@mail.gmail.com>
	<AANLkTinvVqYuZKT6zHAQPW+QdqMoSaabB1WG6DLQOocO@mail.gmail.com>
	<AANLkTi=gPxWQgunBT604zT13gDe4oU8xNzWF3Xj_1aJ+@mail.gmail.com>
	<AANLkTinB6zn9e0ti7NB8y7hWQtuQzWaBjznpUo0+m3aF@mail.gmail.com>
	<AANLkTikqJi+6OMFJw49kfRert6dmFCf5DLi5Y3YzepnS@mail.gmail.com>
	<AANLkTikL=EMSBi57utb0JLOb3TJm5-wJE1rJve3HyAQi@mail.gmail.com>
	<AANLkTim+1+RnKgRi5aWfU9NjXfs2h=P+6HnfiPBhKEtm@mail.gmail.com>
Message-ID: <AANLkTintB=-42GqbwhaQQox5--iTjp-SX9TOcvQxa2iz@mail.gmail.com>

ok. thx a lot!

On Tue, Jan 11, 2011 at 12:34 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Personally I'm not familiar with Oreka. However you shouldn't need
> anything from Asterisk/FreeSWITCH, just put it on the same network
> segment.
>
> http://oreka.sourceforge.net/oreka-user-manual.html#gettingvoiptraffic
>
> -Steve
>
> On 11 January 2011 10:11, Leonid K <pivanet at gmail.com> wrote:
> > thx Steve. I'm going to look at Oreka... is there any specific info that
> I
> > have to know before "merging" FreeSwitch with it? a saw that it has some
> > patch for Asterisk... so i'd need to write smth for FreeSwitch?
> >
> > On Tue, Jan 11, 2011 at 11:33 AM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >>
> >> > 2)what about limitations of the count of concurent calls that
> FreeSwitch
> >> > can
> >> > record? what if i neet it to be huge - like 18.000 calls !!!?
> >>
> >> For that many, you'd want to send calls through a cluster of FS boxes.
> >> Something like orecx would be much better.
> >>
> >> -Steve
> >>
> >>
> >> On 11 January 2011 07:59, Leonid K <pivanet at gmail.com> wrote:
> >> > a few more questions:
> >> > 1) can I rely on orecx in it's openSource view (looks like it hasn't
> >> > been
> >> > updated since 2009) - OR it's better to get their commercial solution?
> >> > 2)what about limitations of the count of concurent calls that
> FreeSwitch
> >> > can
> >> > record? what if i neet it to be huge - like 18.000 calls !!!?
> >> >
> >> > On Mon, Jan 10, 2011 at 7:25 PM, Rupa Schomaker <rupa at rupa.com>
> wrote:
> >> >>
> >> >> Let me clarify. :)
> >> >>
> >> >> If you are ok with routing all your traffic through FreeSWITCH (or
> *),
> >> >> then FreeSWITCH can definitely do what you want.  I use it that way
> >> >> (record all external calls) without issue.  If your volume is high,
> >> >> you'll want to record as PMCU WAV to a ramdisk and then have a cron
> >> >> job that converts completed calls to mp3 or speex or whatever.  I
> have
> >> >> scripts that do this well including preserving the metadata that FS
> >> >> can write to the WAV files for CID info or whatever else you ask it
> to
> >> >> put in there.  (just using sox or lame will loose that metadata from
> >> >> the WAV file since they don't bother to preserve it).
> >> >>
> >> >> That being said.  orecx is designed to record your VOIP traffic
> >> >> without having to do anything put ensure all VOIP traffic ends up on
> >> >> the network segment orecx is attached to.  This "transparent"
> >> >> recording is definitely "the way to go" if you want to separate your
> >> >> phone infrastructure from your recording infrastructure.
> >> >>
> >> >> On Mon, Jan 10, 2011 at 11:16 AM, Rupa Schomaker <rupa at rupa.com>
> wrote:
> >> >> > I think the standard answer to that is www.orecx.com.
> >> >> >
> >> >> > On Mon, Jan 10, 2011 at 3:39 AM, Leonid K <pivanet at gmail.com>
> wrote:
> >> >> >> Hi,
> >> >> >> currently we're looking for voip recording solution - it must
> record
> >> >> >> all
> >> >> >> incoming/outgoing/internal/conference calls within the company.
> >> >> >> later
> >> >> >> on we
> >> >> >> going to develop applications that let us fing/analyze recordings.
> >> >> >> the
> >> >> >> main
> >> >> >> problem as I see at the moment is huge count of calls that is
> going
> >> >> >> through
> >> >> >> switch.
> >> >> >>
> >> >> >> what do u think about using Asterisk or/and FreeSwitch for this
> >> >> >> task?
> >> >> >> thanks
> >> >> >> in advance!
> >> >> >> --
> >> >> >> Sincerely,
> >> >> >> Leonid Kryvoruchko
> >> >> >>
> >> >> >> Mobile   +38 093 7609175
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >>
> >> >> >>
> >> >> >
> >> >> >
> >> >> >
> >> >> > --
> >> >> > -Rupa
> >> >> >
> >> >>
> >> >>
> >> >>
> >> >> --
> >> >> -Rupa
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> >
> >> > --
> >> > Sincerely,
> >> > Leonid Kryvoruchko
> >> >
> >> > Mobile   +38 093 7609175
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Sincerely,
> > Leonid Kryvoruchko
> >
> > Mobile   +38 093 7609175
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,
Leonid Kryvoruchko

Mobile   +38 093 7609175
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From w8hdkim at gmail.com  Tue Jan 11 14:38:31 2011
From: w8hdkim at gmail.com (Kim Culhan)
Date: Tue, 11 Jan 2011 06:38:31 -0500
Subject: [Freeswitch-users] IVR menu not observing timeout value
Message-ID: <AANLkTi=YXnDFeRQB=q3_VxJnC-dG200pg9JNXCyFA3RW@mail.gmail.com>

On Tue, January 11, 2011 1:20 am, Michael Collins wrote:
> I was not able to reproduce this using the demo IVR. Can you try it with
the
> demo IVR and let us know what you get?
Thanks for checking this, describing the problem on irc anthm asked for
debugging output and made a code change a short time later.

After the change the dtmf decode appears to be totally reliable.

-kim
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From hadyn_whx at hotmail.com  Tue Jan 11 14:53:31 2011
From: hadyn_whx at hotmail.com (Alex Wang)
Date: Tue, 11 Jan 2011 06:53:31 -0500
Subject: [Freeswitch-users] Any one use FreeCanadianCalls with
	FreeSwitch?
In-Reply-To: <AANLkTin+smtuNsHmjD3pOtaUu030bxxxwJpzn7Rtc2g_@mail.gmail.com>
References: <BLU0-SMTP84FDB44AA90E9823D8E3B18D0E0@phx.gbl>
	<AANLkTin+smtuNsHmjD3pOtaUu030bxxxwJpzn7Rtc2g_@mail.gmail.com>
Message-ID: <BLU0-SMTP123B69385491571A5AD9C698D0F0@phx.gbl>

Yes. Got it working...

<include>
        <gateway name="fcc">
           <param name="username" value="put freecanadiancall username"/>
           <param name="proxy" value="sip.freecanadiancalls.ca"/>
           <param name="password" value="put password here"/>
    <param name="extension-in-contact" value="true"/>
        </gateway>
</include>

Thanks

Alex

On Mon, 10 Jan 2011 21:09:34 -0800
curriegrad2004 <curriegrad2004 at gmail.com> wrote:

> Just register it as a regular gateway. Shouldn't be too hard to do. As
> for incoming calls, use the destination_number variable instead of
> ${sip_to_user} in the dialplan. Hope this clears things up for you.
> 
> On Mon, Jan 10, 2011 at 6:13 AM, Alex Wang <hadyn_whx at hotmail.com> wrote:
> > Hi
> >
> > Any one is using FreeCanadianCalls with FreeSwitch? Would you mind to
> > share the xml setting?
> >
> > Thanks
> >
> > Alex
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From u2nsam at gmail.com  Tue Jan 11 17:32:07 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 11 Jan 2011 20:02:07 +0530
Subject: [Freeswitch-users] console
In-Reply-To: <AANLkTi=0ZCQYgpR7T4LF11oPvGn75WeinYLPYF-BQXJn@mail.gmail.com>
References: <AANLkTi=_ZNGF1BKsWh6CP8ExrmMYYqwzEx8C5O9OXCZ8@mail.gmail.com>
	<AANLkTi=0ZCQYgpR7T4LF11oPvGn75WeinYLPYF-BQXJn@mail.gmail.com>
Message-ID: <AANLkTinZazjfEZcRHCuj0a9a9r5xg03r-NECCzqNzPeh@mail.gmail.com>

the scenario is i have 2 ips on 1 server for 2 FS instances;

192.168.2.1
192.168.2.2

and the parameters i have set is:-

for
192.168.2.1:-
   <param name="listen-ip" value="127.0.0.1"/>
   <param name="listen-port" value="8021"/>
for
192.168.2.2:-
   <param name="listen-ip" value="127.0.0.1"/>
   <param name="listen-port" value="8022"/>

Ideally it should work  but i am getting console for only 192.168.2.1 FS .


Regards
Sam

On Tue, Jan 11, 2011 at 4:20 PM, Steven Ayre <steveayre at gmail.com> wrote:

>    <param name="listen-ip" value="127.0.0.1"/>
>    <param name="listen-port" value="8021"/>
>
> You can bind both to port 8021 on their individual IPs, or different
> ports on the same IP.
>
> A listen IP of 0.0.0.0 will mean any IP.
>
> -Steve
>
> On 11 January 2011 10:44, Sam <u2nsam at gmail.com> wrote:
> > A query,
> >
> > I have 2 FS running on one server on 2 different ips,
> > so when i do fs_cli going to respective bins , i see console of only the
> > first server.
> >
> > Is there any way to get the console of both the FS on the same server .
> > I tried changing the port of event socket to 8022 but it donot works.
> > <param name="listen-port" value="8022"/>
> >
> > Is there some method to start the console of both the instances.
> >
> > Regds
> > Sam
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Tue Jan 11 17:34:06 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Jan 2011 08:34:06 -0600
Subject: [Freeswitch-users] Sofia fails to detect offline gateway
In-Reply-To: <AANLkTim3PeS3fELJPQ21N38c9v5p8J0wGNkBrUvc9R=t@mail.gmail.com>
References: <AANLkTim3PeS3fELJPQ21N38c9v5p8J0wGNkBrUvc9R=t@mail.gmail.com>
Message-ID: <B16E93D7-3578-4071-8899-DDE70DB4D1DC@freeswitch.org>

what version of FS are you running?

/b

On Jan 11, 2011, at 3:51 AM, Steven Ayre wrote:

> 
> Is there a mistake in my configuration?
> 
> -Steve



From steveayre at gmail.com  Tue Jan 11 17:37:00 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 11 Jan 2011 14:37:00 +0000
Subject: [Freeswitch-users] console
In-Reply-To: <AANLkTinZazjfEZcRHCuj0a9a9r5xg03r-NECCzqNzPeh@mail.gmail.com>
References: <AANLkTi=_ZNGF1BKsWh6CP8ExrmMYYqwzEx8C5O9OXCZ8@mail.gmail.com>
	<AANLkTi=0ZCQYgpR7T4LF11oPvGn75WeinYLPYF-BQXJn@mail.gmail.com>
	<AANLkTinZazjfEZcRHCuj0a9a9r5xg03r-NECCzqNzPeh@mail.gmail.com>
Message-ID: <AANLkTi=qEgee_iH+WLATtXPSCP=2-zb0Wb8aiwFNE_7+@mail.gmail.com>

It should work. Is there anything already listening on port 8022?

$ netstat -a -n -p | grep 8022

Are you also sure that they're not both loading the same config file?

Regards,
-Steve



On 11 January 2011 14:32, Sam <u2nsam at gmail.com> wrote:
> the scenario is i have 2 ips on 1 server for 2 FS instances;
>
> 192.168.2.1
> 192.168.2.2
>
> and the parameters i have set is:-
>
> for
> 192.168.2.1:-
> ?? <param name="listen-ip" value="127.0.0.1"/>
> ? ?<param name="listen-port" value="8021"/>
> for
> 192.168.2.2:-
> ?? <param name="listen-ip" value="127.0.0.1"/>
> ? ?<param name="listen-port" value="8022"/>
>
> Ideally it should work? but i am getting console for only 192.168.2.1 FS .
>
>
> Regards
> Sam
>
> On Tue, Jan 11, 2011 at 4:20 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> ? ?<param name="listen-ip" value="127.0.0.1"/>
>> ? ?<param name="listen-port" value="8021"/>
>>
>> You can bind both to port 8021 on their individual IPs, or different
>> ports on the same IP.
>>
>> A listen IP of 0.0.0.0 will mean any IP.
>>
>> -Steve
>>
>> On 11 January 2011 10:44, Sam <u2nsam at gmail.com> wrote:
>> > A query,
>> >
>> > I have 2 FS running on one server on 2 different ips,
>> > so when i do fs_cli going to respective bins , i see console of only the
>> > first server.
>> >
>> > Is there any way to get the console of both the FS on the same server .
>> > I tried changing the port of event socket to 8022 but it donot works.
>> > <param name="listen-port" value="8022"/>
>> >
>> > Is there some method to start the console of both the instances.
>> >
>> > Regds
>> > Sam
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steveayre at gmail.com  Tue Jan 11 17:37:43 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 11 Jan 2011 14:37:43 +0000
Subject: [Freeswitch-users] Sofia fails to detect offline gateway
In-Reply-To: <B16E93D7-3578-4071-8899-DDE70DB4D1DC@freeswitch.org>
References: <AANLkTim3PeS3fELJPQ21N38c9v5p8J0wGNkBrUvc9R=t@mail.gmail.com>
	<B16E93D7-3578-4071-8899-DDE70DB4D1DC@freeswitch.org>
Message-ID: <AANLkTin2j-+Y11_x4bs0+M6UsVKUN14fXHoZ262vFu26@mail.gmail.com>

freeswitch at internal> version
FreeSWITCH Version 1.0.head (git-a8b2840 2011-01-07 17-53-09 -0600)

-Steve


On 11 January 2011 14:34, Brian West <brian at freeswitch.org> wrote:
> what version of FS are you running?
>
> /b
>
> On Jan 11, 2011, at 3:51 AM, Steven Ayre wrote:
>
>>
>> Is there a mistake in my configuration?
>>
>> -Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From u2nsam at gmail.com  Tue Jan 11 18:04:30 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 11 Jan 2011 20:34:30 +0530
Subject: [Freeswitch-users] console
In-Reply-To: <AANLkTi=qEgee_iH+WLATtXPSCP=2-zb0Wb8aiwFNE_7+@mail.gmail.com>
References: <AANLkTi=_ZNGF1BKsWh6CP8ExrmMYYqwzEx8C5O9OXCZ8@mail.gmail.com>
	<AANLkTi=0ZCQYgpR7T4LF11oPvGn75WeinYLPYF-BQXJn@mail.gmail.com>
	<AANLkTinZazjfEZcRHCuj0a9a9r5xg03r-NECCzqNzPeh@mail.gmail.com>
	<AANLkTi=qEgee_iH+WLATtXPSCP=2-zb0Wb8aiwFNE_7+@mail.gmail.com>
Message-ID: <AANLkTimVk-AZ3dPnE-Kkj4tE8XS7WuLsQzczqTW1fBqf@mail.gmail.com>

Something more here ... i am getting the console for 192.168.2.1 every time
i do fs_cli on both instances .

like
/usr/local/FS_1/bin/fs_cli
/usr/localFS_2/bin/fs_cli
i get the console for the 1st server only

the 2 server are listing to 2 different ips .

Regds
Sam

On Tue, Jan 11, 2011 at 8:07 PM, Steven Ayre <steveayre at gmail.com> wrote:

> It should work. Is there anything already listening on port 8022?
>
> $ netstat -a -n -p | grep 8022
>
> Are you also sure that they're not both loading the same config file?
>
> Regards,
> -Steve
>
>
>
> On 11 January 2011 14:32, Sam <u2nsam at gmail.com> wrote:
> > the scenario is i have 2 ips on 1 server for 2 FS instances;
> >
> > 192.168.2.1
> > 192.168.2.2
> >
> > and the parameters i have set is:-
> >
> > for
> > 192.168.2.1:-
> >    <param name="listen-ip" value="127.0.0.1"/>
> >    <param name="listen-port" value="8021"/>
> > for
> > 192.168.2.2:-
> >    <param name="listen-ip" value="127.0.0.1"/>
> >    <param name="listen-port" value="8022"/>
> >
> > Ideally it should work  but i am getting console for only 192.168.2.1 FS
> .
> >
> >
> > Regards
> > Sam
> >
> > On Tue, Jan 11, 2011 at 4:20 PM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >>
> >>    <param name="listen-ip" value="127.0.0.1"/>
> >>    <param name="listen-port" value="8021"/>
> >>
> >> You can bind both to port 8021 on their individual IPs, or different
> >> ports on the same IP.
> >>
> >> A listen IP of 0.0.0.0 will mean any IP.
> >>
> >> -Steve
> >>
> >> On 11 January 2011 10:44, Sam <u2nsam at gmail.com> wrote:
> >> > A query,
> >> >
> >> > I have 2 FS running on one server on 2 different ips,
> >> > so when i do fs_cli going to respective bins , i see console of only
> the
> >> > first server.
> >> >
> >> > Is there any way to get the console of both the FS on the same server
> .
> >> > I tried changing the port of event socket to 8022 but it donot works.
> >> > <param name="listen-port" value="8022"/>
> >> >
> >> > Is there some method to start the console of both the instances.
> >> >
> >> > Regds
> >> > Sam
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rupa at rupa.com  Tue Jan 11 18:09:46 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 11 Jan 2011 09:09:46 -0600
Subject: [Freeswitch-users] How to disable 5060-5080 with auto-nat at
	upnp?
In-Reply-To: <BLU0-SMTP97E64484D5F1A2D4F0FEFB8D0F0@phx.gbl>
References: <BLU0-SMTP97E64484D5F1A2D4F0FEFB8D0F0@phx.gbl>
Message-ID: <AANLkTi=UeJNn-2bimdA+PkJHL+deLkh9HaQuyL-DmC=4@mail.gmail.com>

Hmmm...  I didn't put anything in the nat_map code to allow some parts
of the sofia profile to participate in the nat mapping while others do
not.

Brian, do you have any ideas?

Alex, I'd suggest just blocking ports 5060-5080 on the firewall.  The
port blocking *should* take precedence over the upnp maps.

On Mon, Jan 10, 2011 at 10:17 PM, Alex Wang <hadyn_whx at hotmail.com> wrote:
> Hi All
>
> Just don't want to open these port to the public, all my extensions are
> in my LAN. If I disable upnp, the outside sip register is not working, I
> mean no sound.
> How to disable the auto-nat publish 5060-5080 on the internet?
>
>
> Thanks
>
> Alex
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa


From hesser4900 at gmail.com  Tue Jan 11 18:09:49 2011
From: hesser4900 at gmail.com (Holger Esser)
Date: Tue, 11 Jan 2011 09:09:49 -0600
Subject: [Freeswitch-users] ESL mod_socket mod_commands
Message-ID: <AANLkTikXOEhiN4zyRkmbTdPnWfBwuyAt9VguUNDCgzQc@mail.gmail.com>

Hi guys,

What would be the best way to play an audio file with the ESL/mod_socket?
I assume it is uuid_broadcast. Would it be possible to use mod_shout with it
like it does in uuid_displace?
Or should I just use uuid_displace for any audio file plays?

Kind regards,
holger
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From rupa at rupa.com  Tue Jan 11 18:14:38 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 11 Jan 2011 09:14:38 -0600
Subject: [Freeswitch-users] Asterisk & FreeSwitch in recording view
In-Reply-To: <AANLkTinB6zn9e0ti7NB8y7hWQtuQzWaBjznpUo0+m3aF@mail.gmail.com>
References: <AANLkTinZT1WiNnae4oMKd+5YdEUO9xq4_cJr6R+4siiK@mail.gmail.com>
	<AANLkTinvVqYuZKT6zHAQPW+QdqMoSaabB1WG6DLQOocO@mail.gmail.com>
	<AANLkTi=gPxWQgunBT604zT13gDe4oU8xNzWF3Xj_1aJ+@mail.gmail.com>
	<AANLkTinB6zn9e0ti7NB8y7hWQtuQzWaBjznpUo0+m3aF@mail.gmail.com>
Message-ID: <AANLkTi=qbSagegawMLFEKZnp0-z9D9Fj4RS09n0yEyF3@mail.gmail.com>

1) Dunno, I don't use it just have heard enough people recommend it.

2) I think you got that one answered...  It is possible to do either
way.  You'll need a serious disk subsystem to handle that volume
though.  Battery backed raid (to deal with inefficient writes) is
gonna be your friend here.  And of course if you do it with FS you'll
want a cluster.  The nice thing about orecx is that if it fails for
whatever reason it will not impact your ability to process calls.
With FS, you can certainly make it fault tolerant but you'll need to
do a bunch more testing and have more confidence that adding recording
impact your ability to process/handle calls.  If this were my setup,
I'd definitely want to decouple the recording from the processing.
Hence the orecx recommendation.

On Tue, Jan 11, 2011 at 1:59 AM, Leonid K <pivanet at gmail.com> wrote:
> a few more questions:
> 1) can I rely on orecx?in it's openSource view (looks like it hasn't been
> updated since 2009) - OR it's better to get their commercial solution?
> 2)what about limitations of the count of concurent calls that FreeSwitch can
> record? what if i neet it to be huge - like 18.000 calls !!!?
>
> On Mon, Jan 10, 2011 at 7:25 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>> Let me clarify. :)
>>
>> If you are ok with routing all your traffic through FreeSWITCH (or *),
>> then FreeSWITCH can definitely do what you want. ?I use it that way
>> (record all external calls) without issue. ?If your volume is high,
>> you'll want to record as PMCU WAV to a ramdisk and then have a cron
>> job that converts completed calls to mp3 or speex or whatever. ?I have
>> scripts that do this well including preserving the metadata that FS
>> can write to the WAV files for CID info or whatever else you ask it to
>> put in there. ?(just using sox or lame will loose that metadata from
>> the WAV file since they don't bother to preserve it).
>>
>> That being said. ?orecx is designed to record your VOIP traffic
>> without having to do anything put ensure all VOIP traffic ends up on
>> the network segment orecx is attached to. ?This "transparent"
>> recording is definitely "the way to go" if you want to separate your
>> phone infrastructure from your recording infrastructure.
>>
>> On Mon, Jan 10, 2011 at 11:16 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>> > I think the standard answer to that is www.orecx.com.
>> >
>> > On Mon, Jan 10, 2011 at 3:39 AM, Leonid K <pivanet at gmail.com> wrote:
>> >> Hi,
>> >> currently we're looking for voip recording solution - it must record
>> >> all
>> >> incoming/outgoing/internal/conference calls within the company. later
>> >> on we
>> >> going to develop applications that let us fing/analyze recordings. the
>> >> main
>> >> problem as I see at the moment is huge count of calls that is going
>> >> through
>> >> switch.
>> >>
>> >> what do u think about using Asterisk or/and FreeSwitch for this task?
>> >> thanks
>> >> in advance!
>> >> --
>> >> Sincerely,
>> >> Leonid Kryvoruchko
>> >>
>> >> Mobile?? +38 093 7609175
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> >
>> >
>> > --
>> > -Rupa
>> >
>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Sincerely,
> Leonid Kryvoruchko
>
> Mobile?? +38 093 7609175
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From peter.olsson at visionutveckling.se  Tue Jan 11 18:19:50 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 11 Jan 2011 16:19:50 +0100
Subject: [Freeswitch-users] ESL mod_socket mod_commands
In-Reply-To: <AANLkTikXOEhiN4zyRkmbTdPnWfBwuyAt9VguUNDCgzQc@mail.gmail.com>
References: <AANLkTikXOEhiN4zyRkmbTdPnWfBwuyAt9VguUNDCgzQc@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57ECF05AB1@cooper>

To play a sound file in ESL I simply use (when intended for one leg only, for instance in an IVR app).

SendMsg <uuid>
call-command: execute
execute-app-name: playback
execute-app-arg: /path/to/file.wav

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Holger Esser
Skickat: den 11 januari 2011 16:10
Till: FreeSWITCH-users at lists.freeswitch.org
?mne: [Freeswitch-users] ESL mod_socket mod_commands

Hi guys,

What would be the best way to play an audio file with the ESL/mod_socket?
I assume it is uuid_broadcast. Would it be possible to use mod_shout with it like it does in uuid_displace?
Or should I just use uuid_displace for any audio file plays?

Kind regards,
holger
!DSPAM:4d2c73e032765465720928!
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From hesser4900 at gmail.com  Tue Jan 11 18:30:20 2011
From: hesser4900 at gmail.com (Holger Esser)
Date: Tue, 11 Jan 2011 09:30:20 -0600
Subject: [Freeswitch-users] ESL mod_socket mod_commands
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57ECF05AB1@cooper>
References: <AANLkTikXOEhiN4zyRkmbTdPnWfBwuyAt9VguUNDCgzQc@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57ECF05AB1@cooper>
Message-ID: <AANLkTinKqQf12VkM3-kBmLeJ+JdQM5WHFgycqcpd1OND@mail.gmail.com>

Many thanks Peter. Sometimes I do not see the forest for the trees.

On Tue, Jan 11, 2011 at 9:19 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> To play a sound file in ESL I simply use (when intended for one leg only,
> for instance in an IVR app).
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: playback
>
> execute-app-arg: /path/to/file.wav
>
>
>
> /Peter
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Holger Esser
> *Skickat:* den 11 januari 2011 16:10
> *Till:* FreeSWITCH-users at lists.freeswitch.org
> *?mne:* [Freeswitch-users] ESL mod_socket mod_commands
>
>
>
> Hi guys,
>
>
>
> What would be the best way to play an audio file with the ESL/mod_socket?
>
> I assume it is uuid_broadcast. Would it be possible to use mod_shout with
> it like it does in uuid_displace?
>
> Or should I just use uuid_displace for any audio file plays?
>
>
>
> Kind regards,
>
> holger
>
> !DSPAM:4d2c73e032765465720928!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From hadyn_whx at hotmail.com  Tue Jan 11 18:51:27 2011
From: hadyn_whx at hotmail.com (Alex Wang)
Date: Tue, 11 Jan 2011 10:51:27 -0500
Subject: [Freeswitch-users] console
In-Reply-To: <AANLkTimVk-AZ3dPnE-Kkj4tE8XS7WuLsQzczqTW1fBqf@mail.gmail.com>
References: <AANLkTi=qEgee_iH+WLATtXPSCP=2-zb0Wb8aiwFNE_7+@mail.gmail.com>
	<AANLkTimVk-AZ3dPnE-Kkj4tE8XS7WuLsQzczqTW1fBqf@mail.gmail.com>
Message-ID: <BLU0-SMTP10376BDBBA6FF74514B30708D0F0@phx.gbl>

Usage: fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x command] [profile]

  -?,-h --help                    Usage Information
  -H, --host=hostname             Host to connect
  -P, --port=port                 Port to connect (1 - 65535)

I think you need to use fs_cli -H 192.168.2.2 to connect to other fs.

Alex


On Tue, 11 Jan 2011 20:34:30 +0530
Sam <u2nsam at gmail.com> wrote:

> Something more here ... i am getting the console for 192.168.2.1 every time
> i do fs_cli on both instances .
> 
> like
> /usr/local/FS_1/bin/fs_cli
> /usr/localFS_2/bin/fs_cli
> i get the console for the 1st server only
> 
> the 2 server are listing to 2 different ips .
> 
> Regds
> Sam
> 
> On Tue, Jan 11, 2011 at 8:07 PM, Steven Ayre <steveayre at gmail.com> wrote:
> 
> > It should work. Is there anything already listening on port 8022?
> >
> > $ netstat -a -n -p | grep 8022
> >
> > Are you also sure that they're not both loading the same config file?
> >
> > Regards,
> > -Steve
> >
> >
> >
> > On 11 January 2011 14:32, Sam <u2nsam at gmail.com> wrote:
> > > the scenario is i have 2 ips on 1 server for 2 FS instances;
> > >
> > > 192.168.2.1
> > > 192.168.2.2
> > >
> > > and the parameters i have set is:-
> > >
> > > for
> > > 192.168.2.1:-
> > >    <param name="listen-ip" value="127.0.0.1"/>
> > >    <param name="listen-port" value="8021"/>
> > > for
> > > 192.168.2.2:-
> > >    <param name="listen-ip" value="127.0.0.1"/>
> > >    <param name="listen-port" value="8022"/>
> > >
> > > Ideally it should work  but i am getting console for only 192.168.2.1 FS
> > .
> > >
> > >
> > > Regards
> > > Sam
> > >
> > > On Tue, Jan 11, 2011 at 4:20 PM, Steven Ayre <steveayre at gmail.com>
> > wrote:
> > >>
> > >>    <param name="listen-ip" value="127.0.0.1"/>
> > >>    <param name="listen-port" value="8021"/>
> > >>
> > >> You can bind both to port 8021 on their individual IPs, or different
> > >> ports on the same IP.
> > >>
> > >> A listen IP of 0.0.0.0 will mean any IP.
> > >>
> > >> -Steve
> > >>
> > >> On 11 January 2011 10:44, Sam <u2nsam at gmail.com> wrote:
> > >> > A query,
> > >> >
> > >> > I have 2 FS running on one server on 2 different ips,
> > >> > so when i do fs_cli going to respective bins , i see console of only
> > the
> > >> > first server.
> > >> >
> > >> > Is there any way to get the console of both the FS on the same server
> > .
> > >> > I tried changing the port of event socket to 8022 but it donot works.
> > >> > <param name="listen-port" value="8022"/>
> > >> >
> > >> > Is there some method to start the console of both the instances.
> > >> >
> > >> > Regds
> > >> > Sam
> > >> >
> > >> > _______________________________________________
> > >> > FreeSWITCH-users mailing list
> > >> > FreeSWITCH-users at lists.freeswitch.org
> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> > UNSUBSCRIBE:
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> > http://www.freeswitch.org
> > >> >
> > >> >
> > >>
> > >> _______________________________________________
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> http://www.freeswitch.org
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >





From steveayre at gmail.com  Tue Jan 11 19:08:13 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 11 Jan 2011 16:08:13 +0000
Subject: [Freeswitch-users] console
In-Reply-To: <AANLkTimVk-AZ3dPnE-Kkj4tE8XS7WuLsQzczqTW1fBqf@mail.gmail.com>
References: <AANLkTi=_ZNGF1BKsWh6CP8ExrmMYYqwzEx8C5O9OXCZ8@mail.gmail.com>
	<AANLkTi=0ZCQYgpR7T4LF11oPvGn75WeinYLPYF-BQXJn@mail.gmail.com>
	<AANLkTinZazjfEZcRHCuj0a9a9r5xg03r-NECCzqNzPeh@mail.gmail.com>
	<AANLkTi=qEgee_iH+WLATtXPSCP=2-zb0Wb8aiwFNE_7+@mail.gmail.com>
	<AANLkTimVk-AZ3dPnE-Kkj4tE8XS7WuLsQzczqTW1fBqf@mail.gmail.com>
Message-ID: <AANLkTiknTsNhvbt-8D185sy_+LbmfJhw6HGR3+C_XgWZ@mail.gmail.com>

/usr/local/FS_1/bin/fs_cli -P 8021
/usr/local/FS_2/bin/fs_cli -P 8022

fs_cll doesn't read any config file. It's not part of the FS server at all,
you can have it on a different machine that doesn't have FS installed. It
entirely relies on the arguments to control where to connect to.

-Steve


On 11 January 2011 15:04, Sam <u2nsam at gmail.com> wrote:

> Something more here ... i am getting the console for 192.168.2.1 every time
> i do fs_cli on both instances .
>
> like
> /usr/local/FS_1/bin/fs_cli
> /usr/localFS_2/bin/fs_cli
> i get the console for the 1st server only
>
> the 2 server are listing to 2 different ips .
>
> Regds
> Sam
>
>
> On Tue, Jan 11, 2011 at 8:07 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> It should work. Is there anything already listening on port 8022?
>>
>> $ netstat -a -n -p | grep 8022
>>
>> Are you also sure that they're not both loading the same config file?
>>
>> Regards,
>> -Steve
>>
>>
>>
>> On 11 January 2011 14:32, Sam <u2nsam at gmail.com> wrote:
>> > the scenario is i have 2 ips on 1 server for 2 FS instances;
>> >
>> > 192.168.2.1
>> > 192.168.2.2
>> >
>> > and the parameters i have set is:-
>> >
>> > for
>> > 192.168.2.1:-
>> >    <param name="listen-ip" value="127.0.0.1"/>
>> >    <param name="listen-port" value="8021"/>
>> > for
>> > 192.168.2.2:-
>> >    <param name="listen-ip" value="127.0.0.1"/>
>> >    <param name="listen-port" value="8022"/>
>> >
>> > Ideally it should work  but i am getting console for only 192.168.2.1 FS
>> .
>> >
>> >
>> > Regards
>> > Sam
>> >
>> > On Tue, Jan 11, 2011 at 4:20 PM, Steven Ayre <steveayre at gmail.com>
>> wrote:
>> >>
>> >>    <param name="listen-ip" value="127.0.0.1"/>
>> >>    <param name="listen-port" value="8021"/>
>> >>
>> >> You can bind both to port 8021 on their individual IPs, or different
>> >> ports on the same IP.
>> >>
>> >> A listen IP of 0.0.0.0 will mean any IP.
>> >>
>> >> -Steve
>> >>
>> >> On 11 January 2011 10:44, Sam <u2nsam at gmail.com> wrote:
>> >> > A query,
>> >> >
>> >> > I have 2 FS running on one server on 2 different ips,
>> >> > so when i do fs_cli going to respective bins , i see console of only
>> the
>> >> > first server.
>> >> >
>> >> > Is there any way to get the console of both the FS on the same server
>> .
>> >> > I tried changing the port of event socket to 8022 but it donot works.
>> >> > <param name="listen-port" value="8022"/>
>> >> >
>> >> > Is there some method to start the console of both the instances.
>> >> >
>> >> > Regds
>> >> > Sam
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Tue Jan 11 19:11:56 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 11 Jan 2011 16:11:56 +0000
Subject: [Freeswitch-users] console
In-Reply-To: <AANLkTiknTsNhvbt-8D185sy_+LbmfJhw6HGR3+C_XgWZ@mail.gmail.com>
References: <AANLkTi=_ZNGF1BKsWh6CP8ExrmMYYqwzEx8C5O9OXCZ8@mail.gmail.com>
	<AANLkTi=0ZCQYgpR7T4LF11oPvGn75WeinYLPYF-BQXJn@mail.gmail.com>
	<AANLkTinZazjfEZcRHCuj0a9a9r5xg03r-NECCzqNzPeh@mail.gmail.com>
	<AANLkTi=qEgee_iH+WLATtXPSCP=2-zb0Wb8aiwFNE_7+@mail.gmail.com>
	<AANLkTimVk-AZ3dPnE-Kkj4tE8XS7WuLsQzczqTW1fBqf@mail.gmail.com>
	<AANLkTiknTsNhvbt-8D185sy_+LbmfJhw6HGR3+C_XgWZ@mail.gmail.com>
Message-ID: <AANLkTimfWAum+7S91Het9EuJ7C8jOP3d+yoix-f7+38H@mail.gmail.com>

Any reason you have FS installed in two locations?

You can save some disk space by two versions from the same path.

ss5:/opt/freeswitch/bin# ./freeswitch --help
these are the optional arguments you can pass to freeswitch
        -nf                    -- no forking
        -u [user]              -- specify user to switch to
        -g [group]             -- specify group to switch to
        -help                  -- this message
        -version               -- print the version and exit
        -waste                 -- allow memory waste
        -core                  -- dump cores
        -hp                    -- enable high priority settings
        -vg                    -- run under valgrind
        -nosql                 -- disable internal sql scoreboard
        -heavy-timer           -- Heavy Timer, possibly more accurate but at
a cost
        -nonat                 -- disable auto nat detection
        -nocal                 -- disable clock calibration
        -nort                  -- disable clock clock_realtime
        -stop                  -- stop freeswitch
        -nc                    -- do not output to a console and background
        -ncwait                -- do not output to a console and background
but wait until the system is ready before exiting (implies -nc)
        -c                     -- output to a console and stay in the
foreground
        -conf [confdir]        -- specify an alternate config dir
        -log [logdir]          -- specify an alternate log dir
        -run [rundir]          -- specify an alternate run dir
        -db [dbdir]            -- specify an alternate db dir
        -mod [moddir]          -- specify an alternate mod dir
        -htdocs [htdocsdir]    -- specify an alternate htdocs dir
        -scripts [scriptsdir]  -- specify an alternate scripts dir

Those last -conf -log -run -db options would let you use different
directories for configs, logs and sqlite databases per-instance but use the
same binaries and modules.

Of course there's no reason not to do it the way you are now - it'll just
save a bit of space this way.

You can also use two SIP profiles to have the same FS process listen on 2
different IPs. I guess you have a reason though for having them separated
into two processes though.

-Steve



On 11 January 2011 16:08, Steven Ayre <steveayre at gmail.com> wrote:

> /usr/local/FS_1/bin/fs_cli -P 8021
> /usr/local/FS_2/bin/fs_cli -P 8022
>
> fs_cll doesn't read any config file. It's not part of the FS server at all,
> you can have it on a different machine that doesn't have FS installed. It
> entirely relies on the arguments to control where to connect to.
>
> -Steve
>
>
>
> On 11 January 2011 15:04, Sam <u2nsam at gmail.com> wrote:
>
>> Something more here ... i am getting the console for 192.168.2.1 every
>> time i do fs_cli on both instances .
>>
>> like
>> /usr/local/FS_1/bin/fs_cli
>> /usr/localFS_2/bin/fs_cli
>> i get the console for the 1st server only
>>
>> the 2 server are listing to 2 different ips .
>>
>> Regds
>> Sam
>>
>>
>> On Tue, Jan 11, 2011 at 8:07 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> It should work. Is there anything already listening on port 8022?
>>>
>>> $ netstat -a -n -p | grep 8022
>>>
>>> Are you also sure that they're not both loading the same config file?
>>>
>>> Regards,
>>> -Steve
>>>
>>>
>>>
>>> On 11 January 2011 14:32, Sam <u2nsam at gmail.com> wrote:
>>> > the scenario is i have 2 ips on 1 server for 2 FS instances;
>>> >
>>> > 192.168.2.1
>>> > 192.168.2.2
>>> >
>>> > and the parameters i have set is:-
>>> >
>>> > for
>>> > 192.168.2.1:-
>>> >    <param name="listen-ip" value="127.0.0.1"/>
>>> >    <param name="listen-port" value="8021"/>
>>> > for
>>> > 192.168.2.2:-
>>> >    <param name="listen-ip" value="127.0.0.1"/>
>>> >    <param name="listen-port" value="8022"/>
>>> >
>>> > Ideally it should work  but i am getting console for only 192.168.2.1
>>> FS .
>>> >
>>> >
>>> > Regards
>>> > Sam
>>> >
>>> > On Tue, Jan 11, 2011 at 4:20 PM, Steven Ayre <steveayre at gmail.com>
>>> wrote:
>>> >>
>>> >>    <param name="listen-ip" value="127.0.0.1"/>
>>> >>    <param name="listen-port" value="8021"/>
>>> >>
>>> >> You can bind both to port 8021 on their individual IPs, or different
>>> >> ports on the same IP.
>>> >>
>>> >> A listen IP of 0.0.0.0 will mean any IP.
>>> >>
>>> >> -Steve
>>> >>
>>> >> On 11 January 2011 10:44, Sam <u2nsam at gmail.com> wrote:
>>> >> > A query,
>>> >> >
>>> >> > I have 2 FS running on one server on 2 different ips,
>>> >> > so when i do fs_cli going to respective bins , i see console of only
>>> the
>>> >> > first server.
>>> >> >
>>> >> > Is there any way to get the console of both the FS on the same
>>> server .
>>> >> > I tried changing the port of event socket to 8022 but it donot
>>> works.
>>> >> > <param name="listen-port" value="8022"/>
>>> >> >
>>> >> > Is there some method to start the console of both the instances.
>>> >> >
>>> >> > Regds
>>> >> > Sam
>>> >> >
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> > http://www.freeswitch.org
>>> >> >
>>> >> >
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From hadyn_whx at hotmail.com  Tue Jan 11 19:26:17 2011
From: hadyn_whx at hotmail.com (Alex Wang)
Date: Tue, 11 Jan 2011 11:26:17 -0500
Subject: [Freeswitch-users] How to disable 5060-5080 with auto-nat at
	upnp?
In-Reply-To: <AANLkTi=UeJNn-2bimdA+PkJHL+deLkh9HaQuyL-DmC=4@mail.gmail.com>
References: <BLU0-SMTP97E64484D5F1A2D4F0FEFB8D0F0@phx.gbl>
	<AANLkTi=UeJNn-2bimdA+PkJHL+deLkh9HaQuyL-DmC=4@mail.gmail.com>
Message-ID: <BLU0-SMTP7A86F22BAE5746316A3478D0F0@phx.gbl>

Hi Rupa

Do you mean create a port map on the router to map to other ip instead
of FS?

update: That seems working. But do you think I can just turn off the
auto-nat and map the rtp port out to get it working? Which way is the
best? In the asterisk, I don't even need turn on upnp and don't need to
map the port and it works fine behind the router, even those ATAs,
normally you don't need to map any port on the router and they just
register from isp and works fine with router(non UPNP & UPNP), why FS
need do those extra step? I try to find the answer in the WiKi but cant
find the explaination for that.

Thanks

Alex


On Tue, 11 Jan 2011 09:09:46 -0600
Rupa Schomaker <rupa at rupa.com> wrote:

> Hmmm...  I didn't put anything in the nat_map code to allow some parts
> of the sofia profile to participate in the nat mapping while others do
> not.
> 
> Brian, do you have any ideas?
> 
> Alex, I'd suggest just blocking ports 5060-5080 on the firewall.  The
> port blocking *should* take precedence over the upnp maps.
> 
> On Mon, Jan 10, 2011 at 10:17 PM, Alex Wang <hadyn_whx at hotmail.com> wrote:
> > Hi All
> >
> > Just don't want to open these port to the public, all my extensions are
> > in my LAN. If I disable upnp, the outside sip register is not working, I
> > mean no sound.
> > How to disable the auto-nat publish 5060-5080 on the internet?
> >
> >
> > Thanks
> >
> > Alex
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> 
> -- 
> -Rupa
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From rupa at rupa.com  Tue Jan 11 19:34:33 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 11 Jan 2011 10:34:33 -0600
Subject: [Freeswitch-users] How to disable 5060-5080 with auto-nat at
	upnp?
In-Reply-To: <BLU0-SMTP7A86F22BAE5746316A3478D0F0@phx.gbl>
References: <BLU0-SMTP97E64484D5F1A2D4F0FEFB8D0F0@phx.gbl>
	<AANLkTi=UeJNn-2bimdA+PkJHL+deLkh9HaQuyL-DmC=4@mail.gmail.com>
	<BLU0-SMTP7A86F22BAE5746316A3478D0F0@phx.gbl>
Message-ID: <AANLkTi=JrAocgCY0uFVAD149R7pz7m4aqgkdkmj1KYHf@mail.gmail.com>

Well, there is no "the RTP port" -- there is a range which is quite
large by default.

I mean setup a rule to block the port, not point it somewhere else.
Though I guess you could point it to a black hole (IP that is never
assigned).

Why/how * works, I dunno.  Maybe someone else can answer that
question.  I don't see how to handle RTP data properly through a
firewall without support for portmapping initiated by FS (using upnp
or nat-pmp) or having a SIP ALG running on the firewall.

On Tue, Jan 11, 2011 at 10:26 AM, Alex Wang <hadyn_whx at hotmail.com> wrote:
> Hi Rupa
>
> Do you mean create a port map on the router to map to other ip instead
> of FS?
>
> update: That seems working. But do you think I can just turn off the
> auto-nat and map the rtp port out to get it working? Which way is the
> best? In the asterisk, I don't even need turn on upnp and don't need to
> map the port and it works fine behind the router, even those ATAs,
> normally you don't need to map any port on the router and they just
> register from isp and works fine with router(non UPNP & UPNP), why FS
> need do those extra step? I try to find the answer in the WiKi but cant
> find the explaination for that.
>
> Thanks
>
> Alex
>
>
> On Tue, 11 Jan 2011 09:09:46 -0600
> Rupa Schomaker <rupa at rupa.com> wrote:
>
>> Hmmm... ?I didn't put anything in the nat_map code to allow some parts
>> of the sofia profile to participate in the nat mapping while others do
>> not.
>>
>> Brian, do you have any ideas?
>>
>> Alex, I'd suggest just blocking ports 5060-5080 on the firewall. ?The
>> port blocking *should* take precedence over the upnp maps.
>>
>> On Mon, Jan 10, 2011 at 10:17 PM, Alex Wang <hadyn_whx at hotmail.com> wrote:
>> > Hi All
>> >
>> > Just don't want to open these port to the public, all my extensions are
>> > in my LAN. If I disable upnp, the outside sip register is not working, I
>> > mean no sound.
>> > How to disable the auto-nat publish 5060-5080 on the internet?
>> >
>> >
>> > Thanks
>> >
>> > Alex
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa


From u2nsam at gmail.com  Tue Jan 11 19:52:23 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 11 Jan 2011 22:22:23 +0530
Subject: [Freeswitch-users] console
In-Reply-To: <AANLkTimfWAum+7S91Het9EuJ7C8jOP3d+yoix-f7+38H@mail.gmail.com>
References: <AANLkTi=_ZNGF1BKsWh6CP8ExrmMYYqwzEx8C5O9OXCZ8@mail.gmail.com>
	<AANLkTi=0ZCQYgpR7T4LF11oPvGn75WeinYLPYF-BQXJn@mail.gmail.com>
	<AANLkTinZazjfEZcRHCuj0a9a9r5xg03r-NECCzqNzPeh@mail.gmail.com>
	<AANLkTi=qEgee_iH+WLATtXPSCP=2-zb0Wb8aiwFNE_7+@mail.gmail.com>
	<AANLkTimVk-AZ3dPnE-Kkj4tE8XS7WuLsQzczqTW1fBqf@mail.gmail.com>
	<AANLkTiknTsNhvbt-8D185sy_+LbmfJhw6HGR3+C_XgWZ@mail.gmail.com>
	<AANLkTimfWAum+7S91Het9EuJ7C8jOP3d+yoix-f7+38H@mail.gmail.com>
Message-ID: <AANLkTimFhgEgzUHdoZqr41Qxy_Dk4+cFaZ60mAjFqus7@mail.gmail.com>

It an experimental setup to check the resource utilization and threading,
also will keep the structure simple and manageable plus different recording
directories.

Regards
Sam

On Tue, Jan 11, 2011 at 9:41 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Any reason you have FS installed in two locations?
>
> You can save some disk space by two versions from the same path.
>
> ss5:/opt/freeswitch/bin# ./freeswitch --help
> these are the optional arguments you can pass to freeswitch
>         -nf                    -- no forking
>         -u [user]              -- specify user to switch to
>         -g [group]             -- specify group to switch to
>         -help                  -- this message
>         -version               -- print the version and exit
>         -waste                 -- allow memory waste
>         -core                  -- dump cores
>         -hp                    -- enable high priority settings
>         -vg                    -- run under valgrind
>         -nosql                 -- disable internal sql scoreboard
>         -heavy-timer           -- Heavy Timer, possibly more accurate but
> at a cost
>         -nonat                 -- disable auto nat detection
>         -nocal                 -- disable clock calibration
>         -nort                  -- disable clock clock_realtime
>         -stop                  -- stop freeswitch
>         -nc                    -- do not output to a console and background
>         -ncwait                -- do not output to a console and background
> but wait until the system is ready before exiting (implies -nc)
>         -c                     -- output to a console and stay in the
> foreground
>         -conf [confdir]        -- specify an alternate config dir
>         -log [logdir]          -- specify an alternate log dir
>         -run [rundir]          -- specify an alternate run dir
>         -db [dbdir]            -- specify an alternate db dir
>         -mod [moddir]          -- specify an alternate mod dir
>         -htdocs [htdocsdir]    -- specify an alternate htdocs dir
>         -scripts [scriptsdir]  -- specify an alternate scripts dir
>
> Those last -conf -log -run -db options would let you use different
> directories for configs, logs and sqlite databases per-instance but use the
> same binaries and modules.
>
> Of course there's no reason not to do it the way you are now - it'll just
> save a bit of space this way.
>
> You can also use two SIP profiles to have the same FS process listen on 2
> different IPs. I guess you have a reason though for having them separated
> into two processes though.
>
> -Steve
>
>
>
>
> On 11 January 2011 16:08, Steven Ayre <steveayre at gmail.com> wrote:
>
>> /usr/local/FS_1/bin/fs_cli -P 8021
>> /usr/local/FS_2/bin/fs_cli -P 8022
>>
>> fs_cll doesn't read any config file. It's not part of the FS server at
>> all, you can have it on a different machine that doesn't have FS installed.
>> It entirely relies on the arguments to control where to connect to.
>>
>> -Steve
>>
>>
>>
>> On 11 January 2011 15:04, Sam <u2nsam at gmail.com> wrote:
>>
>>> Something more here ... i am getting the console for 192.168.2.1 every
>>> time i do fs_cli on both instances .
>>>
>>> like
>>> /usr/local/FS_1/bin/fs_cli
>>> /usr/localFS_2/bin/fs_cli
>>> i get the console for the 1st server only
>>>
>>> the 2 server are listing to 2 different ips .
>>>
>>> Regds
>>> Sam
>>>
>>>
>>> On Tue, Jan 11, 2011 at 8:07 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>>
>>>> It should work. Is there anything already listening on port 8022?
>>>>
>>>> $ netstat -a -n -p | grep 8022
>>>>
>>>> Are you also sure that they're not both loading the same config file?
>>>>
>>>> Regards,
>>>> -Steve
>>>>
>>>>
>>>>
>>>> On 11 January 2011 14:32, Sam <u2nsam at gmail.com> wrote:
>>>> > the scenario is i have 2 ips on 1 server for 2 FS instances;
>>>> >
>>>> > 192.168.2.1
>>>> > 192.168.2.2
>>>> >
>>>> > and the parameters i have set is:-
>>>> >
>>>> > for
>>>> > 192.168.2.1:-
>>>> >    <param name="listen-ip" value="127.0.0.1"/>
>>>> >    <param name="listen-port" value="8021"/>
>>>> > for
>>>> > 192.168.2.2:-
>>>> >    <param name="listen-ip" value="127.0.0.1"/>
>>>> >    <param name="listen-port" value="8022"/>
>>>> >
>>>> > Ideally it should work  but i am getting console for only 192.168.2.1
>>>> FS .
>>>> >
>>>> >
>>>> > Regards
>>>> > Sam
>>>> >
>>>> > On Tue, Jan 11, 2011 at 4:20 PM, Steven Ayre <steveayre at gmail.com>
>>>> wrote:
>>>> >>
>>>> >>    <param name="listen-ip" value="127.0.0.1"/>
>>>> >>    <param name="listen-port" value="8021"/>
>>>> >>
>>>> >> You can bind both to port 8021 on their individual IPs, or different
>>>> >> ports on the same IP.
>>>> >>
>>>> >> A listen IP of 0.0.0.0 will mean any IP.
>>>> >>
>>>> >> -Steve
>>>> >>
>>>> >> On 11 January 2011 10:44, Sam <u2nsam at gmail.com> wrote:
>>>> >> > A query,
>>>> >> >
>>>> >> > I have 2 FS running on one server on 2 different ips,
>>>> >> > so when i do fs_cli going to respective bins , i see console of
>>>> only the
>>>> >> > first server.
>>>> >> >
>>>> >> > Is there any way to get the console of both the FS on the same
>>>> server .
>>>> >> > I tried changing the port of event socket to 8022 but it donot
>>>> works.
>>>> >> > <param name="listen-port" value="8022"/>
>>>> >> >
>>>> >> > Is there some method to start the console of both the instances.
>>>> >> >
>>>> >> > Regds
>>>> >> > Sam
>>>> >> >
>>>> >> > _______________________________________________
>>>> >> > FreeSWITCH-users mailing list
>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> > http://www.freeswitch.org
>>>> >> >
>>>> >> >
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Tue Jan 11 19:56:03 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 11 Jan 2011 22:26:03 +0530
Subject: [Freeswitch-users] console
In-Reply-To: <AANLkTiknTsNhvbt-8D185sy_+LbmfJhw6HGR3+C_XgWZ@mail.gmail.com>
References: <AANLkTi=_ZNGF1BKsWh6CP8ExrmMYYqwzEx8C5O9OXCZ8@mail.gmail.com>
	<AANLkTi=0ZCQYgpR7T4LF11oPvGn75WeinYLPYF-BQXJn@mail.gmail.com>
	<AANLkTinZazjfEZcRHCuj0a9a9r5xg03r-NECCzqNzPeh@mail.gmail.com>
	<AANLkTi=qEgee_iH+WLATtXPSCP=2-zb0Wb8aiwFNE_7+@mail.gmail.com>
	<AANLkTimVk-AZ3dPnE-Kkj4tE8XS7WuLsQzczqTW1fBqf@mail.gmail.com>
	<AANLkTiknTsNhvbt-8D185sy_+LbmfJhw6HGR3+C_XgWZ@mail.gmail.com>
Message-ID: <AANLkTikdAic7TowHGT=2CStif-6zw0QoCyrE8X3KVXp7@mail.gmail.com>

the port and ip donot work for me,
is it that the fs_cli is not reading the config from 192.168.2.2 but it is
reading the config only of 192.168.2.1, though its in the different [FS_1 &
FS_2] path where i am executing.

Regds
Sam


On Tue, Jan 11, 2011 at 9:38 PM, Steven Ayre <steveayre at gmail.com> wrote:

> /usr/local/FS_1/bin/fs_cli -P 8021
> /usr/local/FS_2/bin/fs_cli -P 8022
>
> fs_cll doesn't read any config file. It's not part of the FS server at all,
> you can have it on a different machine that doesn't have FS installed. It
> entirely relies on the arguments to control where to connect to.
>
> -Steve
>
>
>
> On 11 January 2011 15:04, Sam <u2nsam at gmail.com> wrote:
>
>> Something more here ... i am getting the console for 192.168.2.1 every
>> time i do fs_cli on both instances .
>>
>> like
>> /usr/local/FS_1/bin/fs_cli
>> /usr/localFS_2/bin/fs_cli
>> i get the console for the 1st server only
>>
>> the 2 server are listing to 2 different ips .
>>
>> Regds
>> Sam
>>
>>
>> On Tue, Jan 11, 2011 at 8:07 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> It should work. Is there anything already listening on port 8022?
>>>
>>> $ netstat -a -n -p | grep 8022
>>>
>>> Are you also sure that they're not both loading the same config file?
>>>
>>> Regards,
>>> -Steve
>>>
>>>
>>>
>>> On 11 January 2011 14:32, Sam <u2nsam at gmail.com> wrote:
>>> > the scenario is i have 2 ips on 1 server for 2 FS instances;
>>> >
>>> > 192.168.2.1
>>> > 192.168.2.2
>>> >
>>> > and the parameters i have set is:-
>>> >
>>> > for
>>> > 192.168.2.1:-
>>> >    <param name="listen-ip" value="127.0.0.1"/>
>>> >    <param name="listen-port" value="8021"/>
>>> > for
>>> > 192.168.2.2:-
>>> >    <param name="listen-ip" value="127.0.0.1"/>
>>> >    <param name="listen-port" value="8022"/>
>>> >
>>> > Ideally it should work  but i am getting console for only 192.168.2.1
>>> FS .
>>> >
>>> >
>>> > Regards
>>> > Sam
>>> >
>>> > On Tue, Jan 11, 2011 at 4:20 PM, Steven Ayre <steveayre at gmail.com>
>>> wrote:
>>> >>
>>> >>    <param name="listen-ip" value="127.0.0.1"/>
>>> >>    <param name="listen-port" value="8021"/>
>>> >>
>>> >> You can bind both to port 8021 on their individual IPs, or different
>>> >> ports on the same IP.
>>> >>
>>> >> A listen IP of 0.0.0.0 will mean any IP.
>>> >>
>>> >> -Steve
>>> >>
>>> >> On 11 January 2011 10:44, Sam <u2nsam at gmail.com> wrote:
>>> >> > A query,
>>> >> >
>>> >> > I have 2 FS running on one server on 2 different ips,
>>> >> > so when i do fs_cli going to respective bins , i see console of only
>>> the
>>> >> > first server.
>>> >> >
>>> >> > Is there any way to get the console of both the FS on the same
>>> server .
>>> >> > I tried changing the port of event socket to 8022 but it donot
>>> works.
>>> >> > <param name="listen-port" value="8022"/>
>>> >> >
>>> >> > Is there some method to start the console of both the instances.
>>> >> >
>>> >> > Regds
>>> >> > Sam
>>> >> >
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> > http://www.freeswitch.org
>>> >> >
>>> >> >
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Tue Jan 11 19:59:41 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 11 Jan 2011 16:59:41 +0000
Subject: [Freeswitch-users] console
In-Reply-To: <AANLkTikdAic7TowHGT=2CStif-6zw0QoCyrE8X3KVXp7@mail.gmail.com>
References: <AANLkTi=_ZNGF1BKsWh6CP8ExrmMYYqwzEx8C5O9OXCZ8@mail.gmail.com>
	<AANLkTi=0ZCQYgpR7T4LF11oPvGn75WeinYLPYF-BQXJn@mail.gmail.com>
	<AANLkTinZazjfEZcRHCuj0a9a9r5xg03r-NECCzqNzPeh@mail.gmail.com>
	<AANLkTi=qEgee_iH+WLATtXPSCP=2-zb0Wb8aiwFNE_7+@mail.gmail.com>
	<AANLkTimVk-AZ3dPnE-Kkj4tE8XS7WuLsQzczqTW1fBqf@mail.gmail.com>
	<AANLkTiknTsNhvbt-8D185sy_+LbmfJhw6HGR3+C_XgWZ@mail.gmail.com>
	<AANLkTikdAic7TowHGT=2CStif-6zw0QoCyrE8X3KVXp7@mail.gmail.com>
Message-ID: <AANLkTikx+PYeN7c51N+t87eQ8FN6i7vJy7pswE2McLcy@mail.gmail.com>

It does not read any config file.




On 11 January 2011 16:56, Sam <u2nsam at gmail.com> wrote:

> the port and ip donot work for me,
> is it that the fs_cli is not reading the config from 192.168.2.2 but it is
> reading the config only of 192.168.2.1, though its in the different [FS_1 &
> FS_2] path where i am executing.
>
> Regds
> Sam
>
>
>
> On Tue, Jan 11, 2011 at 9:38 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> /usr/local/FS_1/bin/fs_cli -P 8021
>> /usr/local/FS_2/bin/fs_cli -P 8022
>>
>> fs_cll doesn't read any config file. It's not part of the FS server at
>> all, you can have it on a different machine that doesn't have FS installed.
>> It entirely relies on the arguments to control where to connect to.
>>
>> -Steve
>>
>>
>>
>> On 11 January 2011 15:04, Sam <u2nsam at gmail.com> wrote:
>>
>>> Something more here ... i am getting the console for 192.168.2.1 every
>>> time i do fs_cli on both instances .
>>>
>>> like
>>> /usr/local/FS_1/bin/fs_cli
>>> /usr/localFS_2/bin/fs_cli
>>> i get the console for the 1st server only
>>>
>>> the 2 server are listing to 2 different ips .
>>>
>>> Regds
>>> Sam
>>>
>>>
>>> On Tue, Jan 11, 2011 at 8:07 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>>
>>>> It should work. Is there anything already listening on port 8022?
>>>>
>>>> $ netstat -a -n -p | grep 8022
>>>>
>>>> Are you also sure that they're not both loading the same config file?
>>>>
>>>> Regards,
>>>> -Steve
>>>>
>>>>
>>>>
>>>> On 11 January 2011 14:32, Sam <u2nsam at gmail.com> wrote:
>>>> > the scenario is i have 2 ips on 1 server for 2 FS instances;
>>>> >
>>>> > 192.168.2.1
>>>> > 192.168.2.2
>>>> >
>>>> > and the parameters i have set is:-
>>>> >
>>>> > for
>>>> > 192.168.2.1:-
>>>> >    <param name="listen-ip" value="127.0.0.1"/>
>>>> >    <param name="listen-port" value="8021"/>
>>>> > for
>>>> > 192.168.2.2:-
>>>> >    <param name="listen-ip" value="127.0.0.1"/>
>>>> >    <param name="listen-port" value="8022"/>
>>>> >
>>>> > Ideally it should work  but i am getting console for only 192.168.2.1
>>>> FS .
>>>> >
>>>> >
>>>> > Regards
>>>> > Sam
>>>> >
>>>> > On Tue, Jan 11, 2011 at 4:20 PM, Steven Ayre <steveayre at gmail.com>
>>>> wrote:
>>>> >>
>>>> >>    <param name="listen-ip" value="127.0.0.1"/>
>>>> >>    <param name="listen-port" value="8021"/>
>>>> >>
>>>> >> You can bind both to port 8021 on their individual IPs, or different
>>>> >> ports on the same IP.
>>>> >>
>>>> >> A listen IP of 0.0.0.0 will mean any IP.
>>>> >>
>>>> >> -Steve
>>>> >>
>>>> >> On 11 January 2011 10:44, Sam <u2nsam at gmail.com> wrote:
>>>> >> > A query,
>>>> >> >
>>>> >> > I have 2 FS running on one server on 2 different ips,
>>>> >> > so when i do fs_cli going to respective bins , i see console of
>>>> only the
>>>> >> > first server.
>>>> >> >
>>>> >> > Is there any way to get the console of both the FS on the same
>>>> server .
>>>> >> > I tried changing the port of event socket to 8022 but it donot
>>>> works.
>>>> >> > <param name="listen-port" value="8022"/>
>>>> >> >
>>>> >> > Is there some method to start the console of both the instances.
>>>> >> >
>>>> >> > Regds
>>>> >> > Sam
>>>> >> >
>>>> >> > _______________________________________________
>>>> >> > FreeSWITCH-users mailing list
>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> > http://www.freeswitch.org
>>>> >> >
>>>> >> >
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Tue Jan 11 20:06:56 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Jan 2011 11:06:56 -0600
Subject: [Freeswitch-users] How to disable 5060-5080 with auto-nat at
	upnp?
In-Reply-To: <AANLkTi=UeJNn-2bimdA+PkJHL+deLkh9HaQuyL-DmC=4@mail.gmail.com>
References: <BLU0-SMTP97E64484D5F1A2D4F0FEFB8D0F0@phx.gbl>
	<AANLkTi=UeJNn-2bimdA+PkJHL+deLkh9HaQuyL-DmC=4@mail.gmail.com>
Message-ID: <DED97F2D-319E-4073-9141-A463BCA23291@freeswitch.org>

start FreeSWITCH with -nonat .. I can NOT and will NOT change in the way it works.  If you want to manage your mappings yourself do so... I'm not going to cripple the autonat to not AUTO nat.

/b

On Jan 11, 2011, at 9:09 AM, Rupa Schomaker wrote:

> Hmmm...  I didn't put anything in the nat_map code to allow some parts
> of the sofia profile to participate in the nat mapping while others do
> not.
> 
> Brian, do you have any ideas?
> 
> Alex, I'd suggest just blocking ports 5060-5080 on the firewall.  The
> port blocking *should* take precedence over the upnp maps.



From steveayre at gmail.com  Tue Jan 11 20:24:34 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 11 Jan 2011 17:24:34 +0000
Subject: [Freeswitch-users] console
In-Reply-To: <AANLkTikx+PYeN7c51N+t87eQ8FN6i7vJy7pswE2McLcy@mail.gmail.com>
References: <AANLkTi=_ZNGF1BKsWh6CP8ExrmMYYqwzEx8C5O9OXCZ8@mail.gmail.com>
	<AANLkTi=0ZCQYgpR7T4LF11oPvGn75WeinYLPYF-BQXJn@mail.gmail.com>
	<AANLkTinZazjfEZcRHCuj0a9a9r5xg03r-NECCzqNzPeh@mail.gmail.com>
	<AANLkTi=qEgee_iH+WLATtXPSCP=2-zb0Wb8aiwFNE_7+@mail.gmail.com>
	<AANLkTimVk-AZ3dPnE-Kkj4tE8XS7WuLsQzczqTW1fBqf@mail.gmail.com>
	<AANLkTiknTsNhvbt-8D185sy_+LbmfJhw6HGR3+C_XgWZ@mail.gmail.com>
	<AANLkTikdAic7TowHGT=2CStif-6zw0QoCyrE8X3KVXp7@mail.gmail.com>
	<AANLkTikx+PYeN7c51N+t87eQ8FN6i7vJy7pswE2McLcy@mail.gmail.com>
Message-ID: <AANLkTi=6Hwx2Y67Htd+OA2PefQAuVyXUxOGWgEp+F9Ce@mail.gmail.com>

Actually, correction it does - just not the event_socket.conf.xml one. It'll
read .fs_cli_conf in your home directory if it exists, but that isn't
created by default - you create it yourself if you want to use it (it's
optional). The command line arguments -H and -P -will- override the config
file though.

Are you using a capital P? -p is password while -P is port. If there's no
password on the event socket you'd get no error from using a small p by
accident.

-Steve





On 11 January 2011 16:59, Steven Ayre <steveayre at gmail.com> wrote:

> It does not read any config file.
>
>
>
>
>
> On 11 January 2011 16:56, Sam <u2nsam at gmail.com> wrote:
>
>> the port and ip donot work for me,
>> is it that the fs_cli is not reading the config from 192.168.2.2 but it is
>> reading the config only of 192.168.2.1, though its in the different [FS_1 &
>> FS_2] path where i am executing.
>>
>> Regds
>> Sam
>>
>>
>>
>> On Tue, Jan 11, 2011 at 9:38 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> /usr/local/FS_1/bin/fs_cli -P 8021
>>> /usr/local/FS_2/bin/fs_cli -P 8022
>>>
>>> fs_cll doesn't read any config file. It's not part of the FS server at
>>> all, you can have it on a different machine that doesn't have FS installed.
>>> It entirely relies on the arguments to control where to connect to.
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 11 January 2011 15:04, Sam <u2nsam at gmail.com> wrote:
>>>
>>>> Something more here ... i am getting the console for 192.168.2.1 every
>>>> time i do fs_cli on both instances .
>>>>
>>>> like
>>>> /usr/local/FS_1/bin/fs_cli
>>>> /usr/localFS_2/bin/fs_cli
>>>> i get the console for the 1st server only
>>>>
>>>> the 2 server are listing to 2 different ips .
>>>>
>>>> Regds
>>>> Sam
>>>>
>>>>
>>>> On Tue, Jan 11, 2011 at 8:07 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>>>
>>>>> It should work. Is there anything already listening on port 8022?
>>>>>
>>>>> $ netstat -a -n -p | grep 8022
>>>>>
>>>>> Are you also sure that they're not both loading the same config file?
>>>>>
>>>>> Regards,
>>>>> -Steve
>>>>>
>>>>>
>>>>>
>>>>> On 11 January 2011 14:32, Sam <u2nsam at gmail.com> wrote:
>>>>> > the scenario is i have 2 ips on 1 server for 2 FS instances;
>>>>> >
>>>>> > 192.168.2.1
>>>>> > 192.168.2.2
>>>>> >
>>>>> > and the parameters i have set is:-
>>>>> >
>>>>> > for
>>>>> > 192.168.2.1:-
>>>>> >    <param name="listen-ip" value="127.0.0.1"/>
>>>>> >    <param name="listen-port" value="8021"/>
>>>>> > for
>>>>> > 192.168.2.2:-
>>>>> >    <param name="listen-ip" value="127.0.0.1"/>
>>>>> >    <param name="listen-port" value="8022"/>
>>>>> >
>>>>> > Ideally it should work  but i am getting console for only 192.168.2.1
>>>>> FS .
>>>>> >
>>>>> >
>>>>> > Regards
>>>>> > Sam
>>>>> >
>>>>> > On Tue, Jan 11, 2011 at 4:20 PM, Steven Ayre <steveayre at gmail.com>
>>>>> wrote:
>>>>> >>
>>>>> >>    <param name="listen-ip" value="127.0.0.1"/>
>>>>> >>    <param name="listen-port" value="8021"/>
>>>>> >>
>>>>> >> You can bind both to port 8021 on their individual IPs, or different
>>>>> >> ports on the same IP.
>>>>> >>
>>>>> >> A listen IP of 0.0.0.0 will mean any IP.
>>>>> >>
>>>>> >> -Steve
>>>>> >>
>>>>> >> On 11 January 2011 10:44, Sam <u2nsam at gmail.com> wrote:
>>>>> >> > A query,
>>>>> >> >
>>>>> >> > I have 2 FS running on one server on 2 different ips,
>>>>> >> > so when i do fs_cli going to respective bins , i see console of
>>>>> only the
>>>>> >> > first server.
>>>>> >> >
>>>>> >> > Is there any way to get the console of both the FS on the same
>>>>> server .
>>>>> >> > I tried changing the port of event socket to 8022 but it donot
>>>>> works.
>>>>> >> > <param name="listen-port" value="8022"/>
>>>>> >> >
>>>>> >> > Is there some method to start the console of both the instances.
>>>>> >> >
>>>>> >> > Regds
>>>>> >> > Sam
>>>>> >> >
>>>>> >> > _______________________________________________
>>>>> >> > FreeSWITCH-users mailing list
>>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> > http://www.freeswitch.org
>>>>> >> >
>>>>> >> >
>>>>> >>
>>>>> >> _______________________________________________
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> http://www.freeswitch.org
>>>>> >
>>>>> >
>>>>> > _______________________________________________
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>> >
>>>>> >
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From hadyn_whx at hotmail.com  Tue Jan 11 20:27:15 2011
From: hadyn_whx at hotmail.com (Alex Wang)
Date: Tue, 11 Jan 2011 12:27:15 -0500
Subject: [Freeswitch-users] How to disable 5060-5080 with auto-nat at
	upnp?
In-Reply-To: <AANLkTi=JrAocgCY0uFVAD149R7pz7m4aqgkdkmj1KYHf@mail.gmail.com>
References: <BLU0-SMTP7A86F22BAE5746316A3478D0F0@phx.gbl>
	<AANLkTi=JrAocgCY0uFVAD149R7pz7m4aqgkdkmj1KYHf@mail.gmail.com>
Message-ID: <BLU0-SMTP573C29CF8A404620FBB6508D0F0@phx.gbl>

I have trixbox running on my home LAN on the same ESX server, which is
my old pbx system. And I didn't map any port on the router and I use
google voice, freephoneline, voipcheap, etc and they all works fine.

I do need to setup the external ip, my local lan settings on *, but
that's it, nothing relate to the router.

I keep digging. If someone have the answer, pls let me know.

Alex

On Tue, 11 Jan 2011 10:34:33 -0600
Rupa Schomaker <rupa at rupa.com> wrote:

> Well, there is no "the RTP port" -- there is a range which is quite
> large by default.
> 
> I mean setup a rule to block the port, not point it somewhere else.
> Though I guess you could point it to a black hole (IP that is never
> assigned).
> 
> Why/how * works, I dunno.  Maybe someone else can answer that
> question.  I don't see how to handle RTP data properly through a
> firewall without support for portmapping initiated by FS (using upnp
> or nat-pmp) or having a SIP ALG running on the firewall.
> 
> On Tue, Jan 11, 2011 at 10:26 AM, Alex Wang <hadyn_whx at hotmail.com> wrote:
> > Hi Rupa
> >
> > Do you mean create a port map on the router to map to other ip instead
> > of FS?
> >
> > update: That seems working. But do you think I can just turn off the
> > auto-nat and map the rtp port out to get it working? Which way is the
> > best? In the asterisk, I don't even need turn on upnp and don't need to
> > map the port and it works fine behind the router, even those ATAs,
> > normally you don't need to map any port on the router and they just
> > register from isp and works fine with router(non UPNP & UPNP), why FS
> > need do those extra step? I try to find the answer in the WiKi but cant
> > find the explaination for that.
> >
> > Thanks
> >
> > Alex
> >
> >
> > On Tue, 11 Jan 2011 09:09:46 -0600
> > Rupa Schomaker <rupa at rupa.com> wrote:
> >
> >> Hmmm... ?I didn't put anything in the nat_map code to allow some parts
> >> of the sofia profile to participate in the nat mapping while others do
> >> not.
> >>
> >> Brian, do you have any ideas?
> >>
> >> Alex, I'd suggest just blocking ports 5060-5080 on the firewall. ?The
> >> port blocking *should* take precedence over the upnp maps.
> >>
> >> On Mon, Jan 10, 2011 at 10:17 PM, Alex Wang <hadyn_whx at hotmail.com> wrote:
> >> > Hi All
> >> >
> >> > Just don't want to open these port to the public, all my extensions are
> >> > in my LAN. If I disable upnp, the outside sip register is not working, I
> >> > mean no sound.
> >> > How to disable the auto-nat publish 5060-5080 on the internet?
> >> >
> >> >
> >> > Thanks
> >> >
> >> > Alex
> >> >
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >>
> >>
> >>
> >> --
> >> -Rupa
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> 
> -- 
> -Rupa
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From infos at madovsky.org  Tue Jan 11 20:43:45 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 11 Jan 2011 12:43:45 -0500
Subject: [Freeswitch-users] cepstral problem
References: <6DCDF4F8008043CDA4BD7F00005611F4@e1705><693A2C5B2175445CA6FC02821FACFE80@e1705><AANLkTimkDFymovrMg7mX3ZDJT+ubQH=7mz0QZga1Acja@mail.gmail.com>
	<AANLkTinCqzaXDybb62sD4jq=YaocjPnF6OVvx4oG4UAe@mail.gmail.com>
Message-ID: <A8C2094BAD2F4F32A71FD2579B7CB0C0@e1705>

this is really weird.
when I start FS, the console log shows

2011-01-11 12:42:50.029042 [CRIT] switch_loadable_module.c:928 Error Loading 
module /usr/local/freeswitch/mod/mod_cepstral.so
**libswift.so.5: cannot open shared object file: No such file or directory**

but mod_cepstral.so IS in mod folder....


----- Original Message ----- 
From: "Steven Ayre" <steveayre at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, January 11, 2011 4:32 AM
Subject: Re: [Freeswitch-users] cepstral problem


> If in doubt, try 'load mod_cepstral' from the cli
>
> -Steve
>
>
> On 11 January 2011 09:32, Steven Ayre <steveayre at gmail.com> wrote:
>> Check your log from startup - did mod_cepstral fail to load?
>>
>> It could be that you didn't compile it, it can't find a shared library
>> or there's a problem in the config file. It'll show you in the log.
>>
>> 'show modules mod_cepstral' will show you whether it's loaded from cli.
>>
>> -Steve
>>
>> On 11 January 2011 08:33, Madovsky <infos at madovsky.org> wrote:
>>> I'm sure to have set mod_cepstral in modules.conf.xml
>>> but in the log I can see
>>>
>>> 2011-01-11 03:32:14.999468 [ERR] switch_core_speech.c:61 Invalid speech
>>> module [cepstral]!
>>> 2011-01-11 03:32:14.999468 [ERR] switch_ivr_play_say.c:2266 Invalid TTS
>>> module!
>>> I use Cepstral 5.1
>>>
>>> Thanks
>>>
>>> ----- Original Message -----
>>> From: Madovsky
>>> To: freeswitch-users at lists.freeswitch.org
>>> Sent: Tuesday, January 11, 2011 3:12 AM
>>> Subject: [Freeswitch-users] cepstral problem
>>> I have one sentence like this in my dialplan
>>> <action application="speak" data="cepstral|david|want pizza"/>
>>>
>>> so when I start FS from a console directly in foreground the voice 
>>> works.
>>> if I start FS from the init script in background, the voice becomes 
>>> silence
>>> and nothing is running after this in the dialplan.
>>> if I want to shutdown manually FS, it makes a segmentation fault with 
>>> core
>>> dump.
>>>
>>> Any idea ?
>>>
>>> Thanks
>>>
>>> ________________________________
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From marvin.n.dillon at gmail.com  Tue Jan 11 05:30:50 2011
From: marvin.n.dillon at gmail.com (Marvin Dillon)
Date: Mon, 10 Jan 2011 21:30:50 -0500
Subject: [Freeswitch-users] Unable to successfully configure icall
 gateway and route inbound DID
In-Reply-To: <67907271-E9E5-4608-9D3F-41D2D4102BF1@freeswitch.org>
References: <AANLkTimahoPc=gxMD8NR+=y9jHKSfdz0Bbc_BUuLSLkp@mail.gmail.com>
	<67907271-E9E5-4608-9D3F-41D2D4102BF1@freeswitch.org>
Message-ID: <AANLkTik=ie4qQiyP4iaB-G_-ekF=mKh14-VjE+UF=y4B@mail.gmail.com>

Hey Brian,

I checked with icall and i am now able to register to the outbound server. I
am still having a problem when i call my DID, it get an error "Rejected by
acl "domains". Can you say what configuration is missing here?

sofia status
                     Name
Type                                       Data      State
=================================================================================================
                 internal       profile
sip:mod_sofia at 192.168.1.100:5060      RUNNING (0)
            internal-ipv6       profile
sip:mod_sofia@[::1]:5060
RUNNING (0)
                 external       profile
sip:mod_sofia at 192.168.1.100:5080      RUNNING (0)
    external::example.com       gateway
sip:joeuser at example.com <sip%3Ajoeuser at example.com>      NOREG
external::icall_international   gateway
sip:XXX at gw01-car.dal.us.icall.net<sip%3AXXX at gw01-car.dal.us.icall.net>
REGED
 external::icall_outbound       gateway
sip:XXX at sbc01-car.dal.us.icall.net<sip%3AXXX at sbc01-car.dal.us.icall.net>
REGED
  external::icall_inbound       gateway
sip:XXX at 72.249.14.242<sip%3AXXX at 72.249.14.242>
REGED
      external::icall.com       gateway
sip:XXX at 72.249.14.242<sip%3AXXX at 72.249.14.242>
REGED
           208.124.220.35         alias
internal      ALIASED
=================================================================================================
3 profiles 1 alias
freeswitch at debian> 2011-01-10 21:21:51.819275 [WARNING] sofia.c:6075 IP
72.249.14.242 Rejected by acl "domains"
2011-01-10 21:21:52.047119 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
by acl "domains"
2011-01-10 21:21:52.335147 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
by acl "domains"
2011-01-10 21:21:52.552260 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
by acl "domains"
2011-01-10 21:21:53.008400 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
by acl "domains"
2011-01-10 21:21:53.239339 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
by acl "domains"
2011-01-10 21:21:53.553315 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
by acl "domains"
2011-01-10 21:21:53.766360 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
by acl "domains"
2011-01-10 21:21:55.245075 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
by acl "domains"
2011-01-10 21:21:55.485528 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
by acl "domains"
2011-01-10 21:21:55.806639 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
by acl "domains"
2011-01-10 21:21:56.060993 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
by acl "domains"


On Mon, Jan 10, 2011 at 10:14 AM, Brian West <brian at freeswitch.org> wrote:

> Um I don't think you register to the outbound server.
>
>
> /b
>
>  On Jan 10, 2011, at 12:29 AM, Marvin Dillon wrote:
>
>  Hello Team,
>
> I am a rookie running Freeswitch 1.0.6 on Debian Lenny and need some urgent
> help. I have been facing a challenge getting my icall gateways configured
> and being able to route my inbound DID back to my Freeswitch platform. My
> sofia status output is this right now:
>
>  sofia status
>                      Name
> Type                                       Data      State
>
> =================================================================================================
>                  internal       profile
> sip:mod_sofia at 192.168.1.100:5060      RUNNING (0)
>                  external       profile
> sip:mod_sofia at 192.168.1.100:5080      RUNNING (0)
>     external::example.com       gateway
> sip:joeuser at example.com <sip%3Ajoeuser at example.com>      NOREG
> external::icall_international   gateway
> sip:cust_mdillon at gw01-car.dal.us.icall.net<sip%3Acust_mdillon at gw01-car.dal.us.icall.net>
> REGED
>  external::icall_outbound       gateway
> sip:cust_mdillon at sbc01-car.dal.us.icall.net<sip%3Acust_mdillon at sbc01-car.dal.us.icall.net>
> FAIL_WAIT
>   external::icall_inbound       gateway
> sip:cust_mdillon at 72.249.14.242 <sip%3Acust_mdillon at 72.249.14.242>
> REGED
>       external::icall.com       gateway
> sip:cust_mdillon at 72.249.14.242 <sip%3Acust_mdillon at 72.249.14.242>
> REGED
>            208.124.220.35         alias
> internal      ALIASED
>             internal-ipv6       profile
> sip:mod_sofia@[::1]:5060      RUNNING (0)
>
> =================================================================================================
> 3 profiles 1 alias
> but I have no clue why I am getting a busy tone whenever I call my inbound
> DID as the sofia output indicates my inbound gateway is registered. Can
> someone please help me with this.
>
> Thanks,
> MD
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From hwnorman at hotmail.com  Tue Jan 11 13:02:12 2011
From: hwnorman at hotmail.com (Norman Lam)
Date: Tue, 11 Jan 2011 18:02:12 +0800
Subject: [Freeswitch-users] Iksemel msvs compiling
In-Reply-To: <1294673236806-5907287.post@n2.nabble.com>
References: <SNT109-DS13313A1FA906527438E8CC1090@phx.gbl>	<SNT109-DS1155D401A0B9742CF63E13C10B0@phx.gbl>	<1294375438447-5898181.post@n2.nabble.com>	<SNT109-DS11CB0ECE5BD0F9C0384E03C10C0@phx.gbl>	<1294535404901-5903534.post@n2.nabble.com>	<SNT109-DS14B90C48B33126784F294AC10E0@phx.gbl>
	<1294673236806-5907287.post@n2.nabble.com>
Message-ID: <SNT109-DS1726F0FA2FA5318C0CA23BC10F0@phx.gbl>

Hi Jeff 

I am still having the same error, after putting
;..\..\pthreads-w32-2-7-0-release;
at the end of the list

Norman 

28>------ Build started: Project: iksemel, Configuration: Debug Win32 ------
28>Compiling...
26>v3_conf.c
28>dom.c
28>..\..\iksemel\src\dom.c(152) : error C2065: 'ENOENT' : undeclared
identifier
28>filter.c
28>..\..\iksemel\src\filter.c(59) : error C2059: syntax error : 'type'
28>..\..\iksemel\src\filter.c(64) : error C2059: syntax error : 'type'
28>..\..\iksemel\src\filter.c(67) : error C2059: syntax error : 'type'
28>..\..\iksemel\src\filter.c(70) : error C2059: syntax error : 'type'
28>..\..\iksemel\src\filter.c(73) : error C2059: syntax error : 'type'
28>..\..\iksemel\src\filter.c(76) : error C2059: syntax error : 'type'
28>..\..\iksemel\src\filter.c(79) : error C2059: syntax error : 'type'
28>iks.c
28>ikstack.c
28>io-posix.c
26>v3_bitst.c
28>jabber.c
28>md5.c
26>v3_bcons.c
28>sax.c
28>sha.c
28>stream.c
28>..\..\iksemel\src\stream.c(19) : fatal error C1083: Cannot open include
file: 'gnutls/gnutls.h': No such file or directory
28>utility.c
28>base64.c
28>Generating Code...
28>Build log was saved at
"file://c:\FS_GIT2\libs\win32\iksemel\Debug\BuildLog.htm"
28>iksemel - 9 error(s), 0 warning(s)
:
:
125>------ Build started: Project: mod_dingaling, Configuration: Debug Win32
------
125>Compiling...
125>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler
125>mod_dingaling.c
125>Linking...
125>LINK : fatal error LNK1181: cannot open input file
'..\..\..\..\libs\win32\iksemel\debug\iksemel.lib'
125>Build log was saved at
"file://c:\FS_GIT2\src\mod\endpoints\mod_dingaling\Win32\Debug\BuildLog.htm"
125>mod_dingaling - 1 error(s), 1 warning(s)
:
:
====== Build: 135 succeeded, 2 failed, 0 up-to-date, 3 skipped ==========
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff
Lenk
Sent: Monday, January 10, 2011 11:27 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Iksemel msvs compiling


make sure you put the include path for ;..\..\pthreads-w32-2-7-0-release;
at the end of the list. There seems to be some include file conflicts but
this seems to take care of it.

If you are succesfull with all of this please take the time to update the
wiki with more detail.
-- 
View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/Iksemel-msvs-compiling-tp58912
63p5907287.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From efimserg at gmail.com  Tue Jan 11 14:44:19 2011
From: efimserg at gmail.com (Sergii Iefimov)
Date: Tue, 11 Jan 2011 13:44:19 +0200
Subject: [Freeswitch-users] QoS in FreeSwitch
Message-ID: <AANLkTimCm7wwLBePiTtTDY7DLBshW_-HkRwhiStQ9TsX@mail.gmail.com>

*What kind of the QoS (Quality of Service)**do you have in FreeSwitch
(DiffServ, IntServ or other**)?*

*

Does FreeSwitch offer the ability to mark the voice data with the
proper tags so that our switch can prioritize the data through our
network, or if it can't what hard ware device would you suggest to do
this?

Thanks!

*

 Best Regards,

*Sergii Iefimov*

Mobile phone:                +380 (68) 361 4627
E-mail:                          *efimserg at gmail.com*
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From efimserg at gmail.com  Tue Jan 11 16:33:17 2011
From: efimserg at gmail.com (efim_serg)
Date: Tue, 11 Jan 2011 05:33:17 -0800 (PST)
Subject: [Freeswitch-users] DCS in IP telephony communication standards
Message-ID: <1294752797821-5910793.post@n2.nabble.com>


Could somebody tell me what does mean 'DCS' in IP telephony communication
standards?
I have looked at the internet and found that DCS is a distributed control
system, but I think it's not relevant information when we talk about IP
telephony communication standards.
I also need find interrelation between DCS and FreeSwitch.
Thanks.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/DCS-in-IP-telephony-communication-standards-tp5910793p5910793.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From infos at madovsky.org  Tue Jan 11 21:04:19 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 11 Jan 2011 13:04:19 -0500
Subject: [Freeswitch-users] cepstral problem SOLVED
References: <6DCDF4F8008043CDA4BD7F00005611F4@e1705><693A2C5B2175445CA6FC02821FACFE80@e1705><AANLkTimkDFymovrMg7mX3ZDJT+ubQH=7mz0QZga1Acja@mail.gmail.com>
	<AANLkTinCqzaXDybb62sD4jq=YaocjPnF6OVvx4oG4UAe@mail.gmail.com>
Message-ID: <53D3AAAC19144CC698B4023A9E6B904A@e1705>

I put /opt/swift/lib (there was /opt/swift) in ld.so.conf and run again 
ldconfig

thanks

----- Original Message ----- 
From: "Steven Ayre" <steveayre at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, January 11, 2011 4:32 AM
Subject: Re: [Freeswitch-users] cepstral problem


> If in doubt, try 'load mod_cepstral' from the cli
>
> -Steve
>
>
> On 11 January 2011 09:32, Steven Ayre <steveayre at gmail.com> wrote:
>> Check your log from startup - did mod_cepstral fail to load?
>>
>> It could be that you didn't compile it, it can't find a shared library
>> or there's a problem in the config file. It'll show you in the log.
>>
>> 'show modules mod_cepstral' will show you whether it's loaded from cli.
>>
>> -Steve
>>
>> On 11 January 2011 08:33, Madovsky <infos at madovsky.org> wrote:
>>> I'm sure to have set mod_cepstral in modules.conf.xml
>>> but in the log I can see
>>>
>>> 2011-01-11 03:32:14.999468 [ERR] switch_core_speech.c:61 Invalid speech
>>> module [cepstral]!
>>> 2011-01-11 03:32:14.999468 [ERR] switch_ivr_play_say.c:2266 Invalid TTS
>>> module!
>>> I use Cepstral 5.1
>>>
>>> Thanks
>>>
>>> ----- Original Message -----
>>> From: Madovsky
>>> To: freeswitch-users at lists.freeswitch.org
>>> Sent: Tuesday, January 11, 2011 3:12 AM
>>> Subject: [Freeswitch-users] cepstral problem
>>> I have one sentence like this in my dialplan
>>> <action application="speak" data="cepstral|david|want pizza"/>
>>>
>>> so when I start FS from a console directly in foreground the voice 
>>> works.
>>> if I start FS from the init script in background, the voice becomes 
>>> silence
>>> and nothing is running after this in the dialplan.
>>> if I want to shutdown manually FS, it makes a segmentation fault with 
>>> core
>>> dump.
>>>
>>> Any idea ?
>>>
>>> Thanks
>>>
>>> ________________________________
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From norm at voicenetwork.ca  Tue Jan 11 21:04:50 2011
From: norm at voicenetwork.ca (Norman Tomlins)
Date: Tue, 11 Jan 2011 13:04:50 -0500
Subject: [Freeswitch-users] QoS in FreeSwitch
In-Reply-To: <AANLkTimCm7wwLBePiTtTDY7DLBshW_-HkRwhiStQ9TsX@mail.gmail.com>
References: <AANLkTimCm7wwLBePiTtTDY7DLBshW_-HkRwhiStQ9TsX@mail.gmail.com>
Message-ID: <AANLkTi=zCZq6eXPJwM7d4waGKrcN3yzBf0NUL+WzUiTO@mail.gmail.com>

Sergii,

You could use IP tables to set the QOS on the RTP packets.

iptables -t mangle -A OUTPUT -p tcp --sport 5060 -j DSCP
--set-dscp-class cs3 # mark SIP TCP packets with CS3
iptables -t mangle -A OUTPUT -p tcp --sport 5061 -j DSCP
--set-dscp-class cs3 # mark SIP TLS packets with CS3
iptables -t mangle -A OUTPUT -p udp -m udp --sport 16384:32767 -j DSCP
--set-dscp-class ef # mark RTP packets with EF


Norman Tomlins
Voice Network Inc.


On Tue, Jan 11, 2011 at 6:44 AM, Sergii Iefimov <efimserg at gmail.com> wrote:
> What kind of the QoS (Quality of Service)do you have in FreeSwitch
> (DiffServ, IntServ or other)?
>
> Does FreeSwitch offer the ability to mark the voice data with the proper
> tags so that our switch can prioritize the data through our network, or if
> it can't what hard ware device would you suggest to do this?
>
> Thanks!
>
> Best Regards,
>
> Sergii Iefimov
>
> Mobile phone:???? ?????????? +380 (68) 361 4627
> E-mail:???????????????? ?????????efimserg at gmail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From hadyn_whx at hotmail.com  Tue Jan 11 21:22:11 2011
From: hadyn_whx at hotmail.com (Alex Wang)
Date: Tue, 11 Jan 2011 13:22:11 -0500
Subject: [Freeswitch-users] How to disable 5060-5080 with auto-nat at
	upnp?
In-Reply-To: <DED97F2D-319E-4073-9141-A463BCA23291@freeswitch.org>
References: <AANLkTi=UeJNn-2bimdA+PkJHL+deLkh9HaQuyL-DmC=4@mail.gmail.com>
	<DED97F2D-319E-4073-9141-A463BCA23291@freeswitch.org>
Message-ID: <BLU0-SMTP13FBD515FEBDD61F7E04F08D0F0@phx.gbl>

Hi Brian

I think you misunderstood me on this, what I want to say is not every
PBX will face to the public, even in the commercial environment, the PBX
may not need open 5060-5080 to the public, but the sip session will need
upnp to automatically map the port. My question is how to handle this
kind of situation. Just -nonat and map all the other port?

Thanks

Alex
On Tue, 11 Jan 2011 11:06:56 -0600
Brian West <brian at freeswitch.org> wrote:

> start FreeSWITCH with -nonat .. I can NOT and will NOT change in the way it works.  If you want to manage your mappings yourself do so... I'm not going to cripple the autonat to not AUTO nat.
> 
> /b
> 
> On Jan 11, 2011, at 9:09 AM, Rupa Schomaker wrote:
> 
> > Hmmm...  I didn't put anything in the nat_map code to allow some parts
> > of the sofia profile to participate in the nat mapping while others do
> > not.
> > 
> > Brian, do you have any ideas?
> > 
> > Alex, I'd suggest just blocking ports 5060-5080 on the firewall.  The
> > port blocking *should* take precedence over the upnp maps.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From marvin.n.dillon at gmail.com  Tue Jan 11 19:38:44 2011
From: marvin.n.dillon at gmail.com (Marvin Dillon)
Date: Tue, 11 Jan 2011 11:38:44 -0500
Subject: [Freeswitch-users] Fwd: Unable to successfully configure icall
	gateway and route inbound DID
In-Reply-To: <AANLkTik=ie4qQiyP4iaB-G_-ekF=mKh14-VjE+UF=y4B@mail.gmail.com>
References: <AANLkTimahoPc=gxMD8NR+=y9jHKSfdz0Bbc_BUuLSLkp@mail.gmail.com>
	<67907271-E9E5-4608-9D3F-41D2D4102BF1@freeswitch.org>
	<AANLkTik=ie4qQiyP4iaB-G_-ekF=mKh14-VjE+UF=y4B@mail.gmail.com>
Message-ID: <AANLkTi=Hs5NMt1xv2OYx=rELyRULtkTWoabeUmgLGfPt@mail.gmail.com>

Team,

Can anyone help me with the below issue?

Thanks,
MD

---------- Forwarded message ----------
From: Marvin Dillon <marvin.n.dillon at gmail.com>
Date: Mon, 10 Jan 2011 21:30:50 -0500
Subject: Re: [Freeswitch-users] Unable to successfully configure icall
gateway and route inbound DID
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>

Hey Brian,

I checked with icall and i am now able to register to the outbound server. I
am still having a problem when i call my DID, it get an error "Rejected by
acl "domains". Can you say what configuration is missing here?

sofia status
                     Name
Type                                       Data      State
=================================================================================================
                 internal       profile
sip:mod_sofia at 192.168.1.100:5060      RUNNING (0)
            internal-ipv6       profile
sip:mod_sofia@[::1]:5060
RUNNING (0)
                 external       profile
sip:mod_sofia at 192.168.1.100:5080      RUNNING (0)
    external::example.com       gateway
sip:joeuser at example.com <sip%3Ajoeuser at example.com>      NOREG
external::icall_international   gateway
sip:XXX at gw01-car.dal.us.icall.net<sip%3AXXX at gw01-car.dal.us.icall.net>
REGED
 external::icall_outbound       gateway
sip:XXX at sbc01-car.dal.us.icall.net<sip%3AXXX at sbc01-car.dal.us.icall.net>
REGED
  external::icall_inbound       gateway
sip:XXX at 72.249.14.242<sip%3AXXX at 72.249.14.242>
REGED
      external::icall.com       gateway
sip:XXX at 72.249.14.242<sip%3AXXX at 72.249.14.242>
REGED
           208.124.220.35         alias
internal      ALIASED
=================================================================================================
3 profiles 1 alias
freeswitch at debian> 2011-01-10 21:21:51.819275 [WARNING] sofia.c:6075 IP
72.249.14.242 Rejected by acl "domains"
2011-01-10 21:21:52.047119 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
by acl "domains"
2011-01-10 21:21:52.335147 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
by acl "domains"
2011-01-10 21:21:52.552260 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
by acl "domains"
2011-01-10 21:21:53.008400 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
by acl "domains"
2011-01-10 21:21:53.239339 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
by acl "domains"
2011-01-10 21:21:53.553315 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
by acl "domains"
2011-01-10 21:21:53.766360 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
by acl "domains"
2011-01-10 21:21:55.245075 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
by acl "domains"
2011-01-10 21:21:55.485528 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
by acl "domains"
2011-01-10 21:21:55.806639 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
by acl "domains"
2011-01-10 21:21:56.060993 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
by acl "domains"


On Mon, Jan 10, 2011 at 10:14 AM, Brian West <brian at freeswitch.org> wrote:

> Um I don't think you register to the outbound server.
>
>
> /b
>
>  On Jan 10, 2011, at 12:29 AM, Marvin Dillon wrote:
>
>  Hello Team,
>
> I am a rookie running Freeswitch 1.0.6 on Debian Lenny and need some urgent
> help. I have been facing a challenge getting my icall gateways configured
> and being able to route my inbound DID back to my Freeswitch platform. My
> sofia status output is this right now:
>
>  sofia status
>                      Name
> Type                                       Data      State
>
> =================================================================================================
>                  internal       profile
> sip:mod_sofia at 192.168.1.100:5060      RUNNING (0)
>                  external       profile
> sip:mod_sofia at 192.168.1.100:5080      RUNNING (0)
>     external::example.com       gateway
> sip:joeuser at example.com <sip%3Ajoeuser at example.com>      NOREG
> external::icall_international   gateway
> sip:cust_mdillon at gw01-car.dal.us.icall.net<sip%3Acust_mdillon at gw01-car.dal.us.icall.net>
> REGED
>  external::icall_outbound       gateway
> sip:cust_mdillon at sbc01-car.dal.us.icall.net<sip%3Acust_mdillon at sbc01-car.dal.us.icall.net>
> FAIL_WAIT
>   external::icall_inbound       gateway
> sip:cust_mdillon at 72.249.14.242 <sip%3Acust_mdillon at 72.249.14.242>
> REGED
>       external::icall.com       gateway
> sip:cust_mdillon at 72.249.14.242 <sip%3Acust_mdillon at 72.249.14.242>
> REGED
>            208.124.220.35         alias
> internal      ALIASED
>             internal-ipv6       profile
> sip:mod_sofia@[::1]:5060      RUNNING (0)
>
> =================================================================================================
> 3 profiles 1 alias
> but I have no clue why I am getting a busy tone whenever I call my inbound
> DID as the sofia output indicates my inbound gateway is registered. Can
> someone please help me with this.
>
> Thanks,
> MD
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

-- 
Sent from my mobile device


From siobhan.pluggedin at gmail.com  Tue Jan 11 20:39:39 2011
From: siobhan.pluggedin at gmail.com (Siobhan Hamilton)
Date: Tue, 11 Jan 2011 12:39:39 -0500
Subject: [Freeswitch-users] Looking for someone to build Android SIP Client
	To Work W/Freeswitch
Message-ID: <AANLkTi=NkpL8+KiBP6pJ4ubxwvnAkFPVSeWF3UXpf30M@mail.gmail.com>

Hi there,

Hope I'm not totally off-base with this post but I thought this group might
be a good resource to ask.... My company is looking for a SIP/VOIP expert
who can build an integrated SIP client for us on Android to interact with a
Freeswitch server we are building, if anyone is out there who fits the bill
or knows anyone....?

Requirements:
- No GPL'ed SIP Stacks - we are building a commercial app that we wish to
keep closed source.
- Not a standalone, typical SIP client; will use a non-traditional UI to
interact with the SIP server.
- Knowledge of optimization of Android performance and ability to evaluate
and implement best codecs for our purposes.

App will be interfacing with OpenSIPs registration server with a custom
presence module, so must be familiar with presence.  Users will be heavily
interacting in conference mode.

Please email resume/experience to siobhan.pluggedin at gmail.com...

Thank you,

Siobhan Hamilton
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From msc at freeswitch.org  Tue Jan 11 21:23:51 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Jan 2011 18:23:51 +0000
Subject: [Freeswitch-users] bind_digit_action
In-Reply-To: <COL122-W297FC85979AA72D5DBC40B50F0@phx.gbl>
References: <COL122-W20D6C00E627218901B0691B50E0@phx.gbl>
	<AANLkTim-+iLNZAhNaWU0npAfL2Sn++8mCyqwANXXbVax@mail.gmail.com>
	<COL122-W297FC85979AA72D5DBC40B50F0@phx.gbl>
Message-ID: <AANLkTi=sFderd1MSp6aniXvodyZ1nCbE=1ejGXUuNMXb@mail.gmail.com>

If you hang up leg B then what should happen to leg A? Can you not just
transfer the B leg to a hangup extension or something? What's the
application here?
-MC

On Tue, Jan 11, 2011 at 7:57 AM, Rafqat . <rafonline at hotmail.com> wrote:

>
>
> Hi,
>
> I moved the bind_digit_action so my dialplan looks something like this:
>
> <extension name="call-me-service">
>  <condition field="destination_number" expression="^9999$">
>    <action application="answer"/>
>    .........
>    <action application="set"
> data="bridge_pre_execute_bleg_app=execute_extension"/>
>    <action application="set"
> data="bridge_pre_execute_bleg_data=START_LISTENING XML private"/>
>     <action application="bind_digit_action"
> data="start,##,exec:hangup,unknown"/>
>    .......
>  </condition>
> </extension>
>
> <extension name="setup bind_digit_action_listening">
>  <condition field="destination_number" expression="^START_LISTENING$">
>    <action application="log" data="INFO Configuring bind_digit_action to
> listen..."/>
>    <action application="digit_action_set_realm" data="start"/>
>  </condition>
> </extension>
>
> Unfortunalety, now when leg A presses ##, it hangs up leg A. I would like
> to make it hang up leg B only.
>
> I am sure it is my lack of understanding of bind_digit_action.
>
> Apreciate all the help I can get.
>
> Cheers
>
> Raf
>
>
> ________________________________
> > Date: Mon, 10 Jan 2011 23:04:52 +0000
> > From: msc at freeswitch.org
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] bind_digit_action
> >
> > Just to confirm: you want Leg A to press ## to hangup leg B? If so then
> > you need to set the bind_digit_action on Leg A. The way you have it now
> > is that Leg B would need to dial ##.
> >
> > -MC
> >
> > On Mon, Jan 10, 2011 at 10:46 PM, Rafqat .
> > > wrote:
> >
> >
> >
> > Hi,
> >
> > Can someone please help here.
> >
> > I am trying to use bind_digit_action to hangup leg B only, if leg A
> > presses ##, but it doesn't seem to be working. The INFO message is
> > displayed but when I press ##, nothing happens.
> >
> > Any help will be much appreciated.
> >
> >
> >
> >
> >
> >
> > .........
> > > data="bridge_pre_execute_bleg_app=execute_extension"/>
> > > data="bridge_pre_execute_bleg_data=START_LISTENING XML private"/>
> > .......
> >
> >
> >
> >
> >
> >
> >
> > > data="start,##,exec:hangup,unknown"/>
> >
> >
> >
> >
> >
> > Cheers
> >
> > Raf
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________ FreeSWITCH-users
> > mailing list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rafonline at hotmail.com  Tue Jan 11 21:50:57 2011
From: rafonline at hotmail.com (Rafqat .)
Date: Tue, 11 Jan 2011 18:50:57 +0000
Subject: [Freeswitch-users] bind_digit_action
In-Reply-To: <AANLkTi=sFderd1MSp6aniXvodyZ1nCbE=1ejGXUuNMXb@mail.gmail.com>
References: <COL122-W20D6C00E627218901B0691B50E0@phx.gbl>,
	<AANLkTim-+iLNZAhNaWU0npAfL2Sn++8mCyqwANXXbVax@mail.gmail.com>,
	<COL122-W297FC85979AA72D5DBC40B50F0@phx.gbl>,
	<AANLkTi=sFderd1MSp6aniXvodyZ1nCbE=1ejGXUuNMXb@mail.gmail.com>
Message-ID: <COL122-W21FA6DF0EAA7D698DFE055B50F0@phx.gbl>



Hi,

The application is a calling card app, when the user caller (leg A) enters ## it should hang up leg B and allow them to make a follow-on call.

I originally tried to use the following:
<action application="set" data="bind_meta_key=#"/>
<action application="bind_meta_app" data="# a o hangup"/>

but freeswitch was trying to invoke playback(hangup). So I moved onto to using bind_digit_action instead.  But can't get it to work with that either.

Please help.

Cheers

Raf


________________________________
> Date: Tue, 11 Jan 2011 18:23:51 +0000
> From: msc at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] bind_digit_action
>
> If you hang up leg B then what should happen to leg A? Can you not just
> transfer the B leg to a hangup extension or something? What's the
> application here?
> -MC
>
> On Tue, Jan 11, 2011 at 7:57 AM, Rafqat .
> > wrote:
>
>
> Hi,
>
> I moved the bind_digit_action so my dialplan looks something like this:
>
>
>
>
> .........
> > data="bridge_pre_execute_bleg_app=execute_extension"/>
> > data="bridge_pre_execute_bleg_data=START_LISTENING XML private"/>
> > data="start,##,exec:hangup,unknown"/>
> .......
>
>
>
>
>
>
>
>
>
>
> Unfortunalety, now when leg A presses ##, it hangs up leg A. I would
> like to make it hang up leg B only.
>
> I am sure it is my lack of understanding of bind_digit_action.
>
> Apreciate all the help I can get.
>
> Cheers
>
> Raf
>
>
> ________________________________
> > Date: Mon, 10 Jan 2011 23:04:52 +0000
> > From: msc at freeswitch.org
> > To:
> freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] bind_digit_action
> >
> > Just to confirm: you want Leg A to press ## to hangup leg B? If so then
> > you need to set the bind_digit_action on Leg A. The way you have it now
> > is that Leg B would need to dial ##.
> >
> > -MC
> >
> > On Mon, Jan 10, 2011 at 10:46 PM, Rafqat .
> > > wrote:
> >
> >
> >
> > Hi,
> >
> > Can someone please help here.
> >
> > I am trying to use bind_digit_action to hangup leg B only, if leg A
> > presses ##, but it doesn't seem to be working. The INFO message is
> > displayed but when I press ##, nothing happens.
> >
> > Any help will be much appreciated.
> >
> >
> >
> >
> >
> >
> > .........
> > > data="bridge_pre_execute_bleg_app=execute_extension"/>
> > > data="bridge_pre_execute_bleg_data=START_LISTENING XML private"/>
> > .......
> >
> >
> >
> >
> >
> >
> >
> > > data="start,##,exec:hangup,unknown"/>
> >
> >
> >
> >
> >
> > Cheers
> >
> > Raf
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> >
> FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________ FreeSWITCH-users
> > mailing list
> FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________ FreeSWITCH-users
> mailing list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  

From me at nevian.org  Tue Jan 11 21:53:03 2011
From: me at nevian.org (Serge Yuriev)
Date: Tue, 11 Jan 2011 21:53:03 +0300
Subject: [Freeswitch-users] NDLB-force-rport safe
Message-ID: <515521294771983@web140.yandex.ru>

Hi

Trying to use safe settings with no luck

Profile starting w/o problem
http://pastebin.freeswitch.org/14986

FS answers to wrong port
http://pastebin.freeswitch.org/14988

Pls advice

-- 
wbr,
Serge


From djbinter at gmail.com  Tue Jan 11 21:58:12 2011
From: djbinter at gmail.com (DJB International)
Date: Tue, 11 Jan 2011 10:58:12 -0800
Subject: [Freeswitch-users] Sofia fails to detect offline gateway
In-Reply-To: <AANLkTin2j-+Y11_x4bs0+M6UsVKUN14fXHoZ262vFu26@mail.gmail.com>
References: <AANLkTim3PeS3fELJPQ21N38c9v5p8J0wGNkBrUvc9R=t@mail.gmail.com>
	<B16E93D7-3578-4071-8899-DDE70DB4D1DC@freeswitch.org>
	<AANLkTin2j-+Y11_x4bs0+M6UsVKUN14fXHoZ262vFu26@mail.gmail.com>
Message-ID: <AANLkTimTutKEvOg+9+MwrxBOp3yJiEax0S-w_5kkvcwB@mail.gmail.com>

Steve,

I am just curious.  If you issue "sofia status gateway gw3" or "sofia status
gateway gw4", do you see the Status as UP or DOWN.

I would have thought that the State should be either NOREG or REG, and
Status should be either UP or DOWN, but somehow the log displayed as state.

Thanks,
-djbinter

On Tue, Jan 11, 2011 at 6:37 AM, Steven Ayre <steveayre at gmail.com> wrote:

> freeswitch at internal> version
> FreeSWITCH Version 1.0.head (git-a8b2840 2011-01-07 17-53-09 -0600)
>
> -Steve
>
>
> On 11 January 2011 14:34, Brian West <brian at freeswitch.org> wrote:
> > what version of FS are you running?
> >
> > /b
> >
> > On Jan 11, 2011, at 3:51 AM, Steven Ayre wrote:
> >
> >>
> >> Is there a mistake in my configuration?
> >>
> >> -Steve
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From me at nevian.org  Tue Jan 11 22:03:24 2011
From: me at nevian.org (Serge Yuriev)
Date: Tue, 11 Jan 2011 22:03:24 +0300
Subject: [Freeswitch-users] NDLB-force-rport safe
In-Reply-To: <515521294771983@web140.yandex.ru>
References: <515521294771983@web140.yandex.ru>
Message-ID: <522281294772604@web154.yandex.ru>

JFYI..

freeswitch at internal> version 
FreeSWITCH Version 1.0.head (git-6f103ac 2011-01-11 09-40-59 -0600)


11.01.2011, 21:53, "Serge Yuriev" <me at nevian.org>:
> Hi
>
> Trying to use safe settings with no luck
>
> Profile starting w/o problem
> http://pastebin.freeswitch.org/14986
>
> FS answers to wrong port
> http://pastebin.freeswitch.org/14988
>
> Pls advice
>
> --
> wbr,
> Serge
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
wbr,
Serge


From steveayre at gmail.com  Tue Jan 11 22:14:09 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 11 Jan 2011 19:14:09 +0000
Subject: [Freeswitch-users] Sofia fails to detect offline gateway
In-Reply-To: <AANLkTimTutKEvOg+9+MwrxBOp3yJiEax0S-w_5kkvcwB@mail.gmail.com>
References: <AANLkTim3PeS3fELJPQ21N38c9v5p8J0wGNkBrUvc9R=t@mail.gmail.com>
	<B16E93D7-3578-4071-8899-DDE70DB4D1DC@freeswitch.org>
	<AANLkTin2j-+Y11_x4bs0+M6UsVKUN14fXHoZ262vFu26@mail.gmail.com>
	<AANLkTimTutKEvOg+9+MwrxBOp3yJiEax0S-w_5kkvcwB@mail.gmail.com>
Message-ID: <AANLkTikMrAzgwE5Jm4qszE_pYXxqO9bF6t1Qtwi9eoLv@mail.gmail.com>

'sofia status' shows NOREG

'sofia status gateway gw3' shows UP if it's already down when FS starts

'sofia status gateway gw3' shows UP (ping) if it's up for a while and then
disappears

Regards,
-Steve



On 11 January 2011 18:58, DJB International <djbinter at gmail.com> wrote:

> Steve,
>
> I am just curious.  If you issue "sofia status gateway gw3" or "sofia
> status gateway gw4", do you see the Status as UP or DOWN.
>
> I would have thought that the State should be either NOREG or REG, and
> Status should be either UP or DOWN, but somehow the log displayed as state.
>
> Thanks,
> -djbinter
>
>
> On Tue, Jan 11, 2011 at 6:37 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> freeswitch at internal> version
>> FreeSWITCH Version 1.0.head (git-a8b2840 2011-01-07 17-53-09 -0600)
>>
>> -Steve
>>
>>
>> On 11 January 2011 14:34, Brian West <brian at freeswitch.org> wrote:
>> > what version of FS are you running?
>> >
>> > /b
>> >
>> > On Jan 11, 2011, at 3:51 AM, Steven Ayre wrote:
>> >
>> >>
>> >> Is there a mistake in my configuration?
>> >>
>> >> -Steve
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Tue Jan 11 22:17:45 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Jan 2011 13:17:45 -0600
Subject: [Freeswitch-users] NDLB-force-rport safe
In-Reply-To: <522281294772604@web154.yandex.ru>
References: <515521294771983@web140.yandex.ru>
	<522281294772604@web154.yandex.ru>
Message-ID: <A9590149-F446-400B-B69B-8407B51B371A@freeswitch.org>

That only applies to Polycom.

/b

On Jan 11, 2011, at 1:03 PM, Serge Yuriev wrote:

> JFYI..
> 
> freeswitch at internal> version 
> FreeSWITCH Version 1.0.head (git-6f103ac 2011-01-11 09-40-59 -0600)
> 



From djbinter at gmail.com  Tue Jan 11 22:28:53 2011
From: djbinter at gmail.com (DJB International)
Date: Tue, 11 Jan 2011 11:28:53 -0800
Subject: [Freeswitch-users] Sofia fails to detect offline gateway
In-Reply-To: <AANLkTikMrAzgwE5Jm4qszE_pYXxqO9bF6t1Qtwi9eoLv@mail.gmail.com>
References: <AANLkTim3PeS3fELJPQ21N38c9v5p8J0wGNkBrUvc9R=t@mail.gmail.com>
	<B16E93D7-3578-4071-8899-DDE70DB4D1DC@freeswitch.org>
	<AANLkTin2j-+Y11_x4bs0+M6UsVKUN14fXHoZ262vFu26@mail.gmail.com>
	<AANLkTimTutKEvOg+9+MwrxBOp3yJiEax0S-w_5kkvcwB@mail.gmail.com>
	<AANLkTikMrAzgwE5Jm4qszE_pYXxqO9bF6t1Qtwi9eoLv@mail.gmail.com>
Message-ID: <AANLkTimJzFBcMu+npWEs1_fW0UDUdhpaBmfpqEADPteL@mail.gmail.com>

I guess you probably have to open a JIRA on this issue.

-djbinter

On Tue, Jan 11, 2011 at 11:14 AM, Steven Ayre <steveayre at gmail.com> wrote:

> 'sofia status' shows NOREG
>
> 'sofia status gateway gw3' shows UP if it's already down when FS starts
>
> 'sofia status gateway gw3' shows UP (ping) if it's up for a while and then
> disappears
>
> Regards,
> -Steve
>
>
>
>
> On 11 January 2011 18:58, DJB International <djbinter at gmail.com> wrote:
>
>> Steve,
>>
>> I am just curious.  If you issue "sofia status gateway gw3" or "sofia
>> status gateway gw4", do you see the Status as UP or DOWN.
>>
>> I would have thought that the State should be either NOREG or REG, and
>> Status should be either UP or DOWN, but somehow the log displayed as state.
>>
>> Thanks,
>> -djbinter
>>
>>
>> On Tue, Jan 11, 2011 at 6:37 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> freeswitch at internal> version
>>> FreeSWITCH Version 1.0.head (git-a8b2840 2011-01-07 17-53-09 -0600)
>>>
>>> -Steve
>>>
>>>
>>> On 11 January 2011 14:34, Brian West <brian at freeswitch.org> wrote:
>>> > what version of FS are you running?
>>> >
>>> > /b
>>> >
>>> > On Jan 11, 2011, at 3:51 AM, Steven Ayre wrote:
>>> >
>>> >>
>>> >> Is there a mistake in my configuration?
>>> >>
>>> >> -Steve
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Tue Jan 11 23:43:30 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 11 Jan 2011 15:43:30 -0500
Subject: [Freeswitch-users] (no subject)
Message-ID: <A370DA1C463A46F7B2CEC8FF361C4CCD@e1705>

in  mod_cepstral wiki page,

to alter volume example is

<prosody volume='15'>This is pretty softly spoken.</prosody>but where to put this xml in dialplan ?Thanks
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From adminjew at gmail.com  Wed Jan 12 01:22:13 2011
From: adminjew at gmail.com (Yitzchok)
Date: Tue, 11 Jan 2011 17:22:13 -0500
Subject: [Freeswitch-users] New FreeSWITCH IVR coming, but need HELP!
In-Reply-To: <AANLkTi=L+EFi50Q4Mb-=X2HW=SsA_TG8tBcg9k6Dgmth@mail.gmail.com>
References: <AANLkTi=L+EFi50Q4Mb-=X2HW=SsA_TG8tBcg9k6Dgmth@mail.gmail.com>
Message-ID: <AANLkTikPb3d=eYDmai08YxJBB=JweghQXJc14qGH3Ash@mail.gmail.com>

Do you mean that you want that when the IVR is done without a hangup command
it should continue down the xml file?


Yitzchok


On Mon, Jan 10, 2011 at 5:03 PM, Zac Wolfe <zacw at safisystems.com> wrote:

> Hi guys,
>
> First some good news:  we're finally close to releasing our free IVR
> Development platform SafiServer/SafiWorkshop (www.safisystems.com) with
> FreeSWITCH support! It's happening much later than we originally anticipated
> as we've been unexpectedly busy with contracting opportunities but I think
> it will be worth the wait.  Currently everything is working fine with one
> minor exception:  if the user-created script (we call them Saflets) doesn't
> explicitly hang up the call, the call will remain parked until the caller
> hangs up.  Some details:
>
> In Asterisk we invoke our server-side scripting applications via the
> extensions.conf using the following syntax:
>
> exten = 1111,1,Agi(agi://
> 192.168.0.10:3573/safletEngine.agi?saflet=project1/callflow1)
>
> Here '192.168.0.10' is the IP address of the SafiServer and
> project1/callflow1 identifies the Saflet to be executed.  Asterisk creates a
> socket connection to the SafiServer and, once the socket is disconnected,
> the call proceeds to the next entry in the dialplan (typically 'hangup').
>
> For FreeSWITCH, the process is slightly different.  Currently, rather than
> create a separate socket connection for each incoming call, we're invoking
> an event that informs the SafiServer that there is a new incoming call.  The
> event contains the contextual information including the Saflet name.  For
> example:
>
> <extension name="9194">
>       <condition field="destination_number" expression="9194$">
>         <action inline="true" application="set" data="saficall=true"/>
>         <action application="event"
> data="Event-Subclass=saficall::incoming,Event-Name=CUSTOM,saflet_project=test,saflet=flow1,new_saficall=true"/>
>
>        <!-- <action application="set" data="park_timeout=10"/> -->
>         <action application="park" />
>       </condition>
>     </extension>
>
> So once the event is fired, the call is parked to prevent further execution
> within the dialplan.  From there on, SafiServer is controlling the call via
> Inbound Mod event socket.
>
> So this works perfectly, except that if the invoked Saflet doesn't
> explicitly hang-up the call it will remain parked until the caller hangs
> up.  My question is, is there a better way to do this?  Is there some better
> alternative to park in this case?  Ideally I'd like to initiate a 'session'
> of some kind when the SafiServer is "controlling" the call and then exit
> that session as soon as the Saflet is complete, at which point the call
> would continue on to the next entry in the dialplan.  I understand I could
> use Outbound sockets to achieve this but, as I mentioned, I'd like to avoid
> the overhead of a separate socket connection for each incoming call.
>
> I actually have a mod_saficall.c app that does basically the the same thing
> as I described in the dialplan entry.  Perhaps there's something more I
> could do in code that would allow me to be notified when the session is
> complete.  Here's the relevant code I have so far:
>
>     switch_channel_t *channel = NULL;
>     switch_event_t *event;
>     const char *safiCallFlag = NULL;
>     channel = switch_core_session_get_channel(session);
>
>     safiCallFlag = switch_channel_get_variable(channel, "saficall");
>
>     if (!safiCallFlag)
>         switch_channel_set_variable(channel, "saficall", "true");
>
>
>     if (switch_event_create_subclass(&event, SWITCH_EVENT_CUSTOM,
> "saficall::incoming") == SWITCH_STATUS_SUCCESS) {
>
>         switch_event_add_header_string(event, SWITCH_STACK_BOTTOM,
> "new_saficall", safiCallFlag ? "false" : "true");
>
>         switch_channel_event_set_data(channel, event);
>         switch_event_fire(&event);
>         switch_ivr_park(session, NULL);
>     }
>
> Any ideas you might have on this are welcome.
>
> Thanks,
> Zac Wolfe
> Safi Systems LLC
> www.safisystems.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lists at telefaks.de  Wed Jan 12 01:35:31 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Tue, 11 Jan 2011 23:35:31 +0100
Subject: [Freeswitch-users] Try to get rid of Speex codec on invite
Message-ID: <4D2CDB33.3070209@telefaks.de>

Hello,

when our Freeswitch sends an INVITE to a phone, it only offers speex
codec for voice:
 m=audio 12068 RTP/SAVP 8 0 98 3 18 101 13
 a=rtpmap:98 SPEEX/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16

When the phone then answers with ACK and OK and sends back it's own
codecs during OK, then codecs are negociated to PCMA, which is fine.

However our Aastra phones do not like Speex in the INVITE message, so
they send back a "Bad request".
We have set in vars.xml
  <X-PRE-PROCESS cmd="set"
data="global_codec_prefs=PCMA,PCMU,G729,GSM,speex at 8000h@20i,speex at 16000h@20i"/>
  <X-PRE-PROCESS cmd="set"
data="outbound_codec_prefs=PCMA,PCMU,G729,GSM,speex at 8000h@20i,speex at 16000h@20i"/>

and the internal profile contains
  <param name="inbound-codec-prefs" value="PCMA,PCMU,speex,GSM,G729"/>
  <param name="outbound-codec-prefs" value="PCMA,PCMU,speex,GSM,G729"/>
  <param name="inbound-codec-negotiation" value="greedy"/>
  <param name="outbound-codec-negotiation" value="greedy"/>

So Speex is not the preferred codec defined. Eeven 'scrooge' did not
solve the problem.

How can I get rid of this behaviour and force Freeswitch to invite with
PCMA?

-- 

With kind regards
Peter Steinbach 

Telefaks Services GmbH
Theo-Geisel-Strasse 25
D 61250 Usingen, Germany
mailto:lists (att) telefaks.de
Internet: www.telefaks.de




From anthony.minessale at gmail.com  Wed Jan 12 01:45:21 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 11 Jan 2011 16:45:21 -0600
Subject: [Freeswitch-users] New FreeSWITCH IVR coming, but need HELP!
In-Reply-To: <AANLkTi=L+EFi50Q4Mb-=X2HW=SsA_TG8tBcg9k6Dgmth@mail.gmail.com>
References: <AANLkTi=L+EFi50Q4Mb-=X2HW=SsA_TG8tBcg9k6Dgmth@mail.gmail.com>
Message-ID: <AANLkTikGiLtxOXJAbb-3KKzC4M_B+1XzMzimjDDcs_HW@mail.gmail.com>

What condition would you want to use to have the park terminate?

Once you app is controlling the session, it would be up to you to
enforce when it hangs up from the FS side.

Based on what you describe the only issue could be when your remote
application either misses the event or is restarted while calls are up
so what I can suggest is this:

in your C app, you could wait there for some timeout period just
calling switch_ivr_sleep for 1 second up to 10 tries to wait a total
of 10 seconds.

If your app gets the event it can then transfer it to park using
uuid_transfer, this would break the sleep loop and you could do
something at the end of the loop like:

if (switch_channel_ready(channel)) {
   switch_channel_hangup(channel, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER);
}

so when it was already hung or being transfered that could would not
execute but if your loop ended from waiting too long and it was still
active it would hangup.

you could do something similar with just dp logic if you used park_timeout of 10
then in your event handler app, use uuid_setvar to unset park_timeout,
then uuid_transfer it back to park now with now timeout

uuid_transfer <uuid> set:park_timeout=,park inline

You should come and present this at ClueCon if you have it done in time.



On Mon, Jan 10, 2011 at 4:03 PM, Zac Wolfe <zacw at safisystems.com> wrote:
> Hi guys,
>
> First some good news:? we're finally close to releasing our free IVR
> Development platform SafiServer/SafiWorkshop (www.safisystems.com) with
> FreeSWITCH support! It's happening much later than we originally anticipated
> as we've been unexpectedly busy with contracting opportunities but I think
> it will be worth the wait.? Currently everything is working fine with one
> minor exception:? if the user-created script (we call them Saflets) doesn't
> explicitly hang up the call, the call will remain parked until the caller
> hangs up.? Some details:
>
> In Asterisk we invoke our server-side scripting applications via the
> extensions.conf using the following syntax:
>
> exten =
> 1111,1,Agi(agi://192.168.0.10:3573/safletEngine.agi?saflet=project1/callflow1)
>
> Here '192.168.0.10' is the IP address of the SafiServer and
> project1/callflow1 identifies the Saflet to be executed.? Asterisk creates a
> socket connection to the SafiServer and, once the socket is disconnected,
> the call proceeds to the next entry in the dialplan (typically 'hangup').
>
> For FreeSWITCH, the process is slightly different.? Currently, rather than
> create a separate socket connection for each incoming call, we're invoking
> an event that informs the SafiServer that there is a new incoming call.? The
> event contains the contextual information including the Saflet name.? For
> example:
>
> <extension name="9194">
> ????? <condition field="destination_number" expression="9194$">
> ??????? <action inline="true" application="set" data="saficall=true"/>
> ??????? <action application="event"
> data="Event-Subclass=saficall::incoming,Event-Name=CUSTOM,saflet_project=test,saflet=flow1,new_saficall=true"/>
>
> ?????? <!-- <action application="set" data="park_timeout=10"/> -->
> ??????? <action application="park" />
> ????? </condition>
> ??? </extension>
>
> So once the event is fired, the call is parked to prevent further execution
> within the dialplan.? From there on, SafiServer is controlling the call via
> Inbound Mod event socket.
>
> So this works perfectly, except that if the invoked Saflet doesn't
> explicitly hang-up the call it will remain parked until the caller hangs
> up.? My question is, is there a better way to do this?? Is there some better
> alternative to park in this case?? Ideally I'd like to initiate a 'session'
> of some kind when the SafiServer is "controlling" the call and then exit
> that session as soon as the Saflet is complete, at which point the call
> would continue on to the next entry in the dialplan.? I understand I could
> use Outbound sockets to achieve this but, as I mentioned, I'd like to avoid
> the overhead of a separate socket connection for each incoming call.
>
> I actually have a mod_saficall.c app that does basically the the same thing
> as I described in the dialplan entry.? Perhaps there's something more I
> could do in code that would allow me to be notified when the session is
> complete.? Here's the relevant code I have so far:
>
> ??? switch_channel_t *channel = NULL;
> ??? switch_event_t *event;
> ??? const char *safiCallFlag = NULL;
> ??? channel = switch_core_session_get_channel(session);
>
> ??? safiCallFlag = switch_channel_get_variable(channel, "saficall");
>
> ??? if (!safiCallFlag)
> ??????? switch_channel_set_variable(channel, "saficall", "true");
>
>
> ??? if (switch_event_create_subclass(&event, SWITCH_EVENT_CUSTOM,
> "saficall::incoming") == SWITCH_STATUS_SUCCESS) {
>
> ??????? switch_event_add_header_string(event, SWITCH_STACK_BOTTOM,
> "new_saficall", safiCallFlag ? "false" : "true");
>
> ??????? switch_channel_event_set_data(channel, event);
> ??????? switch_event_fire(&event);
> ??????? switch_ivr_park(session, NULL);
> ??? }
>
> Any ideas you might have on this are welcome.
>
> Thanks,
> Zac Wolfe
> Safi Systems LLC
> www.safisystems.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From brian at freeswitch.org  Wed Jan 12 01:54:20 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Jan 2011 16:54:20 -0600
Subject: [Freeswitch-users] Try to get rid of Speex codec on invite
In-Reply-To: <4D2CDB33.3070209@telefaks.de>
References: <4D2CDB33.3070209@telefaks.de>
Message-ID: <C4EED9DD-2971-461D-BF4F-FE7B94CDA222@freeswitch.org>

You are making the wrong conclusion.  The SDP is being verbose.  That SDP is offering PCMA, PCMU, Speex on 98, GSM, DTMF on 101 and CN

What you need to set is the variable verbose_sdp=true so that we do a complete filled out SDP because the device you're speaking to is being intellectually challenged.  See the codec list in the m= line?

/b

On Jan 11, 2011, at 4:35 PM, Peter Steinbach wrote:

> Hello,
> 
> when our Freeswitch sends an INVITE to a phone, it only offers speex
> codec for voice:
> m=audio 12068 RTP/SAVP 8 0 98 3 18 101 13
> a=rtpmap:98 SPEEX/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16



From lists at telefaks.de  Wed Jan 12 01:59:51 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Tue, 11 Jan 2011 23:59:51 +0100
Subject: [Freeswitch-users] FS 502- Bad gateway to Patton SN4960
In-Reply-To: <AANLkTinGL3Rf9Y0B-bauch1aY=--MYj6rDq5-NdSc36m@mail.gmail.com>
References: <AANLkTinGL3Rf9Y0B-bauch1aY=--MYj6rDq5-NdSc36m@mail.gmail.com>
Message-ID: <4D2CE0E7.8070608@telefaks.de>

Hello,

the problem seems to be here:

   1.
      2011-01-10 18:26:50.675121 [DEBUG] sofia_glue.c:2971 AUDIO RTP
      [sofia/pstn/22969927 at 192.168.105.46:5060] 192.168.105.5} port
      26570 -> 192.168.105.46 port 4886 codec: 8 ms: 20
   2.
      2011-01-10 18:26:50.675121 [DEBUG] switch_rtp.c:1418 Starting
      timer [soft] 160 bytes per 20ms
   3.
      2011-01-10 18:26:51.059032 [ERR] sofia_glue.c:3410 AUDIO RTP
      REPORTS ERROR: [Local Address Error!]

There seems to be a problem with the Ip or the UDP port of Freeswitch.

Best regards
Peter


Luis F Urrea schrieb:
> Hi guys,
>
> I have a call coming through a Patton GW in the same LAN as FS, call
> comes in and FS responds with 502 to GW.
>
> Here's a console + siptrace log
>
> http://pastebin.freeswitch.org/14978
>
> I appreciate your input and advice on where to direct my
> troubleshooting efforts!
>
> TIA
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
Theo-Geisel-Strasse 25
D 61250 Usingen, Germany
mailto:lists (att) telefaks.de
Internet: www.telefaks.de




From brian at freeswitch.org  Wed Jan 12 02:04:26 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Jan 2011 17:04:26 -0600
Subject: [Freeswitch-users] FS 502- Bad gateway to Patton SN4960
In-Reply-To: <4D2CE0E7.8070608@telefaks.de>
References: <AANLkTinGL3Rf9Y0B-bauch1aY=--MYj6rDq5-NdSc36m@mail.gmail.com>
	<4D2CE0E7.8070608@telefaks.de>
Message-ID: <50DB7B47-365E-46B5-A333-400F8A89E647@freeswitch.org>

nope it was a stray } in his rtp-ip  :P

/b

On Jan 11, 2011, at 4:59 PM, Peter Steinbach wrote:

> Hello,
> 
> the problem seems to be here:
> 
>   1.
>      2011-01-10 18:26:50.675121 [DEBUG] sofia_glue.c:2971 AUDIO RTP
>      [sofia/pstn/22969927 at 192.168.105.46:5060] 192.168.105.5} port
>      26570 -> 192.168.105.46 port 4886 codec: 8 ms: 20
>   2.
>      2011-01-10 18:26:50.675121 [DEBUG] switch_rtp.c:1418 Starting
>      timer [soft] 160 bytes per 20ms
>   3.
>      2011-01-10 18:26:51.059032 [ERR] sofia_glue.c:3410 AUDIO RTP
>      REPORTS ERROR: [Local Address Error!]
> 
> There seems to be a problem with the Ip or the UDP port of Freeswitch.
> 
> Best regards
> Peter



From lists at telefaks.de  Wed Jan 12 02:10:28 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Wed, 12 Jan 2011 00:10:28 +0100
Subject: [Freeswitch-users] Try to get rid of Speex codec on invite
In-Reply-To: <C4EED9DD-2971-461D-BF4F-FE7B94CDA222@freeswitch.org>
References: <4D2CDB33.3070209@telefaks.de>
	<C4EED9DD-2971-461D-BF4F-FE7B94CDA222@freeswitch.org>
Message-ID: <4D2CE364.5060209@telefaks.de>

Thanks Brian,

that did the trick.

Best regards
Peter


Brian West schrieb:
> You are making the wrong conclusion.  The SDP is being verbose.  That SDP is offering PCMA, PCMU, Speex on 98, GSM, DTMF on 101 and CN
>
> What you need to set is the variable verbose_sdp=true so that we do a complete filled out SDP because the device you're speaking to is being intellectually challenged.  See the codec list in the m= line?
>
> /b
>
> On Jan 11, 2011, at 4:35 PM, Peter Steinbach wrote:
>
>   
>> Hello,
>>
>> when our Freeswitch sends an INVITE to a phone, it only offers speex
>> codec for voice:
>> m=audio 12068 RTP/SAVP 8 0 98 3 18 101 13
>> a=rtpmap:98 SPEEX/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
Theo-Geisel-Strasse 25
D 61250 Usingen, Germany
mailto:lists (att) telefaks.de
Internet: www.telefaks.de




From steveayre at gmail.com  Wed Jan 12 02:12:45 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 11 Jan 2011 23:12:45 +0000
Subject: [Freeswitch-users] Try to get rid of Speex codec on invite
In-Reply-To: <C4EED9DD-2971-461D-BF4F-FE7B94CDA222@freeswitch.org>
References: <4D2CDB33.3070209@telefaks.de>
	<C4EED9DD-2971-461D-BF4F-FE7B94CDA222@freeswitch.org>
Message-ID: <AANLkTikpFMm2_ZKPEZBeX_W=ykhLXXEa7X3nDKoKrea=@mail.gmail.com>

Just to be clear in case you're new to SDP...

The m=audio line gives a list of codecs identified by by number. Many of
them are static numbers meaning they are assigned to particular codecs. The
list is at http://www.iana.org/assignments/rtp-parameters 0=PCMU 8=PCMA etc.

Anything in the 96-127 range is dynamic and identified by name on a a=rtpmap
line.

For static numbers it's allowed to have a a=rtpmap line, but not required.
But some devices are broken and require it anyway. verbose=sdp=true puts
a=rtpmap for all codecs, which will work on all devices but at the risk of
possibly exceeding the MTU.

-Steve





On 11 January 2011 22:54, Brian West <brian at freeswitch.org> wrote:

> You are making the wrong conclusion.  The SDP is being verbose.  That SDP
> is offering PCMA, PCMU, Speex on 98, GSM, DTMF on 101 and CN
>
> What you need to set is the variable verbose_sdp=true so that we do a
> complete filled out SDP because the device you're speaking to is being
> intellectually challenged.  See the codec list in the m= line?
>
> /b
>
> On Jan 11, 2011, at 4:35 PM, Peter Steinbach wrote:
>
> > Hello,
> >
> > when our Freeswitch sends an INVITE to a phone, it only offers speex
> > codec for voice:
> > m=audio 12068 RTP/SAVP 8 0 98 3 18 101 13
> > a=rtpmap:98 SPEEX/8000
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-16
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Wed Jan 12 02:17:35 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Jan 2011 17:17:35 -0600
Subject: [Freeswitch-users] Try to get rid of Speex codec on invite
In-Reply-To: <AANLkTikpFMm2_ZKPEZBeX_W=ykhLXXEa7X3nDKoKrea=@mail.gmail.com>
References: <4D2CDB33.3070209@telefaks.de>
	<C4EED9DD-2971-461D-BF4F-FE7B94CDA222@freeswitch.org>
	<AANLkTikpFMm2_ZKPEZBeX_W=ykhLXXEa7X3nDKoKrea=@mail.gmail.com>
Message-ID: <D05306AF-67F1-4F82-AD61-9F2427815E4D@freeswitch.org>

Funny part is I don't think the people that invented SDP can even understand the monster it has turned into.

/b

On Jan 11, 2011, at 5:12 PM, Steven Ayre wrote:

> Just to be clear in case you're new to SDP...



From matte at ahavaxthuset.se  Tue Jan 11 21:31:46 2011
From: matte at ahavaxthuset.se (Mattias Hemmingsson)
Date: Tue, 11 Jan 2011 19:31:46 +0100 (CET)
Subject: [Freeswitch-users] Two meny qestions
In-Reply-To: <AANLkTikd_A3JCYO3GC0m8KTwWCorQvP+USvO27SBk5Wv@mail.gmail.com>
Message-ID: <24241510.1431294770706120.JavaMail.root@mailserver>

Hi


Sorry can find any command called pasebin in the freeswitch consol ore in the cli.
Will test to do som debugging from the link you sent.


// Matte



----- Ursprungligt meddelande -----
Fr?n: "Michael Collins" <msc at freeswitch.org>
Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Skickat: tisdag, 11 jan 2011 6:39:34
?mne: Re: [Freeswitch-users] Two meny qestions


get a console log and put it in pastebin. That should help us to see what is going on. Look at this page if you need more information on getting debug information: 


http://wiki.freeswitch.org/wiki/Reporting_Bugs 


-MC 


On Mon, Jan 10, 2011 at 6:23 PM, Mattias Hemmingsson < matte at ahavaxthuset.se > wrote: 


Hi 


Still having problems transfering the call to the right extension. 
I have one domain in my directory called www.elino.se and in there all my users all. 
So i test to set upp the meny to transfer the call to my extension like this 

<entry action="menu-exec-app" digits="1" param="transfer 1000 at www.elino.se "/> 

but it dont work. 

i have also test to transfer with 


<entry action="menu-exec-app" digits="1" param="transfer 1000 XML default"/> 

but the i only get an godbye. 


what im i doing wrong ? 

// Matte 


----- Ursprungligt meddelande ----- 
Fr?n: "Brian West" < brian at freeswitch.org > 
Till: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > 
Skickat: s?ndag, 9 jan 2011 22:35:00 
?mne: Re: [Freeswitch-users] Two meny qestions 




This is because you're calling bridge right to the users endpoint... if you were to transfer to extension 1000 or 1001 then voicemail would work exactly like you expect. 

/b 

On Jan 9, 2011, at 1:21 PM, Mattias Hemmingsson wrote: 

> But when the is not online i want the user to be transferd to the users voicemail. 
> I have voicemail working of i call the user from a nother externsion. 
> But i would like it to wokr from the ivr meny as well. 


_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 

_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From lfurrea at gmail.com  Wed Jan 12 04:05:15 2011
From: lfurrea at gmail.com (Luis F Urrea)
Date: Tue, 11 Jan 2011 19:05:15 -0600
Subject: [Freeswitch-users] FS 502- Bad gateway to Patton SN4960
In-Reply-To: <50DB7B47-365E-46B5-A333-400F8A89E647@freeswitch.org>
References: <AANLkTinGL3Rf9Y0B-bauch1aY=--MYj6rDq5-NdSc36m@mail.gmail.com>
	<4D2CE0E7.8070608@telefaks.de>
	<50DB7B47-365E-46B5-A333-400F8A89E647@freeswitch.org>
Message-ID: <AANLkTineN9o6OWubrzL+H-yi4QpdKVJ83fGrWJRbfyow@mail.gmail.com>

Peter thank you very much!

You were right about it being an issue with FS's IP/port.

To my shame, Brian spotted the error in my profile.xml :)

Thank you both guys for your time!





On Tue, Jan 11, 2011 at 5:04 PM, Brian West <brian at freeswitch.org> wrote:

> nope it was a stray } in his rtp-ip  :P
>
> /b
>
> On Jan 11, 2011, at 4:59 PM, Peter Steinbach wrote:
>
> > Hello,
> >
> > the problem seems to be here:
> >
> >   1.
> >      2011-01-10 18:26:50.675121 [DEBUG] sofia_glue.c:2971 AUDIO RTP
> >      [sofia/pstn/22969927 at 192.168.105.46:5060] 192.168.105.5} port
> >      26570 -> 192.168.105.46 port 4886 codec: 8 ms: 20
> >   2.
> >      2011-01-10 18:26:50.675121 [DEBUG] switch_rtp.c:1418 Starting
> >      timer [soft] 160 bytes per 20ms
> >   3.
> >      2011-01-10 18:26:51.059032 [ERR] sofia_glue.c:3410 AUDIO RTP
> >      REPORTS ERROR: [Local Address Error!]
> >
> > There seems to be a problem with the Ip or the UDP port of Freeswitch.
> >
> > Best regards
> > Peter
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Wed Jan 12 05:08:28 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 11 Jan 2011 21:08:28 -0500
Subject: [Freeswitch-users] cepstral and SSML
Message-ID: <B619ED95945044D5934B36A8D1AEBEF2@e1705>

if I use

<action application="speak" data="cepstral|david|<prosody volume='65'>Your credit is too low to reach the landline user phone. Transfer to voicemail.</prosody>"/>

it seems that SSML isn't supported like this,
the voice says "/ prosody" only. the whole sentence is not said.

do I escape the SSML code in the data ?

Thanks
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From rupa at rupa.com  Wed Jan 12 05:36:36 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 11 Jan 2011 20:36:36 -0600
Subject: [Freeswitch-users] How to disable 5060-5080 with auto-nat at
	upnp?
In-Reply-To: <BLU0-SMTP13FBD515FEBDD61F7E04F08D0F0@phx.gbl>
References: <AANLkTi=UeJNn-2bimdA+PkJHL+deLkh9HaQuyL-DmC=4@mail.gmail.com>
	<DED97F2D-319E-4073-9141-A463BCA23291@freeswitch.org>
	<BLU0-SMTP13FBD515FEBDD61F7E04F08D0F0@phx.gbl>
Message-ID: <AANLkTin_f7sm=UphBivKti0nk2721=NaoiciD4PMDs9q@mail.gmail.com>

Again, I would say block the ports at your router.  Or block them with
a host-based firewall on the server.

On Tue, Jan 11, 2011 at 12:22 PM, Alex Wang <hadyn_whx at hotmail.com> wrote:
> Hi Brian
>
> I think you misunderstood me on this, what I want to say is not every
> PBX will face to the public, even in the commercial environment, the PBX
> may not need open 5060-5080 to the public, but the sip session will need
> upnp to automatically map the port. My question is how to handle this
> kind of situation. Just -nonat and map all the other port?
>
> Thanks
>
> Alex
> On Tue, 11 Jan 2011 11:06:56 -0600
> Brian West <brian at freeswitch.org> wrote:
>
>> start FreeSWITCH with -nonat .. I can NOT and will NOT change in the way it works. ?If you want to manage your mappings yourself do so... I'm not going to cripple the autonat to not AUTO nat.
>>
>> /b
>>
>> On Jan 11, 2011, at 9:09 AM, Rupa Schomaker wrote:
>>
>> > Hmmm... ?I didn't put anything in the nat_map code to allow some parts
>> > of the sofia profile to participate in the nat mapping while others do
>> > not.
>> >
>> > Brian, do you have any ideas?
>> >
>> > Alex, I'd suggest just blocking ports 5060-5080 on the firewall. ?The
>> > port blocking *should* take precedence over the upnp maps.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa


From brian at freeswitch.org  Wed Jan 12 05:50:57 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Jan 2011 20:50:57 -0600
Subject: [Freeswitch-users] How to disable 5060-5080 with auto-nat at
	upnp?
In-Reply-To: <AANLkTin_f7sm=UphBivKti0nk2721=NaoiciD4PMDs9q@mail.gmail.com>
References: <AANLkTi=UeJNn-2bimdA+PkJHL+deLkh9HaQuyL-DmC=4@mail.gmail.com>
	<DED97F2D-319E-4073-9141-A463BCA23291@freeswitch.org>
	<BLU0-SMTP13FBD515FEBDD61F7E04F08D0F0@phx.gbl>
	<AANLkTin_f7sm=UphBivKti0nk2721=NaoiciD4PMDs9q@mail.gmail.com>
Message-ID: <0E0757C5-9D2B-484D-A4FA-7425E53E1481@freeswitch.org>

Auto nat isn't or the commercial env.  Its for the small office / home office.

/b

On Jan 11, 2011, at 8:36 PM, Rupa Schomaker wrote:

> Again, I would say block the ports at your router.  Or block them with
> a host-based firewall on the server.
> 
> On Tue, Jan 11, 2011 at 12:22 PM, Alex Wang <hadyn_whx at hotmail.com> wrote:
>> Hi Brian
>> 
>> I think you misunderstood me on this, what I want to say is not every
>> PBX will face to the public, even in the commercial environment, the PBX
>> may not need open 5060-5080 to the public, but the sip session will need
>> upnp to automatically map the port. My question is how to handle this
>> kind of situation. Just -nonat and map all the other port?
>> 
>> Thanks
>> 
>> Alex
>> On Tue, 11 Jan 2011 11:06:56 -0600
>> Brian West <brian at freeswitch.org> wrote:
>> 
>>> start FreeSWITCH with -nonat .. I can NOT and will NOT change in the way it works.  If you want to manage your mappings yourself do so... I'm not going to cripple the autonat to not AUTO nat.
>>> 
>>> /b

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From infos at madovsky.org  Wed Jan 12 06:06:13 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 11 Jan 2011 22:06:13 -0500
Subject: [Freeswitch-users] extract channel vars from another channel
Message-ID: <9E066B8B7262408F8E3B0381D26BF236@e1705>

from xml dialplan I'd like to
get channel vars from another channel but I know only the number of the targetted (internal) user.
is there a way ?

Thanks
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From hadyn_whx at hotmail.com  Wed Jan 12 06:20:41 2011
From: hadyn_whx at hotmail.com (Alex Wang)
Date: Tue, 11 Jan 2011 22:20:41 -0500
Subject: [Freeswitch-users] How to disable 5060-5080 with auto-nat at
	upnp?
In-Reply-To: <0E0757C5-9D2B-484D-A4FA-7425E53E1481@freeswitch.org>
References: <AANLkTin_f7sm=UphBivKti0nk2721=NaoiciD4PMDs9q@mail.gmail.com>
	<0E0757C5-9D2B-484D-A4FA-7425E53E1481@freeswitch.org>
Message-ID: <BLU0-SMTP130EAE2E7CAABEA362202528DF10@phx.gbl>

Well, whatever, I think for now I am just use port forwarding to black
hole to provent it map to FS. It works. Thanks Brian and Rupa.

Alex

On Tue, 11 Jan 2011 20:50:57 -0600
Brian West <brian at freeswitch.org> wrote:

> Auto nat isn't or the commercial env.  Its for the small office / home office.
> 
> /b
> 
> On Jan 11, 2011, at 8:36 PM, Rupa Schomaker wrote:
> 
> > Again, I would say block the ports at your router.  Or block them with
> > a host-based firewall on the server.
> > 
> > On Tue, Jan 11, 2011 at 12:22 PM, Alex Wang <hadyn_whx at hotmail.com> wrote:
> >> Hi Brian
> >> 
> >> I think you misunderstood me on this, what I want to say is not every
> >> PBX will face to the public, even in the commercial environment, the PBX
> >> may not need open 5060-5080 to the public, but the sip session will need
> >> upnp to automatically map the port. My question is how to handle this
> >> kind of situation. Just -nonat and map all the other port?
> >> 
> >> Thanks
> >> 
> >> Alex
> >> On Tue, 11 Jan 2011 11:06:56 -0600
> >> Brian West <brian at freeswitch.org> wrote:
> >> 
> >>> start FreeSWITCH with -nonat .. I can NOT and will NOT change in the way it works.  If you want to manage your mappings yourself do so... I'm not going to cripple the autonat to not AUTO nat.
> >>> 
> >>> /b
> 





From Nabble at slickdeals.endjunk.com  Wed Jan 12 06:47:22 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 11 Jan 2011 19:47:22 -0800 (PST)
Subject: [Freeswitch-users] Asynchronous PTIME
In-Reply-To: <AANLkTim6JbGqr+tid8Unrr5T7n78Zxyr4xUAFtWHJW_i@mail.gmail.com>
References: <4D127733.8010409@nevian.org> <4D15266B.3080903@sns.eu>
	<AANLkTimzmg1kfM4+75e-Y9vG6wErhXjX6f83+4-voNyu@mail.gmail.com>
	<4D1CBC10.8080004@sns.eu>
	<AANLkTim6JbGqr+tid8Unrr5T7n78Zxyr4xUAFtWHJW_i@mail.gmail.com>
Message-ID: <1294804042717-5913375.post@n2.nabble.com>



Saeed Ahmed wrote:
> Thanks Jan, I tried that, but still with ptime: 40 the voice is choppy :(
I just upgraded my FS v1.0.6 to today's git version and experienced this
issue. After reading through the posts on this thread, I did a search for
rtp-autofix-timing variable and found it is commented under
conf/sip_profiles/internal.xml file. I removed the comment and did a sofia
profile internal restart reloadxml under fs_cli and the problem went away.
So far, so good. Thanks to anyone who suggested this option.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Asynchronous-PTIME-tp5860788p5913375.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From infos at madovsky.org  Wed Jan 12 07:48:01 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 11 Jan 2011 23:48:01 -0500
Subject: [Freeswitch-users] mod_db limit and group
Message-ID: <5D26741A84A541D2A9D7AB130EB93086@e1705>


Is there any example of limit and group use ?
I don't understand how to use it.

Thanks
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From u2nsam at gmail.com  Wed Jan 12 08:26:39 2011
From: u2nsam at gmail.com (Sam)
Date: Wed, 12 Jan 2011 10:56:39 +0530
Subject: [Freeswitch-users] console
In-Reply-To: <AANLkTi=6Hwx2Y67Htd+OA2PefQAuVyXUxOGWgEp+F9Ce@mail.gmail.com>
References: <AANLkTi=_ZNGF1BKsWh6CP8ExrmMYYqwzEx8C5O9OXCZ8@mail.gmail.com>
	<AANLkTi=0ZCQYgpR7T4LF11oPvGn75WeinYLPYF-BQXJn@mail.gmail.com>
	<AANLkTinZazjfEZcRHCuj0a9a9r5xg03r-NECCzqNzPeh@mail.gmail.com>
	<AANLkTi=qEgee_iH+WLATtXPSCP=2-zb0Wb8aiwFNE_7+@mail.gmail.com>
	<AANLkTimVk-AZ3dPnE-Kkj4tE8XS7WuLsQzczqTW1fBqf@mail.gmail.com>
	<AANLkTiknTsNhvbt-8D185sy_+LbmfJhw6HGR3+C_XgWZ@mail.gmail.com>
	<AANLkTikdAic7TowHGT=2CStif-6zw0QoCyrE8X3KVXp7@mail.gmail.com>
	<AANLkTikx+PYeN7c51N+t87eQ8FN6i7vJy7pswE2McLcy@mail.gmail.com>
	<AANLkTi=6Hwx2Y67Htd+OA2PefQAuVyXUxOGWgEp+F9Ce@mail.gmail.com>
Message-ID: <AANLkTimN5+VV7Hrz3n8vmrCBrHM7xQKGeyz56C4UVwBG@mail.gmail.com>

Tried this below and received :-


# /usr/local/fs_2/bin/fs_cli -H 127.0.0.2 -P 8082
[ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
Error]

tried by creating profile:-
# /usr/local/fs_2/bin/fs_cli profile1
[ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
Error]

By port:
# /usr/local/fs_2/bin/fs_cli -P 8082
[ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
Error]


*netstat -nlp | grep ; ports 8081 & 8082 are available*

But i could get a console for other server by
# /usr/local/fs_2/bin/fs_cli

Regards
Sam


On Tue, Jan 11, 2011 at 10:54 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Actually, correction it does - just not the event_socket.conf.xml one.
> It'll read .fs_cli_conf in your home directory if it exists, but that isn't
> created by default - you create it yourself if you want to use it (it's
> optional). The command line arguments -H and -P -will- override the config
> file though.
>
> Are you using a capital P? -p is password while -P is port. If there's no
> password on the event socket you'd get no error from using a small p by
> accident.
>
> -Steve
>
>
>
>
>
>
> On 11 January 2011 16:59, Steven Ayre <steveayre at gmail.com> wrote:
>
>> It does not read any config file.
>>
>>
>>
>>
>>
>> On 11 January 2011 16:56, Sam <u2nsam at gmail.com> wrote:
>>
>>> the port and ip donot work for me,
>>> is it that the fs_cli is not reading the config from 192.168.2.2 but it
>>> is reading the config only of 192.168.2.1, though its in the different [FS_1
>>> & FS_2] path where i am executing.
>>>
>>> Regds
>>> Sam
>>>
>>>
>>>
>>> On Tue, Jan 11, 2011 at 9:38 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>>
>>>> /usr/local/FS_1/bin/fs_cli -P 8021
>>>> /usr/local/FS_2/bin/fs_cli -P 8022
>>>>
>>>> fs_cll doesn't read any config file. It's not part of the FS server at
>>>> all, you can have it on a different machine that doesn't have FS installed.
>>>> It entirely relies on the arguments to control where to connect to.
>>>>
>>>> -Steve
>>>>
>>>>
>>>>
>>>> On 11 January 2011 15:04, Sam <u2nsam at gmail.com> wrote:
>>>>
>>>>> Something more here ... i am getting the console for 192.168.2.1 every
>>>>> time i do fs_cli on both instances .
>>>>>
>>>>> like
>>>>> /usr/local/FS_1/bin/fs_cli
>>>>> /usr/localFS_2/bin/fs_cli
>>>>> i get the console for the 1st server only
>>>>>
>>>>> the 2 server are listing to 2 different ips .
>>>>>
>>>>> Regds
>>>>> Sam
>>>>>
>>>>>
>>>>> On Tue, Jan 11, 2011 at 8:07 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>>>>
>>>>>> It should work. Is there anything already listening on port 8022?
>>>>>>
>>>>>> $ netstat -a -n -p | grep 8022
>>>>>>
>>>>>> Are you also sure that they're not both loading the same config file?
>>>>>>
>>>>>> Regards,
>>>>>> -Steve
>>>>>>
>>>>>>
>>>>>>
>>>>>> On 11 January 2011 14:32, Sam <u2nsam at gmail.com> wrote:
>>>>>> > the scenario is i have 2 ips on 1 server for 2 FS instances;
>>>>>> >
>>>>>> > 192.168.2.1
>>>>>> > 192.168.2.2
>>>>>> >
>>>>>> > and the parameters i have set is:-
>>>>>> >
>>>>>> > for
>>>>>> > 192.168.2.1:-
>>>>>> >    <param name="listen-ip" value="127.0.0.1"/>
>>>>>> >    <param name="listen-port" value="8021"/>
>>>>>> > for
>>>>>> > 192.168.2.2:-
>>>>>> >    <param name="listen-ip" value="127.0.0.1"/>
>>>>>> >    <param name="listen-port" value="8022"/>
>>>>>> >
>>>>>> > Ideally it should work  but i am getting console for only
>>>>>> 192.168.2.1 FS .
>>>>>> >
>>>>>> >
>>>>>> > Regards
>>>>>> > Sam
>>>>>> >
>>>>>> > On Tue, Jan 11, 2011 at 4:20 PM, Steven Ayre <steveayre at gmail.com>
>>>>>> wrote:
>>>>>> >>
>>>>>> >>    <param name="listen-ip" value="127.0.0.1"/>
>>>>>> >>    <param name="listen-port" value="8021"/>
>>>>>> >>
>>>>>> >> You can bind both to port 8021 on their individual IPs, or
>>>>>> different
>>>>>> >> ports on the same IP.
>>>>>> >>
>>>>>> >> A listen IP of 0.0.0.0 will mean any IP.
>>>>>> >>
>>>>>> >> -Steve
>>>>>> >>
>>>>>> >> On 11 January 2011 10:44, Sam <u2nsam at gmail.com> wrote:
>>>>>> >> > A query,
>>>>>> >> >
>>>>>> >> > I have 2 FS running on one server on 2 different ips,
>>>>>> >> > so when i do fs_cli going to respective bins , i see console of
>>>>>> only the
>>>>>> >> > first server.
>>>>>> >> >
>>>>>> >> > Is there any way to get the console of both the FS on the same
>>>>>> server .
>>>>>> >> > I tried changing the port of event socket to 8022 but it donot
>>>>>> works.
>>>>>> >> > <param name="listen-port" value="8022"/>
>>>>>> >> >
>>>>>> >> > Is there some method to start the console of both the instances.
>>>>>> >> >
>>>>>> >> > Regds
>>>>>> >> > Sam
>>>>>> >> >
>>>>>> >> > _______________________________________________
>>>>>> >> > FreeSWITCH-users mailing list
>>>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >> > UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> >> > http://www.freeswitch.org
>>>>>> >> >
>>>>>> >> >
>>>>>> >>
>>>>>> >> _______________________________________________
>>>>>> >> FreeSWITCH-users mailing list
>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> >> http://www.freeswitch.org
>>>>>> >
>>>>>> >
>>>>>> > _______________________________________________
>>>>>> > FreeSWITCH-users mailing list
>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> > UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> > http://www.freeswitch.org
>>>>>> >
>>>>>> >
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Wed Jan 12 09:12:20 2011
From: u2nsam at gmail.com (Sam)
Date: Wed, 12 Jan 2011 11:42:20 +0530
Subject: [Freeswitch-users] deflect
Message-ID: <AANLkTiksYLOeki2_XDJC5YwMW6hjUxvdpT7-Mvg7vmKZ@mail.gmail.com>

Hi,

When call comes on 1 server and plays an application and after execution of
the
application the call is bridge to the other server ,but here after bridging
the call
should refer/deflect to other server, how this can be done ?

Here just using the deflect variable is not recommended as there is proxy in
between,
so once the call is bridge the next step would be deflect the leg totally to
another server via proxy.

Regards
Sam
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From curriegrad2004 at gmail.com  Wed Jan 12 09:44:46 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 11 Jan 2011 22:44:46 -0800
Subject: [Freeswitch-users] QoS in FreeSwitch
In-Reply-To: <AANLkTi=zCZq6eXPJwM7d4waGKrcN3yzBf0NUL+WzUiTO@mail.gmail.com>
References: <AANLkTimCm7wwLBePiTtTDY7DLBshW_-HkRwhiStQ9TsX@mail.gmail.com>
	<AANLkTi=zCZq6eXPJwM7d4waGKrcN3yzBf0NUL+WzUiTO@mail.gmail.com>
Message-ID: <AANLkTikTd2ygfqeJbdVhx9PUArawdKe7k87LCveBmvHE@mail.gmail.com>

Is it possible to get tc to prioritize packets using DSCP anyways?
I've been attempting to do this with tc but so far I haven't been
getting too much success out of it

On Tue, Jan 11, 2011 at 10:04 AM, Norman Tomlins <norm at voicenetwork.ca> wrote:
> Sergii,
>
> You could use IP tables to set the QOS on the RTP packets.
>
> iptables -t mangle -A OUTPUT -p tcp --sport 5060 -j DSCP
> --set-dscp-class cs3 # mark SIP TCP packets with CS3
> iptables -t mangle -A OUTPUT -p tcp --sport 5061 -j DSCP
> --set-dscp-class cs3 # mark SIP TLS packets with CS3
> iptables -t mangle -A OUTPUT -p udp -m udp --sport 16384:32767 -j DSCP
> --set-dscp-class ef # mark RTP packets with EF
>
>
> Norman Tomlins
> Voice Network Inc.
>
>
> On Tue, Jan 11, 2011 at 6:44 AM, Sergii Iefimov <efimserg at gmail.com> wrote:
>> What kind of the QoS (Quality of Service)do you have in FreeSwitch
>> (DiffServ, IntServ or other)?
>>
>> Does FreeSwitch offer the ability to mark the voice data with the proper
>> tags so that our switch can prioritize the data through our network, or if
>> it can't what hard ware device would you suggest to do this?
>>
>> Thanks!
>>
>> Best Regards,
>>
>> Sergii Iefimov
>>
>> Mobile phone:???? ?????????? +380 (68) 361 4627
>> E-mail:???????????????? ?????????efimserg at gmail.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From thomas at chaschperli.ch  Wed Jan 12 10:05:46 2011
From: thomas at chaschperli.ch (Thomas Mueller)
Date: Wed, 12 Jan 2011 08:05:46 +0100
Subject: [Freeswitch-users] QoS in FreeSwitch
In-Reply-To: <AANLkTikTd2ygfqeJbdVhx9PUArawdKe7k87LCveBmvHE@mail.gmail.com>
References: <AANLkTimCm7wwLBePiTtTDY7DLBshW_-HkRwhiStQ9TsX@mail.gmail.com>	<AANLkTi=zCZq6eXPJwM7d4waGKrcN3yzBf0NUL+WzUiTO@mail.gmail.com>
	<AANLkTikTd2ygfqeJbdVhx9PUArawdKe7k87LCveBmvHE@mail.gmail.com>
Message-ID: <4D2D52CA.5000201@chaschperli.ch>

On 12.01.2011 07:44, curriegrad2004 wrote:
> Is it possible to get tc to prioritize packets using DSCP anyways?
> I've been attempting to do this with tc but so far I haven't been
> getting too much success out of it
maybe try some sort of iptables/tc frontend like shorewall? shorewall 
has traffic-shaping/paketmarking capabilities. http://shorewall.net/

- Thomas


From tayeb.meftah at gmail.com  Wed Jan 12 11:36:42 2011
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Wed, 12 Jan 2011 09:36:42 +0100
Subject: [Freeswitch-users] mod_db limit and group
In-Reply-To: <5D26741A84A541D2A9D7AB130EB93086@e1705>
References: <5D26741A84A541D2A9D7AB130EB93086@e1705>
Message-ID: <4D2D681A.70201@gmail.com>

http://wiki.freeswitch.org/wiki/limit
see default dialplan have ton of examples
Le 12/01/2011 05:48, Madovsky a ?crit :
> Is there any example of limit and group use ?
> I don't understand how to use it.
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313

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From babak.freeswitch at gmail.com  Wed Jan 12 11:53:09 2011
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Wed, 12 Jan 2011 12:23:09 +0330
Subject: [Freeswitch-users] spandsp t38
Message-ID: <AANLkTinBfHkLZZqsGU2RmaY_aGGysxSokNBYvKbG2DuF@mail.gmail.com>

Hi
I'm trying to send a fax through a cisco gateway like : fs -> cisco gateway
-> pstn -> fax machine
faxes are being sent fine (with some errors ) but there is a reinvite codec
error (seems t38 fails):
http://pastebin.freeswitch.org/14996

and this is the command I'm executing
originate
{origination_uuid=...,fax_ident=...,fax_header=...,fax_enable_t38=true,fax_enable_t38_request=true}sofia/external/...@(cisco
ip) &txfax(...)
shoud I do anything else to use t38?
thanx
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From lists at infosecurity.ch  Wed Jan 12 12:40:20 2011
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Wed, 12 Jan 2011 10:40:20 +0100
Subject: [Freeswitch-users] ZORG,
	new C++ and Java ZRTP implementation public release
Message-ID: <4D2D7704.2060703@infosecurity.ch>

Milan, January 12th 2011  - PrivateWave Italia S.p.A, italian company
engaged in developing technologies for privacy protection and
information security in voice telecommunications, is pleased to announce
the release of ZORG, a new open source ZRTP protocol implementation
available for download from http://www.zrtp.org  .

ZRTP [1] provides end-to-end key exchange with Elliptic Curve
Diffie-Hellmann 384bit and AES-256 SRTP encryption .

ZORG has been originally developed and implemented in PrivateWave's
PrivateGSM voice encryption products available for the following
platforms: Blackberry, Nokia and iOS (iPhone) .

Zorg C++ has been integrated with PJSIP open source VoIP SDK [2] and
it's provided as integration patch against PJSIP 1.8.5. It has been
tested on iPhone, Symbian, Windows, Linux and Mac OS X.

Zorg Java has been integrated within a custom version of MJSIP [3] open
source SDK on Blackberry platform and it includes memory usage
optimizations required to reduce at minimum garbage collector activity.

Both platforms have separated and modular cryptographic back-ends so
that the cryptographic algorithms implementation could be easily swapped
with other ones.

ZORG is licensed under GNU AGPL and source code is available on github
athttps://github.com/privatewave/ZORG.

We are releasing it under open source and in coherence with our approach
to security [4] as we really hope that it can be useful for the open
source ecosystem to create new voice encryption systems in support of
freedom of speech.

More than 20 pjsip-based open source VoIP encryption software (several
written in Java) could directly benefit from ZORG release.

We would be happy to receive proposal of cooperation, new integration,
new cryptographic back-ends, bug scouting and whatever useful to improve
and let ZRTP affirm as voice encryption standard.

Zorg is available from http://www.zrtp.org .

[1] ZRTP: http://http://en.wikipedia.org/wiki/ZRTP
[2] PJSIP: http://www.pjsip.org
[3] MJSIP: http://www.mjsip.org
[4] Security approach:http://www.privatewave.com/security/approch.html
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From bggoutham at gmail.com  Wed Jan 12 14:26:40 2011
From: bggoutham at gmail.com (Goutham BG)
Date: Wed, 12 Jan 2011 16:56:40 +0530
Subject: [Freeswitch-users] Peculiar behavior in FreeSWITCH when media
 streams for SRTP and RTP are offered in the same SDP
Message-ID: <AANLkTim6a7=dq8HA7pVPDjs2kJDJtM_HEy8SYQ9ariA9@mail.gmail.com>

Hi,

I am observing a peculiar behavior in FreeSWITCH-1.0.7 with SRTP.

I have the following entry in my dialplan XML file:

  <extension name="IVR">
    <condition field="destination_number" expression="^IVR$">
      <action application="set" data="sip_secure_media=true"/>
      <action application="bridge" data="loopback/app=socket:
47.152.232.156:8084 async full"/>
    </condition>
  </extension>

A Polycom SoundPoint IP 550 configured in "SRTP best effort" mode dials into
this extension and is connected to the IVR. In this mode, the phone offers
two media streams in the SDP of INVITE; 1st one for SRTP and the 2nd one for
RTP. But the problem is that the media is established in SRTP in one way and
RTP in the other way.

The phone offers the following SDP in the INVITE message:

v=0
o=- 1167766638 1167766638 IN IP4 47.152.232.149
s=Polycom IP Phone
c=IN IP4 47.152.232.149
t=0 0
a=sendrecv
m=audio 5040 RTP/SAVP 9 0 8 18 127
a=crypto:8 AES_CM_128_HMAC_SHA1_32
inline:svWhF9Uh0VonfhtmuvRLM4B9S6+ZEicc3hDd3dAQ
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000
m=audio 5040 RTP/AVP 9 0 8 18 127
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000

As we can see, there are two "m=" lines in the SDP of the offer; the 1st one
for SRTP and the other one for RTP. FreeSWITCH-1.0.7 answers the call by
sending 200OK with the following SDP:

v=0
o=FreeSWITCH 1294817064 1294817065 IN IP4 47.152.232.156
s=FreeSWITCH
c=IN IP4 47.152.232.156
t=0 0
m=audio 11552 RTP/SAVP 9 127
a=rtpmap:9 G722/8000
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=crypto:8 AES_CM_128_HMAC_SHA1_32
inline:z5/m/v2U0negVFjfkCsXR/sNAo9fhBv+dspBIbtx
m=audio 0 RTP/AVP 19

As we can see above, FreeSWITCH accepts the SRTP stream and rejects the RTP
stream (by sending port as 0) in the SDP. The Polycom phone sends the media
in SRTP as expected. But, FreeSWITCH sends the media in RTP to the phone
even though it accepted SRTP in the answer (200OK).

Please let me know if this is a bug or am I missing something here?

The debug log of the call coming into FreeSWITCH is pasted here
http://pastebin.freeswitch.org/14998 . If required I can also send the
wireshark traces of this scenario captured in the system where FreeSITCH is
running.

Thanks
Goutham B G
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From gmaruzz at gmail.com  Wed Jan 12 16:53:01 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 12 Jan 2011 14:53:01 +0100
Subject: [Freeswitch-users] ZORG,
 new C++ and Java ZRTP implementation public release
In-Reply-To: <4D2D7704.2060703@infosecurity.ch>
References: <4D2D7704.2060703@infosecurity.ch>
Message-ID: <AANLkTinZkyNREco5YROXNOMWr-3+10iZstdtd0=mKhi4@mail.gmail.com>

Hi Fabio,

thanks for your company's very valuable contribution!

IANAL, and don't know how the Affero version of GPL is different from
"normal" GPL.

Maybe having double licensing (like AGPL and BSD) would help spreading
the usage of ZORG.

Anyway, yay for the feat!

-giovanni

On Wed, Jan 12, 2011 at 10:40 AM, Fabio Pietrosanti (naif)
<lists at infosecurity.ch> wrote:
> Milan, January 12th 2011 ?- PrivateWave Italia S.p.A, italian company
> engaged in developing technologies for privacy protection and information
> security in voice telecommunications, is pleased to announce the release of
> ZORG, a new open source ZRTP protocol implementation available for download
> from http://www.zrtp.org ?.
>
> ZRTP [1] provides end-to-end key exchange with Elliptic Curve
> Diffie-Hellmann 384bit and AES-256 SRTP encryption .
>
> ZORG has been originally developed and implemented in PrivateWave's
> PrivateGSM voice encryption products available for the following platforms:
> Blackberry, Nokia and iOS (iPhone) .
>
> Zorg C++ has been integrated with PJSIP open source VoIP SDK [2] and it's
> provided as integration patch against PJSIP 1.8.5. It has been tested on
> iPhone, Symbian, Windows, Linux and Mac OS X.
>
> Zorg Java has been integrated within a custom version of MJSIP [3] open
> source SDK on Blackberry platform and it includes memory usage optimizations
> required to reduce at minimum garbage collector activity.
>
> Both platforms have separated and modular cryptographic back-ends so that
> the cryptographic algorithms implementation could be easily swapped with
> other ones.
>
> ZORG is licensed under GNU AGPL and source code is available on github at
> https://github.com/privatewave/ZORG .
>
> We are releasing it under open source and in coherence with our approach to
> security [4] as we really hope that it can be useful for the open source
> ecosystem to create new voice encryption systems in support of freedom of
> speech.
>
> More than 20 pjsip-based open source VoIP encryption software (several
> written in Java) could directly benefit from ZORG release.
>
> We would be happy to receive proposal of cooperation, new integration, new
> cryptographic back-ends, bug scouting and whatever useful to improve and let
> ZRTP affirm as voice encryption standard.
>
> Zorg is available from http://www.zrtp.org .
>
> [1] ZRTP: http://http://en.wikipedia.org/wiki/ZRTP
> [2] PJSIP: http://www.pjsip.org
> [3] MJSIP: http://www.mjsip.org
> [4] Security approach: http://www.privatewave.com/security/approch.html
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gchen00 at insightbb.com  Wed Jan 12 17:51:01 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Wed, 12 Jan 2011 09:51:01 -0500
Subject: [Freeswitch-users] Possible Freeswitch bug with Cisco 7960
Message-ID: <fc78bb9624e42.4d2d7985@insightbb.com>

FreeSWITCH Version 1.0.head (git-9350fb9 2010-12-07 00-20-07 -0600)

I just install freeswitch and have two cisco 7960 phones (firmware sip 7.9) registered.
I can call out to our service provider or make 7960 call each other. They all work fine except this:
If I call into freeswitch echo test extension or check voicemail ?using one of Cisco 7960, the call will be droped after 30 seconds. By looking at SIP messages, it appears that After FS send out 200 OK to Cisco 7960, it never received ACK sip messages back and timeout. Here is the debug info in Cisco 7960:
[11:56:30:41354610] SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.xxx.144:5060;branch=z9hG4bK2be0821e
From: "Line1" <sip:1007 at xxx.xxx.xxx.177>;tag=00082166efcb015d72e42433-5254ac8f
To: <sip:9196 at xxx.xxx.xxx.177;user=phone>;tag=p9H5Qy5r993mB
Call-ID: 00082166-efcb0007-04fe7b08-3c126cb0 at xxx.xxx.xxx.144
CSeq: 102 INVITE
Contact: <sip:9196 at xxx.xxx.xxx.177:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9350fb9 2010-12-07 00-20-07 -0600
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 253
Remote-Party-ID: "9196" <sip:9196 at xxx.xxx.xxx.177>;party=calling;privacy=off;screen=no

v=0
o=FreeSWITCH 1294739962 1294739963 IN IP4 xxx.xxx.xxx.177
s=FreeSWITCH
c=IN IP4 xxx.xxx.xxx.177
t=0 0
m=audio 24992 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
[11:56:30:41354614] SIPTaskProcessSIPMessage: Line filter: Determining destination line...
[11:56:30:41354615] sip_sm_determine_ccb: Matched to_tag
[11:56:30:41354616] sip_sm_ccb_match_branch_cseq: Method index not found
[11:56:30:41354616] SIPTaskProcessSIPMessage: Error: sip_sm_determine_ccb(): bad response. Dropping message.
[11:56:34:41355009] SIPTaskProcessListEvent: cmd = 0x160200
[11:56:34:41355009] SIPProcessUDPMessage: recv UDP message from <xxx.xxx.xxx.177>:<50195>, length=<1185>, message=


Looks like cisco 7960 having trouble to process 200 OK message from FS.
Is this a possible FS bug? Does anybody know how to fix it?
By searching on the WEB, I can see that Asterisk 1.6 also has the same problem in some revisions and been fixed.

Gary


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From brian at freeswitch.org  Wed Jan 12 18:11:10 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Jan 2011 09:11:10 -0600
Subject: [Freeswitch-users] ZORG,
	new C++ and Java ZRTP implementation public release
In-Reply-To: <AANLkTinZkyNREco5YROXNOMWr-3+10iZstdtd0=mKhi4@mail.gmail.com>
References: <4D2D7704.2060703@infosecurity.ch>
	<AANLkTinZkyNREco5YROXNOMWr-3+10iZstdtd0=mKhi4@mail.gmail.com>
Message-ID: <935C4C6C-FB5B-46DA-B1EA-73ECBE875EBB@freeswitch.org>

AGPL is the most evil license you can ever pick for anything... its the one that reaches over the socket and steals your code and soul.

/b

On Jan 12, 2011, at 7:53 AM, Giovanni Maruzzelli wrote:

> Hi Fabio,
> 
> thanks for your company's very valuable contribution!
> 
> IANAL, and don't know how the Affero version of GPL is different from
> "normal" GPL.
> 
> Maybe having double licensing (like AGPL and BSD) would help spreading
> the usage of ZORG.
> 
> Anyway, yay for the feat!
> 
> -giovanni



From ayhkor at gmail.com  Wed Jan 12 07:39:41 2011
From: ayhkor at gmail.com (deniro)
Date: Tue, 11 Jan 2011 23:39:41 -0500
Subject: [Freeswitch-users] conference pin
In-Reply-To: <AANLkTinRMuz2zuspEvp9ZaDugpA5G6GGm0n2HKuMTaiF@mail.gmail.com>
References: <AANLkTi=q5xpnW7HdGVZYpQbUNi2m=dvmFXX6TTKGvxZo@mail.gmail.com>
	<AANLkTin3F-TXJp_Mv-8vwGbkLmAwbi9mE2bobiB2uyQx@mail.gmail.com>
	<AANLkTimSRbCLU_+F3TfR7eK9NXTx=wiVUnVJ-7-yxNqv@mail.gmail.com>
	<AANLkTinRMuz2zuspEvp9ZaDugpA5G6GGm0n2HKuMTaiF@mail.gmail.com>
Message-ID: <AANLkTinObMT4VRwWF7snCrj+XktwJjb9OXTEmLLSYdaw@mail.gmail.com>

Mike
It is working. Thanks again.

On Tue, Jan 11, 2011 at 12:03 AM, Michael Collins <msc at freeswitch.org>wrote:

> Did the example I gave you not work? (http://pastebin.freeswitch.org/14979)
>
> -MC
>
>
> On Tue, Jan 11, 2011 at 12:09 AM, deniro <ayhkor at gmail.com> wrote:
>
>> Hi
>> any expamples of how to store asked pin (phone call) by channel var and
>> re-use it in conferencing to go to that conference?
>> thx
>>
>>
>> On Mon, Jan 10, 2011 at 2:09 AM, Sam <u2nsam at gmail.com> wrote:
>>
>>> use channels variables in freeswitch.
>>>
>>>
>>> http://wiki.freeswitch.org/wiki/Channel_Variables#Channel_Variable_Manipulation
>>>
>>> Regds
>>> Sam
>>>
>>>
>>>
>>>   On Mon, Jan 10, 2011 at 9:35 AM, deniro <ayhkor at gmail.com> wrote:
>>>
>>>>   Hi
>>>> using conferencing software and with the phone dialing,
>>>> entering pin number it will go to a conference  identified by pin
>>>> in its default format  it is "conference at profile+pin"
>>>> in my case it will be "pin at profile+pin" since conference=pin.
>>>> how do I do that? how do I provide a pin that takes me to conference
>>>> which is
>>>> identified by pin?
>>>> thx
>>>> deniro--
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From spambox at haruhiism.net  Wed Jan 12 08:30:08 2011
From: spambox at haruhiism.net (Kamigishi Rei)
Date: Wed, 12 Jan 2011 08:30:08 +0300
Subject: [Freeswitch-users] Profile configuration
Message-ID: <4D2D3C60.607@haruhiism.net>

Hello,

I have the following question regarding FS profile setup: is it
considered normal to only have one profile at all? (The server is not
behind a NAT/firewall; LAN users have direct routes to it.)

From what I gather from the sample configuration, it's pretty much
possible to distinguish between authenticated ("our own") users and
external SIP requests via the dialplan.
Quoting,

    <extension name="check_auth" continue="true">
      <condition field="${sip_authorized}" expression="^true$"
break="never">
        <anti-action application="respond" data="407"/>
      </condition>
    </extension>

    <extension name="transfer_to_default">
      <condition>
        <action application="transfer" data="${destination_number} XML
default"/>
      </condition>
    </extension>

Basically, I'd like FS to accept all calls (internal and external alike)
on 5060, and use 5070 for external providers we register with.
To get the "internal" profile to work like that on 5060, I have to
comment the "apply-inbound-acl" setting (sip_profiles/internal.xml), and
disable auth checking (internal_auth_calls=false in vars.xml).

Would that be the correct solution, or is that somehow considered
"insecure"? According to internal.xml, all non-authenticated (without
user_context defined) calls fall into the public context anyway, and
having an "is the user authenticated?" check in the public context
allows us to transfer the call to the correct dialplan?so what are the
security risks then?

Speaking of which, is there a point in having user_context defined for
users who need the default context, if they can just be redirected to
that context via check_auth extension of the public context?

Thanks in advance.

-- 
Kamigishi Rei
KREI-RIPE


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From vermeulen.deon at gmail.com  Wed Jan 12 17:48:11 2011
From: vermeulen.deon at gmail.com (Deon Vermeulen)
Date: Wed, 12 Jan 2011 16:48:11 +0200
Subject: [Freeswitch-users] Freeswitch eLearning
Message-ID: <71CACB8F-D968-4085-B8C0-B0AB655EFDD4@gmail.com>

Hi


Does anyone know of an online training course for Freeswitch?


Thank you very much

Kind Regards
Deon


From kris at kriskinc.com  Wed Jan 12 18:51:45 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 12 Jan 2011 10:51:45 -0500
Subject: [Freeswitch-users] ZORG,
 new C++ and Java ZRTP implementation public release
In-Reply-To: <935C4C6C-FB5B-46DA-B1EA-73ECBE875EBB@freeswitch.org>
References: <4D2D7704.2060703@infosecurity.ch>
	<AANLkTinZkyNREco5YROXNOMWr-3+10iZstdtd0=mKhi4@mail.gmail.com>
	<935C4C6C-FB5B-46DA-B1EA-73ECBE875EBB@freeswitch.org>
Message-ID: <AANLkTinNZKF6AN1BxSXm2hVkgiZPowKu8Z+5sPzSE6d9@mail.gmail.com>

Brian,

  Not really...

  "IANAL" but the AGPL is virtually identical to the GPL except that
it eliminates the "ASP loophole".

  Let's pretend FreeSWITCH was licensed under the GPL.  Let's say I
make my own modifications to FreeSWITCH and provide network services
to my customers.  Under the GPL I am NOT required to provide these
source changes to anyone because I'm not distributing a binary - I'm
providing network services to customers.  The binary runs in my
datacenter (or wherever); it's not being "distributed".

  If you're providing "network services" to customers and not
distributing software using a GPL license it may as well be BSD or
something else with a weak copyleft.

  Under the AGPL providing network services to customers is
essentially equal to distributing binaries under the GPL.  In my
hypothetical situation above if FreeSWITCH were licensed under the
AGPL I would be required to distribute my  source changes to my
customers even though I never provided them with a binary.

  Of course (under both licenses) once I distribute the binary and
source to my customers I might as well distribute it to the rest of
the world because are free to do so anyway...

On Wed, Jan 12, 2011 at 10:11 AM, Brian West <brian at freeswitch.org> wrote:
> AGPL is the most evil license you can ever pick for anything... its the one that reaches over the socket and steals your code and soul.
>
> /b
>

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


From kris at kriskinc.com  Wed Jan 12 18:54:17 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 12 Jan 2011 10:54:17 -0500
Subject: [Freeswitch-users] ZORG,
 new C++ and Java ZRTP implementation public release
In-Reply-To: <AANLkTinNZKF6AN1BxSXm2hVkgiZPowKu8Z+5sPzSE6d9@mail.gmail.com>
References: <4D2D7704.2060703@infosecurity.ch>
	<AANLkTinZkyNREco5YROXNOMWr-3+10iZstdtd0=mKhi4@mail.gmail.com>
	<935C4C6C-FB5B-46DA-B1EA-73ECBE875EBB@freeswitch.org>
	<AANLkTinNZKF6AN1BxSXm2hVkgiZPowKu8Z+5sPzSE6d9@mail.gmail.com>
Message-ID: <AANLkTim9EUO0+BnDix9zn0T2T_7HRngHb0KPjXTsmYDv@mail.gmail.com>

While I'm not a lawyer the guy that wrote this article (Paul Arne) is
our open source license specialist and him and I have talked about the
AGPL at length (as attorneys usually do):

http://www.mmmtechlaw.com/2010/12/14/gnu-affero-general-public-license-risks-and-opportunities/

On Wed, Jan 12, 2011 at 10:51 AM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> Brian,
>
> ?Not really...
>
> ?"IANAL" but the AGPL is virtually identical to the GPL except that
> it eliminates the "ASP loophole".
>
> ?Let's pretend FreeSWITCH was licensed under the GPL. ?Let's say I
> make my own modifications to FreeSWITCH and provide network services
> to my customers. ?Under the GPL I am NOT required to provide these
> source changes to anyone because I'm not distributing a binary - I'm
> providing network services to customers. ?The binary runs in my
> datacenter (or wherever); it's not being "distributed".
>
> ?If you're providing "network services" to customers and not
> distributing software using a GPL license it may as well be BSD or
> something else with a weak copyleft.
>
> ?Under the AGPL providing network services to customers is
> essentially equal to distributing binaries under the GPL. ?In my
> hypothetical situation above if FreeSWITCH were licensed under the
> AGPL I would be required to distribute my ?source changes to my
> customers even though I never provided them with a binary.
>
> ?Of course (under both licenses) once I distribute the binary and
> source to my customers I might as well distribute it to the rest of
> the world because are free to do so anyway...
>
> On Wed, Jan 12, 2011 at 10:11 AM, Brian West <brian at freeswitch.org> wrote:
>> AGPL is the most evil license you can ever pick for anything... its the one that reaches over the socket and steals your code and soul.
>>
>> /b
>>



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


From peder at networkoblivion.com  Wed Jan 12 19:11:32 2011
From: peder at networkoblivion.com (Peder)
Date: Wed, 12 Jan 2011 10:11:32 -0600
Subject: [Freeswitch-users] Possible Freeswitch bug with Cisco 7960
In-Reply-To: <fc78bb9624e42.4d2d7985@insightbb.com>
References: <fc78bb9624e42.4d2d7985@insightbb.com>
Message-ID: <005301cbb273$60d7ce10$22876a30$@com>

I'd say it is much more likely a Cisco 7960 bug.  Cisco's SIP software is
total crap.  It's too bad because their phones are physically very nice and
very reliable.  I'd verify what release you are on as a search of Cisco's
site doesn't show a 7.9.  It shows up to 7.5.  8 goes all the way to 8.9.
In * land, we never went past 7.4 as anything after that had issues.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gary
Chen
Sent: Wednesday, January 12, 2011 8:51 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Possible Freeswitch bug with Cisco 7960

 

FreeSWITCH Version 1.0.head (git-9350fb9 2010-12-07 00-20-07 -0600)

 

I just install freeswitch and have two cisco 7960 phones (firmware sip 7.9)
registered.

I can call out to our service provider or make 7960 call each other. They
all work fine except this:

If I call into freeswitch echo test extension or check voicemail  using one
of Cisco 7960, the call will be droped after 30 seconds. By looking at SIP
messages, it appears that After FS send out 200 OK to Cisco 7960, it never
received ACK sip messages back and timeout. Here is the debug info in Cisco
7960:

[11:56:30:41354610] SIP/2.0 200 OK

Via: SIP/2.0/UDP xxx.xxx.xxx.144:5060;branch=z9hG4bK2be0821e

From: "Line1"
<sip:1007 at xxx.xxx.xxx.177>;tag=00082166efcb015d72e42433-5254ac8f

To: <sip:9196 at xxx.xxx.xxx.177;user=phone>;tag=p9H5Qy5r993mB

Call-ID: 00082166-efcb0007-04fe7b08-3c126cb0 at xxx.xxx.xxx.144

CSeq: 102 INVITE

Contact: <sip:9196 at xxx.xxx.xxx.177:5060;transport=udp>

User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9350fb9 2010-12-07 00-20-07
-0600

Accept: application/sdp

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE

Supported: timer, precondition, path, replaces

Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer

Content-Type: application/sdp

Content-Disposition: session

Content-Length: 253

Remote-Party-ID: "9196"
<sip:9196 at xxx.xxx.xxx.177>;party=calling;privacy=off;screen=no

 

v=0

o=FreeSWITCH 1294739962 1294739963 IN IP4 xxx.xxx.xxx.177

s=FreeSWITCH

c=IN IP4 xxx.xxx.xxx.177

t=0 0

m=audio 24992 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

[11:56:30:41354614] SIPTaskProcessSIPMessage: Line filter: Determining
destination line...

[11:56:30:41354615] sip_sm_determine_ccb: Matched to_tag

[11:56:30:41354616] sip_sm_ccb_match_branch_cseq: Method index not found

[11:56:30:41354616] SIPTaskProcessSIPMessage: Error: sip_sm_determine_ccb():
bad response. Dropping message.

[11:56:34:41355009] SIPTaskProcessListEvent: cmd = 0x160200

[11:56:34:41355009] SIPProcessUDPMessage: recv UDP message from
<xxx.xxx.xxx.177>:<50195>, length=<1185>, message=

 

 

Looks like cisco 7960 having trouble to process 200 OK message from FS.

Is this a possible FS bug? Does anybody know how to fix it?

By searching on the WEB, I can see that Asterisk 1.6 also has the same
problem in some revisions and been fixed.

 

Gary

 

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From infos at madovsky.org  Wed Jan 12 19:16:15 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 12 Jan 2011 11:16:15 -0500
Subject: [Freeswitch-users] mod_db limit and group
Message-ID: <4E72FEF5D4B74EDD8D976371206246EA@e1705>

Just corrected wiki page from

Insert a group entry:
 <action application="db" data="insert:groupname:sipurl"/>
Delete a group entry:
 <action application="db" data="delete:groupname:sipurl"/>

to

Insert a group entry:
 <action application="group" data="insert:groupname:sipurl"/>
Delete a group entry:
 <action application="group" data="delete:groupname:sipurl"/>
  ----- Original Message ----- 
  From: Madovsky 
  To: Meftah Tayeb 
  Sent: Wednesday, January 12, 2011 10:31 AM
  Subject: Re: [Freeswitch-users] mod_db limit and group


  ha ok mod_db and limit seems to work together now..
    ----- Original Message ----- 
    From: Meftah Tayeb 
    To: FreeSWITCH Users Help 
    Cc: Madovsky 
    Sent: Wednesday, January 12, 2011 3:36 AM
    Subject: Re: [Freeswitch-users] mod_db limit and group


    http://wiki.freeswitch.org/wiki/limit
    see default dialplan have ton of examples
    Le 12/01/2011 05:48, Madovsky a ?crit : 

      Is there any example of limit and group use ?
      I don't understand how to use it.

      Thanks

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
  


-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313
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From gchen00 at insightbb.com  Wed Jan 12 19:29:35 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Wed, 12 Jan 2011 11:29:35 -0500
Subject: [Freeswitch-users] Possible Freeswitch bug with Cisco 7960
Message-ID: <fc4fe9ce264ed.4d2d909f@insightbb.com>

Sorry my mistake. It is Version 8.9 on Cisco 7960 ( POS3-08-9-00).

Gary

?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peder
Sent: Wednesday, January 12, 2011 11:12 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] Possible Freeswitch bug with Cisco 7960

?

I?d say it is much more likely a Cisco 7960 bug.? Cisco?s SIP software is total crap.? It?s too bad because their phones are physically very nice and very reliable.? I?d verify what release you are on as a search of Cisco?s site doesn?t show a 7.9.? It shows up to 7.5.? 8 goes all the way to 8.9.? In * land, we never went past 7.4 as anything after that had issues.

?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gary Chen
Sent: Wednesday, January 12, 2011 8:51 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Possible Freeswitch bug with Cisco 7960

?

FreeSWITCH Version 1.0.head (git-9350fb9 2010-12-07 00-20-07 -0600)

?

I just install freeswitch and have two cisco 7960 phones (firmware sip 7.9) registered.

I can call out to our service provider or make 7960 call each other. They all work fine except this:

If I call into freeswitch echo test extension or check voicemail ?using one of Cisco 7960, the call will be droped after 30 seconds. By looking at SIP messages, it appears that After FS send out 200 OK to Cisco 7960, it never received ACK sip messages back and timeout. Here is the debug info in Cisco 7960:

[11:56:30:41354610] SIP/2.0 200 OK

Via: SIP/2.0/UDP xxx.xxx.xxx.144:5060;branch=z9hG4bK2be0821e

From: "Line1" <sip:1007 at xxx.xxx.xxx.177>;tag=00082166efcb015d72e42433-5254ac8f

To: <sip:9196 at xxx.xxx.xxx.177;user=phone>;tag=p9H5Qy5r993mB

Call-ID: 00082166-efcb0007-04fe7b08-3c126cb0 at xxx.xxx.xxx.144

CSeq: 102 INVITE

Contact: <sip:9196 at xxx.xxx.xxx.177:5060;transport=udp>

User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9350fb9 2010-12-07 00-20-07 -0600

Accept: application/sdp

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

Supported: timer, precondition, path, replaces

Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer

Content-Type: application/sdp

Content-Disposition: session

Content-Length: 253

Remote-Party-ID: "9196" <sip:9196 at xxx.xxx.xxx.177>;party=calling;privacy=off;screen=no

?

v=0

o=FreeSWITCH 1294739962 1294739963 IN IP4 xxx.xxx.xxx.177

s=FreeSWITCH

c=IN IP4 xxx.xxx.xxx.177

t=0 0

m=audio 24992 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

[11:56:30:41354614] SIPTaskProcessSIPMessage: Line filter: Determining destination line...

[11:56:30:41354615] sip_sm_determine_ccb: Matched to_tag

[11:56:30:41354616] sip_sm_ccb_match_branch_cseq: Method index not found

[11:56:30:41354616] SIPTaskProcessSIPMessage: Error: sip_sm_determine_ccb(): bad response. Dropping message.

[11:56:34:41355009] SIPTaskProcessListEvent: cmd = 0x160200

[11:56:34:41355009] SIPProcessUDPMessage: recv UDP message from <xxx.xxx.xxx.177>:<50195>, length=<1185>, message=

?

?

Looks like cisco 7960 having trouble to process 200 OK message from FS.

Is this a possible FS bug? Does anybody know how to fix it?

By searching on the WEB, I can see that Asterisk 1.6 also has the same problem in some revisions and been fixed.

?

Gary

?
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From brian at freeswitch.org  Wed Jan 12 19:42:56 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Jan 2011 10:42:56 -0600
Subject: [Freeswitch-users] Possible Freeswitch bug with Cisco 7960
In-Reply-To: <fc4fe9ce264ed.4d2d909f@insightbb.com>
References: <fc4fe9ce264ed.4d2d909f@insightbb.com>
Message-ID: <6E1A6643-F88E-40B1-8157-B6995708BF54@freeswitch.org>

Where there is your problem... it clearly tells you that in the firmware verison.. POS  :P  No really it should work as long as no nat is involved and force-rport is not set.

/b

On Jan 12, 2011, at 10:29 AM, Gary Chen wrote:

> Sorry my mistake. It is Version 8.9 on Cisco 7960 ( POS3-08-9-00).
> 
> Gary
> 

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From infos at madovsky.org  Wed Jan 12 19:52:52 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 12 Jan 2011 11:52:52 -0500
Subject: [Freeswitch-users] mod_conference end
Message-ID: <0A6E4395607344F08F8300BB1CDCD840@e1705>

How can a xml dialplan guess that a conference is ended ?
I mean the last participant left ?

Thanks
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From msc at freeswitch.org  Wed Jan 12 20:27:44 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 Jan 2011 17:27:44 +0000
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <AANLkTikh8L61qAx8TrWkntCWJ_xc9d=q5xag7xPb2-wF@mail.gmail.com>

Hello all,

We have a light agenda today:

http://wiki.freeswitch.org/wiki/FS_weekly_2011_01_12

Bring your questions and ideas and be ready to share!

-MC
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From msc at freeswitch.org  Wed Jan 12 21:15:16 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 Jan 2011 18:15:16 +0000
Subject: [Freeswitch-users] Vestec Speech Engine: ASR 2.1 Release
In-Reply-To: <4D27665F.4000406@vestec.com>
References: <4D27665F.4000406@vestec.com>
Message-ID: <AANLkTimp3Liu-fQzD=oKsFgM9VCMXD_ORaxpegweF=Y7@mail.gmail.com>

Kashif,

Would you or one of your engineers be willing to join us on our Wednesday
conference call and do a brief presentation on Vestec? If so, please contact
me off list and we'll get you scheduled.

Thanks!
-MC

On Fri, Jan 7, 2011 at 7:15 PM, Kashif Kahn <kahn at vestec.com> wrote:

> Dear All,
>
> We have launched a major upgrade to our ASR engine that offers the best
> deal around for enabling speech recognition with "command and control"
> type IVR applications. The new architecture boasts a number of
> advancements over version 1.1, including:
>
> - Improved US English acoustic model
> - DTMF recognition
> - MRCP support v1 and v2
> - Highly scalable architecture and Redundancy
> - C++ API availability
> - SRGS-XML (.grxml) grammar support
> - Improved logging
> - Bug Fixes
>
> The engine can be integrated with different contact center and soft-PBX
> platforms (such as Freeswitch) using MRCP interface.
>
> A starter kit comprising a specially priced full-function engine is
> available for $25 while a regular one channel (ie. port) license can be
> purchased for $99. Please visit Vestec webstore: http://www.vestec.com/
>
> Regards,
> -Kashif
>
> --
> Kashif Kahn
> VP Business Development
> Vestec Inc
> Waterloo, ON Canada
> phone: +1 519 885-7615
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jrichey at itltd.net  Wed Jan 12 20:55:58 2011
From: jrichey at itltd.net (JRichey)
Date: Wed, 12 Jan 2011 09:55:58 -0800
Subject: [Freeswitch-users] console
Message-ID: <6ECAF1527329364583AB525CF34ABF950B31A4F9@ms.kallback.com>

Did you ever try using netstat like Steven suggested?  What do you get as
output for "netstat -tunlp"?
 
In one of your commands below you show 127.0.0.2 instead of 127.0.0.1 so if
that's not a typo you may want to try it again.
 
-Justin
 

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of Sam
Sent: Tuesday, January 11, 2011 9:27 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] console


Tried this below and received :-


# /usr/local/fs_2/bin/fs_cli -H 127.0.0.2 -P 8082
[ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
Error]

tried by creating profile:-
# /usr/local/fs_2/bin/fs_cli profile1
[ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
Error]

By port:
# /usr/local/fs_2/bin/fs_cli -P 8082
[ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
Error]



netstat -nlp | grep ; ports 8081 & 8082 are available
But i could get a console for other server by
# /usr/local/fs_2/bin/fs_cli

Regards
Sam



On Tue, Jan 11, 2011 at 10:54 PM, Steven Ayre < steveayre at gmail.com
<mailto:steveayre at gmail.com> > wrote:


Actually, correction it does - just not the event_socket.conf.xml one. It'll
read .fs_cli_conf in your home directory if it exists, but that isn't
created by default - you create it yourself if you want to use it (it's
optional). The command line arguments -H and -P -will- override the config
file though.

Are you using a capital P? -p is password while -P is port. If there's no
password on the event socket you'd get no error from using a small p by
accident.

-Steve 






On 11 January 2011 16:59, Steven Ayre < steveayre at gmail.com
<mailto:steveayre at gmail.com> > wrote:


It does not read any config file. 





On 11 January 2011 16:56, Sam < u2nsam at gmail.com <mailto:u2nsam at gmail.com> >
wrote:


the port and ip donot work for me,
is it that the fs_cli is not reading the config from 192.168.2.2 but it is
reading the config only of 192.168.2.1, though its in the different [FS_1 &
FS_2] path where i am executing.

Regds
Sam 



On Tue, Jan 11, 2011 at 9:38 PM, Steven Ayre < steveayre at gmail.com
<mailto:steveayre at gmail.com> > wrote:


/usr/local/FS_1/bin/fs_cli -P 8021
/usr/local/FS_2/bin/fs_cli -P 8022

fs_cll doesn't read any config file. It's not part of the FS server at all,
you can have it on a different machine that doesn't have FS installed. It
entirely relies on the arguments to control where to connect to.

-Steve 



On 11 January 2011 15:04, Sam < u2nsam at gmail.com <mailto:u2nsam at gmail.com> >
wrote:


Something more here ... i am getting the console for 192.168.2.1 every time
i do fs_cli on both instances .

like 
/usr/local/FS_1/bin/fs_cli
/usr/localFS_2/bin/fs_cli
i get the console for the 1st server only

the 2 server are listing to 2 different ips .

Regds
Sam 


On Tue, Jan 11, 2011 at 8:07 PM, Steven Ayre < steveayre at gmail.com
<mailto:steveayre at gmail.com> > wrote:


It should work. Is there anything already listening on port 8022?

$ netstat -a -n -p | grep 8022

Are you also sure that they're not both loading the same config file?

Regards,
-Steve




On 11 January 2011 14:32, Sam < u2nsam at gmail.com <mailto:u2nsam at gmail.com> >
wrote:
> the scenario is i have 2 ips on 1 server for 2 FS instances;
>
> 192.168.2.1
> 192.168.2.2
>
> and the parameters i have set is:-
>
> for
> 192.168.2.1:-
>    <param name="listen-ip" value="127.0.0.1"/>
>    <param name="listen-port" value="8021"/>
> for
> 192.168.2.2:-
>    <param name="listen-ip" value="127.0.0.1"/>
>    <param name="listen-port" value="8022"/>
>
> Ideally it should work  but i am getting console for only 192.168.2.1 FS .
>
>
> Regards
> Sam
>
> On Tue, Jan 11, 2011 at 4:20 PM, Steven Ayre < steveayre at gmail.com
<mailto:steveayre at gmail.com> > wrote:
>>
>>    <param name="listen-ip" value="127.0.0.1"/>
>>    <param name="listen-port" value="8021"/>
>>
>> You can bind both to port 8021 on their individual IPs, or different
>> ports on the same IP.
>>
>> A listen IP of 0.0.0.0 will mean any IP.
>>
>> -Steve
>>
>> On 11 January 2011 10:44, Sam < u2nsam at gmail.com
<mailto:u2nsam at gmail.com> > wrote:
>> > A query,
>> >
>> > I have 2 FS running on one server on 2 different ips,
>> > so when i do fs_cli going to respective bins , i see console of only
the
>> > first server.
>> >
>> > Is there any way to get the console of both the FS on the same server .
>> > I tried changing the port of event socket to 8022 but it donot works.
>> > <param name="listen-port" value="8022"/>
>> >
>> > Is there some method to start the console of both the instances.
>> >
>> > Regds
>> > Sam
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
<mailto:FreeSWITCH-users at lists.freeswitch.org> 
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> 
>> > UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
<http://lists.freeswitch.org/mailman/options/freeswitch-users> 
>> > http://www.freeswitch.org <http://www.freeswitch.org> 
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
<mailto:FreeSWITCH-users at lists.freeswitch.org> 
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> 
>> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users
<http://lists.freeswitch.org/mailman/options/freeswitch-users> 
>> http://www.freeswitch.org <http://www.freeswitch.org> 
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
<mailto:FreeSWITCH-users at lists.freeswitch.org> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> 
> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users
<http://lists.freeswitch.org/mailman/options/freeswitch-users> 
> http://www.freeswitch.org <http://www.freeswitch.org> 
>
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
<mailto:FreeSWITCH-users at lists.freeswitch.org> 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users
<http://lists.freeswitch.org/mailman/options/freeswitch-users> 
http://www.freeswitch.org <http://www.freeswitch.org> 




_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
<mailto:FreeSWITCH-users at lists.freeswitch.org> 
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UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users
<http://lists.freeswitch.org/mailman/options/freeswitch-users> 
http://www.freeswitch.org <http://www.freeswitch.org> 





_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
<mailto:FreeSWITCH-users at lists.freeswitch.org> 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users
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http://www.freeswitch.org <http://www.freeswitch.org> 





_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
<mailto:FreeSWITCH-users at lists.freeswitch.org> 
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UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users
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_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
<mailto:FreeSWITCH-users at lists.freeswitch.org> 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users
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From chat2jesse at gmail.com  Wed Jan 12 21:27:15 2011
From: chat2jesse at gmail.com (jesse)
Date: Wed, 12 Jan 2011 10:27:15 -0800
Subject: [Freeswitch-users] ZORG,
 new C++ and Java ZRTP implementation public release
In-Reply-To: <935C4C6C-FB5B-46DA-B1EA-73ECBE875EBB@freeswitch.org>
References: <4D2D7704.2060703@infosecurity.ch>
	<AANLkTinZkyNREco5YROXNOMWr-3+10iZstdtd0=mKhi4@mail.gmail.com>
	<935C4C6C-FB5B-46DA-B1EA-73ECBE875EBB@freeswitch.org>
Message-ID: <AANLkTimERaLt2ANCsTuV6iVVR_uXiLt2ahtuxDzFTjA7@mail.gmail.com>

insightful comment!

-jesse

On Wed, Jan 12, 2011 at 7:11 AM, Brian West <brian at freeswitch.org> wrote:
> AGPL is the most evil license you can ever pick for anything... its the one that reaches over the socket and steals your code and soul.
>
> /b
>
> On Jan 12, 2011, at 7:53 AM, Giovanni Maruzzelli wrote:
>
>> Hi Fabio,
>>
>> thanks for your company's very valuable contribution!
>>
>> IANAL, and don't know how the Affero version of GPL is different from
>> "normal" GPL.
>>
>> Maybe having double licensing (like AGPL and BSD) would help spreading
>> the usage of ZORG.
>>
>> Anyway, yay for the feat!
>>
>> -giovanni
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From infos at madovsky.org  Thu Jan 13 00:08:39 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 12 Jan 2011 16:08:39 -0500
Subject: [Freeswitch-users] deflect question
Message-ID: <01AF6B571F2B4BAE944C61206687FB19@e1705>


if I use
 <action application="deflect" data="sip:someone at otherFS.com"/>
in the otherFS dialplan should I createan extension with "someone" condition only ?Thanks
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From lists at telefaks.de  Thu Jan 13 00:14:19 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Wed, 12 Jan 2011 22:14:19 +0100
Subject: [Freeswitch-users] Playing Google translation tts
Message-ID: <4D2E19AB.9020908@telefaks.de>

Has anybody tried to play tts files downloaded from Google transation
service?

I can download them as mp3 but FS refused to play it.
  2011-01-12 13:43:46.340378 [ERR] mod_shout.c:800 Error: MPG123 Error
at __FILE__:__LINE__.
  2011-01-12 13:43:46.340378 [ERR] mod_shout.c:803 Error from mpg123:
Invalid mpg123 handle. (code 10)

File format is
  MPEG ADTS, layer III, v2,  32 kbps, 22.05 kHz, Monaural

If I convert it to wav with lame/ffmpeg/mpg123, the quality is of the
played wav file after conversion is not good (distortions).

So anybody had success in playing this on Freeswitch?

-- 
With kind regards
Peter Steinbach 

mailto:lists (att) telefaks.de
Internet: www.telefaks.de




From brian at freeswitch.org  Thu Jan 13 00:22:27 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Jan 2011 15:22:27 -0600
Subject: [Freeswitch-users] deflect question
In-Reply-To: <01AF6B571F2B4BAE944C61206687FB19@e1705>
References: <01AF6B571F2B4BAE944C61206687FB19@e1705>
Message-ID: <11051112-6CBB-4810-A1ED-9F1326B78F4F@freeswitch.org>

What exactly is the question?

/b

On Jan 12, 2011, at 3:08 PM, Madovsky wrote:

>  
> if I use
>  <action application="deflect" data="sip:someone at otherFS.com"/>
> in the otherFS dialplan should I create
> an extension with "someone" condition only ?
>  
> Thanks
> 

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From infos at madovsky.org  Thu Jan 13 00:36:35 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 12 Jan 2011 16:36:35 -0500
Subject: [Freeswitch-users] deflect question
References: <01AF6B571F2B4BAE944C61206687FB19@e1705>
	<11051112-6CBB-4810-A1ED-9F1326B78F4F@freeswitch.org>
Message-ID: <D9EF98037A5049EA9F4AD22172ECDD23@e1705>

I mean is the request arrives as a new INVITE ?
  ----- Original Message ----- 
  From: Brian West 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, January 12, 2011 4:22 PM
  Subject: Re: [Freeswitch-users] deflect question


  What exactly is the question?


  /b


  On Jan 12, 2011, at 3:08 PM, Madovsky wrote:



    if I use
 <action application="deflect" data="sip:someone at otherFS.com"/>
in the otherFS dialplan should I createan extension with "someone" condition only ? Thanks





------------------------------------------------------------------------------


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From kris at livecall.com  Thu Jan 13 00:49:58 2011
From: kris at livecall.com (Kris)
Date: Wed, 12 Jan 2011 13:49:58 -0800
Subject: [Freeswitch-users] mod_conference end
References: <0A6E4395607344F08F8300BB1CDCD840@e1705>
Message-ID: <23D137A4C56A45B08DDAB22C81B9612E@stor1>

At the end of each session, a record is written in the cdr csv file. There 
might be a way to write to ODBC, but I haven't figured it out yet. I doubt 
the dial plan can figure that out, you might need an application or script.

Kris

----- Original Message ----- 
From: "Madovsky" <infos at madovsky.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, January 12, 2011 8:52 AM
Subject: [Freeswitch-users] mod_conference end


How can a xml dialplan guess that a conference is ended ?
I mean the last participant left ?

Thanks 



From infos at madovsky.org  Thu Jan 13 01:03:51 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 12 Jan 2011 17:03:51 -0500
Subject: [Freeswitch-users] mod_conference end
References: <0A6E4395607344F08F8300BB1CDCD840@e1705>
	<23D137A4C56A45B08DDAB22C81B9612E@stor1>
Message-ID: <13D2D20B832543309C7D50150DE44418@e1705>

the hangup is clean if the control user key "#" is typed.
unless,  the BYE request method should be very clean.


----- Original Message ----- 
From: "Kris" <kris at livecall.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, January 12, 2011 4:49 PM
Subject: Re: [Freeswitch-users] mod_conference end


> At the end of each session, a record is written in the cdr csv file. There
> might be a way to write to ODBC, but I haven't figured it out yet. I doubt
> the dial plan can figure that out, you might need an application or 
> script.
>
> Kris
>
> ----- Original Message ----- 
> From: "Madovsky" <infos at madovsky.org>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, January 12, 2011 8:52 AM
> Subject: [Freeswitch-users] mod_conference end
>
>
> How can a xml dialplan guess that a conference is ended ?
> I mean the last participant left ?
>
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From norm at voicenetwork.ca  Thu Jan 13 01:47:36 2011
From: norm at voicenetwork.ca (Norman Tomlins)
Date: Wed, 12 Jan 2011 17:47:36 -0500
Subject: [Freeswitch-users] Playing Google translation tts
In-Reply-To: <4D2E19AB.9020908@telefaks.de>
References: <4D2E19AB.9020908@telefaks.de>
Message-ID: <AANLkTikzqem2YpaDN77FWpBmRavHrd8qADGZP0FrRrWW@mail.gmail.com>

Peter,

I have used this in the past and it has been working.

<extension name="Free_Google_Text_To_Speech">
  <condition field="destination_number" expression="^2115$">
    <action application="answer" data=""/>
    <action application="playback"
data="shout://translate.google.com/translate_tts?tl=en&q=Buy+Cheap+dids+at+www+dot+voice+network+dot+see+eh"/>
  </condition>
</extension>


Norman Tomlins
http://www.VoiceNetwork.ca



On Wed, Jan 12, 2011 at 4:14 PM, Peter Steinbach <lists at telefaks.de> wrote:
> Has anybody tried to play tts files downloaded from Google transation
> service?
>
> I can download them as mp3 but FS refused to play it.
> ?2011-01-12 13:43:46.340378 [ERR] mod_shout.c:800 Error: MPG123 Error
> at __FILE__:__LINE__.
> ?2011-01-12 13:43:46.340378 [ERR] mod_shout.c:803 Error from mpg123:
> Invalid mpg123 handle. (code 10)
>
> File format is
> ?MPEG ADTS, layer III, v2, ?32 kbps, 22.05 kHz, Monaural
>
> If I convert it to wav with lame/ffmpeg/mpg123, the quality is of the
> played wav file after conversion is not good (distortions).
>
> So anybody had success in playing this on Freeswitch?
>
> --
> With kind regards
> Peter Steinbach
>
> mailto:lists (att) telefaks.de
> Internet: www.telefaks.de
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From infos at madovsky.org  Thu Jan 13 01:50:02 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 12 Jan 2011 17:50:02 -0500
Subject: [Freeswitch-users] deflect question
Message-ID: <6FF5500287154041BB8D6209AAA32929@e1705>

I'm stucked.

in the log
the call stalls with this line

EXECUTE sofia/internal/5555555555555 at myfs.org deflect(9999999999999 at otherfs.org)
big silenece and nothing happens

the otherFS doesn't receive anything.

I tried to call manually the deflect sip and it works.

Did I forget something ?


thanks


  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, January 12, 2011 4:36 PM
  Subject: Re: [Freeswitch-users] deflect question


  I mean is the request arrives as a new INVITE ?
    ----- Original Message ----- 
    From: Brian West 
    To: FreeSWITCH Users Help 
    Sent: Wednesday, January 12, 2011 4:22 PM
    Subject: Re: [Freeswitch-users] deflect question


    What exactly is the question? 


    /b


    On Jan 12, 2011, at 3:08 PM, Madovsky wrote:



      if I use
 <action application="deflect" data="sip:someone at otherFS.com"/>
in the otherFS dialplan should I createan extension with "someone" condition only ? Thanks





----------------------------------------------------------------------------


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    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From jack at hotglass.cc  Thu Jan 13 00:38:40 2011
From: jack at hotglass.cc (Jack Loranger)
Date: Wed, 12 Jan 2011 13:38:40 -0800
Subject: [Freeswitch-users] FreeSwitch softphone testing
Message-ID: <4D2E1F60.5010004@hotglass.cc>

I listened in on the conference this morning and if Mitch would like 
someone to use his phone I would be happy to use it while I am testing 
my FreeSwitch app.
Can it be used in a webpage?
Jack

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From lists at telefaks.de  Thu Jan 13 02:06:45 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Thu, 13 Jan 2011 00:06:45 +0100
Subject: [Freeswitch-users] Playing Google translation tts
In-Reply-To: <AANLkTikzqem2YpaDN77FWpBmRavHrd8qADGZP0FrRrWW@mail.gmail.com>
References: <4D2E19AB.9020908@telefaks.de>
	<AANLkTikzqem2YpaDN77FWpBmRavHrd8qADGZP0FrRrWW@mail.gmail.com>
Message-ID: <4D2E3405.10307@telefaks.de>

Norman,

I have the feeling it may be reated to this one, at least the error
mssage looks the same.
  http://jira.freeswitch.org/browse/FS-2942
We have a rather new GIT version of Freeswitch, so maybe it's just in a
newer version?

Best regards
Peter


Norman Tomlins schrieb:
> Peter,
>
> I have used this in the past and it has been working.
>
> <extension name="Free_Google_Text_To_Speech">
>   <condition field="destination_number" expression="^2115$">
>     <action application="answer" data=""/>
>     <action application="playback"
> data="shout://translate.google.com/translate_tts?tl=en&q=Buy+Cheap+dids+at+www+dot+voice+network+dot+see+eh"/>
>   </condition>
> </extension>
>
>
> Norman Tomlins
> http://www.VoiceNetwork.ca
>
>
>
> On Wed, Jan 12, 2011 at 4:14 PM, Peter Steinbach <lists at telefaks.de> wrote:
>   
>> Has anybody tried to play tts files downloaded from Google transation
>> service?
>>
>> I can download them as mp3 but FS refused to play it.
>>  2011-01-12 13:43:46.340378 [ERR] mod_shout.c:800 Error: MPG123 Error
>> at __FILE__:__LINE__.
>>  2011-01-12 13:43:46.340378 [ERR] mod_shout.c:803 Error from mpg123:
>> Invalid mpg123 handle. (code 10)
>>
>> File format is
>>  MPEG ADTS, layer III, v2,  32 kbps, 22.05 kHz, Monaural
>>
>> If I convert it to wav with lame/ffmpeg/mpg123, the quality is of the
>> played wav file after conversion is not good (distortions).
>>
>> So anybody had success in playing this on Freeswitch?
>>
>> --
>> With kind regards
>> Peter Steinbach
>>
>> mailto:lists (att) telefaks.de
>> Internet: www.telefaks.de
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de




From infos at madovsky.org  Thu Jan 13 02:12:39 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 12 Jan 2011 18:12:39 -0500
Subject: [Freeswitch-users] Playing Google translation tts
References: <4D2E19AB.9020908@telefaks.de><AANLkTikzqem2YpaDN77FWpBmRavHrd8qADGZP0FrRrWW@mail.gmail.com>
	<4D2E3405.10307@telefaks.de>
Message-ID: <12CBD39BC5924754A3CE207352885203@e1705>

why don't you use Cepstral ?
it's only USD30

----- Original Message ----- 
From: "Peter Steinbach" <lists at telefaks.de>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, January 12, 2011 6:06 PM
Subject: Re: [Freeswitch-users] Playing Google translation tts


> Norman,
>
> I have the feeling it may be reated to this one, at least the error
> mssage looks the same.
>  http://jira.freeswitch.org/browse/FS-2942
> We have a rather new GIT version of Freeswitch, so maybe it's just in a
> newer version?
>
> Best regards
> Peter
>
>
> Norman Tomlins schrieb:
>> Peter,
>>
>> I have used this in the past and it has been working.
>>
>> <extension name="Free_Google_Text_To_Speech">
>>   <condition field="destination_number" expression="^2115$">
>>     <action application="answer" data=""/>
>>     <action application="playback"
>> data="shout://translate.google.com/translate_tts?tl=en&q=Buy+Cheap+dids+at+www+dot+voice+network+dot+see+eh"/>
>>   </condition>
>> </extension>
>>
>>
>> Norman Tomlins
>> http://www.VoiceNetwork.ca
>>
>>
>>
>> On Wed, Jan 12, 2011 at 4:14 PM, Peter Steinbach <lists at telefaks.de> 
>> wrote:
>>
>>> Has anybody tried to play tts files downloaded from Google transation
>>> service?
>>>
>>> I can download them as mp3 but FS refused to play it.
>>>  2011-01-12 13:43:46.340378 [ERR] mod_shout.c:800 Error: MPG123 Error
>>> at __FILE__:__LINE__.
>>>  2011-01-12 13:43:46.340378 [ERR] mod_shout.c:803 Error from mpg123:
>>> Invalid mpg123 handle. (code 10)
>>>
>>> File format is
>>>  MPEG ADTS, layer III, v2,  32 kbps, 22.05 kHz, Monaural
>>>
>>> If I convert it to wav with lame/ffmpeg/mpg123, the quality is of the
>>> played wav file after conversion is not good (distortions).
>>>
>>> So anybody had success in playing this on Freeswitch?
>>>
>>> --
>>> With kind regards
>>> Peter Steinbach
>>>
>>> mailto:lists (att) telefaks.de
>>> Internet: www.telefaks.de
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> -- 
> With kind regards
> Peter Steinbach
>
> Telefaks Services GmbH
> mailto:lists (att) telefaks.de
> Internet: www.telefaks.de
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From brad at tritelcomm.com  Thu Jan 13 02:12:50 2011
From: brad at tritelcomm.com (Brad Mina)
Date: Wed, 12 Jan 2011 15:12:50 -0800
Subject: [Freeswitch-users] FreeSwitch softphone testing
In-Reply-To: <4D2E1F60.5010004@hotglass.cc>
References: <4D2E1F60.5010004@hotglass.cc>
Message-ID: <AANLkTi=YwA7ThHruFnT1Hg4X7p5jioZEmG0qSW-vC0O8@mail.gmail.com>

>From the sound of it, it's a windows-only softphone and there was no mention
of a web frontend. The only caveat was it didn't have G729 support.

On Wed, Jan 12, 2011 at 1:38 PM, Jack Loranger <jack at hotglass.cc> wrote:

>  I listened in on the conference this morning and if Mitch would like
> someone to use his phone I would be happy to use it while I am testing my
> FreeSwitch app.
> Can it be used in a webpage?
> Jack
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From joaocarlosleme at gmail.com  Thu Jan 13 03:15:01 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Wed, 12 Jan 2011 16:15:01 -0800
Subject: [Freeswitch-users] Caller ID using Fifo
Message-ID: <AANLkTik6pdgXgND6kmzZA4qe8vDZQ0_7e47+ciVWPfub@mail.gmail.com>

Hi there,
I would like to know if there is a way to see the caller ID on my Sip Client
(X-Lite for example) of the caller that I answear from a Fifo queue?
Thanks,
John
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From brian at freeswitch.org  Thu Jan 13 04:40:10 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Jan 2011 19:40:10 -0600
Subject: [Freeswitch-users] deflect question
In-Reply-To: <6FF5500287154041BB8D6209AAA32929@e1705>
References: <6FF5500287154041BB8D6209AAA32929@e1705>
Message-ID: <51CF6810-ECDC-4408-904F-01BE066AE862@freeswitch.org>

try making it sip:something at blah

/b

On Jan 12, 2011, at 4:50 PM, Madovsky wrote:

> I'm stucked.
>  
> in the log
> the call stalls with this line
>  
> EXECUTE sofia/internal/5555555555555 at myfs.org deflect(9999999999999 at otherfs.org)
> big silenece and nothing happens
> the otherFS doesn't receive anything.
>  
> I tried to call manually the deflect sip and it works.
>  
> Did I forget something ?
>  
>  
> thanks

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From jmesquita at freeswitch.org  Thu Jan 13 05:17:43 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 12 Jan 2011 23:17:43 -0300
Subject: [Freeswitch-users] Caller ID using Fifo
In-Reply-To: <AANLkTik6pdgXgND6kmzZA4qe8vDZQ0_7e47+ciVWPfub@mail.gmail.com>
References: <AANLkTik6pdgXgND6kmzZA4qe8vDZQ0_7e47+ciVWPfub@mail.gmail.com>
Message-ID: <AANLkTikKDO3qru-9=nuPEQcW+iCN7ccizyJAh9sqUkCN@mail.gmail.com>

Jo?o Leme,

The caller id is not passed when the phone is ringing because mod_fifo does
not know which call is going to be sent to that channel once it is answered
until it is really answered. I don't know if mod_callcenter does show
anything but you should consider looking at the documentation if you really
need this feature.

Regards,
Jo?o Mesquita


On Wed, Jan 12, 2011 at 9:15 PM, Joao Leme <joaocarlosleme at gmail.com> wrote:

> Hi there,
> I would like to know if there is a way to see the caller ID on my Sip
> Client (X-Lite for example) of the caller that I answear from a Fifo queue?
> Thanks,
> John
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lists at telefaks.de  Thu Jan 13 05:18:16 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Thu, 13 Jan 2011 03:18:16 +0100
Subject: [Freeswitch-users] Playing Google translation tts
In-Reply-To: <12CBD39BC5924754A3CE207352885203@e1705>
References: <4D2E19AB.9020908@telefaks.de><AANLkTikzqem2YpaDN77FWpBmRavHrd8qADGZP0FrRrWW@mail.gmail.com>	<4D2E3405.10307@telefaks.de>
	<12CBD39BC5924754A3CE207352885203@e1705>
Message-ID: <4D2E60E8.3050307@telefaks.de>

We do use Cepstral for German voices. But Google supports a number of
languages right now.

Best regards
Peter

Madovsky schrieb:
> why don't you use Cepstral ?
> it's only USD30
>
> ----- Original Message ----- 
> From: "Peter Steinbach" <lists at telefaks.de>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, January 12, 2011 6:06 PM
> Subject: Re: [Freeswitch-users] Playing Google translation tts
>
>
>   
>> Norman,
>>
>> I have the feeling it may be reated to this one, at least the error
>> mssage looks the same.
>>  http://jira.freeswitch.org/browse/FS-2942
>> We have a rather new GIT version of Freeswitch, so maybe it's just in a
>> newer version?
>>
>> Best regards
>> Peter
>>
>>
>> Norman Tomlins schrieb:
>>     
>>> Peter,
>>>
>>> I have used this in the past and it has been working.
>>>
>>> <extension name="Free_Google_Text_To_Speech">
>>>   <condition field="destination_number" expression="^2115$">
>>>     <action application="answer" data=""/>
>>>     <action application="playback"
>>> data="shout://translate.google.com/translate_tts?tl=en&q=Buy+Cheap+dids+at+www+dot+voice+network+dot+see+eh"/>
>>>   </condition>
>>> </extension>
>>>
>>>
>>> Norman Tomlins
>>> http://www.VoiceNetwork.ca
>>>
>>>
>>>
>>> On Wed, Jan 12, 2011 at 4:14 PM, Peter Steinbach <lists at telefaks.de> 
>>> wrote:
>>>
>>>       
>>>> Has anybody tried to play tts files downloaded from Google transation
>>>> service?
>>>>
>>>> I can download them as mp3 but FS refused to play it.
>>>>  2011-01-12 13:43:46.340378 [ERR] mod_shout.c:800 Error: MPG123 Error
>>>> at __FILE__:__LINE__.
>>>>  2011-01-12 13:43:46.340378 [ERR] mod_shout.c:803 Error from mpg123:
>>>> Invalid mpg123 handle. (code 10)
>>>>
>>>> File format is
>>>>  MPEG ADTS, layer III, v2,  32 kbps, 22.05 kHz, Monaural
>>>>
>>>> If I convert it to wav with lame/ffmpeg/mpg123, the quality is of the
>>>> played wav file after conversion is not good (distortions).
>>>>
>>>> So anybody had success in playing this on Freeswitch?
>>>>
>>>> --
>>>> With kind regards
>>>> Peter Steinbach
>>>>
>>>> mailto:lists (att) telefaks.de
>>>> Internet: www.telefaks.de
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>         
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>       
>> -- 
>> With kind regards
>> Peter Steinbach
>>
>> Telefaks Services GmbH
>> mailto:lists (att) telefaks.de
>> Internet: www.telefaks.de
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org 
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de




From u2nsam at gmail.com  Thu Jan 13 06:16:04 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 13 Jan 2011 08:46:04 +0530
Subject: [Freeswitch-users] console
In-Reply-To: <6ECAF1527329364583AB525CF34ABF950B31A4F9@ms.kallback.com>
References: <6ECAF1527329364583AB525CF34ABF950B31A4F9@ms.kallback.com>
Message-ID: <AANLkTinXiaVGr5bneGCfvXUmbEnJ5Ggjcmvim=Wjqa5O@mail.gmail.com>

Yes the netstat shows the ports 8081 & 8082 not used
Justin, as said earlier i have 2 FS instances running on same server and
having below config:-
for
> 192.168.2.1:-
>    <param name="listen-ip" value="127.0.0.1"/>
>    <param name="listen-port" value="8021"/>
> for
> 192.168.2.2:-
>    <param name="listen-ip" value="127.0.0.1"/>
>    <param name="listen-port" value="8022"/>

Here I tried all the permutation combination changing the IP port but no
sucess.

Regds
Sam

On Wed, Jan 12, 2011 at 11:25 PM, JRichey <jrichey at itltd.net> wrote:

>  Did you ever try using netstat like Steven suggested?  What do you get as
> output for "netstat -tunlp"?
>
> In one of your commands below you show 127.0.0.2 instead of 127.0.0.1 so if
> that's not a typo you may want to try it again.
>
> -Justin
>
>
> -----Original Message-----
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org]*On Behalf Of *Sam
> *Sent:* Tuesday, January 11, 2011 9:27 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] console
>
> Tried this below and received :-
>
>
> # /usr/local/fs_2/bin/fs_cli -H 127.0.0.2 -P 8082
> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
> Error]
>
> tried by creating profile:-
> # /usr/local/fs_2/bin/fs_cli profile1
> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
> Error]
>
> By port:
> # /usr/local/fs_2/bin/fs_cli -P 8082
> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
> Error]
>
>
> *netstat -nlp | grep ; ports 8081 & 8082 are available*
>
> But i could get a console for other server by
> # /usr/local/fs_2/bin/fs_cli
>
> Regards
> Sam
>
>
> On Tue, Jan 11, 2011 at 10:54 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Actually, correction it does - just not the event_socket.conf.xml one.
>> It'll read .fs_cli_conf in your home directory if it exists, but that isn't
>> created by default - you create it yourself if you want to use it (it's
>> optional). The command line arguments -H and -P -will- override the config
>> file though.
>>
>> Are you using a capital P? -p is password while -P is port. If there's no
>> password on the event socket you'd get no error from using a small p by
>> accident.
>>
>> -Steve
>>
>>
>>
>>
>>
>>
>> On 11 January 2011 16:59, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> It does not read any config file.
>>>
>>>
>>>
>>>
>>>
>>> On 11 January 2011 16:56, Sam <u2nsam at gmail.com> wrote:
>>>
>>>> the port and ip donot work for me,
>>>> is it that the fs_cli is not reading the config from 192.168.2.2 but it
>>>> is reading the config only of 192.168.2.1, though its in the different [FS_1
>>>> & FS_2] path where i am executing.
>>>>
>>>> Regds
>>>> Sam
>>>>
>>>>
>>>>
>>>> On Tue, Jan 11, 2011 at 9:38 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>>>
>>>>> /usr/local/FS_1/bin/fs_cli -P 8021
>>>>> /usr/local/FS_2/bin/fs_cli -P 8022
>>>>>
>>>>> fs_cll doesn't read any config file. It's not part of the FS server at
>>>>> all, you can have it on a different machine that doesn't have FS installed.
>>>>> It entirely relies on the arguments to control where to connect to.
>>>>>
>>>>> -Steve
>>>>>
>>>>>
>>>>>
>>>>> On 11 January 2011 15:04, Sam <u2nsam at gmail.com> wrote:
>>>>>
>>>>>> Something more here ... i am getting the console for 192.168.2.1 every
>>>>>> time i do fs_cli on both instances .
>>>>>>
>>>>>> like
>>>>>> /usr/local/FS_1/bin/fs_cli
>>>>>> /usr/localFS_2/bin/fs_cli
>>>>>> i get the console for the 1st server only
>>>>>>
>>>>>> the 2 server are listing to 2 different ips .
>>>>>>
>>>>>> Regds
>>>>>> Sam
>>>>>>
>>>>>>
>>>>>> On Tue, Jan 11, 2011 at 8:07 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>>>>>
>>>>>>> It should work. Is there anything already listening on port 8022?
>>>>>>>
>>>>>>> $ netstat -a -n -p | grep 8022
>>>>>>>
>>>>>>> Are you also sure that they're not both loading the same config file?
>>>>>>>
>>>>>>> Regards,
>>>>>>> -Steve
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On 11 January 2011 14:32, Sam <u2nsam at gmail.com> wrote:
>>>>>>> > the scenario is i have 2 ips on 1 server for 2 FS instances;
>>>>>>> >
>>>>>>> > 192.168.2.1
>>>>>>> > 192.168.2.2
>>>>>>> >
>>>>>>> > and the parameters i have set is:-
>>>>>>> >
>>>>>>> > for
>>>>>>> > 192.168.2.1:-
>>>>>>> >    <param name="listen-ip" value="127.0.0.1"/>
>>>>>>> >    <param name="listen-port" value="8021"/>
>>>>>>> > for
>>>>>>> > 192.168.2.2:-
>>>>>>> >    <param name="listen-ip" value="127.0.0.1"/>
>>>>>>> >    <param name="listen-port" value="8022"/>
>>>>>>> >
>>>>>>> > Ideally it should work  but i am getting console for only
>>>>>>> 192.168.2.1 FS .
>>>>>>> >
>>>>>>> >
>>>>>>> > Regards
>>>>>>> > Sam
>>>>>>> >
>>>>>>> > On Tue, Jan 11, 2011 at 4:20 PM, Steven Ayre <steveayre at gmail.com>
>>>>>>> wrote:
>>>>>>> >>
>>>>>>> >>    <param name="listen-ip" value="127.0.0.1"/>
>>>>>>> >>    <param name="listen-port" value="8021"/>
>>>>>>> >>
>>>>>>> >> You can bind both to port 8021 on their individual IPs, or
>>>>>>> different
>>>>>>> >> ports on the same IP.
>>>>>>> >>
>>>>>>> >> A listen IP of 0.0.0.0 will mean any IP.
>>>>>>> >>
>>>>>>> >> -Steve
>>>>>>> >>
>>>>>>> >> On 11 January 2011 10:44, Sam <u2nsam at gmail.com> wrote:
>>>>>>> >> > A query,
>>>>>>> >> >
>>>>>>> >> > I have 2 FS running on one server on 2 different ips,
>>>>>>> >> > so when i do fs_cli going to respective bins , i see console of
>>>>>>> only the
>>>>>>> >> > first server.
>>>>>>> >> >
>>>>>>> >> > Is there any way to get the console of both the FS on the same
>>>>>>> server .
>>>>>>> >> > I tried changing the port of event socket to 8022 but it donot
>>>>>>> works.
>>>>>>> >> > <param name="listen-port" value="8022"/>
>>>>>>> >> >
>>>>>>> >> > Is there some method to start the console of both the instances.
>>>>>>> >> >
>>>>>>> >> > Regds
>>>>>>> >> > Sam
>>>>>>> >> >
>>>>>>> >> > _______________________________________________
>>>>>>> >> > FreeSWITCH-users mailing list
>>>>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >> > UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> >> > http://www.freeswitch.org
>>>>>>> >> >
>>>>>>> >> >
>>>>>>> >>
>>>>>>> >> _______________________________________________
>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> >> http://www.freeswitch.org
>>>>>>> >
>>>>>>> >
>>>>>>> > _______________________________________________
>>>>>>> > FreeSWITCH-users mailing list
>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> > UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> > http://www.freeswitch.org
>>>>>>> >
>>>>>>> >
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Thu Jan 13 06:32:14 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 12 Jan 2011 22:32:14 -0500
Subject: [Freeswitch-users] console
References: <6ECAF1527329364583AB525CF34ABF950B31A4F9@ms.kallback.com>
	<AANLkTinXiaVGr5bneGCfvXUmbEnJ5Ggjcmvim=Wjqa5O@mail.gmail.com>
Message-ID: <2182C8734A5F48D5858AD492D426DEA0@e1705>

did you try to compile your two FS
with different prefix and .pid path ?
as this you'll be sure that FS are running independtly.
I remember (maybe wrongly) to have this kind of problem before.....
  ----- Original Message ----- 
  From: Sam 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, January 12, 2011 10:16 PM
  Subject: Re: [Freeswitch-users] console


  Yes the netstat shows the ports 8081 & 8082 not used 
  Justin, as said earlier i have 2 FS instances running on same server and having below config:-
  for
  > 192.168.2.1:-
  >    <param name="listen-ip" value="127.0.0.1"/>
  >    <param name="listen-port" value="8021"/>
  > for
  > 192.168.2.2:-
  >    <param name="listen-ip" value="127.0.0.1"/>
  >    <param name="listen-port" value="8022"/>

  Here I tried all the permutation combination changing the IP port but no sucess.

  Regds
  Sam


  On Wed, Jan 12, 2011 at 11:25 PM, JRichey <jrichey at itltd.net> wrote:

    Did you ever try using netstat like Steven suggested?  What do you get as output for "netstat -tunlp"?

    In one of your commands below you show 127.0.0.2 instead of 127.0.0.1 so if that's not a typo you may want to try it again.

    -Justin

      -----Original Message-----
      From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of Sam
      Sent: Tuesday, January 11, 2011 9:27 PM
      To: FreeSWITCH Users Help
      Subject: Re: [Freeswitch-users] console


      Tried this below and received :-


      # /usr/local/fs_2/bin/fs_cli -H 127.0.0.2 -P 8082
      [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection Error]

      tried by creating profile:-
      # /usr/local/fs_2/bin/fs_cli profile1
      [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection Error]

      By port:
      # /usr/local/fs_2/bin/fs_cli -P 8082
      [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection Error]



netstat -nlp | grep ; ports 8081 & 8082 are availableBut i could get a console for other server by
      # /usr/local/fs_2/bin/fs_cli

      Regards
      Sam



      On Tue, Jan 11, 2011 at 10:54 PM, Steven Ayre <steveayre at gmail.com> wrote:

        Actually, correction it does - just not the event_socket.conf.xml one. It'll read .fs_cli_conf in your home directory if it exists, but that isn't created by default - you create it yourself if you want to use it (it's optional). The command line arguments -H and -P -will- override the config file though.

        Are you using a capital P? -p is password while -P is port. If there's no password on the event socket you'd get no error from using a small p by accident.

        -Steve 







        On 11 January 2011 16:59, Steven Ayre <steveayre at gmail.com> wrote:

          It does not read any config file. 






          On 11 January 2011 16:56, Sam <u2nsam at gmail.com> wrote:

            the port and ip donot work for me,
            is it that the fs_cli is not reading the config from 192.168.2.2 but it is reading the config only of 192.168.2.1, though its in the different [FS_1 & FS_2] path where i am executing.

            Regds
            Sam 




            On Tue, Jan 11, 2011 at 9:38 PM, Steven Ayre <steveayre at gmail.com> wrote:

              /usr/local/FS_1/bin/fs_cli -P 8021
              /usr/local/FS_2/bin/fs_cli -P 8022

              fs_cll doesn't read any config file. It's not part of the FS server at all, you can have it on a different machine that doesn't have FS installed. It entirely relies on the arguments to control where to connect to.

              -Steve 




              On 11 January 2011 15:04, Sam <u2nsam at gmail.com> wrote:

                Something more here ... i am getting the console for 192.168.2.1 every time i do fs_cli on both instances .

                like 
                /usr/local/FS_1/bin/fs_cli
                /usr/localFS_2/bin/fs_cli
                i get the console for the 1st server only

                the 2 server are listing to 2 different ips .

                Regds
                Sam 



                On Tue, Jan 11, 2011 at 8:07 PM, Steven Ayre <steveayre at gmail.com> wrote:

                  It should work. Is there anything already listening on port 8022?

                  $ netstat -a -n -p | grep 8022

                  Are you also sure that they're not both loading the same config file?

                  Regards,
                  -Steve




                  On 11 January 2011 14:32, Sam <u2nsam at gmail.com> wrote:
                  > the scenario is i have 2 ips on 1 server for 2 FS instances;
                  >
                  > 192.168.2.1
                  > 192.168.2.2
                  >
                  > and the parameters i have set is:-
                  >
                  > for
                  > 192.168.2.1:-
                  >    <param name="listen-ip" value="127.0.0.1"/>
                  >    <param name="listen-port" value="8021"/>
                  > for
                  > 192.168.2.2:-
                  >    <param name="listen-ip" value="127.0.0.1"/>
                  >    <param name="listen-port" value="8022"/>
                  >
                  > Ideally it should work  but i am getting console for only 192.168.2.1 FS .
                  >
                  >
                  > Regards
                  > Sam
                  >
                  > On Tue, Jan 11, 2011 at 4:20 PM, Steven Ayre <steveayre at gmail.com> wrote:
                  >>
                  >>    <param name="listen-ip" value="127.0.0.1"/>
                  >>    <param name="listen-port" value="8021"/>
                  >>
                  >> You can bind both to port 8021 on their individual IPs, or different
                  >> ports on the same IP.
                  >>
                  >> A listen IP of 0.0.0.0 will mean any IP.
                  >>
                  >> -Steve
                  >>
                  >> On 11 January 2011 10:44, Sam <u2nsam at gmail.com> wrote:
                  >> > A query,
                  >> >
                  >> > I have 2 FS running on one server on 2 different ips,
                  >> > so when i do fs_cli going to respective bins , i see console of only the
                  >> > first server.
                  >> >
                  >> > Is there any way to get the console of both the FS on the same server .
                  >> > I tried changing the port of event socket to 8022 but it donot works.
                  >> > <param name="listen-port" value="8022"/>
                  >> >
                  >> > Is there some method to start the console of both the instances.
                  >> >
                  >> > Regds
                  >> > Sam
                  >> >
                  >> > _______________________________________________
                  >> > FreeSWITCH-users mailing list
                  >> > FreeSWITCH-users at lists.freeswitch.org
                  >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
                  >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
                  >> > http://www.freeswitch.org
                  >> >
                  >> >
                  >>
                  >> _______________________________________________
                  >> FreeSWITCH-users mailing list
                  >> FreeSWITCH-users at lists.freeswitch.org
                  >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
                  >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
                  >> http://www.freeswitch.org
                  >
                  >
                  > _______________________________________________
                  > FreeSWITCH-users mailing list
                  > FreeSWITCH-users at lists.freeswitch.org
                  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
                  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
                  > http://www.freeswitch.org
                  >
                  >

                  _______________________________________________
                  FreeSWITCH-users mailing list
                  FreeSWITCH-users at lists.freeswitch.org
                  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
                  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
                  http://www.freeswitch.org




                _______________________________________________
                FreeSWITCH-users mailing list
                FreeSWITCH-users at lists.freeswitch.org
                http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
                UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
                http://www.freeswitch.org





              _______________________________________________
              FreeSWITCH-users mailing list
              FreeSWITCH-users at lists.freeswitch.org
              http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
              UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
              http://www.freeswitch.org





            _______________________________________________
            FreeSWITCH-users mailing list
            FreeSWITCH-users at lists.freeswitch.org
            http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
            UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
            http://www.freeswitch.org







        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org





    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org






------------------------------------------------------------------------------


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From infos at madovsky.org  Thu Jan 13 06:33:04 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 12 Jan 2011 22:33:04 -0500
Subject: [Freeswitch-users] deflect question
References: <6FF5500287154041BB8D6209AAA32929@e1705>
	<51CF6810-ECDC-4408-904F-01BE066AE862@freeswitch.org>
Message-ID: <21F178F976DD4D9792075BF7507F11E6@e1705>

did it before, no effect.
  ----- Original Message ----- 
  From: Brian West 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, January 12, 2011 8:40 PM
  Subject: Re: [Freeswitch-users] deflect question


  try making it sip:something at blah


  /b


  On Jan 12, 2011, at 4:50 PM, Madovsky wrote:


    I'm stucked.

    in the log
    the call stalls with this line

    EXECUTE sofia/internal/5555555555555 at myfs.org deflect(9999999999999 at otherfs.org)
    big silenece and nothing happens

    the otherFS doesn't receive anything.

    I tried to call manually the deflect sip and it works.

    Did I forget something ?


    thanks




------------------------------------------------------------------------------


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  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Thu Jan 13 06:34:05 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 12 Jan 2011 22:34:05 -0500
Subject: [Freeswitch-users] deflect question
References: <6FF5500287154041BB8D6209AAA32929@e1705>
	<51CF6810-ECDC-4408-904F-01BE066AE862@freeswitch.org>
Message-ID: <0972B70F2B5943F9AB5DDE2E441EB4F8@e1705>

I'm sure also that the call is answered before deflect
  ----- Original Message ----- 
  From: Brian West 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, January 12, 2011 8:40 PM
  Subject: Re: [Freeswitch-users] deflect question


  try making it sip:something at blah


  /b


  On Jan 12, 2011, at 4:50 PM, Madovsky wrote:


    I'm stucked.

    in the log
    the call stalls with this line

    EXECUTE sofia/internal/5555555555555 at myfs.org deflect(9999999999999 at otherfs.org)
    big silenece and nothing happens

    the otherFS doesn't receive anything.

    I tried to call manually the deflect sip and it works.

    Did I forget something ?


    thanks




------------------------------------------------------------------------------


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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From u2nsam at gmail.com  Thu Jan 13 06:59:52 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 13 Jan 2011 09:29:52 +0530
Subject: [Freeswitch-users] console
In-Reply-To: <2182C8734A5F48D5858AD492D426DEA0@e1705>
References: <6ECAF1527329364583AB525CF34ABF950B31A4F9@ms.kallback.com>
	<AANLkTinXiaVGr5bneGCfvXUmbEnJ5Ggjcmvim=Wjqa5O@mail.gmail.com>
	<2182C8734A5F48D5858AD492D426DEA0@e1705>
Message-ID: <AANLkTikgopv7pY4nnhH3h=6bmp7rGe5qpP5FmiUePitb@mail.gmail.com>

Yes both FS are complied with the prefixes and are running quiet
independently , the only problem is with the fs_cli .

Regards
Sam



On Thu, Jan 13, 2011 at 9:02 AM, Madovsky <infos at madovsky.org> wrote:

>  did you try to compile your two FS
> with different prefix and .pid path ?
> as this you'll be sure that FS are running independtly.
> I remember (maybe wrongly) to have this kind of problem before.....
>
> ----- Original Message -----
> *From:* Sam <u2nsam at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, January 12, 2011 10:16 PM
> *Subject:* Re: [Freeswitch-users] console
>
> Yes the netstat shows the ports 8081 & 8082 not used
> Justin, as said earlier i have 2 FS instances running on same server and
> having below config:-
> for
> > 192.168.2.1:-
> >    <param name="listen-ip" value="127.0.0.1"/>
> >    <param name="listen-port" value="8021"/>
> > for
> > 192.168.2.2:-
> >    <param name="listen-ip" value="127.0.0.1"/>
> >    <param name="listen-port" value="8022"/>
>
> Here I tried all the permutation combination changing the IP port but no
> sucess.
>
> Regds
> Sam
>
> On Wed, Jan 12, 2011 at 11:25 PM, JRichey <jrichey at itltd.net> wrote:
>
>>  Did you ever try using netstat like Steven suggested?  What do you get
>> as output for "netstat -tunlp"?
>>
>> In one of your commands below you show 127.0.0.2 instead of 127.0.0.1 so
>> if that's not a typo you may want to try it again.
>>
>> -Justin
>>
>>
>> -----Original Message-----
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org]*On Behalf Of *Sam
>> *Sent:* Tuesday, January 11, 2011 9:27 PM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] console
>>
>> Tried this below and received :-
>>
>>
>> # /usr/local/fs_2/bin/fs_cli -H 127.0.0.2 -P 8082
>> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
>> Error]
>>
>> tried by creating profile:-
>> # /usr/local/fs_2/bin/fs_cli profile1
>> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
>> Error]
>>
>> By port:
>> # /usr/local/fs_2/bin/fs_cli -P 8082
>> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
>> Error]
>>
>>
>> *netstat -nlp | grep ; ports 8081 & 8082 are available*
>>
>> But i could get a console for other server by
>> # /usr/local/fs_2/bin/fs_cli
>>
>> Regards
>> Sam
>>
>>
>> On Tue, Jan 11, 2011 at 10:54 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>
>>> Actually, correction it does - just not the event_socket.conf.xml one.
>>> It'll read .fs_cli_conf in your home directory if it exists, but that isn't
>>> created by default - you create it yourself if you want to use it (it's
>>> optional). The command line arguments -H and -P -will- override the config
>>> file though.
>>>
>>> Are you using a capital P? -p is password while -P is port. If there's no
>>> password on the event socket you'd get no error from using a small p by
>>> accident.
>>>
>>> -Steve
>>>
>>>
>>>
>>>
>>>
>>>
>>> On 11 January 2011 16:59, Steven Ayre <steveayre at gmail.com> wrote:
>>>
>>>> It does not read any config file.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On 11 January 2011 16:56, Sam <u2nsam at gmail.com> wrote:
>>>>
>>>>> the port and ip donot work for me,
>>>>> is it that the fs_cli is not reading the config from 192.168.2.2 but it
>>>>> is reading the config only of 192.168.2.1, though its in the different [FS_1
>>>>> & FS_2] path where i am executing.
>>>>>
>>>>> Regds
>>>>> Sam
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Jan 11, 2011 at 9:38 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>>>>
>>>>>> /usr/local/FS_1/bin/fs_cli -P 8021
>>>>>> /usr/local/FS_2/bin/fs_cli -P 8022
>>>>>>
>>>>>> fs_cll doesn't read any config file. It's not part of the FS server at
>>>>>> all, you can have it on a different machine that doesn't have FS installed.
>>>>>> It entirely relies on the arguments to control where to connect to.
>>>>>>
>>>>>> -Steve
>>>>>>
>>>>>>
>>>>>>
>>>>>> On 11 January 2011 15:04, Sam <u2nsam at gmail.com> wrote:
>>>>>>
>>>>>>> Something more here ... i am getting the console for 192.168.2.1
>>>>>>> every time i do fs_cli on both instances .
>>>>>>>
>>>>>>> like
>>>>>>> /usr/local/FS_1/bin/fs_cli
>>>>>>> /usr/localFS_2/bin/fs_cli
>>>>>>> i get the console for the 1st server only
>>>>>>>
>>>>>>> the 2 server are listing to 2 different ips .
>>>>>>>
>>>>>>> Regds
>>>>>>> Sam
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Jan 11, 2011 at 8:07 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>>>>>>
>>>>>>>> It should work. Is there anything already listening on port 8022?
>>>>>>>>
>>>>>>>> $ netstat -a -n -p | grep 8022
>>>>>>>>
>>>>>>>> Are you also sure that they're not both loading the same config
>>>>>>>> file?
>>>>>>>>
>>>>>>>> Regards,
>>>>>>>> -Steve
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On 11 January 2011 14:32, Sam <u2nsam at gmail.com> wrote:
>>>>>>>> > the scenario is i have 2 ips on 1 server for 2 FS instances;
>>>>>>>> >
>>>>>>>> > 192.168.2.1
>>>>>>>> > 192.168.2.2
>>>>>>>> >
>>>>>>>> > and the parameters i have set is:-
>>>>>>>> >
>>>>>>>> > for
>>>>>>>> > 192.168.2.1:-
>>>>>>>> >    <param name="listen-ip" value="127.0.0.1"/>
>>>>>>>> >    <param name="listen-port" value="8021"/>
>>>>>>>> > for
>>>>>>>> > 192.168.2.2:-
>>>>>>>> >    <param name="listen-ip" value="127.0.0.1"/>
>>>>>>>> >    <param name="listen-port" value="8022"/>
>>>>>>>> >
>>>>>>>> > Ideally it should work  but i am getting console for only
>>>>>>>> 192.168.2.1 FS .
>>>>>>>> >
>>>>>>>> >
>>>>>>>> > Regards
>>>>>>>> > Sam
>>>>>>>> >
>>>>>>>> > On Tue, Jan 11, 2011 at 4:20 PM, Steven Ayre <steveayre at gmail.com>
>>>>>>>> wrote:
>>>>>>>> >>
>>>>>>>> >>    <param name="listen-ip" value="127.0.0.1"/>
>>>>>>>> >>    <param name="listen-port" value="8021"/>
>>>>>>>> >>
>>>>>>>> >> You can bind both to port 8021 on their individual IPs, or
>>>>>>>> different
>>>>>>>> >> ports on the same IP.
>>>>>>>> >>
>>>>>>>> >> A listen IP of 0.0.0.0 will mean any IP.
>>>>>>>> >>
>>>>>>>> >> -Steve
>>>>>>>> >>
>>>>>>>> >> On 11 January 2011 10:44, Sam <u2nsam at gmail.com> wrote:
>>>>>>>> >> > A query,
>>>>>>>> >> >
>>>>>>>> >> > I have 2 FS running on one server on 2 different ips,
>>>>>>>> >> > so when i do fs_cli going to respective bins , i see console of
>>>>>>>> only the
>>>>>>>> >> > first server.
>>>>>>>> >> >
>>>>>>>> >> > Is there any way to get the console of both the FS on the same
>>>>>>>> server .
>>>>>>>> >> > I tried changing the port of event socket to 8022 but it donot
>>>>>>>> works.
>>>>>>>> >> > <param name="listen-port" value="8022"/>
>>>>>>>> >> >
>>>>>>>> >> > Is there some method to start the console of both the
>>>>>>>> instances.
>>>>>>>> >> >
>>>>>>>> >> > Regds
>>>>>>>> >> > Sam
>>>>>>>> >> >
>>>>>>>> >> > _______________________________________________
>>>>>>>> >> > FreeSWITCH-users mailing list
>>>>>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >> > UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> >> > http://www.freeswitch.org
>>>>>>>> >> >
>>>>>>>> >> >
>>>>>>>> >>
>>>>>>>> >> _______________________________________________
>>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> >> http://www.freeswitch.org
>>>>>>>> >
>>>>>>>> >
>>>>>>>> > _______________________________________________
>>>>>>>> > FreeSWITCH-users mailing list
>>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> > UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> > http://www.freeswitch.org
>>>>>>>> >
>>>>>>>> >
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
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From u2nsam at gmail.com  Thu Jan 13 07:05:15 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 13 Jan 2011 09:35:15 +0530
Subject: [Freeswitch-users] deflect question
In-Reply-To: <01AF6B571F2B4BAE944C61206687FB19@e1705>
References: <01AF6B571F2B4BAE944C61206687FB19@e1705>
Message-ID: <AANLkTimGNzQwuWwerGNgnDzzpnyzdMiZn8y9VjkU_AqR@mail.gmail.com>

it originates a call with refer message,when the call lands on the different
server does it releases the first leg ?
and the call bridges the leg b without the leg a on the first server ?

C --> A --> B

A-> server 1
B -> server 2
C-> call initiator

After deflect from A to B will the call flow be C-->B with out A ?

Regards
Sam

On Thu, Jan 13, 2011 at 2:38 AM, Madovsky <infos at madovsky.org> wrote:

>
> if I use
>
>  <action application="deflect" data="sip:someone at otherFS.com"/>
>
> in the otherFS dialplan should I create
>
> an extension with "someone" condition only ?
>
>
>
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Thu Jan 13 07:49:59 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 12 Jan 2011 23:49:59 -0500
Subject: [Freeswitch-users] deflect question
Message-ID: <046F9F9F93F2462C86A467D2B92E5E7A@e1705>

I try with different SIP phones and apparently
it works with x-lite v4.
Is REFER message usually supported as standard ?

  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, January 12, 2011 10:34 PM
  Subject: Re: [Freeswitch-users] deflect question


  I'm sure also that the call is answered before deflect
    ----- Original Message ----- 
    From: Brian West 
    To: FreeSWITCH Users Help 
    Sent: Wednesday, January 12, 2011 8:40 PM
    Subject: Re: [Freeswitch-users] deflect question


    try making it sip:something at blah 


    /b


    On Jan 12, 2011, at 4:50 PM, Madovsky wrote:


      I'm stucked.

      in the log
      the call stalls with this line

      EXECUTE sofia/internal/5555555555555 at myfs.org deflect(9999999999999 at otherfs.org)
      big silenece and nothing happens

      the otherFS doesn't receive anything.

      I tried to call manually the deflect sip and it works.

      Did I forget something ?


      thanks




----------------------------------------------------------------------------


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From infos at madovsky.org  Thu Jan 13 07:58:17 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 12 Jan 2011 23:58:17 -0500
Subject: [Freeswitch-users] deflect question
References: <01AF6B571F2B4BAE944C61206687FB19@e1705>
	<AANLkTimGNzQwuWwerGNgnDzzpnyzdMiZn8y9VjkU_AqR@mail.gmail.com>
Message-ID: <5727624BB9934445B7023220C708B9C7@e1705>

no,
but in fact I just release that my sip phone doesnt' support RFC 3515...
  ----- Original Message ----- 
  From: Sam 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, January 12, 2011 11:05 PM
  Subject: Re: [Freeswitch-users] deflect question


  it originates a call with refer message,when the call lands on the different server does it releases the first leg ?
  and the call bridges the leg b without the leg a on the first server ?

  C --> A --> B

  A-> server 1
  B -> server 2
  C-> call initiator

  After deflect from A to B will the call flow be C-->B with out A ?

  Regards
  Sam


  On Thu, Jan 13, 2011 at 2:38 AM, Madovsky <infos at madovsky.org> wrote:


    if I use
 <action application="deflect" data="sip:someone at otherFS.com"/>
in the otherFS dialplan should I createan extension with "someone" condition only ? Thanks
    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org






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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Thu Jan 13 08:00:21 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 13 Jan 2011 00:00:21 -0500
Subject: [Freeswitch-users] console
References: <6ECAF1527329364583AB525CF34ABF950B31A4F9@ms.kallback.com><AANLkTinXiaVGr5bneGCfvXUmbEnJ5Ggjcmvim=Wjqa5O@mail.gmail.com><2182C8734A5F48D5858AD492D426DEA0@e1705>
	<AANLkTikgopv7pY4nnhH3h=6bmp7rGe5qpP5FmiUePitb@mail.gmail.com>
Message-ID: <A9220D89047744C9A189BA74610DDE52@e1705>

be sure also that you don't have any Fw or IPS rules
that don't block any port/protocol/other on local net.
  ----- Original Message ----- 
  From: Sam 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, January 12, 2011 10:59 PM
  Subject: Re: [Freeswitch-users] console


  Yes both FS are complied with the prefixes and are running quiet independently , the only problem is with the fs_cli .

  Regards
  Sam




  On Thu, Jan 13, 2011 at 9:02 AM, Madovsky <infos at madovsky.org> wrote:

    did you try to compile your two FS
    with different prefix and .pid path ?
    as this you'll be sure that FS are running independtly.
    I remember (maybe wrongly) to have this kind of problem before.....
      ----- Original Message ----- 
      From: Sam 
      To: FreeSWITCH Users Help 
      Sent: Wednesday, January 12, 2011 10:16 PM
      Subject: Re: [Freeswitch-users] console


      Yes the netstat shows the ports 8081 & 8082 not used 
      Justin, as said earlier i have 2 FS instances running on same server and having below config:-
      for
      > 192.168.2.1:-
      >    <param name="listen-ip" value="127.0.0.1"/>
      >    <param name="listen-port" value="8021"/>
      > for
      > 192.168.2.2:-
      >    <param name="listen-ip" value="127.0.0.1"/>
      >    <param name="listen-port" value="8022"/>

      Here I tried all the permutation combination changing the IP port but no sucess.

      Regds
      Sam


      On Wed, Jan 12, 2011 at 11:25 PM, JRichey <jrichey at itltd.net> wrote:

        Did you ever try using netstat like Steven suggested?  What do you get as output for "netstat -tunlp"?

        In one of your commands below you show 127.0.0.2 instead of 127.0.0.1 so if that's not a typo you may want to try it again.

        -Justin

          -----Original Message-----
          From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of Sam
          Sent: Tuesday, January 11, 2011 9:27 PM
          To: FreeSWITCH Users Help
          Subject: Re: [Freeswitch-users] console


          Tried this below and received :-


          # /usr/local/fs_2/bin/fs_cli -H 127.0.0.2 -P 8082
          [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection Error]

          tried by creating profile:-
          # /usr/local/fs_2/bin/fs_cli profile1
          [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection Error]

          By port:
          # /usr/local/fs_2/bin/fs_cli -P 8082
          [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection Error]



netstat -nlp | grep ; ports 8081 & 8082 are availableBut i could get a console for other server by
          # /usr/local/fs_2/bin/fs_cli

          Regards
          Sam



          On Tue, Jan 11, 2011 at 10:54 PM, Steven Ayre <steveayre at gmail.com> wrote:

            Actually, correction it does - just not the event_socket.conf.xml one. It'll read .fs_cli_conf in your home directory if it exists, but that isn't created by default - you create it yourself if you want to use it (it's optional). The command line arguments -H and -P -will- override the config file though.

            Are you using a capital P? -p is password while -P is port. If there's no password on the event socket you'd get no error from using a small p by accident.

            -Steve 







            On 11 January 2011 16:59, Steven Ayre <steveayre at gmail.com> wrote:

              It does not read any config file. 






              On 11 January 2011 16:56, Sam <u2nsam at gmail.com> wrote:

                the port and ip donot work for me,
                is it that the fs_cli is not reading the config from 192.168.2.2 but it is reading the config only of 192.168.2.1, though its in the different [FS_1 & FS_2] path where i am executing.

                Regds
                Sam 




                On Tue, Jan 11, 2011 at 9:38 PM, Steven Ayre <steveayre at gmail.com> wrote:

                  /usr/local/FS_1/bin/fs_cli -P 8021
                  /usr/local/FS_2/bin/fs_cli -P 8022

                  fs_cll doesn't read any config file. It's not part of the FS server at all, you can have it on a different machine that doesn't have FS installed. It entirely relies on the arguments to control where to connect to.

                  -Steve 




                  On 11 January 2011 15:04, Sam <u2nsam at gmail.com> wrote:

                    Something more here ... i am getting the console for 192.168.2.1 every time i do fs_cli on both instances .

                    like 
                    /usr/local/FS_1/bin/fs_cli
                    /usr/localFS_2/bin/fs_cli
                    i get the console for the 1st server only

                    the 2 server are listing to 2 different ips .

                    Regds
                    Sam 



                    On Tue, Jan 11, 2011 at 8:07 PM, Steven Ayre <steveayre at gmail.com> wrote:

                      It should work. Is there anything already listening on port 8022?

                      $ netstat -a -n -p | grep 8022

                      Are you also sure that they're not both loading the same config file?

                      Regards,
                      -Steve




                      On 11 January 2011 14:32, Sam <u2nsam at gmail.com> wrote:
                      > the scenario is i have 2 ips on 1 server for 2 FS instances;
                      >
                      > 192.168.2.1
                      > 192.168.2.2
                      >
                      > and the parameters i have set is:-
                      >
                      > for
                      > 192.168.2.1:-
                      >    <param name="listen-ip" value="127.0.0.1"/>
                      >    <param name="listen-port" value="8021"/>
                      > for
                      > 192.168.2.2:-
                      >    <param name="listen-ip" value="127.0.0.1"/>
                      >    <param name="listen-port" value="8022"/>
                      >
                      > Ideally it should work  but i am getting console for only 192.168.2.1 FS .
                      >
                      >
                      > Regards
                      > Sam
                      >
                      > On Tue, Jan 11, 2011 at 4:20 PM, Steven Ayre <steveayre at gmail.com> wrote:
                      >>
                      >>    <param name="listen-ip" value="127.0.0.1"/>
                      >>    <param name="listen-port" value="8021"/>
                      >>
                      >> You can bind both to port 8021 on their individual IPs, or different
                      >> ports on the same IP.
                      >>
                      >> A listen IP of 0.0.0.0 will mean any IP.
                      >>
                      >> -Steve
                      >>
                      >> On 11 January 2011 10:44, Sam <u2nsam at gmail.com> wrote:
                      >> > A query,
                      >> >
                      >> > I have 2 FS running on one server on 2 different ips,
                      >> > so when i do fs_cli going to respective bins , i see console of only the
                      >> > first server.
                      >> >
                      >> > Is there any way to get the console of both the FS on the same server .
                      >> > I tried changing the port of event socket to 8022 but it donot works.
                      >> > <param name="listen-port" value="8022"/>
                      >> >
                      >> > Is there some method to start the console of both the instances.
                      >> >
                      >> > Regds
                      >> > Sam
                      >> >
                      >> > _______________________________________________
                      >> > FreeSWITCH-users mailing list
                      >> > FreeSWITCH-users at lists.freeswitch.org
                      >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
                      >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
                      >> > http://www.freeswitch.org
                      >> >
                      >> >
                      >>
                      >> _______________________________________________
                      >> FreeSWITCH-users mailing list
                      >> FreeSWITCH-users at lists.freeswitch.org
                      >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
                      >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
                      >> http://www.freeswitch.org
                      >
                      >
                      > _______________________________________________
                      > FreeSWITCH-users mailing list
                      > FreeSWITCH-users at lists.freeswitch.org
                      > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
                      > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
                      > http://www.freeswitch.org
                      >
                      >

                      _______________________________________________
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        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From vermeulen.deon at gmail.com  Thu Jan 13 08:31:27 2011
From: vermeulen.deon at gmail.com (Deon Vermeulen)
Date: Thu, 13 Jan 2011 07:31:27 +0200
Subject: [Freeswitch-users] Freeswitch eLearning
Message-ID: <E04DC7C5-478D-4404-89C0-E273D4F5A0E5@gmail.com>

Hi


Does anyone know of an online training course for Freeswitch?



Thank you very much

Kind Regards
Deon


From u2nsam at gmail.com  Thu Jan 13 08:50:56 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 13 Jan 2011 11:20:56 +0530
Subject: [Freeswitch-users] console
In-Reply-To: <A9220D89047744C9A189BA74610DDE52@e1705>
References: <6ECAF1527329364583AB525CF34ABF950B31A4F9@ms.kallback.com>
	<AANLkTinXiaVGr5bneGCfvXUmbEnJ5Ggjcmvim=Wjqa5O@mail.gmail.com>
	<2182C8734A5F48D5858AD492D426DEA0@e1705>
	<AANLkTikgopv7pY4nnhH3h=6bmp7rGe5qpP5FmiUePitb@mail.gmail.com>
	<A9220D89047744C9A189BA74610DDE52@e1705>
Message-ID: <AANLkTi=djXbjZbNtC6D-N016O3xg9Fz7F_xpiT_HE_y_@mail.gmail.com>

no its all opened ... i am getting the console for one instans of FS but not
for the another

On Thu, Jan 13, 2011 at 10:30 AM, Madovsky <infos at madovsky.org> wrote:

>  be sure also that you don't have any Fw or IPS rules
> that don't block any port/protocol/other on local net.
>
> ----- Original Message -----
> *From:* Sam <u2nsam at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, January 12, 2011 10:59 PM
> *Subject:* Re: [Freeswitch-users] console
>
> Yes both FS are complied with the prefixes and are running quiet
> independently , the only problem is with the fs_cli .
>
> Regards
> Sam
>
>
>
> On Thu, Jan 13, 2011 at 9:02 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  did you try to compile your two FS
>> with different prefix and .pid path ?
>> as this you'll be sure that FS are running independtly.
>> I remember (maybe wrongly) to have this kind of problem before.....
>>
>>   ----- Original Message -----
>> *From:* Sam <u2nsam at gmail.com>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Sent:* Wednesday, January 12, 2011 10:16 PM
>> *Subject:* Re: [Freeswitch-users] console
>>
>> Yes the netstat shows the ports 8081 & 8082 not used
>> Justin, as said earlier i have 2 FS instances running on same server and
>> having below config:-
>> for
>> > 192.168.2.1:-
>> >    <param name="listen-ip" value="127.0.0.1"/>
>> >    <param name="listen-port" value="8021"/>
>> > for
>> > 192.168.2.2:-
>> >    <param name="listen-ip" value="127.0.0.1"/>
>> >    <param name="listen-port" value="8022"/>
>>
>> Here I tried all the permutation combination changing the IP port but no
>> sucess.
>>
>> Regds
>> Sam
>>
>> On Wed, Jan 12, 2011 at 11:25 PM, JRichey <jrichey at itltd.net> wrote:
>>
>>>  Did you ever try using netstat like Steven suggested?  What do you get
>>> as output for "netstat -tunlp"?
>>>
>>> In one of your commands below you show 127.0.0.2 instead of 127.0.0.1 so
>>> if that's not a typo you may want to try it again.
>>>
>>> -Justin
>>>
>>>
>>> -----Original Message-----
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org]*On Behalf Of *Sam
>>> *Sent:* Tuesday, January 11, 2011 9:27 PM
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* Re: [Freeswitch-users] console
>>>
>>> Tried this below and received :-
>>>
>>>
>>> # /usr/local/fs_2/bin/fs_cli -H 127.0.0.2 -P 8082
>>> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
>>> Error]
>>>
>>> tried by creating profile:-
>>> # /usr/local/fs_2/bin/fs_cli profile1
>>> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
>>> Error]
>>>
>>> By port:
>>> # /usr/local/fs_2/bin/fs_cli -P 8082
>>> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
>>> Error]
>>>
>>>
>>> *netstat -nlp | grep ; ports 8081 & 8082 are available*
>>>
>>> But i could get a console for other server by
>>> # /usr/local/fs_2/bin/fs_cli
>>>
>>> Regards
>>> Sam
>>>
>>>
>>> On Tue, Jan 11, 2011 at 10:54 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>>
>>>> Actually, correction it does - just not the event_socket.conf.xml one.
>>>> It'll read .fs_cli_conf in your home directory if it exists, but that isn't
>>>> created by default - you create it yourself if you want to use it (it's
>>>> optional). The command line arguments -H and -P -will- override the config
>>>> file though.
>>>>
>>>> Are you using a capital P? -p is password while -P is port. If there's
>>>> no password on the event socket you'd get no error from using a small p by
>>>> accident.
>>>>
>>>> -Steve
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On 11 January 2011 16:59, Steven Ayre <steveayre at gmail.com> wrote:
>>>>
>>>>> It does not read any config file.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On 11 January 2011 16:56, Sam <u2nsam at gmail.com> wrote:
>>>>>
>>>>>> the port and ip donot work for me,
>>>>>> is it that the fs_cli is not reading the config from 192.168.2.2 but
>>>>>> it is reading the config only of 192.168.2.1, though its in the different
>>>>>> [FS_1 & FS_2] path where i am executing.
>>>>>>
>>>>>> Regds
>>>>>> Sam
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Jan 11, 2011 at 9:38 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>>>>>
>>>>>>> /usr/local/FS_1/bin/fs_cli -P 8021
>>>>>>> /usr/local/FS_2/bin/fs_cli -P 8022
>>>>>>>
>>>>>>> fs_cll doesn't read any config file. It's not part of the FS server
>>>>>>> at all, you can have it on a different machine that doesn't have FS
>>>>>>> installed. It entirely relies on the arguments to control where to connect
>>>>>>> to.
>>>>>>>
>>>>>>> -Steve
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On 11 January 2011 15:04, Sam <u2nsam at gmail.com> wrote:
>>>>>>>
>>>>>>>> Something more here ... i am getting the console for 192.168.2.1
>>>>>>>> every time i do fs_cli on both instances .
>>>>>>>>
>>>>>>>> like
>>>>>>>> /usr/local/FS_1/bin/fs_cli
>>>>>>>> /usr/localFS_2/bin/fs_cli
>>>>>>>> i get the console for the 1st server only
>>>>>>>>
>>>>>>>> the 2 server are listing to 2 different ips .
>>>>>>>>
>>>>>>>> Regds
>>>>>>>> Sam
>>>>>>>>
>>>>>>>>
>>>>>>>> On Tue, Jan 11, 2011 at 8:07 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>>>>>>>
>>>>>>>>> It should work. Is there anything already listening on port 8022?
>>>>>>>>>
>>>>>>>>> $ netstat -a -n -p | grep 8022
>>>>>>>>>
>>>>>>>>> Are you also sure that they're not both loading the same config
>>>>>>>>> file?
>>>>>>>>>
>>>>>>>>> Regards,
>>>>>>>>> -Steve
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On 11 January 2011 14:32, Sam <u2nsam at gmail.com> wrote:
>>>>>>>>> > the scenario is i have 2 ips on 1 server for 2 FS instances;
>>>>>>>>> >
>>>>>>>>> > 192.168.2.1
>>>>>>>>> > 192.168.2.2
>>>>>>>>> >
>>>>>>>>> > and the parameters i have set is:-
>>>>>>>>> >
>>>>>>>>> > for
>>>>>>>>> > 192.168.2.1:-
>>>>>>>>> >    <param name="listen-ip" value="127.0.0.1"/>
>>>>>>>>> >    <param name="listen-port" value="8021"/>
>>>>>>>>> > for
>>>>>>>>> > 192.168.2.2:-
>>>>>>>>> >    <param name="listen-ip" value="127.0.0.1"/>
>>>>>>>>> >    <param name="listen-port" value="8022"/>
>>>>>>>>> >
>>>>>>>>> > Ideally it should work  but i am getting console for only
>>>>>>>>> 192.168.2.1 FS .
>>>>>>>>> >
>>>>>>>>> >
>>>>>>>>> > Regards
>>>>>>>>> > Sam
>>>>>>>>> >
>>>>>>>>> > On Tue, Jan 11, 2011 at 4:20 PM, Steven Ayre <
>>>>>>>>> steveayre at gmail.com> wrote:
>>>>>>>>> >>
>>>>>>>>> >>    <param name="listen-ip" value="127.0.0.1"/>
>>>>>>>>> >>    <param name="listen-port" value="8021"/>
>>>>>>>>> >>
>>>>>>>>> >> You can bind both to port 8021 on their individual IPs, or
>>>>>>>>> different
>>>>>>>>> >> ports on the same IP.
>>>>>>>>> >>
>>>>>>>>> >> A listen IP of 0.0.0.0 will mean any IP.
>>>>>>>>> >>
>>>>>>>>> >> -Steve
>>>>>>>>> >>
>>>>>>>>> >> On 11 January 2011 10:44, Sam <u2nsam at gmail.com> wrote:
>>>>>>>>> >> > A query,
>>>>>>>>> >> >
>>>>>>>>> >> > I have 2 FS running on one server on 2 different ips,
>>>>>>>>> >> > so when i do fs_cli going to respective bins , i see console
>>>>>>>>> of only the
>>>>>>>>> >> > first server.
>>>>>>>>> >> >
>>>>>>>>> >> > Is there any way to get the console of both the FS on the same
>>>>>>>>> server .
>>>>>>>>> >> > I tried changing the port of event socket to 8022 but it donot
>>>>>>>>> works.
>>>>>>>>> >> > <param name="listen-port" value="8022"/>
>>>>>>>>> >> >
>>>>>>>>> >> > Is there some method to start the console of both the
>>>>>>>>> instances.
>>>>>>>>> >> >
>>>>>>>>> >> > Regds
>>>>>>>>> >> > Sam
>>>>>>>>> >> >
>>>>>>>>> >> > _______________________________________________
>>>>>>>>> >> > FreeSWITCH-users mailing list
>>>>>>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> >> > UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> >> > http://www.freeswitch.org
>>>>>>>>> >> >
>>>>>>>>> >> >
>>>>>>>>> >>
>>>>>>>>> >> _______________________________________________
>>>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> >> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> >> http://www.freeswitch.org
>>>>>>>>> >
>>>>>>>>> >
>>>>>>>>> > _______________________________________________
>>>>>>>>> > FreeSWITCH-users mailing list
>>>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> > UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> > http://www.freeswitch.org
>>>>>>>>> >
>>>>>>>>> >
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>  ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Thu Jan 13 10:43:06 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 13 Jan 2011 13:13:06 +0530
Subject: [Freeswitch-users] Compile error
Message-ID: <AANLkTimuZWE79y15=uT7RxD9PPEWOCxuYFjs5g_Qj2gG@mail.gmail.com>

Hello ,

Got the compile error for latest git ,





making all mod_spandsp

Making all in src

libtool: compile:  gcc -DHAVE_CONFIG_H -I. -I.. -I/usr/include/libxml2
-I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -msse2
-std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings
-Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden
-DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -MT v17rx.lo -MD -MP
-MF .deps/v17rx.Tpo -c v17rx.c  -fPIC -DPIC -o v17rx.o

v17rx.c: In function ?v17_rx_symbol_timing_correction?:

v17rx.c:169: error: ?RX_PULSESHAPER_COEFF_SETS? undeclared (first use in
this function)

v17rx.c:169: error: (Each undeclared identifier is reported only once

v17rx.c:169: error: for each function it appears in.)

v17rx.c: In function ?equalizer_restore?:

v17rx.c:229: error: ?RX_PULSESHAPER_COEFF_SETS? undeclared (first use in
this function)

v17rx.c: In function ?equalizer_reset?:

v17rx.c:250: error: ?RX_PULSESHAPER_COEFF_SETS? undeclared (first use in
this function)

v17rx.c: In function ?v17_rx?:

v17rx.c:1151: error: ?RX_PULSESHAPER_COEFF_SETS? undeclared (first use in
this function)

v17rx.c:1162: error: ?rx_pulseshaper_re? undeclared (first use in this
function)

v17rx.c:1181: error: ?RX_PULSESHAPER_GAIN? undeclared (first use in this
function)

v17rx.c:1198: error: ?rx_pulseshaper_im? undeclared (first use in this
function)

v17rx.c: In function ?v17_rx_fillin?:

v17rx.c:1235: error: ?RX_PULSESHAPER_COEFF_SETS? undeclared (first use in
this function)

v17rx.c: In function ?v17_rx_restart?:

v17rx.c:1368: error: ?RX_PULSESHAPER_GAIN? undeclared (first use in this
function)

make[7]: *** [v17rx.lo] Error 1

make[6]: *** [all] Error 2

make[5]: *** [all-recursive] Error 1

make[4]: *** [/usr/local/src/freeswitch/libs/spandsp/src/libspandsp.la]
Error 2

make[3]: *** [mod_spandsp-all] Error 1

make[2]: *** [all-recursive] Error 1

make[1]: *** [all-recursive] Error 1

make: *** [all] Error 2


Regds
Sam
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From david.ponzone at ipeva.fr  Thu Jan 13 11:16:48 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 13 Jan 2011 09:16:48 +0100
Subject: [Freeswitch-users] Freeswitch eLearning
In-Reply-To: <E04DC7C5-478D-4404-89C0-E273D4F5A0E5@gmail.com>
References: <E04DC7C5-478D-4404-89C0-E273D4F5A0E5@gmail.com>
Message-ID: <033F9ACA-07E3-4E91-8C5C-8240A8182F82@ipeva.fr>

Deon,

I think the lack of answer to your first mail obivously meant that either:
-noone has any answers
-there is no such thing
-noone with an answer had time to reply yet

So it's probably useless to resend the same mail.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 13/01/2011 ? 06:31, Deon Vermeulen a ?crit :

> Hi
> 
> 
> Does anyone know of an online training course for Freeswitch?
> 
> 
> 
> Thank you very much
> 
> Kind Regards
> Deon
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From steveayre at gmail.com  Thu Jan 13 11:38:23 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 13 Jan 2011 08:38:23 +0000
Subject: [Freeswitch-users] console
In-Reply-To: <AANLkTinXiaVGr5bneGCfvXUmbEnJ5Ggjcmvim=Wjqa5O@mail.gmail.com>
References: <6ECAF1527329364583AB525CF34ABF950B31A4F9@ms.kallback.com>
	<AANLkTinXiaVGr5bneGCfvXUmbEnJ5Ggjcmvim=Wjqa5O@mail.gmail.com>
Message-ID: <B17A6BC7-4704-436B-A4F0-A62A687E3D5F@gmail.com>

Neither are listening yet you can connect to one if them? That doesn't sound right?

Steve on iPhone

On 13 Jan 2011, at 03:16, Sam <u2nsam at gmail.com> wrote:

> Yes the netstat shows the ports 8081 & 8082 not used 
> Justin, as said earlier i have 2 FS instances running on same server and having below config:-
> for
> > 192.168.2.1:-
> >    <param name="listen-ip" value="127.0.0.1"/>
> >    <param name="listen-port" value="8021"/>
> > for
> > 192.168.2.2:-
> >    <param name="listen-ip" value="127.0.0.1"/>
> >    <param name="listen-port" value="8022"/>
> 
> Here I tried all the permutation combination changing the IP port but no sucess.
> 
> Regds
> Sam
> 
> On Wed, Jan 12, 2011 at 11:25 PM, JRichey <jrichey at itltd.net> wrote:
> Did you ever try using netstat like Steven suggested?  What do you get as output for "netstat -tunlp"?
>  
> In one of your commands below you show 127.0.0.2 instead of 127.0.0.1 so if that's not a typo you may want to try it again.
>  
> -Justin
>  
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of Sam
> Sent: Tuesday, January 11, 2011 9:27 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] console
> 
> Tried this below and received :-
> 
> 
> # /usr/local/fs_2/bin/fs_cli -H 127.0.0.2 -P 8082
> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection Error]
> 
> tried by creating profile:-
> # /usr/local/fs_2/bin/fs_cli profile1
> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection Error]
> 
> By port:
> # /usr/local/fs_2/bin/fs_cli -P 8082
> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection Error]
> 
> 
> netstat -nlp | grep ; ports 8081 & 8082 are available
> But i could get a console for other server by
> # /usr/local/fs_2/bin/fs_cli
> 
> Regards
> Sam
> 
> 
> On Tue, Jan 11, 2011 at 10:54 PM, Steven Ayre <steveayre at gmail.com> wrote:
> Actually, correction it does - just not the event_socket.conf.xml one. It'll read .fs_cli_conf in your home directory if it exists, but that isn't created by default - you create it yourself if you want to use it (it's optional). The command line arguments -H and -P -will- override the config file though.
> 
> Are you using a capital P? -p is password while -P is port. If there's no password on the event socket you'd get no error from using a small p by accident.
> 
> -Steve
> 
> 
> 
> 
> 
> 
> On 11 January 2011 16:59, Steven Ayre <steveayre at gmail.com> wrote:
> It does not read any config file.
> 
> 
> 
> 
> 
> On 11 January 2011 16:56, Sam <u2nsam at gmail.com> wrote:
> the port and ip donot work for me,
> is it that the fs_cli is not reading the config from 192.168.2.2 but it is reading the config only of 192.168.2.1, though its in the different [FS_1 & FS_2] path where i am executing.
> 
> Regds
> Sam
> 
> 
> 
> On Tue, Jan 11, 2011 at 9:38 PM, Steven Ayre <steveayre at gmail.com> wrote:
> /usr/local/FS_1/bin/fs_cli -P 8021
> /usr/local/FS_2/bin/fs_cli -P 8022
> 
> fs_cll doesn't read any config file. It's not part of the FS server at all, you can have it on a different machine that doesn't have FS installed. It entirely relies on the arguments to control where to connect to.
> 
> -Steve
> 
> 
> 
> On 11 January 2011 15:04, Sam <u2nsam at gmail.com> wrote:
> Something more here ... i am getting the console for 192.168.2.1 every time i do fs_cli on both instances .
> 
> like 
> /usr/local/FS_1/bin/fs_cli
> /usr/localFS_2/bin/fs_cli
> i get the console for the 1st server only
> 
> the 2 server are listing to 2 different ips .
> 
> Regds
> Sam
> 
> 
> On Tue, Jan 11, 2011 at 8:07 PM, Steven Ayre <steveayre at gmail.com> wrote:
> It should work. Is there anything already listening on port 8022?
> 
> $ netstat -a -n -p | grep 8022
> 
> Are you also sure that they're not both loading the same config file?
> 
> Regards,
> -Steve
> 
> 
> 
> On 11 January 2011 14:32, Sam <u2nsam at gmail.com> wrote:
> > the scenario is i have 2 ips on 1 server for 2 FS instances;
> >
> > 192.168.2.1
> > 192.168.2.2
> >
> > and the parameters i have set is:-
> >
> > for
> > 192.168.2.1:-
> >    <param name="listen-ip" value="127.0.0.1"/>
> >    <param name="listen-port" value="8021"/>
> > for
> > 192.168.2.2:-
> >    <param name="listen-ip" value="127.0.0.1"/>
> >    <param name="listen-port" value="8022"/>
> >
> > Ideally it should work  but i am getting console for only 192.168.2.1 FS .
> >
> >
> > Regards
> > Sam
> >
> > On Tue, Jan 11, 2011 at 4:20 PM, Steven Ayre <steveayre at gmail.com> wrote:
> >>
> >>    <param name="listen-ip" value="127.0.0.1"/>
> >>    <param name="listen-port" value="8021"/>
> >>
> >> You can bind both to port 8021 on their individual IPs, or different
> >> ports on the same IP.
> >>
> >> A listen IP of 0.0.0.0 will mean any IP.
> >>
> >> -Steve
> >>
> >> On 11 January 2011 10:44, Sam <u2nsam at gmail.com> wrote:
> >> > A query,
> >> >
> >> > I have 2 FS running on one server on 2 different ips,
> >> > so when i do fs_cli going to respective bins , i see console of only the
> >> > first server.
> >> >
> >> > Is there any way to get the console of both the FS on the same server .
> >> > I tried changing the port of event socket to 8022 but it donot works.
> >> > <param name="listen-port" value="8022"/>
> >> >
> >> > Is there some method to start the console of both the instances.
> >> >
> >> > Regds
> >> > Sam
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From vermeulen.deon at gmail.com  Thu Jan 13 12:21:10 2011
From: vermeulen.deon at gmail.com (Deon Vermeulen)
Date: Thu, 13 Jan 2011 11:21:10 +0200
Subject: [Freeswitch-users] Freeswitch eLearning
In-Reply-To: <033F9ACA-07E3-4E91-8C5C-8240A8182F82@ipeva.fr>
References: <E04DC7C5-478D-4404-89C0-E273D4F5A0E5@gmail.com>
	<033F9ACA-07E3-4E91-8C5C-8240A8182F82@ipeva.fr>
Message-ID: <69E76AD8-7363-49D1-AC0B-7A5324198F20@gmail.com>

David

Thanks for replying.

I sent the mail again, as for some reason I saw it lying in my Drafts box and thought it was never sent.


Rgds
Deon


On Jan 13, 2011, at 10:16 AM, David Ponzone wrote:

> Deon,
> 
> I think the lack of answer to your first mail obivously meant that either:
> -noone has any answers
> -there is no such thing
> -noone with an answer had time to reply yet
> 
> So it's probably useless to resend the same mail.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 13/01/2011 ? 06:31, Deon Vermeulen a ?crit :
> 
>> Hi
>> 
>> 
>> Does anyone know of an online training course for Freeswitch?
>> 
>> 
>> 
>> Thank you very much
>> 
>> Kind Regards
>> Deon
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From u2nsam at gmail.com  Thu Jan 13 12:52:30 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 13 Jan 2011 15:22:30 +0530
Subject: [Freeswitch-users] metadigits
Message-ID: <AANLkTik6yP0Kry1n6TyjakGg=9WfvF0uGXGNzdJ2ZHUx@mail.gmail.com>

Any Ideas,

Why I am getting this logs below:-

2011-01-13 15:13:51.821614 [DEBUG] switch_rtp.c:3105 RTP RECV DTMF #:960
2011-01-13 15:13:53.001254 [DEBUG] switch_rtp.c:3105 RTP RECV DTMF 3:1280
2011-01-13 15:13:53.001254 [WARNING] switch_ivr_async.c:2882 sofia/internal/
7001 at 192.168.2.190 Ignoring meta digit '3' not mapped


I have in my dialplan :-


<action application="set" data="bind_meta_key=#"/>
<action application="bind_meta_app" data="1 ab s execute_extension::dx XML
features"/>
<action application="bind_meta_app" data="2 ab s execute_extension::att_xfer
XML features"/>
<action application="bind_meta_app" data="3 ab s
record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>



the other (1&2) meta keys are working.


Regds
Sam
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From bernhard.suttner at winet.ch  Thu Jan 13 15:40:21 2011
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Thu, 13 Jan 2011 13:40:21 +0100
Subject: [Freeswitch-users] Outgoing registrations within a fs pool
Message-ID: <3ff41827-ea0f-4302-8d70-c7ee16feec83@winet.ch>

Hi,

I want to do outgoing registrations (gateways) to a provider. The scenario is like that:

Provider
   \
    \

SBC (Kamailio)
      + 
+-----+-------+
|     |       |
FS1  FS2     FS3

FS does share the same database with ODBC. The question is now, could FS1 register to the provider that the call will reach the SBC later and then maybe forwaded to FS3 instead of FS1? The best way would be, that the "pool" of FreeSWITCH Server does register to the Provider. So, one of the FreeSWITCH server decides that it is necessary to register to the provider again (because of expires) and all the other will not register for this time. Next time maybe another server does register. It should be not bound to one specific FreeSWITCH server. 

Thanks in advance!

Best regards,
Bernhard




From steveayre at gmail.com  Thu Jan 13 17:06:19 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 13 Jan 2011 14:06:19 +0000
Subject: [Freeswitch-users] Sofia fails to detect offline gateway
In-Reply-To: <AANLkTimJzFBcMu+npWEs1_fW0UDUdhpaBmfpqEADPteL@mail.gmail.com>
References: <AANLkTim3PeS3fELJPQ21N38c9v5p8J0wGNkBrUvc9R=t@mail.gmail.com>
	<B16E93D7-3578-4071-8899-DDE70DB4D1DC@freeswitch.org>
	<AANLkTin2j-+Y11_x4bs0+M6UsVKUN14fXHoZ262vFu26@mail.gmail.com>
	<AANLkTimTutKEvOg+9+MwrxBOp3yJiEax0S-w_5kkvcwB@mail.gmail.com>
	<AANLkTikMrAzgwE5Jm4qszE_pYXxqO9bF6t1Qtwi9eoLv@mail.gmail.com>
	<AANLkTimJzFBcMu+npWEs1_fW0UDUdhpaBmfpqEADPteL@mail.gmail.com>
Message-ID: <AANLkTinzgfX97Xb52c66KWGV-=WQR1VGDtj=7+w-EOLE@mail.gmail.com>

Ok, I've figured it out. It turns out the problem was
  <param name="ping-min" value="1"/>

This was stopping the ping count dropping below 1, so it never reached 0 so
never marked the gateway as down.

ping-min and ping-max aren't very well documented on the wiki. I can see
what ping-max is for (determines the number of failed pings before the
gateway is marked as down, so it doesn't matter is an occasional request
fails), but what's the purpose of ping-min?

-Steve



On 11 January 2011 19:28, DJB International <djbinter at gmail.com> wrote:

> I guess you probably have to open a JIRA on this issue.
>
> -djbinter
>
>
> On Tue, Jan 11, 2011 at 11:14 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> 'sofia status' shows NOREG
>>
>> 'sofia status gateway gw3' shows UP if it's already down when FS starts
>>
>> 'sofia status gateway gw3' shows UP (ping) if it's up for a while and then
>> disappears
>>
>> Regards,
>> -Steve
>>
>>
>>
>>
>> On 11 January 2011 18:58, DJB International <djbinter at gmail.com> wrote:
>>
>>> Steve,
>>>
>>> I am just curious.  If you issue "sofia status gateway gw3" or "sofia
>>> status gateway gw4", do you see the Status as UP or DOWN.
>>>
>>> I would have thought that the State should be either NOREG or REG, and
>>> Status should be either UP or DOWN, but somehow the log displayed as state.
>>>
>>> Thanks,
>>> -djbinter
>>>
>>>
>>> On Tue, Jan 11, 2011 at 6:37 AM, Steven Ayre <steveayre at gmail.com>wrote:
>>>
>>>> freeswitch at internal> version
>>>> FreeSWITCH Version 1.0.head (git-a8b2840 2011-01-07 17-53-09 -0600)
>>>>
>>>> -Steve
>>>>
>>>>
>>>> On 11 January 2011 14:34, Brian West <brian at freeswitch.org> wrote:
>>>> > what version of FS are you running?
>>>> >
>>>> > /b
>>>> >
>>>> > On Jan 11, 2011, at 3:51 AM, Steven Ayre wrote:
>>>> >
>>>> >>
>>>> >> Is there a mistake in my configuration?
>>>> >>
>>>> >> -Steve
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Thu Jan 13 17:47:29 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 13 Jan 2011 15:47:29 +0100
Subject: [Freeswitch-users] Sofia fails to detect offline gateway
In-Reply-To: <AANLkTinzgfX97Xb52c66KWGV-=WQR1VGDtj=7+w-EOLE@mail.gmail.com>
References: <AANLkTim3PeS3fELJPQ21N38c9v5p8J0wGNkBrUvc9R=t@mail.gmail.com>
	<B16E93D7-3578-4071-8899-DDE70DB4D1DC@freeswitch.org>
	<AANLkTin2j-+Y11_x4bs0+M6UsVKUN14fXHoZ262vFu26@mail.gmail.com>
	<AANLkTimTutKEvOg+9+MwrxBOp3yJiEax0S-w_5kkvcwB@mail.gmail.com>
	<AANLkTikMrAzgwE5Jm4qszE_pYXxqO9bF6t1Qtwi9eoLv@mail.gmail.com>
	<AANLkTimJzFBcMu+npWEs1_fW0UDUdhpaBmfpqEADPteL@mail.gmail.com>
	<AANLkTinzgfX97Xb52c66KWGV-=WQR1VGDtj=7+w-EOLE@mail.gmail.com>
Message-ID: <99BC6722-8718-4D80-826C-A3984B88A0AC@ipeva.fr>

Min number of positive pings before marking the gw up, perhaps ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 13/01/2011 ? 15:06, Steven Ayre a ?crit :

> Ok, I've figured it out. It turns out the problem was
>   <param name="ping-min" value="1"/>
> 
> This was stopping the ping count dropping below 1, so it never reached 0 so never marked the gateway as down.
> 
> ping-min and ping-max aren't very well documented on the wiki. I can see what ping-max is for (determines the number of failed pings before the gateway is marked as down, so it doesn't matter is an occasional request fails), but what's the purpose of ping-min?
> 
> -Steve
> 
> 
> 
> On 11 January 2011 19:28, DJB International <djbinter at gmail.com> wrote:
> I guess you probably have to open a JIRA on this issue.
> 
> -djbinter
> 
> 
> On Tue, Jan 11, 2011 at 11:14 AM, Steven Ayre <steveayre at gmail.com> wrote:
> 'sofia status' shows NOREG
> 
> 'sofia status gateway gw3' shows UP if it's already down when FS starts
> 
> 'sofia status gateway gw3' shows UP (ping) if it's up for a while and then disappears
> 
> Regards,
> -Steve
> 
> 
> 
> 
> On 11 January 2011 18:58, DJB International <djbinter at gmail.com> wrote:
> Steve,
> 
> I am just curious.  If you issue "sofia status gateway gw3" or "sofia status gateway gw4", do you see the Status as UP or DOWN.
> 
> I would have thought that the State should be either NOREG or REG, and Status should be either UP or DOWN, but somehow the log displayed as state.
> 
> Thanks,
> -djbinter
> 
> 
> On Tue, Jan 11, 2011 at 6:37 AM, Steven Ayre <steveayre at gmail.com> wrote:
> freeswitch at internal> version
> FreeSWITCH Version 1.0.head (git-a8b2840 2011-01-07 17-53-09 -0600)
> 
> -Steve
> 
> 
> On 11 January 2011 14:34, Brian West <brian at freeswitch.org> wrote:
> > what version of FS are you running?
> >
> > /b
> >
> > On Jan 11, 2011, at 3:51 AM, Steven Ayre wrote:
> >
> >>
> >> Is there a mistake in my configuration?
> >>
> >> -Steve
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From steveayre at gmail.com  Thu Jan 13 17:54:59 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 13 Jan 2011 14:54:59 +0000
Subject: [Freeswitch-users] Jira emails not arriving
Message-ID: <AANLkTikv6MT+cDsfOatex0VaCOWZCEPRauHGwk3jmRs5@mail.gmail.com>

Does anyone know who looks after jira.freeswitch.org?

I've used the forgotten password link several times this week but none of
the emails have arrived.

-Steve
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From steveayre at gmail.com  Thu Jan 13 17:57:05 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 13 Jan 2011 14:57:05 +0000
Subject: [Freeswitch-users] Sofia fails to detect offline gateway
In-Reply-To: <99BC6722-8718-4D80-826C-A3984B88A0AC@ipeva.fr>
References: <AANLkTim3PeS3fELJPQ21N38c9v5p8J0wGNkBrUvc9R=t@mail.gmail.com>
	<B16E93D7-3578-4071-8899-DDE70DB4D1DC@freeswitch.org>
	<AANLkTin2j-+Y11_x4bs0+M6UsVKUN14fXHoZ262vFu26@mail.gmail.com>
	<AANLkTimTutKEvOg+9+MwrxBOp3yJiEax0S-w_5kkvcwB@mail.gmail.com>
	<AANLkTikMrAzgwE5Jm4qszE_pYXxqO9bF6t1Qtwi9eoLv@mail.gmail.com>
	<AANLkTimJzFBcMu+npWEs1_fW0UDUdhpaBmfpqEADPteL@mail.gmail.com>
	<AANLkTinzgfX97Xb52c66KWGV-=WQR1VGDtj=7+w-EOLE@mail.gmail.com>
	<99BC6722-8718-4D80-826C-A3984B88A0AC@ipeva.fr>
Message-ID: <AANLkTikNKvT+bkwyf-5NW8qjGyNE054=VjGJOf35vZTN@mail.gmail.com>

Done some testing with ping-max=10 and ping-min=3, and the gateway was
marked as UP on the first successful ping.

-Steve



On 13 January 2011 14:47, David Ponzone <david.ponzone at ipeva.fr> wrote:

> Min number of positive pings before marking the gw up, perhaps ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 13/01/2011 ? 15:06, Steven Ayre a ?crit :
>
> Ok, I've figured it out. It turns out the problem was
>   <param name="ping-min" value="1"/>
>
> This was stopping the ping count dropping below 1, so it never reached 0 so
> never marked the gateway as down.
>
> ping-min and ping-max aren't very well documented on the wiki. I can see
> what ping-max is for (determines the number of failed pings before the
> gateway is marked as down, so it doesn't matter is an occasional request
> fails), but what's the purpose of ping-min?
>
> -Steve
>
>
>
> On 11 January 2011 19:28, DJB International <djbinter at gmail.com> wrote:
>
>> I guess you probably have to open a JIRA on this issue.
>>
>> -djbinter
>>
>>
>> On Tue, Jan 11, 2011 at 11:14 AM, Steven Ayre <steveayre at gmail.com>wrote:
>>
>>> 'sofia status' shows NOREG
>>>
>>> 'sofia status gateway gw3' shows UP if it's already down when FS starts
>>>
>>> 'sofia status gateway gw3' shows UP (ping) if it's up for a while and
>>> then disappears
>>>
>>> Regards,
>>> -Steve
>>>
>>>
>>>
>>>
>>> On 11 January 2011 18:58, DJB International <djbinter at gmail.com> wrote:
>>>
>>>> Steve,
>>>>
>>>> I am just curious.  If you issue "sofia status gateway gw3" or "sofia
>>>> status gateway gw4", do you see the Status as UP or DOWN.
>>>>
>>>> I would have thought that the State should be either NOREG or REG, and
>>>> Status should be either UP or DOWN, but somehow the log displayed as state.
>>>>
>>>> Thanks,
>>>> -djbinter
>>>>
>>>>
>>>> On Tue, Jan 11, 2011 at 6:37 AM, Steven Ayre <steveayre at gmail.com>wrote:
>>>>
>>>>> freeswitch at internal> version
>>>>> FreeSWITCH Version 1.0.head (git-a8b2840 2011-01-07 17-53-09 -0600)
>>>>>
>>>>> -Steve
>>>>>
>>>>>
>>>>> On 11 January 2011 14:34, Brian West <brian at freeswitch.org> wrote:
>>>>> > what version of FS are you running?
>>>>> >
>>>>> > /b
>>>>> >
>>>>> > On Jan 11, 2011, at 3:51 AM, Steven Ayre wrote:
>>>>> >
>>>>> >>
>>>>> >> Is there a mistake in my configuration?
>>>>> >>
>>>>> >> -Steve
>>>>> >
>>>>> >
>>>>> > _______________________________________________
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>> >
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Thu Jan 13 18:00:34 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 13 Jan 2011 15:00:34 +0000
Subject: [Freeswitch-users] Sofia fails to detect offline gateway
In-Reply-To: <AANLkTikNKvT+bkwyf-5NW8qjGyNE054=VjGJOf35vZTN@mail.gmail.com>
References: <AANLkTim3PeS3fELJPQ21N38c9v5p8J0wGNkBrUvc9R=t@mail.gmail.com>
	<B16E93D7-3578-4071-8899-DDE70DB4D1DC@freeswitch.org>
	<AANLkTin2j-+Y11_x4bs0+M6UsVKUN14fXHoZ262vFu26@mail.gmail.com>
	<AANLkTimTutKEvOg+9+MwrxBOp3yJiEax0S-w_5kkvcwB@mail.gmail.com>
	<AANLkTikMrAzgwE5Jm4qszE_pYXxqO9bF6t1Qtwi9eoLv@mail.gmail.com>
	<AANLkTimJzFBcMu+npWEs1_fW0UDUdhpaBmfpqEADPteL@mail.gmail.com>
	<AANLkTinzgfX97Xb52c66KWGV-=WQR1VGDtj=7+w-EOLE@mail.gmail.com>
	<99BC6722-8718-4D80-826C-A3984B88A0AC@ipeva.fr>
	<AANLkTikNKvT+bkwyf-5NW8qjGyNE054=VjGJOf35vZTN@mail.gmail.com>
Message-ID: <AANLkTinHdBZ9UzwhVS286ob==5i71kS6OFXan6sedapS@mail.gmail.com>

<gateway name="gw1">
  <param name="realm" value="a.b.c.d"/>
  <param name="username" value="nouser"/>
  <param name="password" value="nopass"/>
  <param name="register" value="false"/>
  <param name="caller-id-in-from" value="true"/>
  <param name="ping" value="5"/>
  <param name="ping-max" value="10"/>
  <param name="ping-min" value="3"/>
  <param name="retry-seconds" value="5"/>
  <param name="expire-seconds" value="60"/>
</gateway>

2011-01-13 14:59:08.762286 [WARNING] sofia.c:3962 Ping succeeded gw1 with
code 200 - count 3/1/10, state UP
2011-01-13 14:59:13.768365 [WARNING] sofia.c:3962 Ping succeeded gw1 with
code 200 - count 3/2/10, state UP
2011-01-13 14:59:18.771748 [WARNING] sofia.c:3962 Ping succeeded gw1 with
code 200 - count 3/3/10, state UP
2011-01-13 14:59:23.776471 [WARNING] sofia.c:3962 Ping succeeded gw1 with
code 200 - count 3/4/10, state UP
2011-01-13 14:59:28.781190 [WARNING] sofia.c:3962 Ping succeeded gw1 with
code 200 - count 3/5/10, state UP
2011-01-13 14:59:34.784801 [WARNING] sofia.c:3962 Ping succeeded gw1 with
code 200 - count 3/6/10, state UP
2011-01-13 14:59:39.791909 [WARNING] sofia.c:3962 Ping succeeded gw1 with
code 200 - count 3/7/10, state UP
2011-01-13 14:59:44.835272 [WARNING] sofia.c:3962 Ping succeeded gw1 with
code 200 - count 3/8/10, state UP
2011-01-13 14:59:49.838109 [WARNING] sofia.c:3962 Ping succeeded gw1 with
code 200 - count 3/9/10, state UP
2011-01-13 14:59:55.840360 [WARNING] sofia.c:3962 Ping succeeded gw1 with
code 200 - count 3/10/10, state UP


On 13 January 2011 14:57, Steven Ayre <steveayre at gmail.com> wrote:

> Done some testing with ping-max=10 and ping-min=3, and the gateway was
> marked as UP on the first successful ping.
>
> -Steve
>
>
>
>
> On 13 January 2011 14:47, David Ponzone <david.ponzone at ipeva.fr> wrote:
>
>> Min number of positive pings before marking the gw up, perhaps ?
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 13/01/2011 ? 15:06, Steven Ayre a ?crit :
>>
>> Ok, I've figured it out. It turns out the problem was
>>   <param name="ping-min" value="1"/>
>>
>> This was stopping the ping count dropping below 1, so it never reached 0
>> so never marked the gateway as down.
>>
>> ping-min and ping-max aren't very well documented on the wiki. I can see
>> what ping-max is for (determines the number of failed pings before the
>> gateway is marked as down, so it doesn't matter is an occasional request
>> fails), but what's the purpose of ping-min?
>>
>> -Steve
>>
>>
>>
>> On 11 January 2011 19:28, DJB International <djbinter at gmail.com> wrote:
>>
>>> I guess you probably have to open a JIRA on this issue.
>>>
>>> -djbinter
>>>
>>>
>>> On Tue, Jan 11, 2011 at 11:14 AM, Steven Ayre <steveayre at gmail.com>wrote:
>>>
>>>> 'sofia status' shows NOREG
>>>>
>>>> 'sofia status gateway gw3' shows UP if it's already down when FS starts
>>>>
>>>> 'sofia status gateway gw3' shows UP (ping) if it's up for a while and
>>>> then disappears
>>>>
>>>> Regards,
>>>> -Steve
>>>>
>>>>
>>>>
>>>>
>>>> On 11 January 2011 18:58, DJB International <djbinter at gmail.com> wrote:
>>>>
>>>>> Steve,
>>>>>
>>>>> I am just curious.  If you issue "sofia status gateway gw3" or "sofia
>>>>> status gateway gw4", do you see the Status as UP or DOWN.
>>>>>
>>>>> I would have thought that the State should be either NOREG or REG, and
>>>>> Status should be either UP or DOWN, but somehow the log displayed as state.
>>>>>
>>>>> Thanks,
>>>>> -djbinter
>>>>>
>>>>>
>>>>> On Tue, Jan 11, 2011 at 6:37 AM, Steven Ayre <steveayre at gmail.com>wrote:
>>>>>
>>>>>> freeswitch at internal> version
>>>>>> FreeSWITCH Version 1.0.head (git-a8b2840 2011-01-07 17-53-09 -0600)
>>>>>>
>>>>>> -Steve
>>>>>>
>>>>>>
>>>>>> On 11 January 2011 14:34, Brian West <brian at freeswitch.org> wrote:
>>>>>> > what version of FS are you running?
>>>>>> >
>>>>>> > /b
>>>>>> >
>>>>>> > On Jan 11, 2011, at 3:51 AM, Steven Ayre wrote:
>>>>>> >
>>>>>> >>
>>>>>> >> Is there a mistake in my configuration?
>>>>>> >>
>>>>>> >> -Steve
>>>>>> >
>>>>>> >
>>>>>> > _______________________________________________
>>>>>> > FreeSWITCH-users mailing list
>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> > UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> > http://www.freeswitch.org
>>>>>> >
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From Nabble at slickdeals.endjunk.com  Thu Jan 13 19:22:34 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 13 Jan 2011 08:22:34 -0800 (PST)
Subject: [Freeswitch-users] enum to inum
Message-ID: <1294935754041-5918795.post@n2.nabble.com>


If I called an iNUM number, the eNUM dialplan picked up and it dumped the
following on fs_cli:

2011-01-12 14:14:33.973951 [DEBUG] mod_enum.c:576 ENUM Lookup on
883510074743246
2011-01-12 14:14:34.081638 [DEBUG] mod_enum.c:204 Unable to lookup NAPTR
record for 6.4.2.3.4.7.4.7.0.0.1.5.3.8.8.e164.org: valid domain but no data
of requested type
If I do a dig naptr 6.4.2.3.4.7.4.7.0.0.1.5.3.8.8.e164.org on my FS hosted
on a Seagate DockStar, this is what I will get:
; <<>> DiG 9.7.2-P3 <<>> naptr 6.4.2.3.4.7.4.7.0.0.1.5.3.8.8.e164.org
;; global options: +cmd
;; Got answer:
;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 23298
;; flags: qr rd ra; QUERY: 1, ANSWER: 0, AUTHORITY: 1, ADDITIONAL: 0

;; QUESTION SECTION:
;6.4.2.3.4.7.4.7.0.0.1.5.3.8.8.e164.org.        IN NAPTR

;; AUTHORITY SECTION:
6.4.2.3.4.7.4.7.0.0.1.5.3.8.8.e164.org. 166 IN SOA na.e164.biz.
support.e164.org. 2011011318 86400 1800 3600000 300

;; Query time: 16 msec
;; SERVER: 192.168.1.1#53(192.168.1.1)
;; WHEN: Thu Jan 13 11:03:40 2011
;; MSG SIZE  rcvd: 111
If I do a  dig naptr 6.4.2.3.4.7.4.7.0.0.1.5.3.8.8.e164.arpa on my FS hosted
on a Seagate DockStar, this is what I will get:
; <<>> DiG 9.7.2-P3 <<>> naptr 6.4.2.3.4.7.4.7.0.0.1.5.3.8.8.e164.arpa
;; global options: +cmd
;; Got answer:
;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 30195
;; flags: qr rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 0, ADDITIONAL: 0

;; QUESTION SECTION:
;6.4.2.3.4.7.4.7.0.0.1.5.3.8.8.e164.arpa. IN NAPTR

;; ANSWER SECTION:
6.4.2.3.4.7.4.7.0.0.1.5.3.8.8.e164.arpa. 86400 IN NAPTR 10 100 "u" "E2U+sip"
"!^(.*)$!sip:\\1 at 81.201.82.25!" .

;; Query time: 277 msec
;; SERVER: 192.168.1.1#53(192.168.1.1)
;; WHEN: Thu Jan 13 11:05:29 2011
;; MSG SIZE  rcvd: 113
So, I replaced the e164.org with e164.arpa on the
conf/autoload_configs/enum.conf.xml as shown below, and then eNUM is able to
find the iNUM number to call without a problem.<configuration
name="enum.conf" description="ENUM Module">
  <settings>
    
    
    
    
  </settings>

  <routes>
    <route service="E2U+SIP" regex="sip:(.*)"
replace="sofia/${use_profile}/$1"/>
    <!--<route service="E2U+XMPP" regex="XMPP:(.*)"
replace="dingaling/$${xmpp_server_profile}/$1"/
  </routes>
</configuration>
In this case, why not update the conf/autoload_configs/enum.conf.xml file on
git repository to reflect e164.arpa?

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/enum-to-inum-tp5918795p5918795.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From oseslija at gmail.com  Thu Jan 13 20:00:28 2011
From: oseslija at gmail.com (Ognjen Seslija)
Date: Thu, 13 Jan 2011 18:00:28 +0100
Subject: [Freeswitch-users] Looking for someone to build Android SIP
 Client To Work W/Freeswitch
In-Reply-To: <AANLkTi=NkpL8+KiBP6pJ4ubxwvnAkFPVSeWF3UXpf30M@mail.gmail.com>
References: <AANLkTi=NkpL8+KiBP6pJ4ubxwvnAkFPVSeWF3UXpf30M@mail.gmail.com>
Message-ID: <AANLkTimUAtJ+CK1tXfGtoi6RMPWfQu20zV_de2Z8VMrB@mail.gmail.com>

Hmm...Android 2.3 features full SIP stack out-of-the-box.


On Tue, Jan 11, 2011 at 6:39 PM, Siobhan Hamilton <
siobhan.pluggedin at gmail.com> wrote:

> Hi there,
>
> Hope I'm not totally off-base with this post but I thought this group might
> be a good resource to ask.... My company is looking for a SIP/VOIP expert
> who can build an integrated SIP client for us on Android to interact with a
> Freeswitch server we are building, if anyone is out there who fits the bill
> or knows anyone....?
>
> Requirements:
> - No GPL'ed SIP Stacks - we are building a commercial app that we wish to
> keep closed source.
> - Not a standalone, typical SIP client; will use a non-traditional UI to
> interact with the SIP server.
> - Knowledge of optimization of Android performance and ability to evaluate
> and implement best codecs for our purposes.
>
> App will be interfacing with OpenSIPs registration server with a custom
> presence module, so must be familiar with presence.  Users will be heavily
> interacting in conference mode.
>
> Please email resume/experience to siobhan.pluggedin at gmail.com...
>
> Thank you,
>
> Siobhan Hamilton
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Thu Jan 13 20:40:26 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 13 Jan 2011 11:40:26 -0600
Subject: [Freeswitch-users] console
In-Reply-To: <AANLkTinXiaVGr5bneGCfvXUmbEnJ5Ggjcmvim=Wjqa5O@mail.gmail.com>
References: <6ECAF1527329364583AB525CF34ABF950B31A4F9@ms.kallback.com>
	<AANLkTinXiaVGr5bneGCfvXUmbEnJ5Ggjcmvim=Wjqa5O@mail.gmail.com>
Message-ID: <AANLkTinnKUY9xkWAfLGgs+sf41A-nZ0xH9C05ogeBHqW@mail.gmail.com>

8081 8082 vs 8021 8022



On Wed, Jan 12, 2011 at 9:16 PM, Sam <u2nsam at gmail.com> wrote:
> Yes the netstat shows the ports 8081 & 8082 not used
> Justin, as said earlier i have 2 FS instances running on same server and
> having below config:-
> for
>> 192.168.2.1:-
>> ?? <param name="listen-ip" value="127.0.0.1"/>
>> ? ?<param name="listen-port" value="8021"/>
>> for
>> 192.168.2.2:-
>> ?? <param name="listen-ip" value="127.0.0.1"/>
>> ? ?<param name="listen-port" value="8022"/>
>
> Here I tried all the permutation combination changing the IP port but no
> sucess.
>
> Regds
> Sam
>
> On Wed, Jan 12, 2011 at 11:25 PM, JRichey <jrichey at itltd.net> wrote:
>>
>> Did you ever try?using netstat like Steven suggested?? What do you get as
>> output for "netstat -tunlp"?
>>
>> In one of your commands below you show 127.0.0.2 instead of 127.0.0.1 so
>> if that's not a typo you may want to try it again.
>>
>> -Justin
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of Sam
>> Sent: Tuesday, January 11, 2011 9:27 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] console
>>
>> Tried this below and received :-
>>
>>
>> # /usr/local/fs_2/bin/fs_cli -H 127.0.0.2 -P 8082
>> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
>> Error]
>>
>> tried by creating profile:-
>> # /usr/local/fs_2/bin/fs_cli profile1
>> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
>> Error]
>>
>> By port:
>> # /usr/local/fs_2/bin/fs_cli -P 8082
>> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
>> Error]
>>
>>
>> netstat -nlp | grep ; ports 8081 & 8082 are available
>>
>> But i could get a console for other server by
>> # /usr/local/fs_2/bin/fs_cli
>>
>> Regards
>> Sam
>>
>>
>> On Tue, Jan 11, 2011 at 10:54 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>>
>>> Actually, correction it does - just not the event_socket.conf.xml one.
>>> It'll read .fs_cli_conf in your home directory if it exists, but that isn't
>>> created by default - you create it yourself if you want to use it (it's
>>> optional). The command line arguments -H and -P -will- override the config
>>> file though.
>>>
>>> Are you using a capital P? -p is password while -P is port. If there's no
>>> password on the event socket you'd get no error from using a small p by
>>> accident.
>>>
>>> -Steve
>>>
>>>
>>>
>>>
>>>
>>> On 11 January 2011 16:59, Steven Ayre <steveayre at gmail.com> wrote:
>>>>
>>>> It does not read any config file.
>>>>
>>>>
>>>>
>>>>
>>>> On 11 January 2011 16:56, Sam <u2nsam at gmail.com> wrote:
>>>>>
>>>>> the port and ip donot work for me,
>>>>> is it that the fs_cli is not reading the config from 192.168.2.2 but it
>>>>> is reading the config only of 192.168.2.1, though its in the different [FS_1
>>>>> & FS_2] path where i am executing.
>>>>>
>>>>> Regds
>>>>> Sam
>>>>>
>>>>>
>>>>> On Tue, Jan 11, 2011 at 9:38 PM, Steven Ayre <steveayre at gmail.com>
>>>>> wrote:
>>>>>>
>>>>>> /usr/local/FS_1/bin/fs_cli -P 8021
>>>>>> /usr/local/FS_2/bin/fs_cli -P 8022
>>>>>>
>>>>>> fs_cll doesn't read any config file. It's not part of the FS server at
>>>>>> all, you can have it on a different machine that doesn't have FS installed.
>>>>>> It entirely relies on the arguments to control where to connect to.
>>>>>>
>>>>>> -Steve
>>>>>>
>>>>>>
>>>>>> On 11 January 2011 15:04, Sam <u2nsam at gmail.com> wrote:
>>>>>>>
>>>>>>> Something more here ... i am getting the console for 192.168.2.1
>>>>>>> every time i do fs_cli on both instances .
>>>>>>>
>>>>>>> like
>>>>>>> /usr/local/FS_1/bin/fs_cli
>>>>>>> /usr/localFS_2/bin/fs_cli
>>>>>>> i get the console for the 1st server only
>>>>>>>
>>>>>>> the 2 server are listing to 2 different ips .
>>>>>>>
>>>>>>> Regds
>>>>>>> Sam
>>>>>>>
>>>>>>> On Tue, Jan 11, 2011 at 8:07 PM, Steven Ayre <steveayre at gmail.com>
>>>>>>> wrote:
>>>>>>>>
>>>>>>>> It should work. Is there anything already listening on port 8022?
>>>>>>>>
>>>>>>>> $ netstat -a -n -p | grep 8022
>>>>>>>>
>>>>>>>> Are you also sure that they're not both loading the same config
>>>>>>>> file?
>>>>>>>>
>>>>>>>> Regards,
>>>>>>>> -Steve
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On 11 January 2011 14:32, Sam <u2nsam at gmail.com> wrote:
>>>>>>>> > the scenario is i have 2 ips on 1 server for 2 FS instances;
>>>>>>>> >
>>>>>>>> > 192.168.2.1
>>>>>>>> > 192.168.2.2
>>>>>>>> >
>>>>>>>> > and the parameters i have set is:-
>>>>>>>> >
>>>>>>>> > for
>>>>>>>> > 192.168.2.1:-
>>>>>>>> > ?? <param name="listen-ip" value="127.0.0.1"/>
>>>>>>>> > ? ?<param name="listen-port" value="8021"/>
>>>>>>>> > for
>>>>>>>> > 192.168.2.2:-
>>>>>>>> > ?? <param name="listen-ip" value="127.0.0.1"/>
>>>>>>>> > ? ?<param name="listen-port" value="8022"/>
>>>>>>>> >
>>>>>>>> > Ideally it should work? but i am getting console for only
>>>>>>>> > 192.168.2.1 FS .
>>>>>>>> >
>>>>>>>> >
>>>>>>>> > Regards
>>>>>>>> > Sam
>>>>>>>> >
>>>>>>>> > On Tue, Jan 11, 2011 at 4:20 PM, Steven Ayre <steveayre at gmail.com>
>>>>>>>> > wrote:
>>>>>>>> >>
>>>>>>>> >> ? ?<param name="listen-ip" value="127.0.0.1"/>
>>>>>>>> >> ? ?<param name="listen-port" value="8021"/>
>>>>>>>> >>
>>>>>>>> >> You can bind both to port 8021 on their individual IPs, or
>>>>>>>> >> different
>>>>>>>> >> ports on the same IP.
>>>>>>>> >>
>>>>>>>> >> A listen IP of 0.0.0.0 will mean any IP.
>>>>>>>> >>
>>>>>>>> >> -Steve
>>>>>>>> >>
>>>>>>>> >> On 11 January 2011 10:44, Sam <u2nsam at gmail.com> wrote:
>>>>>>>> >> > A query,
>>>>>>>> >> >
>>>>>>>> >> > I have 2 FS running on one server on 2 different ips,
>>>>>>>> >> > so when i do fs_cli going to respective bins , i see console of
>>>>>>>> >> > only the
>>>>>>>> >> > first server.
>>>>>>>> >> >
>>>>>>>> >> > Is there any way to get the console of both the FS on the same
>>>>>>>> >> > server .
>>>>>>>> >> > I tried changing the port of event socket to 8022 but it donot
>>>>>>>> >> > works.
>>>>>>>> >> > <param name="listen-port" value="8022"/>
>>>>>>>> >> >
>>>>>>>> >> > Is there some method to start the console of both the
>>>>>>>> >> > instances.
>>>>>>>> >> >
>>>>>>>> >> > Regds
>>>>>>>> >> > Sam
>>>>>>>> >> >
>>>>>>>> >> > _______________________________________________
>>>>>>>> >> > FreeSWITCH-users mailing list
>>>>>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >> >
>>>>>>>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> >> > http://www.freeswitch.org
>>>>>>>> >> >
>>>>>>>> >> >
>>>>>>>> >>
>>>>>>>> >> _______________________________________________
>>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >>
>>>>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> >> http://www.freeswitch.org
>>>>>>>> >
>>>>>>>> >
>>>>>>>> > _______________________________________________
>>>>>>>> > FreeSWITCH-users mailing list
>>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >
>>>>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> > http://www.freeswitch.org
>>>>>>>> >
>>>>>>>> >
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From steveayre at gmail.com  Thu Jan 13 20:48:41 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 13 Jan 2011 17:48:41 +0000
Subject: [Freeswitch-users] console
In-Reply-To: <AANLkTinnKUY9xkWAfLGgs+sf41A-nZ0xH9C05ogeBHqW@mail.gmail.com>
References: <6ECAF1527329364583AB525CF34ABF950B31A4F9@ms.kallback.com>
	<AANLkTinXiaVGr5bneGCfvXUmbEnJ5Ggjcmvim=Wjqa5O@mail.gmail.com>
	<AANLkTinnKUY9xkWAfLGgs+sf41A-nZ0xH9C05ogeBHqW@mail.gmail.com>
Message-ID: <AANLkTikVECT9BAXCTLLkxXbfdi0wcVv1yvjteyigTqh=@mail.gmail.com>

Sam,

Run these command as root with both versions of FS started. If FreeSWITCH is
running you should see 2 lines like below.

$ netstat -anp | grep 8021
tcp        0      0 127.0.0.1:8021          0.0.0.0:*
LISTEN      15394/freeswitch
$ netstat -anp | grep 8022
tcp        0      0 127.0.0.1:8022          0.0.0.0:*
LISTEN      15395/freeswitch

-Steve



On 13 January 2011 17:40, Anthony Minessale <anthony.minessale at gmail.com>wrote:

> 8081 8082 vs 8021 8022
>
>
>
> On Wed, Jan 12, 2011 at 9:16 PM, Sam <u2nsam at gmail.com> wrote:
> > Yes the netstat shows the ports 8081 & 8082 not used
> > Justin, as said earlier i have 2 FS instances running on same server and
> > having below config:-
> > for
> >> 192.168.2.1:-
> >>    <param name="listen-ip" value="127.0.0.1"/>
> >>    <param name="listen-port" value="8021"/>
> >> for
> >> 192.168.2.2:-
> >>    <param name="listen-ip" value="127.0.0.1"/>
> >>    <param name="listen-port" value="8022"/>
> >
> > Here I tried all the permutation combination changing the IP port but no
> > sucess.
> >
> > Regds
> > Sam
> >
> > On Wed, Jan 12, 2011 at 11:25 PM, JRichey <jrichey at itltd.net> wrote:
> >>
> >> Did you ever try using netstat like Steven suggested?  What do you get
> as
> >> output for "netstat -tunlp"?
> >>
> >> In one of your commands below you show 127.0.0.2 instead of 127.0.0.1 so
> >> if that's not a typo you may want to try it again.
> >>
> >> -Justin
> >>
> >>
> >> -----Original Message-----
> >> From: freeswitch-users-bounces at lists.freeswitch.org
> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of Sam
> >> Sent: Tuesday, January 11, 2011 9:27 PM
> >> To: FreeSWITCH Users Help
> >> Subject: Re: [Freeswitch-users] console
> >>
> >> Tried this below and received :-
> >>
> >>
> >> # /usr/local/fs_2/bin/fs_cli -H 127.0.0.2 -P 8082
> >> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
> Connection
> >> Error]
> >>
> >> tried by creating profile:-
> >> # /usr/local/fs_2/bin/fs_cli profile1
> >> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
> Connection
> >> Error]
> >>
> >> By port:
> >> # /usr/local/fs_2/bin/fs_cli -P 8082
> >> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
> Connection
> >> Error]
> >>
> >>
> >> netstat -nlp | grep ; ports 8081 & 8082 are available
> >>
> >> But i could get a console for other server by
> >> # /usr/local/fs_2/bin/fs_cli
> >>
> >> Regards
> >> Sam
> >>
> >>
> >> On Tue, Jan 11, 2011 at 10:54 PM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >>>
> >>> Actually, correction it does - just not the event_socket.conf.xml one.
> >>> It'll read .fs_cli_conf in your home directory if it exists, but that
> isn't
> >>> created by default - you create it yourself if you want to use it (it's
> >>> optional). The command line arguments -H and -P -will- override the
> config
> >>> file though.
> >>>
> >>> Are you using a capital P? -p is password while -P is port. If there's
> no
> >>> password on the event socket you'd get no error from using a small p by
> >>> accident.
> >>>
> >>> -Steve
> >>>
> >>>
> >>>
> >>>
> >>>
> >>> On 11 January 2011 16:59, Steven Ayre <steveayre at gmail.com> wrote:
> >>>>
> >>>> It does not read any config file.
> >>>>
> >>>>
> >>>>
> >>>>
> >>>> On 11 January 2011 16:56, Sam <u2nsam at gmail.com> wrote:
> >>>>>
> >>>>> the port and ip donot work for me,
> >>>>> is it that the fs_cli is not reading the config from 192.168.2.2 but
> it
> >>>>> is reading the config only of 192.168.2.1, though its in the
> different [FS_1
> >>>>> & FS_2] path where i am executing.
> >>>>>
> >>>>> Regds
> >>>>> Sam
> >>>>>
> >>>>>
> >>>>> On Tue, Jan 11, 2011 at 9:38 PM, Steven Ayre <steveayre at gmail.com>
> >>>>> wrote:
> >>>>>>
> >>>>>> /usr/local/FS_1/bin/fs_cli -P 8021
> >>>>>> /usr/local/FS_2/bin/fs_cli -P 8022
> >>>>>>
> >>>>>> fs_cll doesn't read any config file. It's not part of the FS server
> at
> >>>>>> all, you can have it on a different machine that doesn't have FS
> installed.
> >>>>>> It entirely relies on the arguments to control where to connect to.
> >>>>>>
> >>>>>> -Steve
> >>>>>>
> >>>>>>
> >>>>>> On 11 January 2011 15:04, Sam <u2nsam at gmail.com> wrote:
> >>>>>>>
> >>>>>>> Something more here ... i am getting the console for 192.168.2.1
> >>>>>>> every time i do fs_cli on both instances .
> >>>>>>>
> >>>>>>> like
> >>>>>>> /usr/local/FS_1/bin/fs_cli
> >>>>>>> /usr/localFS_2/bin/fs_cli
> >>>>>>> i get the console for the 1st server only
> >>>>>>>
> >>>>>>> the 2 server are listing to 2 different ips .
> >>>>>>>
> >>>>>>> Regds
> >>>>>>> Sam
> >>>>>>>
> >>>>>>> On Tue, Jan 11, 2011 at 8:07 PM, Steven Ayre <steveayre at gmail.com>
> >>>>>>> wrote:
> >>>>>>>>
> >>>>>>>> It should work. Is there anything already listening on port 8022?
> >>>>>>>>
> >>>>>>>> $ netstat -a -n -p | grep 8022
> >>>>>>>>
> >>>>>>>> Are you also sure that they're not both loading the same config
> >>>>>>>> file?
> >>>>>>>>
> >>>>>>>> Regards,
> >>>>>>>> -Steve
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>
> >>>>>>>> On 11 January 2011 14:32, Sam <u2nsam at gmail.com> wrote:
> >>>>>>>> > the scenario is i have 2 ips on 1 server for 2 FS instances;
> >>>>>>>> >
> >>>>>>>> > 192.168.2.1
> >>>>>>>> > 192.168.2.2
> >>>>>>>> >
> >>>>>>>> > and the parameters i have set is:-
> >>>>>>>> >
> >>>>>>>> > for
> >>>>>>>> > 192.168.2.1:-
> >>>>>>>> >    <param name="listen-ip" value="127.0.0.1"/>
> >>>>>>>> >    <param name="listen-port" value="8021"/>
> >>>>>>>> > for
> >>>>>>>> > 192.168.2.2:-
> >>>>>>>> >    <param name="listen-ip" value="127.0.0.1"/>
> >>>>>>>> >    <param name="listen-port" value="8022"/>
> >>>>>>>> >
> >>>>>>>> > Ideally it should work  but i am getting console for only
> >>>>>>>> > 192.168.2.1 FS .
> >>>>>>>> >
> >>>>>>>> >
> >>>>>>>> > Regards
> >>>>>>>> > Sam
> >>>>>>>> >
> >>>>>>>> > On Tue, Jan 11, 2011 at 4:20 PM, Steven Ayre <
> steveayre at gmail.com>
> >>>>>>>> > wrote:
> >>>>>>>> >>
> >>>>>>>> >>    <param name="listen-ip" value="127.0.0.1"/>
> >>>>>>>> >>    <param name="listen-port" value="8021"/>
> >>>>>>>> >>
> >>>>>>>> >> You can bind both to port 8021 on their individual IPs, or
> >>>>>>>> >> different
> >>>>>>>> >> ports on the same IP.
> >>>>>>>> >>
> >>>>>>>> >> A listen IP of 0.0.0.0 will mean any IP.
> >>>>>>>> >>
> >>>>>>>> >> -Steve
> >>>>>>>> >>
> >>>>>>>> >> On 11 January 2011 10:44, Sam <u2nsam at gmail.com> wrote:
> >>>>>>>> >> > A query,
> >>>>>>>> >> >
> >>>>>>>> >> > I have 2 FS running on one server on 2 different ips,
> >>>>>>>> >> > so when i do fs_cli going to respective bins , i see console
> of
> >>>>>>>> >> > only the
> >>>>>>>> >> > first server.
> >>>>>>>> >> >
> >>>>>>>> >> > Is there any way to get the console of both the FS on the
> same
> >>>>>>>> >> > server .
> >>>>>>>> >> > I tried changing the port of event socket to 8022 but it
> donot
> >>>>>>>> >> > works.
> >>>>>>>> >> > <param name="listen-port" value="8022"/>
> >>>>>>>> >> >
> >>>>>>>> >> > Is there some method to start the console of both the
> >>>>>>>> >> > instances.
> >>>>>>>> >> >
> >>>>>>>> >> > Regds
> >>>>>>>> >> > Sam
> >>>>>>>> >> >
> >>>>>>>> >> > _______________________________________________
> >>>>>>>> >> > FreeSWITCH-users mailing list
> >>>>>>>> >> > FreeSWITCH-users at lists.freeswitch.org
> >>>>>>>> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>>> >> >
> >>>>>>>> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>>> >> > http://www.freeswitch.org
> >>>>>>>> >> >
> >>>>>>>> >> >
> >>>>>>>> >>
> >>>>>>>> >> _______________________________________________
> >>>>>>>> >> FreeSWITCH-users mailing list
> >>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>>> >>
> >>>>>>>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>>> >> http://www.freeswitch.org
> >>>>>>>> >
> >>>>>>>> >
> >>>>>>>> > _______________________________________________
> >>>>>>>> > FreeSWITCH-users mailing list
> >>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
> >>>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>>> >
> >>>>>>>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>>> > http://www.freeswitch.org
> >>>>>>>> >
> >>>>>>>> >
> >>>>>>>>
> >>>>>>>> _______________________________________________
> >>>>>>>> FreeSWITCH-users mailing list
> >>>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>>>
> >>>>>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>>> http://www.freeswitch.org
> >>>>>>>
> >>>>>>>
> >>>>>>> _______________________________________________
> >>>>>>> FreeSWITCH-users mailing list
> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>>
> >>>>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> http://www.freeswitch.org
> >>>>>>>
> >>>>>>
> >>>>>>
> >>>>>> _______________________________________________
> >>>>>> FreeSWITCH-users mailing list
> >>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>
> >>>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>> http://www.freeswitch.org
> >>>>>>
> >>>>>
> >>>>>
> >>>>> _______________________________________________
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>
> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>>
> >>>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Jan 13 22:27:53 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 13 Jan 2011 11:27:53 -0800
Subject: [Freeswitch-users] Sofia fails to detect offline gateway
In-Reply-To: <99BC6722-8718-4D80-826C-A3984B88A0AC@ipeva.fr>
References: <AANLkTim3PeS3fELJPQ21N38c9v5p8J0wGNkBrUvc9R=t@mail.gmail.com>
	<B16E93D7-3578-4071-8899-DDE70DB4D1DC@freeswitch.org>
	<AANLkTin2j-+Y11_x4bs0+M6UsVKUN14fXHoZ262vFu26@mail.gmail.com>
	<AANLkTimTutKEvOg+9+MwrxBOp3yJiEax0S-w_5kkvcwB@mail.gmail.com>
	<AANLkTikMrAzgwE5Jm4qszE_pYXxqO9bF6t1Qtwi9eoLv@mail.gmail.com>
	<AANLkTimJzFBcMu+npWEs1_fW0UDUdhpaBmfpqEADPteL@mail.gmail.com>
	<AANLkTinzgfX97Xb52c66KWGV-=WQR1VGDtj=7+w-EOLE@mail.gmail.com>
	<99BC6722-8718-4D80-826C-A3984B88A0AC@ipeva.fr>
Message-ID: <AANLkTik7taMRY-K81Wb7j+2F1udHEK4q3Yc=E0j421hE@mail.gmail.com>

If anyone knows for sure the proper information on ping-min and ping-max
please let me know and I will make sure that they get documented on the
wiki.

Thanks,
MC

On Thu, Jan 13, 2011 at 6:47 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Min number of positive pings before marking the gw up, perhaps ?
>
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From Nabble at slickdeals.endjunk.com  Thu Jan 13 22:30:47 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 13 Jan 2011 11:30:47 -0800 (PST)
Subject: [Freeswitch-users] spindermonkey: Host execution of a
 cross-compilation of libs/js/nsprpub/config/nsinstall.c
In-Reply-To: <1293570293681-5872965.post@n2.nabble.com>
References: <1293570293681-5872965.post@n2.nabble.com>
Message-ID: <1294947047337-5919447.post@n2.nabble.com>


Bumped!

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/spindermonkey-Host-execution-of-a-cross-compilation-of-libs-js-nsprpub-config-nsinstall-c-tp5872965p5919447.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kapil.rastogi at telemune.net  Thu Jan 13 17:31:11 2011
From: kapil.rastogi at telemune.net (kapil.rastogi)
Date: Thu, 13 Jan 2011 06:31:11 -0800 (PST)
Subject: [Freeswitch-users] How to kick a member from conference using java
 application (Kapil Rastogi)
Message-ID: <1294929071769-5918355.post@n2.nabble.com>


Dear All,

I am using java script as a scripting language to develop conference system.
I get a request from WEB to Kick (disconnect) a member from conference room
and I process that request in java application. To Kick a member i am using
the following function in my java application:
 
         con.api("conference", room_name + " kick " + member_id);

But it is not working. The system hangs on this statement.

Please tell me how to kick a member from conference. Please give me a sample
code for this.


Thanks,


-----
Regards,
Kapil Rastogi
Telemune Software Solutions P Ltd.
kapil.rastogi at telemune.net
+919013204760
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-kick-a-member-from-conference-using-java-application-Kapil-Rastogi-tp5918355p5918355.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From paul at cupis.co.uk  Thu Jan 13 23:58:59 2011
From: paul at cupis.co.uk (Paul Cupis)
Date: Thu, 13 Jan 2011 20:58:59 +0000
Subject: [Freeswitch-users] Outgoing registrations within a fs pool
In-Reply-To: <3ff41827-ea0f-4302-8d70-c7ee16feec83@winet.ch>
References: <3ff41827-ea0f-4302-8d70-c7ee16feec83@winet.ch>
Message-ID: <4D2F6793.4010801@cupis.co.uk>

Perhaps look at getting the kamailio server to register to the provider
rather than one of the FS from the pool?



From rafonline at hotmail.com  Fri Jan 14 00:14:19 2011
From: rafonline at hotmail.com (Rafqat .)
Date: Thu, 13 Jan 2011 21:14:19 +0000
Subject: [Freeswitch-users] bridge_terminate_key...z
Message-ID: <COL122-W16C6F460AD6565DA890716B5F00@phx.gbl>


Hi

I would like to terminate a bridge when leg A enters ##.? 

What would be a quick and efficient way of doing this?

I was hoping for something similar to what the 'bridge_terminate_key' session variable offers but with the obvious exception of accepting ## instead of just #.

Cheers

Raf
 		 	   		  

From peter.olsson at visionutveckling.se  Fri Jan 14 00:22:14 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 13 Jan 2011 22:22:14 +0100
Subject: [Freeswitch-users] How to kick a member from conference using
 java application (Kapil Rastogi)
In-Reply-To: <1294929071769-5918355.post@n2.nabble.com>
References: <1294929071769-5918355.post@n2.nabble.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57EC81C50A@cooper>

I'm doing excactly like this (using ESL), and it works, so I can't find a reason why it shouldn't work for you...

Not much help though :)

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r kapil.rastogi [kapil.rastogi at telemune.net]
Skickat: den 13 januari 2011 15:31
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] How to kick a member from conference using java application (Kapil Rastogi)

Dear All,

I am using java script as a scripting language to develop conference system.
I get a request from WEB to Kick (disconnect) a member from conference room
and I process that request in java application. To Kick a member i am using
the following function in my java application:

         con.api("conference", room_name + " kick " + member_id);

But it is not working. The system hangs on this statement.

Please tell me how to kick a member from conference. Please give me a sample
code for this.


Thanks,


-----
Regards,
Kapil Rastogi
Telemune Software Solutions P Ltd.
kapil.rastogi at telemune.net
+919013204760
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-kick-a-member-from-conference-using-java-application-Kapil-Rastogi-tp5918355p5918355.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4d2f5b5532761157274465!



From bernhard.suttner at winet.ch  Fri Jan 14 00:25:46 2011
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Thu, 13 Jan 2011 22:25:46 +0100
Subject: [Freeswitch-users] Outgoing registrations within a fs pool
Message-ID: <20110113222546.6a7df03a@mail.winet.ch>

no idea?

----- Original Message -----
From: Bernhard Suttner [mailto:bernhard.suttner at winet.ch]
To: FreeSWITCH-users at lists.freeswitch.org
Sent: Thu, 13 Jan 2011 13:40:21 +0100
Subject: [Freeswitch-users] Outgoing registrations within a fs pool


> Hi,
> 
> I want to do outgoing registrations (gateways) to a provider. The scenario
> is like that:
> 
> Provider
>    \
>     \
> 
> SBC (Kamailio)
>       + 
> +-----+-------+
> |     |       |
> FS1  FS2     FS3
> 
> FS does share the same database with ODBC. The question is now, could FS1
> register to the provider that the call will reach the SBC later and then
> maybe forwaded to FS3 instead of FS1? The best way would be, that the "pool"
> of FreeSWITCH Server does register to the Provider. So, one of the
> FreeSWITCH server decides that it is necessary to register to the provider
> again (because of expires) and all the other will not register for this
> time. Next time maybe another server does register. It should be not bound
> to one specific FreeSWITCH server. 
> 
> Thanks in advance!
> 
> Best regards,
> Bernhard
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From kris at livecall.com  Fri Jan 14 01:10:30 2011
From: kris at livecall.com (Kris)
Date: Thu, 13 Jan 2011 14:10:30 -0800
Subject: [Freeswitch-users] Jira emails not arriving
References: <AANLkTikv6MT+cDsfOatex0VaCOWZCEPRauHGwk3jmRs5@mail.gmail.com>
Message-ID: <273D0046728E4CC5891C185C9BAAF6AB@stor1>

I also didn't get my password. It changed it so now I can't get in on a 
comuter it was auto-filled.
May have to create a new account...
Kris
----- Original Message ----- 
From: "Steven Ayre" <steveayre at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, January 13, 2011 6:54 AM
Subject: [Freeswitch-users] Jira emails not arriving


> Does anyone know who looks after jira.freeswitch.org?
>
> I've used the forgotten password link several times this week but none of
> the emails have arrived.
>
> -Steve
> 



From steveayre at gmail.com  Fri Jan 14 01:35:45 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 13 Jan 2011 22:35:45 +0000
Subject: [Freeswitch-users] bridge_terminate_key...z
In-Reply-To: <COL122-W16C6F460AD6565DA890716B5F00@phx.gbl>
References: <COL122-W16C6F460AD6565DA890716B5F00@phx.gbl>
Message-ID: <AANLkTimaR7kB3-ihiG2PsTfCQRGGPmbtPPusuC9DNgMu@mail.gmail.com>

Possibly using bind_digit_action, something perhaps like:

<action application="bind_digit_action"
data="bridgeterminator,##,exec:hangup"/>
<action application="digit_action_set_realm" data="bridgeterminator"/>

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_digit_action

-Steve



On 13 January 2011 21:14, Rafqat . <rafonline at hotmail.com> wrote:

>
> Hi
>
> I would like to terminate a bridge when leg A enters ##.
>
> What would be a quick and efficient way of doing this?
>
> I was hoping for something similar to what the 'bridge_terminate_key'
> session variable offers but with the obvious exception of accepting ##
> instead of just #.
>
> Cheers
>
> Raf
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From joaocarlosleme at gmail.com  Fri Jan 14 01:47:01 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Thu, 13 Jan 2011 14:47:01 -0800
Subject: [Freeswitch-users] Caller ID using Fifo
In-Reply-To: <AANLkTikKDO3qru-9=nuPEQcW+iCN7ccizyJAh9sqUkCN@mail.gmail.com>
References: <AANLkTik6pdgXgND6kmzZA4qe8vDZQ0_7e47+ciVWPfub@mail.gmail.com>
	<AANLkTikKDO3qru-9=nuPEQcW+iCN7ccizyJAh9sqUkCN@mail.gmail.com>
Message-ID: <AANLkTi=p7_CsUDWS62+weZv50+PBZ8RLop0Q04y-c4hB@mail.gmail.com>

What about showing the Caller ID after it is answered? Any way to do that?

2011/1/12 Jo?o Mesquita <jmesquita at freeswitch.org>

> Jo?o Leme,
>
> The caller id is not passed when the phone is ringing because mod_fifo does
> not know which call is going to be sent to that channel once it is answered
> until it is really answered. I don't know if mod_callcenter does show
> anything but you should consider looking at the documentation if you really
> need this feature.
>
> Regards,
> Jo?o Mesquita
>
>
> On Wed, Jan 12, 2011 at 9:15 PM, Joao Leme <joaocarlosleme at gmail.com>wrote:
>
>> Hi there,
>> I would like to know if there is a way to see the caller ID on my Sip
>> Client (X-Lite for example) of the caller that I answear from a Fifo queue?
>> Thanks,
>> John
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Fri Jan 14 04:11:13 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 13 Jan 2011 19:11:13 -0600
Subject: [Freeswitch-users] Jira emails not arriving
In-Reply-To: <273D0046728E4CC5891C185C9BAAF6AB@stor1>
References: <AANLkTikv6MT+cDsfOatex0VaCOWZCEPRauHGwk3jmRs5@mail.gmail.com>
	<273D0046728E4CC5891C185C9BAAF6AB@stor1>
Message-ID: <9FF81869-6947-4ED9-869E-80815F495638@freeswitch.org>

Try again... I fixed the mail server today.

/b

On Jan 13, 2011, at 4:10 PM, Kris wrote:

> I also didn't get my password. It changed it so now I can't get in on a 
> comuter it was auto-filled.
> May have to create a new account...
> Kris



From infos at madovsky.org  Fri Jan 14 04:19:27 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 13 Jan 2011 20:19:27 -0500
Subject: [Freeswitch-users] (no subject)
Message-ID: <0A487614A85048019FF252D4DABB2260@e1705>


Is REFER method works trhough a proxy ?
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From Nabble at slickdeals.endjunk.com  Fri Jan 14 04:47:13 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 13 Jan 2011 17:47:13 -0800 (PST)
Subject: [Freeswitch-users] Playing Google translation tts
In-Reply-To: <AANLkTikzqem2YpaDN77FWpBmRavHrd8qADGZP0FrRrWW@mail.gmail.com>
References: <4D2E19AB.9020908@telefaks.de>
	<AANLkTikzqem2YpaDN77FWpBmRavHrd8qADGZP0FrRrWW@mail.gmail.com>
Message-ID: <1294969633389-5920422.post@n2.nabble.com>



Norman Tomlins wrote:
> 
> Peter,
> 
> I have used this in the past and it has been working.
> 
> <extension name="Free_Google_Text_To_Speech">
>   <condition field="destination_number" expression="^2115$">
>     <action application="answer" data=""/>
>     <action application="playback"
> data="shout://translate.google.com/translate_tts?tl=en&q=Buy+Cheap+dids+at+www+dot+voice+network+dot+see+eh"/>
>   </condition>
> </extension>
Thanks Norman. I tried your dialplan above and got the following even though
I have mod_shout loaded.
EXECUTE sofia/internal/1003 at 192.168.1.15
playback(shout://translate.google.com/translate_tts?tl=en&q=Hello+,+It+is+me)
2011-01-13 20:41:57.388295 [ERR] mod_shout.c:800 Error: MPG123 Error at
__FILE__:__LINE__.
2011-01-13 20:41:57.389383 [ERR] mod_shout.c:803 Error from mpg123: Invalid
mpg123 handle. (code 10)
2011-01-13 20:41:57.397162 [DEBUG] sofia.c:4626 Channel
sofia/internal/1003 at 192.168.1.15 entering state [ready][200]
2011-01-13 20:41:57.402589 [NOTICE] switch_core_state_machine.c:189
sofia/internal/1003 at 192.168.1.15 has executed the last dialplan instruction,
hanging up.
2011-01-13 20:41:57.402589 [DEBUG] switch_channel.c:2535
(sofia/internal/1003 at 192.168.1.15) Callstate Change ACTIVE -> HANGUP

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Playing-Google-translation-tts-tp5916138p5920422.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From lists at telefaks.de  Fri Jan 14 05:08:28 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Fri, 14 Jan 2011 03:08:28 +0100
Subject: [Freeswitch-users] Playing Google translation tts
In-Reply-To: <1294969633389-5920422.post@n2.nabble.com>
References: <4D2E19AB.9020908@telefaks.de>	<AANLkTikzqem2YpaDN77FWpBmRavHrd8qADGZP0FrRrWW@mail.gmail.com>
	<1294969633389-5920422.post@n2.nabble.com>
Message-ID: <4D2FB01C.8010504@telefaks.de>

Yesterday I updated a test FS (make current on Ubuntu) and that one
played the file successfully.
I have not tried it with our main FS (on Debian where the problem
occured) yet. So I don't know yet, whether it's from FS or from any
library used.

Best regards
Peter
mazilo schrieb:
> Norman Tomlins wrote:
>   
>> Peter,
>>
>> I have used this in the past and it has been working.
>>
>> <extension name="Free_Google_Text_To_Speech">
>>   <condition field="destination_number" expression="^2115$">
>>     <action application="answer" data=""/>
>>     <action application="playback"
>> data="shout://translate.google.com/translate_tts?tl=en&q=Buy+Cheap+dids+at+www+dot+voice+network+dot+see+eh"/>
>>   </condition>
>> </extension>
>>     
> Thanks Norman. I tried your dialplan above and got the following even though
> I have mod_shout loaded.
> EXECUTE sofia/internal/1003 at 192.168.1.15
> playback(shout://translate.google.com/translate_tts?tl=en&q=Hello+,+It+is+me)
> 2011-01-13 20:41:57.388295 [ERR] mod_shout.c:800 Error: MPG123 Error at
> __FILE__:__LINE__.
> 2011-01-13 20:41:57.389383 [ERR] mod_shout.c:803 Error from mpg123: Invalid
> mpg123 handle. (code 10)
> 2011-01-13 20:41:57.397162 [DEBUG] sofia.c:4626 Channel
> sofia/internal/1003 at 192.168.1.15 entering state [ready][200]
> 2011-01-13 20:41:57.402589 [NOTICE] switch_core_state_machine.c:189
> sofia/internal/1003 at 192.168.1.15 has executed the last dialplan instruction,
> hanging up.
> 2011-01-13 20:41:57.402589 [DEBUG] switch_channel.c:2535
> (sofia/internal/1003 at 192.168.1.15) Callstate Change ACTIVE -> HANGUP
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
>   


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de




From ewin.hogan at gmail.com  Fri Jan 14 05:47:42 2011
From: ewin.hogan at gmail.com (Edwin Moedano)
Date: Thu, 13 Jan 2011 20:47:42 -0600
Subject: [Freeswitch-users] deflect question
In-Reply-To: <01AF6B571F2B4BAE944C61206687FB19@e1705>
References: <01AF6B571F2B4BAE944C61206687FB19@e1705>
Message-ID: <0546EC45-A71B-43A2-BE08-A8B43853DC52@gmail.com>

Could be an extension on the dp or a registered extension to fs or a PBX or a GW

:)

Edwin Moedano

El 12/01/2011, a las 03:08 p.m., "Madovsky" <infos at madovsky.org> escribi?:

>  
> if I use
>  <action application="deflect" data="sip:someone at otherFS.com"/>
> in the otherFS dialplan should I create
> an extension with "someone" condition only ?
>  
> Thanks
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From bernhard.suttner at winet.ch  Fri Jan 14 12:42:10 2011
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Fri, 14 Jan 2011 10:42:10 +0100
Subject: [Freeswitch-users] Outgoing registrations within a fs pool
In-Reply-To: <4D2F6793.4010801@cupis.co.uk>
References: <3ff41827-ea0f-4302-8d70-c7ee16feec83@winet.ch>
	<4D2F6793.4010801@cupis.co.uk>
Message-ID: <3b82d49f-a3be-4a5f-925b-b125fe378645@winet.ch>

Hi,

Kamailio is able to do a registration with the uacreg module, but it is not as stable as on FreeSWITCH (its not the main task of Kamailio to implement B2BUA features). Therefore I asked if FreeSWITCH is able to register "from a pool of FS servers".

Best regards,
Bernhard

-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Paul Cupis
Gesendet: Donnerstag, 13. Januar 2011 21:59
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Outgoing registrations within a fs pool

Perhaps look at getting the kamailio server to register to the provider
rather than one of the FS from the pool?


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From jalsot at gmail.com  Fri Jan 14 13:14:56 2011
From: jalsot at gmail.com (Tamas Jalsovszky)
Date: Fri, 14 Jan 2011 11:14:56 +0100
Subject: [Freeswitch-users] Removing inband DTMF
Message-ID: <AANLkTikBZPbeKLO_fX=s4p1pZxPCzwcAet1win8eaUMn@mail.gmail.com>

Hello,

Is there a way to remove an inband DTMF signal from the RTP stream with
FreeSWITCH?
We have a partner with ugly Cirpack which can not remove inband DTMF when
the codec is g711. The problem is, that our other party wants RFC2833 only -
thus we have to remove inband dtmf. Unfortunately Cirpack is not open to do
this.

Any advice?

Kind regards,
   T.
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From andy at fabulous4.co.uk  Fri Jan 14 13:16:08 2011
From: andy at fabulous4.co.uk (Andy Ayers)
Date: Fri, 14 Jan 2011 10:16:08 -0000
Subject: [Freeswitch-users] RTCP Keep Alive issue - hangup after 60 seconds
	of silence
Message-ID: <006001cbb3d4$11487130$33d95390$@fabulous4.co.uk>

Hi,

 

Have encountered a problem recently where the company responsible for
forwarding sip calls to my switch has started hanging up calls when there is
more than 60 seconds of silence on one side of the call. As my service is
primarily used for recording incoming messages, this means that any message
being recorded longer than 60 seconds gets cut off. My provider says I need
to configure freeswitch to send rtcp keep-alive packets to prevent them from
hanging up the call.

 

Can anyone tell me how I can do this, I've checked the docs and can't seem
to find the right setting?

 

Many thanks

Andy

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From Avi at aMarcus.com  Fri Jan 14 13:23:45 2011
From: Avi at aMarcus.com (Avi Marcus)
Date: Fri, 14 Jan 2011 12:23:45 +0200
Subject: [Freeswitch-users] Removing inband DTMF
In-Reply-To: <AANLkTikBZPbeKLO_fX=s4p1pZxPCzwcAet1win8eaUMn@mail.gmail.com>
References: <AANLkTikBZPbeKLO_fX=s4p1pZxPCzwcAet1win8eaUMn@mail.gmail.com>
Message-ID: <AANLkTimYd8PZumEY1GzkcAj-kZ1auH8Lvr-7OBm7r3fB@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf
start_dtmf does detection from the inband. I'm not sure that it removes them
from the rtp, though...
-Avi

On Fri, Jan 14, 2011 at 12:14 PM, Tamas Jalsovszky <jalsot at gmail.com> wrote:
> Hello,
>
> Is there a way to remove an inband DTMF signal from the RTP stream with
> FreeSWITCH?
> We have a partner with ugly Cirpack which can not remove inband DTMF when
> the codec is g711. The problem is, that our other party wants RFC2833 only
-
> thus we have to remove inband dtmf. Unfortunately Cirpack is not open to
do
> this.
>
> Any advice?
>
> Kind regards,
>    T.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cmrienzo at gmail.com  Fri Jan 14 15:49:08 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Fri, 14 Jan 2011 07:49:08 -0500
Subject: [Freeswitch-users] RTCP Keep Alive issue - hangup after 60
 seconds of silence
In-Reply-To: <006001cbb3d4$11487130$33d95390$@fabulous4.co.uk>
References: <006001cbb3d4$11487130$33d95390$@fabulous4.co.uk>
Message-ID: <AANLkTinAKvgAdUsFOci-ORR8t5KFGEWNZ95b9hcJHpH3@mail.gmail.com>

I don't know what RTCP keep alive is, but if they just mean to turn on RTCP,
you can do it with the following params in your sofia configuration:

 <!-- enable rtcp on every channel also can be done per leg basis
      with rtcp_audio_interval_msec variable set to passthru to pass
      it across a call -->
 <param name="rtcp-audio-interval-msec" value="5000"/>

or, set the rtcp_audio_interval_msec channel variable.

See http://wiki.freeswitch.org/wiki/RTCP




On Fri, Jan 14, 2011 at 5:16 AM, Andy Ayers <andy at fabulous4.co.uk> wrote:

> Hi,
>
>
>
> Have encountered a problem recently where the company responsible for
> forwarding sip calls to my switch has started hanging up calls when there is
> more than 60 seconds of silence on one side of the call. As my service is
> primarily used for recording incoming messages, this means that any message
> being recorded longer than 60 seconds gets cut off. My provider says I need
> to configure freeswitch to send rtcp keep-alive packets to prevent them from
> hanging up the call.
>
>
>
> Can anyone tell me how I can do this, I?ve checked the docs and can?t seem
> to find the right setting?
>
>
>
> Many thanks
>
> Andy
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Fri Jan 14 16:40:13 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 14 Jan 2011 14:40:13 +0100
Subject: [Freeswitch-users] Freeswitch & SNMP ( again ,
 but with a difference )
In-Reply-To: <AANLkTimv8tLJuaQf2=7_c+8UxLtOa_HPtOz7NkHoeD2Z@mail.gmail.com>
References: <ec2ce17e1003040038q153b5b8dgdc0137b3f152dca3@mail.gmail.com>
	<65d96fc81003041244t363450f2p8bf853c788d2295d@mail.gmail.com>
	<ec2ce17e1003041425p4e70995crf1e7b447a981e283@mail.gmail.com>
	<65d96fc81003050144k76ffdf70rea30b109e2b19392@mail.gmail.com>
	<AANLkTimv8tLJuaQf2=7_c+8UxLtOa_HPtOz7NkHoeD2Z@mail.gmail.com>
Message-ID: <AANLkTimzgyZn8AC6jcR=OTorCAtTUkJxCaZazgwFTAge@mail.gmail.com>

hello Jay,

i hope its not too late for this :P
updated the page: http://wiki.freeswitch.org/wiki/Snmp


Tihomir.



On Thu, Oct 7, 2010 at 11:48 PM, jay binks <jaybinks at gmail.com> wrote:

> sorry to bug you, but
> did you manage to get anywhere with this ??
>
> I think we need to try and expand this sort of stuff in the wiki...
> maybe Michael Collins would like stuff like this for a few miscellaneous
> recopies for his cook book ? :)
>
> Jay
>
>
> On Fri, Mar 5, 2010 at 7:44 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>>
>>
>> On Thu, Mar 4, 2010 at 11:25 PM, jay binks <jaybinks at gmail.com> wrote:
>>
>>> HUH, well there you go..   exactly what I was after...
>>> someone has done exactly what I was thinking.
>>>
>>> I know you've sent it to me by email, but lets get this in the wiki.
>>> Ill put this up from your email, but if you have more id encourage you to
>>> share what youve done.
>>>
>>>
>>
>> let me finish the current project ..."wholesale routing machine". and i
>> will put everything on wiki
>>
>>
>>
>> T.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Sincerely
>
> Jay
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Fri Jan 14 18:00:15 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 14 Jan 2011 09:00:15 -0600
Subject: [Freeswitch-users] RTCP Keep Alive issue - hangup after 60
	seconds of silence
In-Reply-To: <AANLkTinAKvgAdUsFOci-ORR8t5KFGEWNZ95b9hcJHpH3@mail.gmail.com>
References: <006001cbb3d4$11487130$33d95390$@fabulous4.co.uk>
	<AANLkTinAKvgAdUsFOci-ORR8t5KFGEWNZ95b9hcJHpH3@mail.gmail.com>
Message-ID: <5CE50C9B-6F53-4A1A-8695-88B6E4073FDC@freeswitch.org>

While this will turn on RTCP your provider needs to be beaten for requiring such a resource wasting process.

/b

On Jan 14, 2011, at 6:49 AM, Christopher Rienzo wrote:

> I don't know what RTCP keep alive is, but if they just mean to turn on RTCP, you can do it with the following params in your sofia configuration:
>  <!-- enable rtcp on every channel also can be done per leg basis 
> 
>       with rtcp_audio_interval_msec variable set to passthru to pass 
>       it across a call -->
>  <param name="rtcp-audio-interval-msec" value="5000"/>
> or, set the rtcp_audio_interval_msec channel variable.
> 
> See http://wiki.freeswitch.org/wiki/RTCP

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From brian at freeswitch.org  Fri Jan 14 18:33:15 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 14 Jan 2011 09:33:15 -0600
Subject: [Freeswitch-users] Peculiar behavior in FreeSWITCH when media
	streams for SRTP and RTP are offered in the same SDP
In-Reply-To: <AANLkTim6a7=dq8HA7pVPDjs2kJDJtM_HEy8SYQ9ariA9@mail.gmail.com>
References: <AANLkTim6a7=dq8HA7pVPDjs2kJDJtM_HEy8SYQ9ariA9@mail.gmail.com>
Message-ID: <48C09B53-A630-489B-8B52-4CCDE9FB0ECA@freeswitch.org>

You're forgetting to set sip_secure_media=true before the call is answered.

/b

On Jan 12, 2011, at 5:26 AM, Goutham BG wrote:

> A Polycom SoundPoint IP 550 configured in "SRTP best effort" mode dials into this extension and is connected to the IVR. In this mode, the phone offers two media streams in the SDP of INVITE; 1st one for SRTP and the 2nd one for RTP. But the problem is that the media is established in SRTP in one way and RTP in the other way.



From Joshua.Foshee at LogixCom.com  Fri Jan 14 18:42:58 2011
From: Joshua.Foshee at LogixCom.com (Joshua Foshee)
Date: Fri, 14 Jan 2011 09:42:58 -0600
Subject: [Freeswitch-users] CDR Frontend Report/GUI
Message-ID: <06502C073AD9394AADB3CA7FD94931BC0519E179@okc1x1.Logixcom.com>

I have a client that is wanting a polished commercial CDR GUI frontend.
They have been looking at Fonality and trixbox pro reports and want
something similar. I have attached address to some screen shots below.
Does anyone have any suggestions on commercial or open source? If not I
might be looking to put out a bounty if we can't find one that has all
the features the client is looking for.
 
Thanks,
Josh
 
http://cdnso.pbxtra.fonality.com/images/screenshots/800/ss-callcenter-ag
entreport.jpg
 
http://cdnso.pbxtra.fonality.com/images/screenshots/800/ss-powerfulrepor
ting-calldistribution.jpg
 
http://cdnso.pbxtra.fonality.com/images/screenshots/800/ss_powerful_repo
rting.jpg
 
http://web2.fonality.com/images/call_dist-snapshot.gif
 
http://web2.fonality.com/images/call_dist-month.gif
 
http://web2.fonality.com/images/call_dist-day.gif
 
http://web2.fonality.com/images/call_dist-weekday.gif
 
http://web2.fonality.com/images/call_dist-hour.gif
 
http://web2.fonality.com/images/abandoned-detail-(as-admin).gif
 
 
 
 
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From rupa at rupa.com  Fri Jan 14 18:46:34 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 14 Jan 2011 09:46:34 -0600
Subject: [Freeswitch-users] mod_lcr and extra_vars
In-Reply-To: <4D2B452D.8060004@tagnet.ru>
References: <4D2B452D.8060004@tagnet.ru>
Message-ID: <AANLkTimu-qW6BRJC3crQLy8uYavepzuxRVcarBP1h6Lg@mail.gmail.com>

You aren't quite using it right.  You need to create a distinct sql
field for each var you want imported.  Not a single field with a list
of var=value like you show below.

On Mon, Jan 10, 2011 at 11:43 AM, Boris Kovalenko <boris at tagnet.ru> wrote:
> Hello!
>
> ? ? I need to set extra vars with mod_lcr. I did as wiki recomended:
> 1) created sql column
> 2) modified sql quer
> 3) added <param name="export_fields" value="lcr_gw_extra_vars"/> to profile
> So, may lcr output looks nice:
> | Digit Match | Carrier ? ? | Rate ? ? | Codec | CID Regexp ? ? ? ? ?|
> Limit |
> Dialstring
> |
> ?| 734353 ? ? ?| tagnet.ru ? | 0.00000 ?| ? ? ? |
> | ? ? ? |
> [lcr_carrier=tagnet.ru,lcr_rate=0.00000,lcr_gw_extra_vars=,v_tagnet_ats_dstport=50000]sofia/epbx/73435350101 at X.X.X.X:5060
>
> Unfortunatelly I can't see (and can't use) v_tagnet_ats_dstport variable
> in my cdr records (even with b-leg the variable isn't present) until I
> set import=v_tagnet_ats_dstport variable. But I thinked mod_lcr should
> do this?
> Something wrong with my configuration?
>
>
> --
> ? ?????????,
> ? ????? ?????????
> ? ??? "??????"
> ? (3435) 494991
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa


From jmesquita at freeswitch.org  Fri Jan 14 19:10:40 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 14 Jan 2011 13:10:40 -0300
Subject: [Freeswitch-users] CDR Frontend Report/GUI
In-Reply-To: <06502C073AD9394AADB3CA7FD94931BC0519E179@okc1x1.Logixcom.com>
References: <06502C073AD9394AADB3CA7FD94931BC0519E179@okc1x1.Logixcom.com>
Message-ID: <AANLkTi=LzhoGe5-Zu8Ra6vhmTGRnST6PDLVOXc3+H45f@mail.gmail.com>

You should contact consulting at freeswitchsolutions.com and I bet they can
find someone to develop those. The question is, you are talking about a CDR
system based on FreeSWITCH's CDR, right?

Most of the reports you've sent are based on the crappy app_queue cdr that
stinks and I bet most of us that have played with that won't play with it
EVER again. LOL.

Regards,
Jo?o Mesquita


On Fri, Jan 14, 2011 at 12:42 PM, Joshua Foshee
<Joshua.Foshee at logixcom.com>wrote:

>  I have a client that is wanting a polished commercial CDR GUI frontend.
> They have been looking at Fonality and trixbox pro reports and want
> something similar. I have attached address to some screen shots below. Does
> anyone have any suggestions on commercial or open source? If not I might be
> looking to put out a bounty if we can't find one that has all the features
> the client is looking for.
>
> Thanks,
> Josh
>
>
> http://cdnso.pbxtra.fonality.com/images/screenshots/800/ss-callcenter-agentreport.jpg
>
>
> http://cdnso.pbxtra.fonality.com/images/screenshots/800/ss-powerfulreporting-calldistribution.jpg
>
>
> http://cdnso.pbxtra.fonality.com/images/screenshots/800/ss_powerful_reporting.jpg
>
> http://web2.fonality.com/images/call_dist-snapshot.gif
>
> http://web2.fonality.com/images/call_dist-month.gif
>
> http://web2.fonality.com/images/call_dist-day.gif
>
> http://web2.fonality.com/images/call_dist-weekday.gif
>
> http://web2.fonality.com/images/call_dist-hour.gif
>
> http://web2.fonality.com/images/abandoned-detail-(as-admin).gif
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kapil.rastogi at telemune.net  Fri Jan 14 14:13:36 2011
From: kapil.rastogi at telemune.net (kapil.rastogi)
Date: Fri, 14 Jan 2011 03:13:36 -0800 (PST)
Subject: [Freeswitch-users] How to kick a member from conference using
 java application (Kapil Rastogi)
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57EC81C50A@cooper>
References: <1294929071769-5918355.post@n2.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57EC81C50A@cooper>
Message-ID: <1295003615334-5921362.post@n2.nabble.com>


Dear Peter,

Thanx for your reply. Can u please me your sample code to kick a member from
conference. May be i am doing other mistake in my code. 

Also plz send me configuration (if any) related to this request in
freeswitch.

Thanks in advance.

-----
Regards,
Kapil Rastogi
Telemune Software Solutions P Ltd.
kapil.rastogi at telemune.net
+919013204760
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-kick-a-member-from-conference-using-java-application-Kapil-Rastogi-tp5918355p5921362.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From hwnorman at hotmail.com  Fri Jan 14 10:06:57 2011
From: hwnorman at hotmail.com (Norman Lam)
Date: Fri, 14 Jan 2011 15:06:57 +0800
Subject: [Freeswitch-users] Iksemel msvs compiling
References: <SNT109-DS13313A1FA906527438E8CC1090@phx.gbl>	<SNT109-DS1155D401A0B9742CF63E13C10B0@phx.gbl>	<1294375438447-5898181.post@n2.nabble.com>	<SNT109-DS11CB0ECE5BD0F9C0384E03C10C0@phx.gbl>	<1294535404901-5903534.post@n2.nabble.com>	<SNT109-DS14B90C48B33126784F294AC10E0@phx.gbl>
	<1294673236806-5907287.post@n2.nabble.com> 
Message-ID: <SNT109-DS227CE87422187AA91545CFC1F30@phx.gbl>

Hi jeff , I didn't modify the project , I follow your instruction and still
getting the error. Also follow the wiki mod_dingaling document. 

Look forward to your guidance

Norman lam 

-----Original Message-----
From: Norman Lam [mailto:hwnorman at hotmail.com] 
Sent: Tuesday, January 11, 2011 6:02 PM
To: 'FreeSWITCH Users Help'
Subject: RE: [Freeswitch-users] Iksemel msvs compiling

Hi Jeff 

I am still having the same error, after putting
;..\..\pthreads-w32-2-7-0-release;
at the end of the list

Norman 

28>------ Build started: Project: iksemel, Configuration: Debug Win32 
28>------ Compiling...
26>v3_conf.c
28>dom.c
28>..\..\iksemel\src\dom.c(152) : error C2065: 'ENOENT' : undeclared 
28>identifier filter.c
28>..\..\iksemel\src\filter.c(59) : error C2059: syntax error : 'type'
28>..\..\iksemel\src\filter.c(64) : error C2059: syntax error : 'type'
28>..\..\iksemel\src\filter.c(67) : error C2059: syntax error : 'type'
28>..\..\iksemel\src\filter.c(70) : error C2059: syntax error : 'type'
28>..\..\iksemel\src\filter.c(73) : error C2059: syntax error : 'type'
28>..\..\iksemel\src\filter.c(76) : error C2059: syntax error : 'type'
28>..\..\iksemel\src\filter.c(79) : error C2059: syntax error : 'type'
28>iks.c
28>ikstack.c
28>io-posix.c
26>v3_bitst.c
28>jabber.c
28>md5.c
26>v3_bcons.c
28>sax.c
28>sha.c
28>stream.c
28>..\..\iksemel\src\stream.c(19) : fatal error C1083: Cannot open 
28>include file: 'gnutls/gnutls.h': No such file or directory utility.c 
28>base64.c Generating Code...
28>Build log was saved at
"file://c:\FS_GIT2\libs\win32\iksemel\Debug\BuildLog.htm"
28>iksemel - 9 error(s), 0 warning(s)
:
:
125>------ Build started: Project: mod_dingaling, Configuration: Debug 
125>Win32 ------ Compiling...
125>cl : Command line warning D9040 : ignoring option '/analyze'; Code 
125>Analysis warnings are not available in this edition of the compiler 
125>mod_dingaling.c Linking...
125>LINK : fatal error LNK1181: cannot open input file
'..\..\..\..\libs\win32\iksemel\debug\iksemel.lib'
125>Build log was saved at
"file://c:\FS_GIT2\src\mod\endpoints\mod_dingaling\Win32\Debug\BuildLog.htm"
125>mod_dingaling - 1 error(s), 1 warning(s)
:
:
====== Build: 135 succeeded, 2 failed, 0 up-to-date, 3 skipped ==========
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff
Lenk
Sent: Monday, January 10, 2011 11:27 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Iksemel msvs compiling


make sure you put the include path for ;..\..\pthreads-w32-2-7-0-release;
at the end of the list. There seems to be some include file conflicts but
this seems to take care of it.

If you are succesfull with all of this please take the time to update the
wiki with more detail.
--
View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/Iksemel-msvs-compiling-tp58912
63p5907287.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From txbitrot at hotmail.com  Fri Jan 14 15:06:24 2011
From: txbitrot at hotmail.com (Bit Rot)
Date: Fri, 14 Jan 2011 12:06:24 +0000
Subject: [Freeswitch-users] =?windows-1252?q?SIP/Sofia_create_channel_503_?=
 =?windows-1252?q?error=85?=
Message-ID: <SNT124-W6526226B1E4BE6817AF9D7D5F30@phx.gbl>


 
Hey y'all,
 
We have FreeSWITCH and Microsoft Speech Server 2007 running fine on separate boxes.  However when we try to consolidate onto a single box as per instructions for inbound DID calls in Brian Campbell?s most excellent blog: 
 
http://gotspeech.net/blogs/verbalinput/archive/2010/01/08/speech-server-2007-marries-freeswitch-part-4-receiving-calls.aspx 
 
We receive a 503 error ? terminate and SIP returns Error Cause: 41 [NORMAL_TEMPORARY_FAILURE].
 
The first leg from the provider to FreeSwitch works fine and the dialplan is parsed correctly, also outbound calls through FreeSWITCH work.  The problem occurs when the second leg of the call is trying to create a channel from FreeSWITCH to the Speech Server.  Here is the trace:
 
EXECUTE sofia/external/106308_1 at 74.54.54.178 bridge(sofia/internal/2146130149 at 127.0.0.1:5060;transport=tcp)
2011-01-13 22:23:27.387971 [NOTICE] switch_channel.c:784 New Channel sofia/internal/2146130149 at 127.0.0.1:5060 [1df88e9c-16e6-415a-ac1b-83e29d25b3b0]
2011-01-13 22:23:27.387971 [DEBUG] mod_sofia.c:3976 (sofia/internal/2146130149 at 127.0.0.1:5060) State Change CS_NEW -> CS_INIT
2011-01-13 22:23:27.387971 [DEBUG] switch_core_session.c:1064 Send signal sofia/internal/2146130149 at 127.0.0.1:5060 [BREAK]
2011-01-13 22:23:27.387971 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/2146130149 at 127.0.0.1:5060) Running State Change CS_INIT
2011-01-13 22:23:27.387971 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/2146130149 at 127.0.0.1:5060) State INIT
2011-01-13 22:23:27.387971 [DEBUG] mod_sofia.c:86 sofia/internal/2146130149 at 127.0.0.1:5060 SOFIA INIT
2011-01-13 22:23:27.388971 [DEBUG] mod_sofia.c:126 (sofia/internal/2146130149 at 127.0.0.1:5060) State Change CS_INIT -> CS_ROUTING
2011-01-13 22:23:27.388971 [DEBUG] switch_core_session.c:1064 Send signal sofia/internal/2146130149 at 127.0.0.1:5060 [BREAK]
2011-01-13 22:23:27.388971 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/2146130149 at 127.0.0.1:5060) State INIT going to sleep
2011-01-13 22:23:27.388971 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/2146130149 at 127.0.0.1:5060) Running State Change CS_ROUTING
2011-01-13 22:23:27.388971 [DEBUG] switch_channel.c:1615 (sofia/internal/2146130149 at 127.0.0.1:5060) Callstate Change DOWN -> RINGING
2011-01-13 22:23:27.388971 [DEBUG] switch_core_state_machine.c:359 (sofia/internal/2146130149 at 127.0.0.1:5060) State ROUTING
2011-01-13 22:23:27.388971 [DEBUG] mod_sofia.c:149 sofia/internal/2146130149 at 127.0.0.1:5060 SOFIA ROUTING
2011-01-13 22:23:27.388971 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/2146130149 at 127.0.0.1:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2011-01-13 22:23:27.388971 [DEBUG] switch_core_session.c:1064 Send signal sofia/internal/2146130149 at 127.0.0.1:5060 [BREAK]
2011-01-13 22:23:27.388971 [DEBUG] switch_core_state_machine.c:359 (sofia/internal/2146130149 at 127.0.0.1:5060) State ROUTING going to sleep
2011-01-13 22:23:27.388971 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/2146130149 at 127.0.0.1:5060) Running State Change CS_CONSUME_MEDIA
2011-01-13 22:23:27.388971 [DEBUG] switch_core_state_machine.c:378 (sofia/internal/2146130149 at 127.0.0.1:5060) State CONSUME_MEDIA
2011-01-13 22:23:27.388971 [DEBUG] switch_core_state_machine.c:378 (sofia/internal/2146130149 at 127.0.0.1:5060) State CONSUME_MEDIA going to sleep
 
2011-01-13 22:23:27.389971 [DEBUG] sofia.c:4604 Channel sofia/internal/2146130149 at 127.0.0.1:5060 entering state [calling][0]
2011-01-13 22:23:27.389971 [DEBUG] sofia.c:4604 Channel sofia/internal/2146130149 at 127.0.0.1:5060 entering state [terminated][503]
 
2011-01-13 22:23:27.389971 [DEBUG] switch_channel.c:2455 (sofia/internal/2146130149 at 127.0.0.1:5060) Callstate Change RINGING -> HANGUP
2011-01-13 22:23:27.389971 [NOTICE] sofia.c:5244 Hangup sofia/internal/2146130149 at 127.0.0.1:5060 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
2011-01-13 22:23:27.389971 [DEBUG] switch_channel.c:2471 Send signal sofia/internal/2146130149 at 127.0.0.1:5060 [KILL]
 
The Speech Server is listening at TCP 0.0.0.0:5060 and Sofia Internal at TCP IP Addy of box also on port 5060.
 
Any help or insight would be much appreciated.
 
Cheers.
 
bit
 		 	   		  
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From fabio.bigliardi at gmail.com  Fri Jan 14 19:22:23 2011
From: fabio.bigliardi at gmail.com (Fabio Bigliardi)
Date: Fri, 14 Jan 2011 17:22:23 +0100
Subject: [Freeswitch-users] How to set sip_callee_id_name in a conference
Message-ID: <AANLkTiksge_egVYtW=iLcMWrbd4E+X0AdGDd+Dnsr7A-@mail.gmail.com>

Hi all,
I would like to configure a dialplan so that a tag with the name of the
conference appears in the calling phone's display.

Thanks a lot for your support.

Best regards,
F. Bigliardi
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From brian at freeswitch.org  Fri Jan 14 19:25:41 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 14 Jan 2011 10:25:41 -0600
Subject: [Freeswitch-users] FreeSWITCH 1.0.7
Message-ID: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org>

http://latest.freeswitch.org/

Enjoy!

/b

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From marcdecorny at gmail.com  Fri Jan 14 19:35:21 2011
From: marcdecorny at gmail.com (Marc de Corny)
Date: Fri, 14 Jan 2011 16:35:21 +0000
Subject: [Freeswitch-users] Caller ID using Fifo
In-Reply-To: <AANLkTi=p7_CsUDWS62+weZv50+PBZ8RLop0Q04y-c4hB@mail.gmail.com>
References: <AANLkTik6pdgXgND6kmzZA4qe8vDZQ0_7e47+ciVWPfub@mail.gmail.com>
	<AANLkTikKDO3qru-9=nuPEQcW+iCN7ccizyJAh9sqUkCN@mail.gmail.com>
	<AANLkTi=p7_CsUDWS62+weZv50+PBZ8RLop0Q04y-c4hB@mail.gmail.com>
Message-ID: <AANLkTi=m_hW9vQV5nbC2jwcos0bKy2rCf7YAr6aA23oz@mail.gmail.com>

Just to follow up on this subject.

I have done a lot of testing on the fifo trying to get the caller_id_name
changed on the outbound call to the agent and to be honest I cannot
understand the explanation.

If mod_fifo does not know which call it will connect until the agent
answers, how come it displays the CLI correctly, jsut won;t let me change
it.

Still seems strange. I am looking into the Mod_callcentre to check if it
sends caller_id information. but the same logic if valid could apply

Also maybe someone should change the Wiki ( I would but do not have enough
expertise on the subject) because the following is a bit misleading

 "Note: If you wish to specify the caller ID presented when a fifo calls an
agent, set the origination_caller_id_name and origination_caller_id_num
variables to the values desired. These could be set within the {} of the
dialstring, or they could be set using the set application in the dialplan
which places the caller into the fifo (before the 'fifo in' executed on the
caller). "
thanks
Marc
On Thu, Jan 13, 2011 at 10:47 PM, Joao Leme <joaocarlosleme at gmail.com>wrote:

> What about showing the Caller ID after it is answered? Any way to do that?
>
> 2011/1/12 Jo?o Mesquita <jmesquita at freeswitch.org>
>
> Jo?o Leme,
>>
>> The caller id is not passed when the phone is ringing because mod_fifo
>> does not know which call is going to be sent to that channel once it is
>> answered until it is really answered. I don't know if mod_callcenter does
>> show anything but you should consider looking at the documentation if you
>> really need this feature.
>>
>> Regards,
>> Jo?o Mesquita
>>
>>
>>   On Wed, Jan 12, 2011 at 9:15 PM, Joao Leme <joaocarlosleme at gmail.com>wrote:
>>
>>>  Hi there,
>>> I would like to know if there is a way to see the caller ID on my Sip
>>> Client (X-Lite for example) of the caller that I answear from a Fifo queue?
>>> Thanks,
>>> John
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Fri Jan 14 19:38:05 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 14 Jan 2011 16:38:05 +0000
Subject: [Freeswitch-users] How to set sip_callee_id_name in a conference
In-Reply-To: <AANLkTiksge_egVYtW=iLcMWrbd4E+X0AdGDd+Dnsr7A-@mail.gmail.com>
References: <AANLkTiksge_egVYtW=iLcMWrbd4E+X0AdGDd+Dnsr7A-@mail.gmail.com>
Message-ID: <AANLkTikRdk2ZMEvbJb_J=UAoX-o0o0NW35bLwTcqiESK@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_send_display

For example:

<extension name="test_conference">
   <condition field="destination_number" expression="^(3000)$">
      <action application="answer"/>
      <action application="send_display data="confname"/>
      <action application="conference" data="confname at confprofile"/>
   </condition>
</extension>

-Steve

On 14 January 2011 16:22, Fabio Bigliardi <fabio.bigliardi at gmail.com> wrote:

> Hi all,
> I would like to configure a dialplan so that a tag with the name of the
> conference appears in the calling phone's display.
>
> Thanks a lot for your support.
>
> Best regards,
> F. Bigliardi
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Fri Jan 14 19:40:42 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 14 Jan 2011 16:40:42 +0000
Subject: [Freeswitch-users]
	=?windows-1252?q?SIP/Sofia_create_channel_503_?=
	=?windows-1252?q?error=85?=
In-Reply-To: <SNT124-W6526226B1E4BE6817AF9D7D5F30@phx.gbl>
References: <SNT124-W6526226B1E4BE6817AF9D7D5F30@phx.gbl>
Message-ID: <AANLkTi=YzFBSvqf=_sdooGD2+kvpgre6kQ41LDXrvMZ8@mail.gmail.com>

Does it help if they're listening on different ports, e.g. Speech Server
port 5080 and FreeSWITCH port 5060?

Also try enabling the siptrace (sofia global siptrace on) and include that
in your logs.

-Steve




On 14 January 2011 12:06, Bit Rot <txbitrot at hotmail.com> wrote:

>
> Hey y'all,
>
> We have FreeSWITCH and Microsoft Speech Server 2007 running fine on
> separate boxes.  However when we try to consolidate onto a single box as per
> instructions for inbound DID calls in Brian Campbell?s most excellent blog:
>
>
> http://gotspeech.net/blogs/verbalinput/archive/2010/01/08/speech-server-2007-marries-freeswitch-part-4-receiving-calls.aspx
>
> We receive a 503 error ? terminate and SIP returns Error Cause: 41
> [NORMAL_TEMPORARY_FAILURE].
>
> The first leg from the provider to FreeSwitch works fine and the dialplan
> is parsed correctly, also outbound calls through FreeSWITCH work.  The
> problem occurs when the second leg of the call is trying to create a channel
> from FreeSWITCH to the Speech Server.  Here is the trace:
>
> EXECUTE sofia/external/106308_1 at 74.54.54.178 bridge(
> sofia/internal/2146130149 at 127.0.0.1:5060;transport=tcp)
> 2011-01-13 22:23:27.387971 [NOTICE] switch_channel.c:784 New Channel
> sofia/internal/2146130149 at 127.0.0.1:5060[1df88e9c-16e6-415a-ac1b-83e29d25b3b0]
> 2011-01-13 22:23:27.387971 [DEBUG] mod_sofia.c:3976 (
> sofia/internal/2146130149 at 127.0.0.1:5060) State Change CS_NEW -> CS_INIT
> 2011-01-13 22:23:27.387971 [DEBUG] switch_core_session.c:1064 Send signal
> sofia/internal/2146130149 at 127.0.0.1:5060 [BREAK]
> 2011-01-13 22:23:27.387971 [DEBUG] switch_core_state_machine.c:320 (
> sofia/internal/2146130149 at 127.0.0.1:5060) Running State Change CS_INIT
> 2011-01-13 22:23:27.387971 [DEBUG] switch_core_state_machine.c:356 (
> sofia/internal/2146130149 at 127.0.0.1:5060) State INIT
> 2011-01-13 22:23:27.387971 [DEBUG] mod_sofia.c:86
> sofia/internal/2146130149 at 127.0.0.1:5060 SOFIA INIT
> 2011-01-13 22:23:27.388971 [DEBUG] mod_sofia.c:126 (
> sofia/internal/2146130149 at 127.0.0.1:5060) State Change CS_INIT ->
> CS_ROUTING
> 2011-01-13 22:23:27.388971 [DEBUG] switch_core_session.c:1064 Send signal
> sofia/internal/2146130149 at 127.0.0.1:5060 [BREAK]
> 2011-01-13 22:23:27.388971 [DEBUG] switch_core_state_machine.c:356 (
> sofia/internal/2146130149 at 127.0.0.1:5060) State INIT going to sleep
> 2011-01-13 22:23:27.388971 [DEBUG] switch_core_state_machine.c:320 (
> sofia/internal/2146130149 at 127.0.0.1:5060) Running State Change CS_ROUTING
> 2011-01-13 22:23:27.388971 [DEBUG] switch_channel.c:1615 (
> sofia/internal/2146130149 at 127.0.0.1:5060) Callstate Change DOWN -> RINGING
> 2011-01-13 22:23:27.388971 [DEBUG] switch_core_state_machine.c:359 (
> sofia/internal/2146130149 at 127.0.0.1:5060) State ROUTING
> 2011-01-13 22:23:27.388971 [DEBUG] mod_sofia.c:149
> sofia/internal/2146130149 at 127.0.0.1:5060 SOFIA ROUTING
> 2011-01-13 22:23:27.388971 [DEBUG] switch_ivr_originate.c:66 (
> sofia/internal/2146130149 at 127.0.0.1:5060) State Change CS_ROUTING ->
> CS_CONSUME_MEDIA
> 2011-01-13 22:23:27.388971 [DEBUG] switch_core_session.c:1064 Send signal
> sofia/internal/2146130149 at 127.0.0.1:5060 [BREAK]
> 2011-01-13 22:23:27.388971 [DEBUG] switch_core_state_machine.c:359 (
> sofia/internal/2146130149 at 127.0.0.1:5060) State ROUTING going to sleep
> 2011-01-13 22:23:27.388971 [DEBUG] switch_core_state_machine.c:320 (
> sofia/internal/2146130149 at 127.0.0.1:5060) Running State Change
> CS_CONSUME_MEDIA
> 2011-01-13 22:23:27.388971 [DEBUG] switch_core_state_machine.c:378 (
> sofia/internal/2146130149 at 127.0.0.1:5060) State CONSUME_MEDIA
> 2011-01-13 22:23:27.388971 [DEBUG] switch_core_state_machine.c:378 (
> sofia/internal/2146130149 at 127.0.0.1:5060) State CONSUME_MEDIA going to
> sleep
>
> 2011-01-13 22:23:27.389971 [DEBUG] sofia.c:4604 Channel
> sofia/internal/2146130149 at 127.0.0.1:5060 entering state [calling][0]
> 2011-01-13 22:23:27.389971 [DEBUG] sofia.c:4604 Channel
> sofia/internal/2146130149 at 127.0.0.1:5060 entering state [terminated][503]
>
> 2011-01-13 22:23:27.389971 [DEBUG] switch_channel.c:2455 (
> sofia/internal/2146130149 at 127.0.0.1:5060) Callstate Change RINGING ->
> HANGUP
> 2011-01-13 22:23:27.389971 [NOTICE] sofia.c:5244 Hangup
> sofia/internal/2146130149 at 127.0.0.1:5060 [CS_CONSUME_MEDIA]
> [NORMAL_TEMPORARY_FAILURE]
> 2011-01-13 22:23:27.389971 [DEBUG] switch_channel.c:2471 Send signal
> sofia/internal/2146130149 at 127.0.0.1:5060 [KILL]
>
> The Speech Server is listening at TCP 0.0.0.0:5060 and Sofia Internal at
> TCP IP Addy of box also on port 5060.
>
> Any help or insight would be much appreciated.
>
> Cheers.
>
> bit
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Nabble at slickdeals.endjunk.com  Fri Jan 14 19:57:39 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 14 Jan 2011 08:57:39 -0800 (PST)
Subject: [Freeswitch-users] FreeSWITCH 1.0.7
In-Reply-To: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org>
References: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org>
Message-ID: <1295024259219-5922533.post@n2.nabble.com>


 Finally a new released version! 

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-1-0-7-tp5922408p5922533.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From infos at madovsky.org  Fri Jan 14 20:16:55 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 14 Jan 2011 12:16:55 -0500
Subject: [Freeswitch-users] FS as sip proxy registration
Message-ID: <DA6646E5B03140F2B20660208359F9DC@e1705>

if I use FS as SIP proxy registrar,
does the REFER method will go through the client transparently ?
I'm facing problem of this method with this configuration

Thanks
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From manieq at wp.eu  Fri Jan 14 19:31:33 2011
From: manieq at wp.eu (Mariusz Czulada)
Date: Fri, 14 Jan 2011 17:31:33 +0100
Subject: [Freeswitch-users] bind_meta_app blocks DTMFs?
Message-ID: <4d307a65a43472.49337517@wp.pl>

Hi All,

I'm trying to implement subconference feature, to allow some 
participants go (or rather "gosub") to a another conference room to 
discuss something then get back to the main conference. I use FreeSWITCH 
Version 1.0.head (hacked-20110112T172836Z). Bellow some configure 
extracts:

>From conference.conf.xml:
  ...
  <caller-controls>
    ...
    <group name="just_exit">
      <control action="hangup" digits="#"/>
      <control action="hangup" digits="1"/>
      <control action="mute" digits="0"/>
    </group>
    ...
  </caller-controls>
  ...
  <profiles>
    ..
    <profile name="confX_M">
      ...
      <param name="caller-controls" value="just_exit"/>
      ...
    </profile>
    <profile name="confX_S">
      ...
      <param name="caller-controls" value="just_exit"/>
      ...
    </profile>
  </profiles>
  ...


>From dialplan (own) public/confX.conf.xml:
    ...
    <extension name="confX_join">
      <condition field="destination_number" expression="^confX_join$"/>
      <condition field="${confX_num}" expression="^1(\d{4})$">
        <action application="bind_meta_app" data="1 a s 
execute_extension::subX_1"/>
        <action application="bind_meta_app" data="2 a s 
execute_extension::subX_2"/>
        <action application="bind_meta_app" data="3 a s 
execute_extension::subX_3"/>
        <action application="conference" data="confX_${cpnfX_num}@confX_M"/>
      </condition>
    </extension>
    ...
    <extension name="subX_1">
      <condition field="destination_number" expression="^subX_1$">
        <action application="set" data="pass_rfc2833=true"/>
        <action application="conference" data="confX_${cpnfX_num}_1 at confX_S"/>
      </condition>
    </extension>
    <extension name="subX_2">
      <condition field="destination_number" expression="^subX_2$">
        <action application="set" data="pass_rfc2833=true"/>
        <action application="conference" data="confX_${cpnfX_num}_2 at confX_S"/>
      </condition>
    </extension>
    <extension name="subX_3">
      <condition field="destination_number" expression="^subX_3$">
        <action application="set" data="pass_rfc2833=true"/>
        <action application="conference" data="confX_${cpnfX_num}_3 at confX_S"/>
      </condition>
    </extension>
    ...


During connection user is first handled by IVR, where he enters 
conference number (assume: "1234") which is stored in "confX_num" 
variable. Then call is transfered to "confX_join". Here I assign actions 
for *1, *2 and *3 DTMF sequences then join main conference 
("confX_1234_M"). When some user presses i.e. "*2" then he also enters 
subconf 2 ("confX_1234_2"). Those users attached to "confX_1234_2" are 
at the same time still attached also to "confX_1234_M", but found muted 
and deaf to other participants. Then, leaving subconference they will be 
back "alive" in master conference, which I tested by kicking them from 
"confX_1234_2" manually.

Now my problem begins. Then user is on "confX_1234_M" dialing "#" forces 
user to leave conference, as defined in "just-exit" controls. But when 
user enters also "confX_1234_2" DTMF codes are no longer processed, 
neither by master conference nor by subconference. I set 
"pass_rfc2833=true" but it is not helpful. Did I missed to set something 
[1] in the dialplan or other config files, or [2] actually no DTMF codes 
can be processed while in "*2" leg? And if [2]==true: would it be 
possible to develop such DTMF handling?

BTW: DTMFs are not processed while in subconference but if user is 
kicked off from subconf back to main conference, one can use DTMF codes 
again to (un)mute (0), join sub (*2) or leave (# or 1).

TIA for your help or tips.

Regards,
Mariusz




From michelhabib at gmail.com  Fri Jan 14 19:44:40 2011
From: michelhabib at gmail.com (Michel Habib)
Date: Fri, 14 Jan 2011 18:44:40 +0200
Subject: [Freeswitch-users] Processing Live Audio during a call between 2
	extensions
Message-ID: <AANLkTikTMazyD_Q9FOZj=Ky+sC+JAK4aTKE+cV6NgEWf@mail.gmail.com>

Dear Freeswitch Developers, i am relatively new to freeswitch and i am
seeking some advice.

I am creating an External Application that does some [live] processing on
the Audio of the freeswitch call before sending it to the second Call Leg,
and vice versa.
Can you please direct me to the best way to do that and how to
access/capture the live audio channel in both legs while it is streaming,
and how to resubmit it again to the other leg?
I read the book about freeswitch, but still couldnt figure the best approach
to do that, any hint is appreciated.

Best Regards,
Michel Habib.
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From louis.huppenbauer at gmail.com  Fri Jan 14 21:02:00 2011
From: louis.huppenbauer at gmail.com (Louis Huppenbauer)
Date: Fri, 14 Jan 2011 19:02:00 +0100
Subject: [Freeswitch-users] FreeSWITCH 1.0.7
In-Reply-To: <1295024259219-5922533.post@n2.nabble.com>
References: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org>
	<1295024259219-5922533.post@n2.nabble.com>
Message-ID: <AANLkTik6iOqpe=fBMOo9Z6JHx3eDg_6-UR6AVoSZ1if2@mail.gmail.com>

Thank's a bunch!

2011/1/14 mazilo <Nabble at slickdeals.endjunk.com>

>
>  Finally a new released version!
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-1-0-7-tp5922408p5922533.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From peter.olsson at visionutveckling.se  Fri Jan 14 21:29:18 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 14 Jan 2011 19:29:18 +0100
Subject: [Freeswitch-users] FS as sip proxy registration
Message-ID: <23834DF3-F6D6-48C0-AE24-77981C92D204@visionutveckling.se>

FS is not a proxy...

/Peter

----- Reply message -----
Fr?n: "Madovsky" <infos at madovsky.org>
Datum: fre, jan 14, 2011 18:24
Rubrik: [Freeswitch-users] FS as sip proxy registration
Till: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>

if I use FS as SIP proxy registrar,
does the REFER method will go through the client transparently ?
I'm facing problem of this method with this configuration

Thanks
!DSPAM:4d30866d32761881510302!


From msc at freeswitch.org  Fri Jan 14 21:39:18 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 14 Jan 2011 10:39:18 -0800
Subject: [Freeswitch-users] bind_meta_app blocks DTMFs?
In-Reply-To: <4d307a65a43472.49337517@wp.pl>
References: <4d307a65a43472.49337517@wp.pl>
Message-ID: <AANLkTinAx-cxv_a5b8+VA89duWY=xHjOOmMcZ604Z8R8@mail.gmail.com>

Mariusz,

After moving from the main conference to the second conference, when the
user presses digits, do any of them work? The way to test would be to call
in to the first conference then press #1 to go to the sub conference, then
press 0 a few times to see if you mute/unmute. Also, try pressing  1, 2, 3,
etc. and watch the FS console. Be sure that you are in debug mode on the
console. (If you use fs_cli then you will be at loglevel debug by default.)

Question - what customizations have you made? I see that your version is
"hacked" so most likely you've done some tinkering. One thing you can do to
test is to update to latest git and don't do any "hacking" to the code. Test
your dialplan and see if the DTMFs work as expected. If your DTMFs work on a
plain install then you know that your customizations are doing something. If
not, then pastebin your output (pastebin.freeswitch.org) and we'll take a
look.

-MC

On Fri, Jan 14, 2011 at 8:31 AM, Mariusz Czulada <manieq at wp.eu> wrote:

> Hi All,
>
> I'm trying to implement subconference feature, to allow some
> participants go (or rather "gosub") to a another conference room to
> discuss something then get back to the main conference. I use FreeSWITCH
> Version 1.0.head (hacked-20110112T172836Z). Bellow some configure
> extracts:
>
> >From conference.conf.xml:
>  ...
>  <caller-controls>
>    ...
>    <group name="just_exit">
>      <control action="hangup" digits="#"/>
>      <control action="hangup" digits="1"/>
>      <control action="mute" digits="0"/>
>    </group>
>    ...
>  </caller-controls>
>  ...
>  <profiles>
>    ..
>    <profile name="confX_M">
>      ...
>      <param name="caller-controls" value="just_exit"/>
>      ...
>    </profile>
>    <profile name="confX_S">
>      ...
>      <param name="caller-controls" value="just_exit"/>
>      ...
>    </profile>
>  </profiles>
>  ...
>
>
> >From dialplan (own) public/confX.conf.xml:
>    ...
>    <extension name="confX_join">
>      <condition field="destination_number" expression="^confX_join$"/>
>      <condition field="${confX_num}" expression="^1(\d{4})$">
>        <action application="bind_meta_app" data="1 a s
> execute_extension::subX_1"/>
>        <action application="bind_meta_app" data="2 a s
> execute_extension::subX_2"/>
>        <action application="bind_meta_app" data="3 a s
> execute_extension::subX_3"/>
>        <action application="conference" data="confX_${cpnfX_num}@confX_M
> "/>
>      </condition>
>    </extension>
>    ...
>    <extension name="subX_1">
>      <condition field="destination_number" expression="^subX_1$">
>        <action application="set" data="pass_rfc2833=true"/>
>        <action application="conference" data="confX_${cpnfX_num}_1 at confX_S
> "/>
>      </condition>
>    </extension>
>    <extension name="subX_2">
>      <condition field="destination_number" expression="^subX_2$">
>        <action application="set" data="pass_rfc2833=true"/>
>        <action application="conference" data="confX_${cpnfX_num}_2 at confX_S
> "/>
>      </condition>
>    </extension>
>    <extension name="subX_3">
>      <condition field="destination_number" expression="^subX_3$">
>        <action application="set" data="pass_rfc2833=true"/>
>        <action application="conference" data="confX_${cpnfX_num}_3 at confX_S
> "/>
>      </condition>
>    </extension>
>    ...
>
>
> During connection user is first handled by IVR, where he enters
> conference number (assume: "1234") which is stored in "confX_num"
> variable. Then call is transfered to "confX_join". Here I assign actions
> for *1, *2 and *3 DTMF sequences then join main conference
> ("confX_1234_M"). When some user presses i.e. "*2" then he also enters
> subconf 2 ("confX_1234_2"). Those users attached to "confX_1234_2" are
> at the same time still attached also to "confX_1234_M", but found muted
> and deaf to other participants. Then, leaving subconference they will be
> back "alive" in master conference, which I tested by kicking them from
> "confX_1234_2" manually.
>
> Now my problem begins. Then user is on "confX_1234_M" dialing "#" forces
> user to leave conference, as defined in "just-exit" controls. But when
> user enters also "confX_1234_2" DTMF codes are no longer processed,
> neither by master conference nor by subconference. I set
> "pass_rfc2833=true" but it is not helpful. Did I missed to set something
> [1] in the dialplan or other config files, or [2] actually no DTMF codes
> can be processed while in "*2" leg? And if [2]==true: would it be
> possible to develop such DTMF handling?
>
> BTW: DTMFs are not processed while in subconference but if user is
> kicked off from subconf back to main conference, one can use DTMF codes
> again to (un)mute (0), join sub (*2) or leave (# or 1).
>
> TIA for your help or tips.
>
> Regards,
> Mariusz
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Fri Jan 14 21:51:09 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 14 Jan 2011 13:51:09 -0500
Subject: [Freeswitch-users] FS as sip proxy registration
References: <23834DF3-F6D6-48C0-AE24-77981C92D204@visionutveckling.se>
Message-ID: <9341C51152A541A28E924C6D7BFE4289@e1705>

I know, but it can work as a registrar kind, I use it like this and it works

----- Original Message ----- 
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Friday, January 14, 2011 1:29 PM
Subject: Re: [Freeswitch-users] FS as sip proxy registration


FS is not a proxy...

/Peter

----- Reply message -----
Fr?n: "Madovsky" <infos at madovsky.org>
Datum: fre, jan 14, 2011 18:24
Rubrik: [Freeswitch-users] FS as sip proxy registration
Till: "freeswitch-users at lists.freeswitch.org" 
<freeswitch-users at lists.freeswitch.org>

if I use FS as SIP proxy registrar,
does the REFER method will go through the client transparently ?
I'm facing problem of this method with this configuration

Thanks
!DSPAM:4d30866d32761881510302!

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 



From brian at freeswitch.org  Fri Jan 14 21:53:54 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 14 Jan 2011 12:53:54 -0600
Subject: [Freeswitch-users] bind_meta_app blocks DTMFs?
In-Reply-To: <AANLkTinAx-cxv_a5b8+VA89duWY=xHjOOmMcZ604Z8R8@mail.gmail.com>
References: <4d307a65a43472.49337517@wp.pl>
	<AANLkTinAx-cxv_a5b8+VA89duWY=xHjOOmMcZ604Z8R8@mail.gmail.com>
Message-ID: <D5B07991-6832-475F-B56A-794D85BA039F@freeswitch.org>

learn to use GIT and perform a git pull if at all possible or download the 1.0.7 tarball from latest.freeswitch.org

/b

On Jan 14, 2011, at 12:39 PM, Michael Collins wrote:

> Mariusz,
> 
> After moving from the main conference to the second conference, when the user presses digits, do any of them work? The way to test would be to call in to the first conference then press #1 to go to the sub conference, then press 0 a few times to see if you mute/unmute. Also, try pressing  1, 2, 3, etc. and watch the FS console. Be sure that you are in debug mode on the console. (If you use fs_cli then you will be at loglevel debug by default.)
> 
> Question - what customizations have you made? I see that your version is "hacked" so most likely you've done some tinkering. One thing you can do to test is to update to latest git and don't do any "hacking" to the code. Test your dialplan and see if the DTMFs work as expected. If your DTMFs work on a plain install then you know that your customizations are doing something. If not, then pastebin your output (pastebin.freeswitch.org) and we'll take a look.
> 
> -MC

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From max.clark at gmail.com  Fri Jan 14 22:55:10 2011
From: max.clark at gmail.com (Max Clark)
Date: Fri, 14 Jan 2011 11:55:10 -0800
Subject: [Freeswitch-users] FreeSWITCH 1.0.7
In-Reply-To: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org>
References: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org>
Message-ID: <AANLkTimdEX_Hd=Hp8dtPxC1BZbkM4W8-Eiui4wrOrhf_@mail.gmail.com>

sweet!

On Fri, Jan 14, 2011 at 8:25 AM, Brian West <brian at freeswitch.org> wrote:
> http://latest.freeswitch.org/
>
> Enjoy!
> /b
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mastermind202 at gmail.com  Fri Jan 14 22:59:51 2011
From: mastermind202 at gmail.com (mm_202)
Date: Fri, 14 Jan 2011 14:59:51 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.0.7
In-Reply-To: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org>
References: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org>
Message-ID: <AANLkTimO0=ZG7Ck1aFJMXg+vwUqveX_yjw6sGxEgvXua@mail.gmail.com>

Any hope for a changelog of the changes since 1.0.6? :D

On Fri, Jan 14, 2011 at 11:25, Brian West <brian at freeswitch.org> wrote:
> http://latest.freeswitch.org/
>
> Enjoy!
> /b
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From ritzalam at gmail.com  Fri Jan 14 23:04:46 2011
From: ritzalam at gmail.com (Richard Alam)
Date: Fri, 14 Jan 2011 15:04:46 -0500
Subject: [Freeswitch-users] Confirming if conference is being recorded
Message-ID: <AANLkTi=o0wtYYdY8dbP=gsSkMgdBpetdSqM3Mkjr7g-r@mail.gmail.com>

Hi,

I am planning on recording a conference.

Is there are way to regularly check if the conference is being
recorded aside from regularly checking the recorded file increasing in
size?

Can I tell the conference app to send out events to ESL regularly
saying that it is recording?

Thanks in advance.

Richard

-- 
---
BigBlueButton
http://www.bigbluebutton.org
http://code.google.com/p/bigbluebutton


From brian at freeswitch.org  Fri Jan 14 23:07:19 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 14 Jan 2011 14:07:19 -0600
Subject: [Freeswitch-users] FreeSWITCH 1.0.7
In-Reply-To: <AANLkTimO0=ZG7Ck1aFJMXg+vwUqveX_yjw6sGxEgvXua@mail.gmail.com>
References: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org>
	<AANLkTimO0=ZG7Ck1aFJMXg+vwUqveX_yjw6sGxEgvXua@mail.gmail.com>
Message-ID: <2DBA22E6-61E5-4552-8C00-E47A7CD4E06A@freeswitch.org>

That is in there and maintained .

/b

On Jan 14, 2011, at 1:59 PM, mm_202 wrote:

> Any hope for a changelog of the changes since 1.0.6? :D
> 
> On Fri, Jan 14, 2011 at 11:25, Brian West <brian at freeswitch.org> wrote:
>> http://latest.freeswitch.org/
>> 
>> Enjoy!
>> /b

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From jmesquita at freeswitch.org  Fri Jan 14 23:08:54 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 14 Jan 2011 17:08:54 -0300
Subject: [Freeswitch-users] Confirming if conference is being recorded
In-Reply-To: <AANLkTi=o0wtYYdY8dbP=gsSkMgdBpetdSqM3Mkjr7g-r@mail.gmail.com>
References: <AANLkTi=o0wtYYdY8dbP=gsSkMgdBpetdSqM3Mkjr7g-r@mail.gmail.com>
Message-ID: <AANLkTimd0bMAqx-HGa2fgihQgONePwSHfOupog-wXesP@mail.gmail.com>

Why not use inbound esl and listen in for the start-recording and
stop-recording events?

I've added a couple of more event fires on the code lately so you won't miss
a single stop-recording even if it is at the end of the conference with
auto-record enabled.

Regards,
Jo?o Mesquita


On Fri, Jan 14, 2011 at 5:04 PM, Richard Alam <ritzalam at gmail.com> wrote:

> Hi,
>
> I am planning on recording a conference.
>
> Is there are way to regularly check if the conference is being
> recorded aside from regularly checking the recorded file increasing in
> size?
>
> Can I tell the conference app to send out events to ESL regularly
> saying that it is recording?
>
> Thanks in advance.
>
> Richard
>
> --
> ---
> BigBlueButton
> http://www.bigbluebutton.org
> http://code.google.com/p/bigbluebutton
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ritzalam at gmail.com  Fri Jan 14 23:15:18 2011
From: ritzalam at gmail.com (Richard Alam)
Date: Fri, 14 Jan 2011 15:15:18 -0500
Subject: [Freeswitch-users] Confirming if conference is being recorded
In-Reply-To: <AANLkTimd0bMAqx-HGa2fgihQgONePwSHfOupog-wXesP@mail.gmail.com>
References: <AANLkTi=o0wtYYdY8dbP=gsSkMgdBpetdSqM3Mkjr7g-r@mail.gmail.com>
	<AANLkTimd0bMAqx-HGa2fgihQgONePwSHfOupog-wXesP@mail.gmail.com>
Message-ID: <AANLkTinKY-D_TWm2av-VoL=S2YiAULwvQXur21G2eJ2A@mail.gmail.com>

Perfect!

Did not see the start-recording and stop-recording events (under Event
Socket Use) here http://wiki.freeswitch.org/wiki/Mod_conference

Perhaps I missed them too on the other wiki pages.

Thanks.

Richard

2011/1/14 Jo?o Mesquita <jmesquita at freeswitch.org>:
> Why not use inbound esl and listen in for the start-recording and
> stop-recording events?
> I've added a couple of more event fires on the code lately so you won't miss
> a single stop-recording even if it is at the end of the conference with
> auto-record enabled.
> Regards,
> Jo?o Mesquita
>
>
> On Fri, Jan 14, 2011 at 5:04 PM, Richard Alam <ritzalam at gmail.com> wrote:
>>
>> Hi,
>>
>> I am planning on recording a conference.
>>
>> Is there are way to regularly check if the conference is being
>> recorded aside from regularly checking the recorded file increasing in
>> size?
>>
>> Can I tell the conference app to send out events to ESL regularly
>> saying that it is recording?
>>
>> Thanks in advance.
>>
>> Richard
>>
>> --
>> ---
>> BigBlueButton
>> http://www.bigbluebutton.org
>> http://code.google.com/p/bigbluebutton
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
---
BigBlueButton
http://www.bigbluebutton.org
http://code.google.com/p/bigbluebutton


From infos at madovsky.org  Fri Jan 14 23:22:05 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 14 Jan 2011 15:22:05 -0500
Subject: [Freeswitch-users] empty var expression
Message-ID: <E2E64EFB13D6470BB46E3F74AE6142DE@e1705>

I tried this for an var marked as "UNDEF" in log

<condition field="${var}" expression="^UNDEF$" break="never">
<condition field="${var}" expression="UNDEF" break="never">
<condition field="${var}" expression="undef" break="never">
<condition field="${var}" expression="^undef$" break="never">
<condition field="${var}" expression="" break="never">
<condition field="${var}" expression="^$" break="never">
<condition field="${var}" expression="[UNDEF]" break="never">
<condition field="${var}" expression="[]" break="never">

without success.
which is the right expression for empty var please ?

Thanks



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From jmesquita at freeswitch.org  Fri Jan 14 23:22:14 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 14 Jan 2011 17:22:14 -0300
Subject: [Freeswitch-users] Confirming if conference is being recorded
In-Reply-To: <AANLkTinKY-D_TWm2av-VoL=S2YiAULwvQXur21G2eJ2A@mail.gmail.com>
References: <AANLkTi=o0wtYYdY8dbP=gsSkMgdBpetdSqM3Mkjr7g-r@mail.gmail.com>
	<AANLkTimd0bMAqx-HGa2fgihQgONePwSHfOupog-wXesP@mail.gmail.com>
	<AANLkTinKY-D_TWm2av-VoL=S2YiAULwvQXur21G2eJ2A@mail.gmail.com>
Message-ID: <AANLkTin2ekWEtKHqOmiL1+x2TCSaPuzeTOkqz1c+4tTX@mail.gmail.com>

The Event-Name header will always be  conference::maintenance but the Action
header is what you are looking for.

Regards,
Jo?o Mesquita


On Fri, Jan 14, 2011 at 5:15 PM, Richard Alam <ritzalam at gmail.com> wrote:

> Perfect!
>
> Did not see the start-recording and stop-recording events (under Event
> Socket Use) here http://wiki.freeswitch.org/wiki/Mod_conference
>
> Perhaps I missed them too on the other wiki pages.
>
> Thanks.
>
> Richard
>
> 2011/1/14 Jo?o Mesquita <jmesquita at freeswitch.org>:
> > Why not use inbound esl and listen in for the start-recording and
> > stop-recording events?
> > I've added a couple of more event fires on the code lately so you won't
> miss
> > a single stop-recording even if it is at the end of the conference with
> > auto-record enabled.
> > Regards,
> > Jo?o Mesquita
> >
> >
> > On Fri, Jan 14, 2011 at 5:04 PM, Richard Alam <ritzalam at gmail.com>
> wrote:
> >>
> >> Hi,
> >>
> >> I am planning on recording a conference.
> >>
> >> Is there are way to regularly check if the conference is being
> >> recorded aside from regularly checking the recorded file increasing in
> >> size?
> >>
> >> Can I tell the conference app to send out events to ESL regularly
> >> saying that it is recording?
> >>
> >> Thanks in advance.
> >>
> >> Richard
> >>
> >> --
> >> ---
> >> BigBlueButton
> >> http://www.bigbluebutton.org
> >> http://code.google.com/p/bigbluebutton
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> ---
> BigBlueButton
> http://www.bigbluebutton.org
> http://code.google.com/p/bigbluebutton
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jmesquita at freeswitch.org  Fri Jan 14 23:36:02 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 14 Jan 2011 17:36:02 -0300
Subject: [Freeswitch-users] empty var expression
In-Reply-To: <E2E64EFB13D6470BB46E3F74AE6142DE@e1705>
References: <E2E64EFB13D6470BB46E3F74AE6142DE@e1705>
Message-ID: <AANLkTi=Gg0ukr0vJapeVUwTGN3rkQ4w_mV43Fdhy3sqM@mail.gmail.com>

I thought it was __undef__, ain't it? I haven't digged into the source code,
but should be easy enough to give it a try...

Regards,
Jo?o Mesquita


On Fri, Jan 14, 2011 at 5:22 PM, Madovsky <infos at madovsky.org> wrote:

>  I tried this for an var marked as "UNDEF" in log
>
> <condition field="${var}" expression="^UNDEF$" break="never">
> <condition field="${var}" expression="UNDEF" break="never">
> <condition field="${var}" expression="undef" break="never">
> <condition field="${var}" expression="^undef$" break="never">
> <condition field="${var}" expression="" break="never">
> <condition field="${var}" expression="^$" break="never">
> <condition field="${var}" expression="[UNDEF]" break="never">
> <condition field="${var}" expression="[]" break="never">
>
> without success.
> which is the right expression for empty var please ?
>
> Thanks
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Sat Jan 15 00:01:03 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 14 Jan 2011 16:01:03 -0500
Subject: [Freeswitch-users] empty var expression
References: <E2E64EFB13D6470BB46E3F74AE6142DE@e1705>
	<AANLkTi=Gg0ukr0vJapeVUwTGN3rkQ4w_mV43Fdhy3sqM@mail.gmail.com>
Message-ID: <AF174A28158A4725A2CC231E3E64BFF7@e1705>

__undef__ doesnt' work either.

Thanks
  ----- Original Message ----- 
  From: Jo?o Mesquita 
  To: FreeSWITCH Users Help 
  Sent: Friday, January 14, 2011 3:36 PM
  Subject: Re: [Freeswitch-users] empty var expression


  I thought it was __undef__, ain't it? I haven't digged into the source code, but should be easy enough to give it a try...


  Regards,
  Jo?o Mesquita



  On Fri, Jan 14, 2011 at 5:22 PM, Madovsky <infos at madovsky.org> wrote:

    I tried this for an var marked as "UNDEF" in log

    <condition field="${var}" expression="^UNDEF$" break="never">
    <condition field="${var}" expression="UNDEF" break="never">
    <condition field="${var}" expression="undef" break="never">
    <condition field="${var}" expression="^undef$" break="never">
    <condition field="${var}" expression="" break="never">
    <condition field="${var}" expression="^$" break="never">
    <condition field="${var}" expression="[UNDEF]" break="never">
    <condition field="${var}" expression="[]" break="never">

    without success.
    which is the right expression for empty var please ?

    Thanks





    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org






------------------------------------------------------------------------------


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  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From cjbujold at accra.ca  Sat Jan 15 00:00:47 2011
From: cjbujold at accra.ca (Charles Bujold)
Date: Fri, 14 Jan 2011 17:00:47 -0400
Subject: [Freeswitch-users] Newbie Dial Plan question
Message-ID: <006301cbb42e$1e063650$5a12a2f0$@accra.ca>

Newbie question,  

 

I have a ht503(FXO) connected to freeswitch to make local calls (7digits)
The HT503-FXO is registered as extension 510.  If I dial 510 (Bria
softphone) via freeswitch  I get dial tone and can dial my local number.
However I can't seem to get it working via my dial plan.

 

I created a gateway to connect to the HT503 see below, and I created a local
7digit dial plan.   When I try to connect I get the error
"Network_out_of_Order".   Can somebody point out my error.  Thanks

 

Charles 

 

Gateway: /sip_profiles/external

 

<include>

    <gateway name="8580444">

      <param name="username" value="510"/>

      <param name="password" value="510"/>

      <param name="proxy" value="192.168.10.82"/>

      <param name="expire-seconds" value="800"/>

      <param name="register" value="false"/>

      <param name="retry-seconds" value="60"/>

      <param name="context" value="public"/>

    </gateway>

</include>

 

 

Dial Plan: /dialplan/default

 

<extension name="LocalCalls.7d" >

   <condition field="destination_number" expression="^\d{7}$">

       <action application="set"
data="effective_caller_id_name=${outbound_caller_id_name}"/>

       <action application="set"
data="effective_caller_id_number=${outbound_caller_id_number}"/>

       <action application="bridge" data="sofia/gateway/8580444/$1"/>

   </condition>

</extension>

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From chat2jesse at gmail.com  Sat Jan 15 01:30:52 2011
From: chat2jesse at gmail.com (jesse)
Date: Fri, 14 Jan 2011 14:30:52 -0800
Subject: [Freeswitch-users] FreeSWITCH 1.0.7
In-Reply-To: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org>
References: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org>
Message-ID: <AANLkTik+hPwuvDtDOQ-6pPg-wskP_BTCn2=gm9vtV688@mail.gmail.com>

congratulations!  well done!

-jesse

On Fri, Jan 14, 2011 at 8:25 AM, Brian West <brian at freeswitch.org> wrote:
> http://latest.freeswitch.org/
>
> Enjoy!
> /b
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From gcd at i.ph  Sat Jan 15 03:17:26 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Sat, 15 Jan 2011 08:17:26 +0800
Subject: [Freeswitch-users] Newbie Dial Plan question
In-Reply-To: <006301cbb42e$1e063650$5a12a2f0$@accra.ca>
References: <006301cbb42e$1e063650$5a12a2f0$@accra.ca>
Message-ID: <AANLkTimHdozkENz4Cfmr3Y4F=6tQDMQqDNtFE76i-9-o@mail.gmail.com>

hi charles,

in your bridge data, try this format instead:
sofia/gateway/8580444/$1 at fxo_ip_address:fxo_port

cheers!

On Sat, Jan 15, 2011 at 5:00 AM, Charles Bujold <cjbujold at accra.ca> wrote:

> Newbie question,
>
>
>
> I have a ht503(FXO) connected to freeswitch to make local calls (7digits)
> The HT503-FXO is registered as extension 510.  If I dial 510 (Bria
> softphone) via freeswitch  I get dial tone and can dial my local number.
> However I can?t seem to get it working via my dial plan.
>
>
>
> I created a gateway to connect to the HT503 see below, and I created a
> local 7digit dial plan.   When I try to connect I get the error
> ?Network_out_of_Order?.   Can somebody point out my error.  Thanks
>
>
>
> *Charles *
>
>
>
> Gateway: /sip_profiles/external
>
>
>
> <include>
>
>     <gateway name="8580444">
>
>       <param name="username" value="510"/>
>
>       <param name="password" value="510"/>
>
>       <param name="proxy" value="192.168.10.82"/>
>
>       <param name="expire-seconds" value="800"/>
>
>       <param name="register" value="false"/>
>
>       <param name="retry-seconds" value="60"/>
>
>       <param name="context" value="public"/>
>
>     </gateway>
>
> </include>
>
>
>
>
>
> Dial Plan: /dialplan/default
>
>
>
> <extension name="LocalCalls.7d" >
>
>    <condition field="destination_number" expression="^\d{7}$">
>
>        <action application="set"
> data="effective_caller_id_name=${outbound_caller_id_name}"/>
>
>        <action application="set"
> data="effective_caller_id_number=${outbound_caller_id_number}"/>
>
>        <action application="bridge" data="sofia/gateway/8580444/$1"/>
>
>    </condition>
>
> </extension>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Nabble at slickdeals.endjunk.com  Sat Jan 15 05:47:31 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 14 Jan 2011 18:47:31 -0800 (PST)
Subject: [Freeswitch-users] Newbie Dial Plan question
In-Reply-To: <AANLkTimHdozkENz4Cfmr3Y4F=6tQDMQqDNtFE76i-9-o@mail.gmail.com>
References: <006301cbb42e$1e063650$5a12a2f0$@accra.ca>
	<AANLkTimHdozkENz4Cfmr3Y4F=6tQDMQqDNtFE76i-9-o@mail.gmail.com>
Message-ID: <1295059651627-5924001.post@n2.nabble.com>



Nandy Dagondon wrote:
> in your bridge data, try this format instead:
> sofia/gateway/8580444/$1 at fxo_ip_address:fxo_port
With expression="^\d{7}$", the regex will match into array[0] and not
array[1]. As such, the $1 on the above line needs be replaced with $0. Or,
to keep the above line as is, one needs to change the regex expression to 
expression="^(\d{7})$" (notice the () block).



-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Newbie-Dial-Plan-question-tp5923355p5924001.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From infos at madovsky.org  Sat Jan 15 06:38:51 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 14 Jan 2011 22:38:51 -0500
Subject: [Freeswitch-users] SOFIA_REFER_TO_VARIABLE
Message-ID: <FE8BF1788FF04027A7D52B9685F4ACDE@e1705>

can anyone what the rul of this var ?

Thanks
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From dome at tel.co.th  Sat Jan 15 06:46:56 2011
From: dome at tel.co.th (dome at tel.co.th)
Date: Sat, 15 Jan 2011 10:46:56 +0700
Subject: [Freeswitch-users] FreeSWITCH 1.0.7
In-Reply-To: <AANLkTik+hPwuvDtDOQ-6pPg-wskP_BTCn2=gm9vtV688@mail.gmail.com>
References: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org>
	<AANLkTik+hPwuvDtDOQ-6pPg-wskP_BTCn2=gm9vtV688@mail.gmail.com>
Message-ID: <AANLkTikNCfjHWhQWB9C1THUNdUWhRdP0rcs1JNQOqdmO@mail.gmail.com>

I use (from git) for my production about 2 month ago :)
I can confirm very stable :)

freeswitch at internal> version
FreeSWITCH Version 1.0.head (git-6faa4c9 2010-12-02 17-11-04 -0600)

freeswitch at internal> status
UP 0 years, 42 days, 13 hours, 26 minutes, 11 seconds, 371
milliseconds, 850 microseconds
365465 session(s) since startup
13 session(s) 0/30
1000 session(s) max
min idle cpu 0.00/98.00



Dome C.


2011/1/15 jesse <chat2jesse at gmail.com>:
> congratulations! ?well done!
>
> -jesse
>
> On Fri, Jan 14, 2011 at 8:25 AM, Brian West <brian at freeswitch.org> wrote:
>> http://latest.freeswitch.org/
>>
>> Enjoy!
>> /b
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From boris at tagnet.ru  Sat Jan 15 09:02:40 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 15 Jan 2011 11:02:40 +0500
Subject: [Freeswitch-users] mod_lcr and extra_vars
In-Reply-To: <AANLkTimu-qW6BRJC3crQLy8uYavepzuxRVcarBP1h6Lg@mail.gmail.com>
References: <4D2B452D.8060004@tagnet.ru>
	<AANLkTimu-qW6BRJC3crQLy8uYavepzuxRVcarBP1h6Lg@mail.gmail.com>
Message-ID: <4D313880.3020903@tagnet.ru>

So, the wiki is incorrect?
http://wiki.freeswitch.org/wiki/Mod_lcr#Adding_extra_channel_variables
My lcr output look similar
> You aren't quite using it right.  You need to create a distinct sql
> field for each var you want imported.  Not a single field with a list
> of var=value like you show below.
>
> On Mon, Jan 10, 2011 at 11:43 AM, Boris Kovalenko<boris at tagnet.ru>  wrote:
>> Hello!
>>
>>      I need to set extra vars with mod_lcr. I did as wiki recomended:
>> 1) created sql column
>> 2) modified sql quer
>> 3) added<param name="export_fields" value="lcr_gw_extra_vars"/>  to profile
>> So, may lcr output looks nice:
>> | Digit Match | Carrier     | Rate     | Codec | CID Regexp          |
>> Limit |
>> Dialstring
>> |
>>   | 734353      | tagnet.ru   | 0.00000  |       |
>> |       |
>> [lcr_carrier=tagnet.ru,lcr_rate=0.00000,lcr_gw_extra_vars=,v_tagnet_ats_dstport=50000]sofia/epbx/73435350101 at X.X.X.X:5060
>>
>> Unfortunatelly I can't see (and can't use) v_tagnet_ats_dstport variable
>> in my cdr records (even with b-leg the variable isn't present) until I
>> set import=v_tagnet_ats_dstport variable. But I thinked mod_lcr should
>> do this?
>> Something wrong with my configuration?
>>
>>
>> --
>> ? ?????????,
>>    ????? ?????????
>>    ??? "??????"
>>    (3435) 494991
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From rafonline at hotmail.com  Sat Jan 15 17:44:48 2011
From: rafonline at hotmail.com (Rafqat .)
Date: Sat, 15 Jan 2011 14:44:48 +0000
Subject: [Freeswitch-users] DTMF events issue
Message-ID: <COL122-W49E961E8ADD843FE0C0E0BB5F20@phx.gbl>



Hi,

I am using ESL to listen to DTMF events.? In most part I can listen to these events and respond to them fine.

However, when leg A attempts to bridge to leg B, and presses some keys whilst waiting for leg b to answer the call, no DTMF events are received.? When leg b answers the call all the DTMF events that leg A generated whilst waiting for leg B to answer are received in one batch.

What am i doing wrong here? Is this normal behavour? surely not.

Cheers

Raf
 		 	   		  

From rupa at rupa.com  Sat Jan 15 18:11:44 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 15 Jan 2011 09:11:44 -0600
Subject: [Freeswitch-users] mod_lcr and extra_vars
In-Reply-To: <4D313880.3020903@tagnet.ru>
References: <4D2B452D.8060004@tagnet.ru>
	<AANLkTimu-qW6BRJC3crQLy8uYavepzuxRVcarBP1h6Lg@mail.gmail.com>
	<4D313880.3020903@tagnet.ru>
Message-ID: <AANLkTikrRQKSs6e3dgWQTyiKMAyLirm1PrsTCQ5n+2Kc@mail.gmail.com>

It isn't how I'd do it nor how the feature was designed.  Look at the
nibblebill section for the right way to do it.  I'll see if I can fix
the wiki docs....

On Sat, Jan 15, 2011 at 12:02 AM, Boris Kovalenko <boris at tagnet.ru> wrote:
> So, the wiki is incorrect?
> http://wiki.freeswitch.org/wiki/Mod_lcr#Adding_extra_channel_variables
> My lcr output look similar
>> You aren't quite using it right. ?You need to create a distinct sql
>> field for each var you want imported. ?Not a single field with a list
>> of var=value like you show below.
>>
>> On Mon, Jan 10, 2011 at 11:43 AM, Boris Kovalenko<boris at tagnet.ru> ?wrote:
>>> Hello!
>>>
>>> ? ? ?I need to set extra vars with mod_lcr. I did as wiki recomended:
>>> 1) created sql column
>>> 2) modified sql quer
>>> 3) added<param name="export_fields" value="lcr_gw_extra_vars"/> ?to profile
>>> So, may lcr output looks nice:
>>> | Digit Match | Carrier ? ? | Rate ? ? | Codec | CID Regexp ? ? ? ? ?|
>>> Limit |
>>> Dialstring
>>> |
>>> ? | 734353 ? ? ?| tagnet.ru ? | 0.00000 ?| ? ? ? |
>>> | ? ? ? |
>>> [lcr_carrier=tagnet.ru,lcr_rate=0.00000,lcr_gw_extra_vars=,v_tagnet_ats_dstport=50000]sofia/epbx/73435350101 at X.X.X.X:5060
>>>
>>> Unfortunatelly I can't see (and can't use) v_tagnet_ats_dstport variable
>>> in my cdr records (even with b-leg the variable isn't present) until I
>>> set import=v_tagnet_ats_dstport variable. But I thinked mod_lcr should
>>> do this?
>>> Something wrong with my configuration?
>>>
>>>
>>> --
>>> ? ?????????,
>>> ? ?????? ?????????
>>> ? ???? "??????"
>>> ? ?(3435) 494991
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
>
> --
> ? ?????????,
> ? ????? ?????????
> ? ??? "??????"
> ? (3435) 494991
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa


From boris at tagnet.ru  Sat Jan 15 18:12:02 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 15 Jan 2011 20:12:02 +0500
Subject: [Freeswitch-users] mod_lcr and extra_vars
In-Reply-To: <AANLkTimu-qW6BRJC3crQLy8uYavepzuxRVcarBP1h6Lg@mail.gmail.com>
References: <4D2B452D.8060004@tagnet.ru>
	<AANLkTimu-qW6BRJC3crQLy8uYavepzuxRVcarBP1h6Lg@mail.gmail.com>
Message-ID: <4D31B942.20709@tagnet.ru>

Hello!

     I've tested as You recommended. No success. Would You please 
provide an example?
> You aren't quite using it right.  You need to create a distinct sql
> field for each var you want imported.  Not a single field with a list
> of var=value like you show below.
>
> On Mon, Jan 10, 2011 at 11:43 AM, Boris Kovalenko<boris at tagnet.ru>  wrote:
>> Hello!
>>
>>      I need to set extra vars with mod_lcr. I did as wiki recomended:
>> 1) created sql column
>> 2) modified sql quer
>> 3) added<param name="export_fields" value="lcr_gw_extra_vars"/>  to profile
>> So, may lcr output looks nice:
>> | Digit Match | Carrier     | Rate     | Codec | CID Regexp          |
>> Limit |
>> Dialstring
>> |
>>   | 734353      | tagnet.ru   | 0.00000  |       |
>> |       |
>> [lcr_carrier=tagnet.ru,lcr_rate=0.00000,lcr_gw_extra_vars=,v_tagnet_ats_dstport=50000]sofia/epbx/73435350101 at X.X.X.X:5060
>>
>> Unfortunatelly I can't see (and can't use) v_tagnet_ats_dstport variable
>> in my cdr records (even with b-leg the variable isn't present) until I
>> set import=v_tagnet_ats_dstport variable. But I thinked mod_lcr should
>> do this?
>> Something wrong with my configuration?
>>
>>
>> --
>> ? ?????????,
>>    ????? ?????????
>>    ??? "??????"
>>    (3435) 494991
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From rupa at rupa.com  Sat Jan 15 18:42:15 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 15 Jan 2011 09:42:15 -0600
Subject: [Freeswitch-users] mod_lcr and extra_vars
In-Reply-To: <4D31B942.20709@tagnet.ru>
References: <4D2B452D.8060004@tagnet.ru>
	<AANLkTimu-qW6BRJC3crQLy8uYavepzuxRVcarBP1h6Lg@mail.gmail.com>
	<4D31B942.20709@tagnet.ru>
Message-ID: <AANLkTimdwanzVg7Ux90PRaYfjSWrNsaRmEbvQ0mW5SB-@mail.gmail.com>

The nibblebill example works.  I use it all the time.

a) are you on current git?

b) can you provide your profile and debug logs?

On Sat, Jan 15, 2011 at 9:12 AM, Boris Kovalenko <boris at tagnet.ru> wrote:
> Hello!
>
> ? ? I've tested as You recommended. No success. Would You please
> provide an example?
>> You aren't quite using it right. ?You need to create a distinct sql
>> field for each var you want imported. ?Not a single field with a list
>> of var=value like you show below.
>>
>> On Mon, Jan 10, 2011 at 11:43 AM, Boris Kovalenko<boris at tagnet.ru> ?wrote:
>>> Hello!
>>>
>>> ? ? ?I need to set extra vars with mod_lcr. I did as wiki recomended:
>>> 1) created sql column
>>> 2) modified sql quer
>>> 3) added<param name="export_fields" value="lcr_gw_extra_vars"/> ?to profile
>>> So, may lcr output looks nice:
>>> | Digit Match | Carrier ? ? | Rate ? ? | Codec | CID Regexp ? ? ? ? ?|
>>> Limit |
>>> Dialstring
>>> |
>>> ? | 734353 ? ? ?| tagnet.ru ? | 0.00000 ?| ? ? ? |
>>> | ? ? ? |
>>> [lcr_carrier=tagnet.ru,lcr_rate=0.00000,lcr_gw_extra_vars=,v_tagnet_ats_dstport=50000]sofia/epbx/73435350101 at X.X.X.X:5060
>>>
>>> Unfortunatelly I can't see (and can't use) v_tagnet_ats_dstport variable
>>> in my cdr records (even with b-leg the variable isn't present) until I
>>> set import=v_tagnet_ats_dstport variable. But I thinked mod_lcr should
>>> do this?
>>> Something wrong with my configuration?
>>>
>>>
>>> --
>>> ? ?????????,
>>> ? ?????? ?????????
>>> ? ???? "??????"
>>> ? ?(3435) 494991
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
>
> --
> ? ?????????,
> ? ????? ?????????
> ? ??? "??????"
> ? (3435) 494991
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa


From infos at madovsky.org  Sat Jan 15 19:06:17 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 15 Jan 2011 11:06:17 -0500
Subject: [Freeswitch-users] BLIND_TRANFER dialplan
Message-ID: <13C9ECCC43834B309130668C170F0D53@e1705>

is it possible to create a blind transfer manually without deflect ?

Thanks
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From infos at madovsky.org  Sat Jan 15 19:57:08 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 15 Jan 2011 11:57:08 -0500
Subject: [Freeswitch-users] watchdog
Message-ID: <0CFFA71FF3DC475DB0CC476A30C7BBE6@e1705>

I tried to add this on sip profile

    <!--<param name="watchdog-enabled" value="yes"/>-->
    <!--<param name="watchdog-step-timeout" value="30000"/>-->
    <!--<param name="watchdog-event-timeout" value="30000"/>-->

but after that freeswitch shutdown saying STACK error.

Thanks
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From infos at madovsky.org  Sat Jan 15 20:02:26 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 15 Jan 2011 12:02:26 -0500
Subject: [Freeswitch-users] deflect question
Message-ID: <AA67D36EA3FB4D0F82DF57DC8E23E647@e1705>

Not sure that what I said below is the answer of the problem.
I made other tests that show that
if the caller is not registered user and call from external sip profile,
deflect works flawessly.
if I try to deflect from a registered user (internal.xml) to an external or internal clustered user
it locks after execute deflect about 5 mn and hangup.

  ----- Original Message ----- 

  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, January 12, 2011 11:58 PM
  Subject: Re: [Freeswitch-users] deflect question


  no,
  but in fact I just release that my sip phone doesnt' support RFC 3515...
    ----- Original Message ----- 
    From: Sam 
    To: FreeSWITCH Users Help 
    Sent: Wednesday, January 12, 2011 11:05 PM
    Subject: Re: [Freeswitch-users] deflect question


    it originates a call with refer message,when the call lands on the different server does it releases the first leg ?
    and the call bridges the leg b without the leg a on the first server ?

    C --> A --> B

    A-> server 1
    B -> server 2
    C-> call initiator

    After deflect from A to B will the call flow be C-->B with out A ?

    Regards
    Sam


    On Thu, Jan 13, 2011 at 2:38 AM, Madovsky <infos at madovsky.org> wrote:


      if I use
 <action application="deflect" data="sip:someone at otherFS.com"/>
in the otherFS dialplan should I createan extension with "someone" condition only ? Thanks
      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org






----------------------------------------------------------------------------


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org
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From boris at tagnet.ru  Sat Jan 15 20:09:46 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 15 Jan 2011 22:09:46 +0500
Subject: [Freeswitch-users] mod_lcr and extra_vars
In-Reply-To: <AANLkTimdwanzVg7Ux90PRaYfjSWrNsaRmEbvQ0mW5SB-@mail.gmail.com>
References: <4D2B452D.8060004@tagnet.ru>	<AANLkTimu-qW6BRJC3crQLy8uYavepzuxRVcarBP1h6Lg@mail.gmail.com>	<4D31B942.20709@tagnet.ru>
	<AANLkTimdwanzVg7Ux90PRaYfjSWrNsaRmEbvQ0mW5SB-@mail.gmail.com>
Message-ID: <4D31D4DA.3060102@tagnet.ru>

Hello!

     I'll read nibblebill example more carefull and will do some tests 
and ask You if  i'll need help still.
Thank You!
> The nibblebill example works.  I use it all the time.
>
> a) are you on current git?
>
> b) can you provide your profile and debug logs?
>
> On Sat, Jan 15, 2011 at 9:12 AM, Boris Kovalenko<boris at tagnet.ru>  wrote:
>> Hello!
>>
>>      I've tested as You recommended. No success. Would You please
>> provide an example?
>>> You aren't quite using it right.  You need to create a distinct sql
>>> field for each var you want imported.  Not a single field with a list
>>> of var=value like you show below.
>>>
>>> On Mon, Jan 10, 2011 at 11:43 AM, Boris Kovalenko<boris at tagnet.ru>    wrote:
>>>> Hello!
>>>>
>>>>       I need to set extra vars with mod_lcr. I did as wiki recomended:
>>>> 1) created sql column
>>>> 2) modified sql quer
>>>> 3) added<param name="export_fields" value="lcr_gw_extra_vars"/>    to profile
>>>> So, may lcr output looks nice:
>>>> | Digit Match | Carrier     | Rate     | Codec | CID Regexp          |
>>>> Limit |
>>>> Dialstring
>>>> |
>>>>    | 734353      | tagnet.ru   | 0.00000  |       |
>>>> |       |
>>>> [lcr_carrier=tagnet.ru,lcr_rate=0.00000,lcr_gw_extra_vars=,v_tagnet_ats_dstport=50000]sofia/epbx/73435350101 at X.X.X.X:5060
>>>>
>>>> Unfortunatelly I can't see (and can't use) v_tagnet_ats_dstport variable
>>>> in my cdr records (even with b-leg the variable isn't present) until I
>>>> set import=v_tagnet_ats_dstport variable. But I thinked mod_lcr should
>>>> do this?
>>>> Something wrong with my configuration?
>>>>
>>>>
>>>> --
>>>> ? ?????????,
>>>>     ????? ?????????
>>>>     ??? "??????"
>>>>     (3435) 494991
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>
>> --
>> ? ?????????,
>>    ????? ?????????
>>    ??? "??????"
>>    (3435) 494991
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From infos at madovsky.org  Sat Jan 15 20:18:10 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 15 Jan 2011 12:18:10 -0500
Subject: [Freeswitch-users] deflect question
Message-ID: <190D58D5175F4AE9AACB0642C53C9012@e1705>

now I can see in log from a registered user deflect:

EXECUTE sofia/internal/9999999999999 at domain deflect(sip:conf_9999999999999 at 12.34.56.78)
2011-01-15 12:14:14.303271 [DEBUG] sofia.c:4604 Channel sofia/internal/9999999999999 at domain entering state [terminated][401]
2011-01-15 12:14:14.303271 [DEBUG] switch_channel.c:2455 (sofia/internal/9999999999999 at domain) Callstate Change RINGING -> HANGUP
2011-01-15 12:14:14.303271 [NOTICE] sofia.c:5244 Hangup sofia/internal/9999999999999 at domain [CS_EXECUTE] [CALL_REJECTED]
2011-01-15 12:14:14.303271 [DEBUG] switch_channel.c:2471 Send signal sofia/internal/9999999999999 at domain [KILL]
2011-01-15 12:14:14.303271 [DEBUG] switch_core_session.c:1083 Send signal sofia/internal/9999999999999 at domain [BREAK]
2011-01-15 12:14:14.353430 [INFO] sofia_presence.c:785 IN START_PRESENCE_SQL (internal)
2011-01-15 12:14:14.353430 [ERR] sofia_presence.c:796 EVENT DUMP:



  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, January 15, 2011 12:02 PM
  Subject: Re: [Freeswitch-users] deflect question


  Not sure that what I said below is the answer of the problem.
  I made other tests that show that
  if the caller is not registered user and call from external sip profile,
  deflect works flawessly.
  if I try to deflect from a registered user (internal.xml) to an external or internal clustered user
  it locks after execute deflect about 5 mn and hangup.

    ----- Original Message ----- 

    From: Madovsky 
    To: FreeSWITCH Users Help 
    Sent: Wednesday, January 12, 2011 11:58 PM
    Subject: Re: [Freeswitch-users] deflect question


    no,
    but in fact I just release that my sip phone doesnt' support RFC 3515...
      ----- Original Message ----- 
      From: Sam 
      To: FreeSWITCH Users Help 
      Sent: Wednesday, January 12, 2011 11:05 PM
      Subject: Re: [Freeswitch-users] deflect question


      it originates a call with refer message,when the call lands on the different server does it releases the first leg ?
      and the call bridges the leg b without the leg a on the first server ?

      C --> A --> B

      A-> server 1
      B -> server 2
      C-> call initiator

      After deflect from A to B will the call flow be C-->B with out A ?

      Regards
      Sam


      On Thu, Jan 13, 2011 at 2:38 AM, Madovsky <infos at madovsky.org> wrote:


        if I use
 <action application="deflect" data="sip:someone at otherFS.com"/>
in the otherFS dialplan should I createan extension with "someone" condition only ? Thanks
        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org






--------------------------------------------------------------------------


      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org
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From jeff at jefflenk.com  Sat Jan 15 20:53:10 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sat, 15 Jan 2011 09:53:10 -0800 (PST)
Subject: [Freeswitch-users] Iksemel msvs compiling
In-Reply-To: <SNT109-DS227CE87422187AA91545CFC1F30@phx.gbl>
References: <SNT109-DS13313A1FA906527438E8CC1090@phx.gbl>
	<SNT109-DS1155D401A0B9742CF63E13C10B0@phx.gbl>
	<1294375438447-5898181.post@n2.nabble.com>
	<SNT109-DS11CB0ECE5BD0F9C0384E03C10C0@phx.gbl>
	<1294535404901-5903534.post@n2.nabble.com>
	<SNT109-DS14B90C48B33126784F294AC10E0@phx.gbl>
	<1294673236806-5907287.post@n2.nabble.com>
	<SNT109-DS227CE87422187AA91545CFC1F30@phx.gbl>
Message-ID: <1295113990176-5925152.post@n2.nabble.com>


Hi Norman,

I thought you were trying to modify IKSemel to support gnutls - if so how
can you not have modifed the project?

Compare your project includes to the following line.

..\..\iksemel\include;.;..\..\pthreads-w32-2-7-0-release;

the line looks like this when pulled from git :
..\..\iksemel\include;.

Jeff
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Iksemel-msvs-compiling-tp5891263p5925152.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From boris at tagnet.ru  Sat Jan 15 21:01:04 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 15 Jan 2011 23:01:04 +0500
Subject: [Freeswitch-users] mod_lcr and extra_vars
In-Reply-To: <AANLkTimdwanzVg7Ux90PRaYfjSWrNsaRmEbvQ0mW5SB-@mail.gmail.com>
References: <4D2B452D.8060004@tagnet.ru>	<AANLkTimu-qW6BRJC3crQLy8uYavepzuxRVcarBP1h6Lg@mail.gmail.com>	<4D31B942.20709@tagnet.ru>
	<AANLkTimdwanzVg7Ux90PRaYfjSWrNsaRmEbvQ0mW5SB-@mail.gmail.com>
Message-ID: <4D31E0E0.6010909@tagnet.ru>

Hello!

     Thank You, Rupa, I got it working the nibblebill way. But, imho, 
the way with one extra_vars field is more simple.
> The nibblebill example works.  I use it all the time.
>
> a) are you on current git?
>
> b) can you provide your profile and debug logs?
>
> On Sat, Jan 15, 2011 at 9:12 AM, Boris Kovalenko<boris at tagnet.ru>  wrote:
>> Hello!
>>
>>      I've tested as You recommended. No success. Would You please
>> provide an example?
>>> You aren't quite using it right.  You need to create a distinct sql
>>> field for each var you want imported.  Not a single field with a list
>>> of var=value like you show below.
>>>
>>> On Mon, Jan 10, 2011 at 11:43 AM, Boris Kovalenko<boris at tagnet.ru>    wrote:
>>>> Hello!
>>>>
>>>>       I need to set extra vars with mod_lcr. I did as wiki recomended:
>>>> 1) created sql column
>>>> 2) modified sql quer
>>>> 3) added<param name="export_fields" value="lcr_gw_extra_vars"/>    to profile
>>>> So, may lcr output looks nice:
>>>> | Digit Match | Carrier     | Rate     | Codec | CID Regexp          |
>>>> Limit |
>>>> Dialstring
>>>> |
>>>>    | 734353      | tagnet.ru   | 0.00000  |       |
>>>> |       |
>>>> [lcr_carrier=tagnet.ru,lcr_rate=0.00000,lcr_gw_extra_vars=,v_tagnet_ats_dstport=50000]sofia/epbx/73435350101 at X.X.X.X:5060
>>>>
>>>> Unfortunatelly I can't see (and can't use) v_tagnet_ats_dstport variable
>>>> in my cdr records (even with b-leg the variable isn't present) until I
>>>> set import=v_tagnet_ats_dstport variable. But I thinked mod_lcr should
>>>> do this?
>>>> Something wrong with my configuration?
>>>>
>>>>
>>>> --
>>>> ? ?????????,
>>>>     ????? ?????????
>>>>     ??? "??????"
>>>>     (3435) 494991
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>
>> --
>> ? ?????????,
>>    ????? ?????????
>>    ??? "??????"
>>    (3435) 494991
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From rupa at rupa.com  Sat Jan 15 21:34:14 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 15 Jan 2011 12:34:14 -0600
Subject: [Freeswitch-users] mod_lcr and extra_vars
In-Reply-To: <4D31E0E0.6010909@tagnet.ru>
References: <4D2B452D.8060004@tagnet.ru>
	<AANLkTimu-qW6BRJC3crQLy8uYavepzuxRVcarBP1h6Lg@mail.gmail.com>
	<4D31B942.20709@tagnet.ru>
	<AANLkTimdwanzVg7Ux90PRaYfjSWrNsaRmEbvQ0mW5SB-@mail.gmail.com>
	<4D31E0E0.6010909@tagnet.ru>
Message-ID: <AANLkTikqj3gCrZ5P7PyN5LZ9XjskiOR8cmUucJv5rwZ+@mail.gmail.com>

It may be more simple but it is less flexible.  You can probably still
do it the other way but it isn't how I designed the feature.

On Sat, Jan 15, 2011 at 12:01 PM, Boris Kovalenko <boris at tagnet.ru> wrote:
> Hello!
>
> ? ? Thank You, Rupa, I got it working the nibblebill way. But, imho,
> the way with one extra_vars field is more simple.
>> The nibblebill example works. ?I use it all the time.
>>
>> a) are you on current git?
>>
>> b) can you provide your profile and debug logs?
>>
>> On Sat, Jan 15, 2011 at 9:12 AM, Boris Kovalenko<boris at tagnet.ru> ?wrote:
>>> Hello!
>>>
>>> ? ? ?I've tested as You recommended. No success. Would You please
>>> provide an example?
>>>> You aren't quite using it right. ?You need to create a distinct sql
>>>> field for each var you want imported. ?Not a single field with a list
>>>> of var=value like you show below.
>>>>
>>>> On Mon, Jan 10, 2011 at 11:43 AM, Boris Kovalenko<boris at tagnet.ru> ? ?wrote:
>>>>> Hello!
>>>>>
>>>>> ? ? ? I need to set extra vars with mod_lcr. I did as wiki recomended:
>>>>> 1) created sql column
>>>>> 2) modified sql quer
>>>>> 3) added<param name="export_fields" value="lcr_gw_extra_vars"/> ? ?to profile
>>>>> So, may lcr output looks nice:
>>>>> | Digit Match | Carrier ? ? | Rate ? ? | Codec | CID Regexp ? ? ? ? ?|
>>>>> Limit |
>>>>> Dialstring
>>>>> |
>>>>> ? ?| 734353 ? ? ?| tagnet.ru ? | 0.00000 ?| ? ? ? |
>>>>> | ? ? ? |
>>>>> [lcr_carrier=tagnet.ru,lcr_rate=0.00000,lcr_gw_extra_vars=,v_tagnet_ats_dstport=50000]sofia/epbx/73435350101 at X.X.X.X:5060
>>>>>
>>>>> Unfortunatelly I can't see (and can't use) v_tagnet_ats_dstport variable
>>>>> in my cdr records (even with b-leg the variable isn't present) until I
>>>>> set import=v_tagnet_ats_dstport variable. But I thinked mod_lcr should
>>>>> do this?
>>>>> Something wrong with my configuration?
>>>>>
>>>>>
>>>>> --
>>>>> ? ?????????,
>>>>> ? ? ????? ?????????
>>>>> ? ? ??? "??????"
>>>>> ? ? (3435) 494991
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>
>>> --
>>> ? ?????????,
>>> ? ?????? ?????????
>>> ? ???? "??????"
>>> ? ?(3435) 494991
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
>
> --
> ? ?????????,
> ? ????? ?????????
> ? ??? "??????"
> ? (3435) 494991
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa


From curriegrad2004 at gmail.com  Sat Jan 15 23:51:00 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 15 Jan 2011 12:51:00 -0800
Subject: [Freeswitch-users] FS as sip proxy registration
In-Reply-To: <9341C51152A541A28E924C6D7BFE4289@e1705>
References: <23834DF3-F6D6-48C0-AE24-77981C92D204@visionutveckling.se>
	<9341C51152A541A28E924C6D7BFE4289@e1705>
Message-ID: <AANLkTin5FMq2nn6Boc9ss=BE1Y9pBVwrqyayUQixi7AC@mail.gmail.com>

As far as I can tell, no FS can't work as that either. It's a softswitch

On Fri, Jan 14, 2011 at 10:51 AM, Madovsky <infos at madovsky.org> wrote:
> I know, but it can work as a registrar kind, I use it like this and it works
>
> ----- Original Message -----
> From: "Peter Olsson" <peter.olsson at visionutveckling.se>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Friday, January 14, 2011 1:29 PM
> Subject: Re: [Freeswitch-users] FS as sip proxy registration
>
>
> FS is not a proxy...
>
> /Peter
>
> ----- Reply message -----
> Fr?n: "Madovsky" <infos at madovsky.org>
> Datum: fre, jan 14, 2011 18:24
> Rubrik: [Freeswitch-users] FS as sip proxy registration
> Till: "freeswitch-users at lists.freeswitch.org"
> <freeswitch-users at lists.freeswitch.org>
>
> if I use FS as SIP proxy registrar,
> does the REFER method will go through the client transparently ?
> I'm facing problem of this method with this configuration
>
> Thanks
> !DSPAM:4d30866d32761881510302!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From Nabble at slickdeals.endjunk.com  Sun Jan 16 01:16:26 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sat, 15 Jan 2011 14:16:26 -0800 (PST)
Subject: [Freeswitch-users] Iksemel msvs compiling
In-Reply-To: <SNT109-DS14B90C48B33126784F294AC10E0@phx.gbl>
References: <SNT109-DS13313A1FA906527438E8CC1090@phx.gbl>
	<SNT109-DS1155D401A0B9742CF63E13C10B0@phx.gbl>
	<1294375438447-5898181.post@n2.nabble.com>
	<SNT109-DS11CB0ECE5BD0F9C0384E03C10C0@phx.gbl>
	<1294535404901-5903534.post@n2.nabble.com>
	<SNT109-DS14B90C48B33126784F294AC10E0@phx.gbl>
Message-ID: <1295129786597-5925550.post@n2.nabble.com>



Norman Lam wrote:
> 
> 28>..\..\iksemel\src\dom.c(152) : error C2065: 'ENOENT' : undeclared
Apparently, your compiler doesn't know what 'ENOENT' is.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Iksemel-msvs-compiling-tp5891263p5925550.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From infos at madovsky.org  Sun Jan 16 02:36:01 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 15 Jan 2011 18:36:01 -0500
Subject: [Freeswitch-users] mod_conference member-flags
Message-ID: <69D8E2E35ADF4A9EB28E9EE4BF4970F7@e1705>

How a "moderator" can be
also "endconf" in same time ?

Thanks
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From avi at avimarcus.net  Sun Jan 16 03:27:42 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 16 Jan 2011 02:27:42 +0200
Subject: [Freeswitch-users] IP Failover with Sofia Recover - not recovering?!
In-Reply-To: <mailman.1927.1295137414.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.1927.1295137414.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <AANLkTinA9f19XPq6d+TzeXt3iE7PW500yBxPGAS+R3GH@mail.gmail.com>

I finally figured out how to get pacemaker to start profiles/trigger a sofia
recover on moving the floating IP (but an FS crash it still can't handle)
Anyway - I
1) set up odbc for the core, and mysql master-master replication, so I see
calls & channels being logged in both
2) I set track calls in all the profiles
So when I set it to transfer from one node to another, I'm kinda surprised
I'm not getting a recovery of the calls.
I do see that the DB has a hostname stored - "sip1" or "sip2" depending on
where the call originated, but even if I change it manually, a sofia recover
seems to ways say "no calls to recover"
Am I missing something?!

-Avi Marcus
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From infos at madovsky.org  Sun Jan 16 06:54:51 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 15 Jan 2011 22:54:51 -0500
Subject: [Freeswitch-users] bridge a call to remote conference
Message-ID: <66485ECAF27B488AB461375999D3EA9B@e1705>

everyhting is ok unless that conference ivr can't be heard (like fraction of sec of sound every 2/3 sec)
but the hold music (in 8000hz) works well.
should I set any special codec in the legB conference bridge ?

Thanks
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From vetali100 at gmail.com  Sun Jan 16 12:13:42 2011
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 16 Jan 2011 11:13:42 +0200
Subject: [Freeswitch-users] [freeswitch-users] sofia_presence.c:404 Cannot
	find profile [my.domain.com]
Message-ID: <AANLkTi=edzD3Q=omSnVwqji-kzY0_Zo7mYdmicgc=3Dq@mail.gmail.com>

Hi,

After installing 1.0.7 I observed in the log the following message in error
verbosity:

     sofia_presence.c:404 Cannot find profile [my.domain.com]

It looks like everything works good, however I never saw this error in
1.0.6.

Could you please hint what this error means and what should be configured in
order to fix it?

Thank you,
Vitalie
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From steveayre at gmail.com  Sun Jan 16 12:18:53 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 16 Jan 2011 09:18:53 +0000
Subject: [Freeswitch-users] IP Failover with Sofia Recover - not
	recovering?!
In-Reply-To: <AANLkTinA9f19XPq6d+TzeXt3iE7PW500yBxPGAS+R3GH@mail.gmail.com>
References: <mailman.1927.1295137414.24555.freeswitch-users@lists.freeswitch.org>
	<AANLkTinA9f19XPq6d+TzeXt3iE7PW500yBxPGAS+R3GH@mail.gmail.com>
Message-ID: <AANLkTi=FOkWq9pRFqUhGN-gUVjvajGH1iSxU9u1nu+X-@mail.gmail.com>

I believe that hostname is the hostname of the server and set on FS startup.

-Steve


On 16 January 2011 00:27, Avi Marcus <avi at avimarcus.net> wrote:

> I finally figured out how to get pacemaker to start profiles/trigger a
> sofia recover on moving the floating IP (but an FS crash it still can't
> handle)
> Anyway - I
> 1) set up odbc for the core, and mysql master-master replication, so I see
> calls & channels being logged in both
> 2) I set track calls in all the profiles
> So when I set it to transfer from one node to another, I'm kinda surprised
> I'm not getting a recovery of the calls.
> I do see that the DB has a hostname stored - "sip1" or "sip2" depending on
> where the call originated, but even if I change it manually, a sofia recover
> seems to ways say "no calls to recover"
> Am I missing something?!
>
> -Avi Marcus
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vetali100 at gmail.com  Sun Jan 16 12:20:14 2011
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 16 Jan 2011 11:20:14 +0200
Subject: [Freeswitch-users] [freeswitch-users] SIP auth challenge (REGISTER)
 on sofia profile 'internal' for [1000@my.domain.com] from [some ip]
Message-ID: <AANLkTik7LbV5ZJxB14yp7hsR6-UwK9nYxECv6grDbwmL@mail.gmail.com>

Hi,

After upgrading to 1.0.7 I am getting WARNING message for every register
attempt:

     [WARNING] sofia_reg.c:1241 SIP auth challenge (REGISTER) on sofia
profile 'internal' for [1000 at my.domain.com] from [some ip]

Is it OK?
If yes, than why it is WARNING and not INFO for example?

Thank you,
Vitalie
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From steveayre at gmail.com  Sun Jan 16 12:20:52 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 16 Jan 2011 09:20:52 +0000
Subject: [Freeswitch-users] bridge a call to remote conference
In-Reply-To: <66485ECAF27B488AB461375999D3EA9B@e1705>
References: <66485ECAF27B488AB461375999D3EA9B@e1705>
Message-ID: <AANLkTi=UaTQ70AS2t+OYFrZrHMRW8=Mm5qKP8OyW+OK+@mail.gmail.com>

Try answering the call before you enter the conference, possibly with a
small sleep inbetween too. That might give the media time to startup before
the conference IVR starts.

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_answer
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sleep

-Steve


On 16 January 2011 03:54, Madovsky <infos at madovsky.org> wrote:

>  everyhting is ok unless that conference ivr can't be heard (like fraction
> of sec of sound every 2/3 sec)
> but the hold music (in 8000hz) works well.
> should I set any special codec in the legB conference bridge ?
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Sun Jan 16 12:26:53 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 16 Jan 2011 09:26:53 +0000
Subject: [Freeswitch-users] Processing Live Audio during a call between
	2 extensions
In-Reply-To: <AANLkTikTMazyD_Q9FOZj=Ky+sC+JAK4aTKE+cV6NgEWf@mail.gmail.com>
References: <AANLkTikTMazyD_Q9FOZj=Ky+sC+JAK4aTKE+cV6NgEWf@mail.gmail.com>
Message-ID: <AANLkTimoNgqmDaGzWuC3oLjSpYgjoLrkRrPAcgzBbHp5@mail.gmail.com>

It was hinted on IRC that media bugs may be the way to go. It all depends on
what exactly you're trying to do.

Read FreeSWITCH -> Modules -> Core Library -> Media Bugs on
http://docs.freeswitch.org/ and see if they sound the way to go.

In particular this may be what you need:

 SWITCH_DECLARE<http://docs.freeswitch.org/group__mb1.html#gf0ebe1244a6a3937e9fa039eb91b3325>(
switch_frame_t <http://docs.freeswitch.org/structswitch__frame.html> *)
switch_core_media_bug_get_write_replace_frame(_In_
switch_media_bug_t<http://docs.freeswitch.org/structswitch__media__bug.html>*
bug<http://docs.freeswitch.org/group__mb1.html#g2a620e89890bd1e0e9ccd08954d361b1>
)  Obtain a replace frame from a media bug.
You'll write a module in C that when loaded registeres a callback with FS.
Any channel using the media bug will get the audio decoded, given to the
media bug. Your app will process the audio and replace it, then FS will
encode and send the replacement. Decoded audio will be a data buffer
containing L16 (WAV) samples.

-Steve


On 14 January 2011 16:44, Michel Habib <michelhabib at gmail.com> wrote:

> Dear Freeswitch Developers, i am relatively new to freeswitch and i am
> seeking some advice.
>
> I am creating an External Application that does some [live] processing on
> the Audio of the freeswitch call before sending it to the second Call Leg,
> and vice versa.
> Can you please direct me to the best way to do that and how to
> access/capture the live audio channel in both legs while it is streaming,
> and how to resubmit it again to the other leg?
> I read the book about freeswitch, but still couldnt figure the best
> approach to do that, any hint is appreciated.
>
> Best Regards,
> Michel Habib.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Sun Jan 16 12:28:01 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 16 Jan 2011 09:28:01 +0000
Subject: [Freeswitch-users] [freeswitch-users] SIP auth challenge
 (REGISTER) on sofia profile 'internal' for [1000@my.domain.com] from [some
 ip]
In-Reply-To: <AANLkTik7LbV5ZJxB14yp7hsR6-UwK9nYxECv6grDbwmL@mail.gmail.com>
References: <AANLkTik7LbV5ZJxB14yp7hsR6-UwK9nYxECv6grDbwmL@mail.gmail.com>
Message-ID: <AANLkTinX91gdzrPPp0wxhcUtWLU7YSQStuXVy3RtoM1h@mail.gmail.com>

It's logging that a register request is occuring. It's neither succeeded nor
failed at this point.

It's there to provide enough logging for fail2ban to work with.

-Steve


On 16 January 2011 09:20, Vitalii Colosov <vetali100 at gmail.com> wrote:

> Hi,
>
> After upgrading to 1.0.7 I am getting WARNING message for every register
> attempt:
>
>      [WARNING] sofia_reg.c:1241 SIP auth challenge (REGISTER) on sofia
> profile 'internal' for [1000 at my.domain.com] from [some ip]
>
> Is it OK?
> If yes, than why it is WARNING and not INFO for example?
>
> Thank you,
> Vitalie
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From saeedahmad1981 at gmail.com  Sun Jan 16 15:24:50 2011
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Sun, 16 Jan 2011 13:24:50 +0100
Subject: [Freeswitch-users] Another: xml_curl vs Lua
Message-ID: <AANLkTik8RRmhN+-LYAG6o-ssuf4RYEYrine8Jj9s21PK@mail.gmail.com>

Dear all,

i know that lua is preferable by FS devs and community, but here i want to
ask questions particular to my use case

*1. Current Setup*
*
*1.1 currently i am using xml_curl - for dialplan only
1.2 xml_curl.conf has two bindings so i am safe if first one dies
1.3 i am using apache + mono (.net) + mysql on another server (other than FS
server)
1.4 Intel(R) Xeon(R) CPU  E5420  @ 2.50GHz 8 core with 8 gm ram (for both FS
and apache+mysql server)

*2. Call Life Cycle*

2.1. call comes on internal profile
2.2. xml_curl ask xml from webserver
2.3 i do database query for each call and, return back possible supplier(s)
based on dialed number and customer id
2.4 i return xml to bridge the call on external profile

Questiosn & Concerns:

1. So with above setup i am not able to reach more than 60 cps (most
probably issue with my xml_curl backend), so i am thinking to try LUA
2. but one point is clicking my mind that, in case of xml_curl the webserver
can be on external server, and xml_curl conf could have primary and backup
binding, can i achieve that in LUA too?
3. Is LUA still be usefull as i am not doing any ivr etc.. its just xml conf
which i have to return.
4. xml_curl based setup has good feature that i can serve  more FS servers,
can i also do it in LUA?

Please ask me if i miss some information

Thanks
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From saeedahmad1981 at gmail.com  Sun Jan 16 16:24:10 2011
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Sun, 16 Jan 2011 14:24:10 +0100
Subject: [Freeswitch-users] Another: xml_curl vs Lua
In-Reply-To: <AANLkTik8RRmhN+-LYAG6o-ssuf4RYEYrine8Jj9s21PK@mail.gmail.com>
References: <AANLkTik8RRmhN+-LYAG6o-ssuf4RYEYrine8Jj9s21PK@mail.gmail.com>
Message-ID: <AANLkTinq=Sh2wQO+0YTF9V5mM51mYFVArd+LE5Dgi2Q-@mail.gmail.com>

i want to handle 2000 concurrent calls and minimum 100 cps.

possible with lua? or should i stay with xml_curl?

On Sun, Jan 16, 2011 at 1:24 PM, Saeed Ahmed <saeedahmad1981 at gmail.com>wrote:

> Dear all,
>
> i know that lua is preferable by FS devs and community, but here i want to
> ask questions particular to my use case
>
> *1. Current Setup*
> *
> *1.1 currently i am using xml_curl - for dialplan only
> 1.2 xml_curl.conf has two bindings so i am safe if first one dies
> 1.3 i am using apache + mono (.net) + mysql on another server (other than
> FS server)
> 1.4 Intel(R) Xeon(R) CPU  E5420  @ 2.50GHz 8 core with 8 gm ram (for both
> FS and apache+mysql server)
>
> *2. Call Life Cycle*
>
> 2.1. call comes on internal profile
> 2.2. xml_curl ask xml from webserver
> 2.3 i do database query for each call and, return back possible supplier(s)
> based on dialed number and customer id
> 2.4 i return xml to bridge the call on external profile
>
> Questiosn & Concerns:
>
> 1. So with above setup i am not able to reach more than 60 cps (most
> probably issue with my xml_curl backend), so i am thinking to try LUA
> 2. but one point is clicking my mind that, in case of xml_curl the
> webserver can be on external server, and xml_curl conf could have primary
> and backup binding, can i achieve that in LUA too?
> 3. Is LUA still be usefull as i am not doing any ivr etc.. its just xml
> conf which i have to return.
> 4. xml_curl based setup has good feature that i can serve  more FS servers,
> can i also do it in LUA?
>
> Please ask me if i miss some information
>
> Thanks
>
>
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From dome at tel.co.th  Sun Jan 16 17:16:43 2011
From: dome at tel.co.th (dome at tel.co.th)
Date: Sun, 16 Jan 2011 21:16:43 +0700
Subject: [Freeswitch-users] Another: xml_curl vs Lua
In-Reply-To: <AANLkTinq=Sh2wQO+0YTF9V5mM51mYFVArd+LE5Dgi2Q-@mail.gmail.com>
References: <AANLkTik8RRmhN+-LYAG6o-ssuf4RYEYrine8Jj9s21PK@mail.gmail.com>
	<AANLkTinq=Sh2wQO+0YTF9V5mM51mYFVArd+LE5Dgi2Q-@mail.gmail.com>
Message-ID: <AANLkTikGqn_oXe56BZuqZK==aF=mSJQS7sfgrQJc2Gv_@mail.gmail.com>

xml_curl -> app server
my app server use nginx + luajit + tokyotylant


Dome C.


2011/1/16 Saeed Ahmed <saeedahmad1981 at gmail.com>:
> i want to handle 2000 concurrent calls and minimum 100 cps.
> possible with lua? or should i stay with xml_curl?
>
> On Sun, Jan 16, 2011 at 1:24 PM, Saeed Ahmed <saeedahmad1981 at gmail.com>
> wrote:
>>
>> Dear all,
>>
>> i know that lua is preferable by FS devs and community, but here i want to
>> ask questions particular to my use case
>> 1. Current Setup
>> 1.1 currently i am using xml_curl - for dialplan only
>> 1.2 xml_curl.conf has two bindings so i am safe if first one dies
>> 1.3 i am using apache + mono (.net) + mysql on another server (other than
>> FS server)
>> 1.4?Intel(R) Xeon(R) CPU ?E5420 ?@ 2.50GHz 8 core with 8 gm ram (for both
>> FS and apache+mysql server)
>> 2. Call Life Cycle
>> 2.1. call comes on internal profile
>> 2.2. xml_curl ask xml from webserver
>> 2.3 i do database query for each call and, return back possible
>> supplier(s) based on dialed number and customer id
>> 2.4 i return xml to bridge the call on external profile
>> Questiosn & Concerns:
>>
>> 1. So with above setup i am not able to reach more than 60 cps (most
>> probably issue with my xml_curl backend), so i am thinking to try LUA
>> 2. but one point is clicking my mind that, in case of xml_curl the
>> webserver can be on external server, and xml_curl conf could have primary
>> and backup binding, can i achieve that in LUA too?
>> 3. Is LUA still be usefull as i am not doing any ivr etc.. its just xml
>> conf which i have to return.
>> 4. xml_curl based setup has good feature that i can serve ?more FS
>> servers, can i also do it in LUA?
>>
>> Please ask me if i miss some information
>>
>> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From jalsot at gmail.com  Sun Jan 16 18:33:57 2011
From: jalsot at gmail.com (Tamas Jalsovszky)
Date: Sun, 16 Jan 2011 16:33:57 +0100
Subject: [Freeswitch-users] Removing inband DTMF
In-Reply-To: <AANLkTimYd8PZumEY1GzkcAj-kZ1auH8Lvr-7OBm7r3fB@mail.gmail.com>
References: <AANLkTikBZPbeKLO_fX=s4p1pZxPCzwcAet1win8eaUMn@mail.gmail.com>
	<AANLkTimYd8PZumEY1GzkcAj-kZ1auH8Lvr-7OBm7r3fB@mail.gmail.com>
Message-ID: <AANLkTimgMxuC=FEQ0UFerTLE1wRCKUx5L1Kr-_KMWJ3b@mail.gmail.com>

Hello,

Thanks.
Yeah, I know that command, as far as I know, it is only for detection, it
does not change the media.
Anybody else?

Regards,
   T.

On Fri, Jan 14, 2011 at 11:23 AM, Avi Marcus <Avi at amarcus.com> wrote:

> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf
> start_dtmf does detection from the inband. I'm not sure that it removes
> them from the rtp, though...
> -Avi
>
>
> On Fri, Jan 14, 2011 at 12:14 PM, Tamas Jalsovszky <jalsot at gmail.com>
> wrote:
> > Hello,
> >
> > Is there a way to remove an inband DTMF signal from the RTP stream with
> > FreeSWITCH?
> > We have a partner with ugly Cirpack which can not remove inband DTMF when
> > the codec is g711. The problem is, that our other party wants RFC2833
> only -
> > thus we have to remove inband dtmf. Unfortunately Cirpack is not open to
> do
> > this.
> >
> > Any advice?
> >
> > Kind regards,
> >    T.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Sun Jan 16 19:13:52 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 16 Jan 2011 11:13:52 -0500
Subject: [Freeswitch-users] [freeswitch-users] sofia_presence.c:404
	Cannotfind profile [my.domain.com]
References: <AANLkTi=edzD3Q=omSnVwqji-kzY0_Zo7mYdmicgc=3Dq@mail.gmail.com>
Message-ID: <2560F9AC02E244CEA258689845671F27@e1705>

maybe some file settings changed so you have now
default config
  ----- Original Message ----- 
  From: Vitalii Colosov 
  To: FreeSWITCH Users Help 
  Sent: Sunday, January 16, 2011 4:13 AM
  Subject: [Freeswitch-users] [freeswitch-users] sofia_presence.c:404 Cannotfind profile [my.domain.com]


  Hi,


  After installing 1.0.7 I observed in the log the following message in error verbosity:


       sofia_presence.c:404 Cannot find profile [my.domain.com]


  It looks like everything works good, however I never saw this error in 1.0.6.


  Could you please hint what this error means and what should be configured in order to fix it?


  Thank you,
  Vitalie


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From steveayre at gmail.com  Sun Jan 16 19:50:38 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 16 Jan 2011 16:50:38 +0000
Subject: [Freeswitch-users] Removing inband DTMF
In-Reply-To: <AANLkTimgMxuC=FEQ0UFerTLE1wRCKUx5L1Kr-_KMWJ3b@mail.gmail.com>
References: <AANLkTikBZPbeKLO_fX=s4p1pZxPCzwcAet1win8eaUMn@mail.gmail.com>
	<AANLkTimYd8PZumEY1GzkcAj-kZ1auH8Lvr-7OBm7r3fB@mail.gmail.com>
	<AANLkTimgMxuC=FEQ0UFerTLE1wRCKUx5L1Kr-_KMWJ3b@mail.gmail.com>
Message-ID: <AANLkTi=BQGJJrVgyPKxX2aP3kq_xWRHDzzsf1-qQd9X3@mail.gmail.com>

First of, AFAIK there's nothing in FS that can currently do it.

I'm not even sure it's possible at all. You can subtract a noise from audio,
but the noisy nature of an encoded signal would mean it wouldn't work well
(you'd still hear something), and DTMF can't be detected until some of it
will have already been sent to the other endpoint so you'd still have the
start of each keypress heard.  For the same reason at the end until the
detector spots the dtmf has ended you'll still be subtracting from the audio
which will actually generate noise. The only way to avoid that may be to
introduce a delay but that's not something FS can do.

-Steve



On 16 January 2011 15:33, Tamas Jalsovszky <jalsot at gmail.com> wrote:

> Hello,
>
> Thanks.
> Yeah, I know that command, as far as I know, it is only for detection, it
> does not change the media.
> Anybody else?
>
> Regards,
>    T.
>
>
> On Fri, Jan 14, 2011 at 11:23 AM, Avi Marcus <Avi at amarcus.com> wrote:
>
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf
>> start_dtmf does detection from the inband. I'm not sure that it removes
>> them from the rtp, though...
>> -Avi
>>
>>
>> On Fri, Jan 14, 2011 at 12:14 PM, Tamas Jalsovszky <jalsot at gmail.com>
>> wrote:
>> > Hello,
>> >
>> > Is there a way to remove an inband DTMF signal from the RTP stream with
>> > FreeSWITCH?
>> > We have a partner with ugly Cirpack which can not remove inband DTMF
>> when
>> > the codec is g711. The problem is, that our other party wants RFC2833
>> only -
>> > thus we have to remove inband dtmf. Unfortunately Cirpack is not open to
>> do
>> > this.
>> >
>> > Any advice?
>> >
>> > Kind regards,
>> >    T.
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Sun Jan 16 20:04:51 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 16 Jan 2011 12:04:51 -0500
Subject: [Freeswitch-users] bridge a call to remote conference
References: <66485ECAF27B488AB461375999D3EA9B@e1705>
	<AANLkTi=UaTQ70AS2t+OYFrZrHMRW8=Mm5qKP8OyW+OK+@mail.gmail.com>
Message-ID: <DC34326AE42D4E6C808129CD961675F8@e1705>

I tried it and it's the same.
the thing I don't understand is why ivr for conference is starting with this codec

switch_ivr_play_say.c:1236 Codec Activated L16 at 16000hz 1 channels 20ms

so the default conference is in 8000hz ? that's the problem I think

Thanks
  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Sunday, January 16, 2011 4:20 AM
  Subject: Re: [Freeswitch-users] bridge a call to remote conference


  Try answering the call before you enter the conference, possibly with a small sleep inbetween too. That might give the media time to startup before the conference IVR starts.

  http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_answer
  http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sleep

  -Steve



  On 16 January 2011 03:54, Madovsky <infos at madovsky.org> wrote:

    everyhting is ok unless that conference ivr can't be heard (like fraction of sec of sound every 2/3 sec)
    but the hold music (in 8000hz) works well.
    should I set any special codec in the legB conference bridge ?

    Thanks

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Sun Jan 16 20:10:56 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 16 Jan 2011 12:10:56 -0500
Subject: [Freeswitch-users] bridge a call to remote conference
References: <66485ECAF27B488AB461375999D3EA9B@e1705>
	<AANLkTi=UaTQ70AS2t+OYFrZrHMRW8=Mm5qKP8OyW+OK+@mail.gmail.com>
Message-ID: <AE1D75D22E6E4755A666C38404C76C32@e1705>

after some test I confirm that it should force to offer a codec
at the same rate of the conference, example

<action application="bridge" data="{absolute_codec_string=GSM,ignore_early_media=true}sofia/internal/${conference}"/>

as GSM is a 8000hz code so it works now...


thanks
  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Sunday, January 16, 2011 4:20 AM
  Subject: Re: [Freeswitch-users] bridge a call to remote conference


  Try answering the call before you enter the conference, possibly with a small sleep inbetween too. That might give the media time to startup before the conference IVR starts.

  http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_answer
  http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sleep

  -Steve



  On 16 January 2011 03:54, Madovsky <infos at madovsky.org> wrote:

    everyhting is ok unless that conference ivr can't be heard (like fraction of sec of sound every 2/3 sec)
    but the hold music (in 8000hz) works well.
    should I set any special codec in the legB conference bridge ?

    Thanks

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org






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From infos at madovsky.org  Sun Jan 16 20:23:30 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 16 Jan 2011 12:23:30 -0500
Subject: [Freeswitch-users] mod_conference with cluster ODBC
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705><AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com><AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com><AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com><AANLkTimutwOyeubcS=tax5EbDEniCD+Nc4iHMydQxpFb@mail.gmail.com><37761574FEE44D13BEBE4A7DE0089083@e1705><AANLkTi=VZ8JZ+PuyXoz_EYNVovY-AOuPyz9+yfKoUc3k@mail.gmail.com><337274A8CA2343259DF18D8135AF2DCE@e1705><AANLkTimVaxdbwVHCTqpYC5w3q+Vi+t7-oMekrkxu4Kfd@mail.gmail.com><5B0F4D23C27A47A08D36C36B78AB981B@e1705><A7B3B85184604E1DB648F9AD0F99CD7C@stor1><5961547B8D8C429E8B6F7A938CC3E955@e1705><8AC5322D48564DFAA3D3D68B335C5963@stor1>
	<E0E6652F-800E-4562-8967-4DE0461363C9@gmail.com>
Message-ID: <140B7E83F3854BAFB0B6C4381AB68466@e1705>

for those who want a unique conference name on one node only,
I use db application to insert the conference at node_ip when the first caller 
enters (but it depends your conference rules also)
so when other callers are coming I check the db and compare the ip with the 
current node ip of the caller.
if not the same I create a bridge to the conference (with audio codec with 
the same freq rate of the conference).
if it's the same so I only put conference application in my dialplan.
it's almost the same as Joao said unless you bridge a user to the conference 
and not a conference to another conference.
hope this helps

Regards

Franck

----- Original Message ----- 
From: "Chris Rienzo" <cmrienzo at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Cc: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, January 08, 2011 10:28 AM
Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


>I think circularly linked conferences cause echo.  Daisy chained 
>conferences have high latency, though are simple to build.
>
>
> On Jan 8, 2011, at 8:37, "Kris" <kris at livecall.com> wrote:
>
>> On second thought, Joao may be right. If the conference is spread accross 
>> 10
>> servers, and one crashes, it keeps the other users chatting. 
>> A->B->C->D->A.
>> C crashes, the conference on B has to become aware, and immediately 
>> connect
>> to D. I don't know if such failover exists in mod_conference or even if
>> having conferences circularly linked would cause a feedback. If they are
>> circularly linked and only one fails, it should still work even without
>> failover.
>>
>> ----- Original Message ----- 
>> From: "Madovsky" <infos at madovsky.org>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Friday, January 07, 2011 3:31 PM
>> Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
>>
>>
>> it's what I thought first, but Joao is not hot for that apparently.
>> for now I had another idea as I don't want to spread the same
>> conference in several servers.
>>
>> Franck
>>
>> ----- Original Message ----- 
>> From: "Kris" <kris at livecall.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Friday, January 07, 2011 4:43 PM
>> Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
>>
>>
>> Just an idea..soon I will have to put people that are answered on 
>> multiple
>> servers into the same conference. I am thinking about having a table on 
>> the
>> central SQLServer like this: ConferenceName, ServerName. . I would lookup
>> the server a particular conference is on and then transfer the caller to
>> that server and extension that will put the caller into the appropriate
>> conference (dial something.. at SERVER)- I guess. I've seen the export word
>> that maybe the way to pass on variables to the other server such as the
>> ConferenceName, UserName
>>
>> Then the server hosting the conference will have an extension that has 
>> the
>> forums profile and controls
>>      <action application="conference" data="${ConferenceName}@forums"/>
>> That way all the users are in the same conference and can be controlled
>> there instead of having only one link to a bunch of callers on another
>> server.
>>
>> If you get it going, could you email the dial strings, extensions you
>> used.etc.I am curious..
>> Kris
>>
>> ----- Original Message ----- 
>> From: "Madovsky" <infos at madovsky.org>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Friday, January 07, 2011 10:28 AM
>> Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
>>
>>
>> I got it thanks,
>> but do you think it would be more interesting to reduce
>> bandwidth and latency between nodes and centralize the conference on one
>> node only
>> by transferring the incoming user to the right node ?
>>  ----- Original Message ----- 
>>  From: Jo?o Mesquita
>>  To: FreeSWITCH Users Help
>>  Sent: Thursday, January 06, 2011 4:33 PM
>>  Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
>>
>>
>>  Ok, let me see if I can get this into your head. (giggles)
>>
>>
>>  A conference means that the audio needs to mixed in together so that all
>> participants can talk/hear each other, right? If you implement something 
>> in
>> C on mod_conference, you are going to essentially do the same as what an 
>> ESL
>> app does. You _need_ to call in from one server to the other so that you 
>> can
>> mix the audio of all the participants. The real advantage would be the
>> management API being only one for everything and the challenge is exactly
>> that. How to mute certain users on a conference that is spanning over 10
>> servers or deaf them, etc...
>>
>>
>>  A SIP "user" is easier because you don't have to bridge audio from 
>> another
>> server necessarily. Got it?
>>
>>
>>  Regards,
>>  Jo?o Mesquita
>>
>>
>>
>>  On Thu, Jan 6, 2011 at 6:25 PM, Madovsky <infos at madovsky.org> wrote:
>>
>>    Rupa,
>>
>>    I don't want bother anyone with this thread but why not
>>    to manage conference as SIP user ?
>>    if someone from server A call an other who is registered on server B, 
>> so
>>    FS do it automatically, why not with conference ? Or maybe create a
>> param
>>    in mod_conference that let the choice of the admin to manage unique 
>> name
>> in
>>    all cluster or not.
>>    like <param name="cluster_unique_name" value="true"/>
>>    I will try to understand the C code to hack something like this...
>>
>>
>>    ----- Original Message -----
>>    From: "Rupa Schomaker" <rupa at rupa.com>
>>    To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>
>>    Sent: Thursday, January 06, 2011 3:01 PM
>>    Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
>>
>>
>>    Yes
>>
>>    On Thu, Jan 6, 2011 at 1:40 PM, Madovsky <infos at madovsky.org> wrote:
>>> in case of you have 8 servers you have to do it for each ?
>>>
>>> Thanks
>>>
>>> ----- Original Message -----
>>> From: joy this
>>> To: FreeSWITCH Users Help
>>> Sent: Thursday, January 06, 2011 2:51 AM
>>> Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
>>> It works. Thank you everyone.
>>>
>>> 2011/1/5 Rupa Schomaker <rupa at rupa.com>
>>>>
>>>> Use the api: conference <confname> dial [{dial string
>>>> options}]<endpoint_module_name>/<destination> [<callerid_number>
>>>> [<callerid_name>]]
>>>> To initiate the call from within conference A on server 1. Have a
>>>> corresponding dialplan entry on server 2 to accept the call and add
>> it
>>>> into
>>>> the conference A on server 2. You've now bridged the two conferences
>> in
>>>> the
>>>> two servers.
>>>
>>> ________________________________
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> --
>>> This message has been scanned for viruses and
>>> dangerous content by MailScanner, and is
>>> believed to be clean.
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>>    --
>>    -Rupa
>>
>>    _______________________________________________
>>    FreeSWITCH-users mailing list
>>    FreeSWITCH-users at lists.freeswitch.org
>>    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>    http://www.freeswitch.org
>>
>>
>>    _______________________________________________
>>    FreeSWITCH-users mailing list
>>    FreeSWITCH-users at lists.freeswitch.org
>>    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>    http://www.freeswitch.org
>>
>>
>>
>>
>>
>> ------------------------------------------------------------------------------
>>
>>
>>  _______________________________________________
>>  FreeSWITCH-users mailing list
>>  FreeSWITCH-users at lists.freeswitch.org
>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From infos at madovsky.org  Sun Jan 16 22:33:15 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 16 Jan 2011 14:33:15 -0500
Subject: [Freeswitch-users] bridge a call to remote conference
Message-ID: <57626E7D11064552BF446636E880CA8F@e1705>

more weird,

if I choose a codec from the sip phone based on 8000hz it doesn't work.
it's like a rate conversion has to be forced to make the audio work on the conference bridge...
  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Sunday, January 16, 2011 12:10 PM
  Subject: Re: [Freeswitch-users] bridge a call to remote conference


  after some test I confirm that it should force to offer a codec
  at the same rate of the conference, example

  <action application="bridge" data="{absolute_codec_string=GSM,ignore_early_media=true}sofia/internal/${conference}"/>

  as GSM is a 8000hz code so it works now...


  thanks
    ----- Original Message ----- 
    From: Steven Ayre 
    To: FreeSWITCH Users Help 
    Sent: Sunday, January 16, 2011 4:20 AM
    Subject: Re: [Freeswitch-users] bridge a call to remote conference


    Try answering the call before you enter the conference, possibly with a small sleep inbetween too. That might give the media time to startup before the conference IVR starts.

    http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_answer
    http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sleep

    -Steve



    On 16 January 2011 03:54, Madovsky <infos at madovsky.org> wrote:

      everyhting is ok unless that conference ivr can't be heard (like fraction of sec of sound every 2/3 sec)
      but the hold music (in 8000hz) works well.
      should I set any special codec in the legB conference bridge ?

      Thanks

      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org






----------------------------------------------------------------------------


    _______________________________________________
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    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From ayhkor at gmail.com  Sun Jan 16 22:46:43 2011
From: ayhkor at gmail.com (deniro)
Date: Sun, 16 Jan 2011 14:46:43 -0500
Subject: [Freeswitch-users] read conference PIN from a file or db
Message-ID: <AANLkTik3goWsqBVSjXBSD8HJLPHV4FWUFyVPsJXsycdh@mail.gmail.com>

Hi
Regarding providing PIN numbers to a conference
how can I read PINs from a file or database and allow them to go to a
meeting
thx
deniro--
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From steveayre at gmail.com  Sun Jan 16 23:49:03 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 16 Jan 2011 20:49:03 +0000
Subject: [Freeswitch-users] read conference PIN from a file or db
In-Reply-To: <AANLkTik3goWsqBVSjXBSD8HJLPHV4FWUFyVPsJXsycdh@mail.gmail.com>
References: <AANLkTik3goWsqBVSjXBSD8HJLPHV4FWUFyVPsJXsycdh@mail.gmail.com>
Message-ID: <AANLkTi=PFQTZbeNCm7fjE9LrZNFi0V+idB5ypjQeoY=S@mail.gmail.com>

There's several ways, you can write your own module, a script, use xml curl
etc.

The simplest option is probably to use lua - you can access odbc if you
install luasql. I'm sure file access is possible too.

-Steve


On 16 January 2011 19:46, deniro <ayhkor at gmail.com> wrote:

> Hi
> Regarding providing PIN numbers to a conference
> how can I read PINs from a file or database and allow them to go to a
> meeting
> thx
> deniro--
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cmrienzo at gmail.com  Sun Jan 16 23:54:29 2011
From: cmrienzo at gmail.com (Chris Rienzo)
Date: Sun, 16 Jan 2011 15:54:29 -0500
Subject: [Freeswitch-users] Removing inband DTMF
In-Reply-To: <AANLkTi=BQGJJrVgyPKxX2aP3kq_xWRHDzzsf1-qQd9X3@mail.gmail.com>
References: <AANLkTikBZPbeKLO_fX=s4p1pZxPCzwcAet1win8eaUMn@mail.gmail.com>
	<AANLkTimYd8PZumEY1GzkcAj-kZ1auH8Lvr-7OBm7r3fB@mail.gmail.com>
	<AANLkTimgMxuC=FEQ0UFerTLE1wRCKUx5L1Kr-_KMWJ3b@mail.gmail.com>
	<AANLkTi=BQGJJrVgyPKxX2aP3kq_xWRHDzzsf1-qQd9X3@mail.gmail.com>
Message-ID: <A7150993-F25F-474D-A6A9-5EFA84C7ACDB@gmail.com>

That's not true.  A detector can spot the start of dtmf in less than 40ms.  The audio frames can then be replaced with 2833 packets.  This is how it works on media gateways.  I'm sure FS could be patched to do this if it uses the spandsp dtmf detector, which can detect dtmf duration.

On Jan 16, 2011, at 11:50, Steven Ayre <steveayre at gmail.com> wrote:

> First of, AFAIK there's nothing in FS that can currently do it.
> 
> I'm not even sure it's possible at all. You can subtract a noise from audio, but the noisy nature of an encoded signal would mean it wouldn't work well (you'd still hear something), and DTMF can't be detected until some of it will have already been sent to the other endpoint so you'd still have the start of each keypress heard.  For the same reason at the end until the detector spots the dtmf has ended you'll still be subtracting from the audio which will actually generate noise. The only way to avoid that may be to introduce a delay but that's not something FS can do.
> 
> -Steve
> 
> 
> 
> On 16 January 2011 15:33, Tamas Jalsovszky <jalsot at gmail.com> wrote:
> Hello,
> 
> Thanks.
> Yeah, I know that command, as far as I know, it is only for detection, it does not change the media.
> Anybody else?
> 
> Regards,
>    T.
> 
> 
> On Fri, Jan 14, 2011 at 11:23 AM, Avi Marcus <Avi at amarcus.com> wrote:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf
> start_dtmf does detection from the inband. I'm not sure that it removes them from the rtp, though...
> -Avi
> 
> 
> On Fri, Jan 14, 2011 at 12:14 PM, Tamas Jalsovszky <jalsot at gmail.com> wrote:
> > Hello,
> >
> > Is there a way to remove an inband DTMF signal from the RTP stream with
> > FreeSWITCH?
> > We have a partner with ugly Cirpack which can not remove inband DTMF when
> > the codec is g711. The problem is, that our other party wants RFC2833 only -
> > thus we have to remove inband dtmf. Unfortunately Cirpack is not open to do
> > this.
> >
> > Any advice?
> >
> > Kind regards,
> >    T.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From diego.viola at gmail.com  Sun Jan 16 23:59:06 2011
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 16 Jan 2011 17:59:06 -0300
Subject: [Freeswitch-users] Codecs issue
Message-ID: <AANLkTin=U-+LOeJ4JdeL2u3gXapaPdnZDMGk7-kAFAYe@mail.gmail.com>

Hello,

I'm experiencing some strange issue with codecs. I have the following
in my vars.xml file:

  <X-PRE-PROCESS cmd="set"
data="global_codec_prefs=G729,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>

"inbound-late-negotiation" and "disable-transcoding" are commented in
my internal SIP profile. So I guess I'm in Early Negotiation (default
behavior) mode.

However, when I send a call to my provider, and I look at the SIP
trace I see that FS is sending another codec, not G729 as I specified
in the global_codec_prefs / outbound_codec_prefs parameters.

I'm sending calls like this:

        <action application="bridge" data="sofia/internal/$1 at 38.102.93.70"/>

Here is a SIP trace of a call:

http://pastebin.freeswitch.org/15042

I'm not understanding why FS is sending an INVITE with the G7221 codec
in line 240, if I'm telling it explicitly that I want G729 as the
priority when possible in the codec prefs options. But I see G729 in
the 200 OK in line 291.

I've been told to use absolute_codec_string=G729 in my dialplan or
enable late negotiation, but why if I'm already telling it to use G729
in the codec prefs?

my softphone IP: 190.23.80.10
provider IP: 38.102.93.70
FS IP: 77.92.65.126

calls flow like this:

softphone -> FS -> provider

Any help appreciated.


From rafonline at hotmail.com  Mon Jan 17 00:36:58 2011
From: rafonline at hotmail.com (Rafqat .)
Date: Sun, 16 Jan 2011 21:36:58 +0000
Subject: [Freeswitch-users] nibblebill no bal action
Message-ID: <COL122-W14D36811E53F5598D6E855B5F50@phx.gbl>



Hi,

I would like leg A and leg B transferred to different extensions when the no balance amount is triggered.? 

I have tried to set a different nobal_action in leg A's session to what I set in the dial string for leg b. But it seems whatever action i put into the dial string gets executed for leg A aswell as leg B.

After having a look at the wiki, it seems that this is not currently supported.? Can any suggest any work arounds please?


Cheers

Raf




 		 	   		  

From steveayre at gmail.com  Mon Jan 17 00:45:15 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 16 Jan 2011 21:45:15 +0000
Subject: [Freeswitch-users] Codecs issue
In-Reply-To: <AANLkTin=U-+LOeJ4JdeL2u3gXapaPdnZDMGk7-kAFAYe@mail.gmail.com>
References: <AANLkTin=U-+LOeJ4JdeL2u3gXapaPdnZDMGk7-kAFAYe@mail.gmail.com>
Message-ID: <AANLkTimCkD97JC90ewd8uH03bspt_EGpTC_GZuGgn2cB@mail.gmail.com>

Are those variables being used in the outbound-codec-prefs or codec-prefs
params in the sip profile?

What does the "sofia status profile NAME" output show for "CODECS OUT"?

-Steve



On 16 January 2011 20:59, Diego Viola <diego.viola at gmail.com> wrote:

> Hello,
>
> I'm experiencing some strange issue with codecs. I have the following
> in my vars.xml file:
>
>  <X-PRE-PROCESS cmd="set"
> data="global_codec_prefs=G729,G7221 at 32000h,G7221 at 16000h
> ,G722,PCMU,PCMA,GSM"/>
>  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
>
> "inbound-late-negotiation" and "disable-transcoding" are commented in
> my internal SIP profile. So I guess I'm in Early Negotiation (default
> behavior) mode.
>
> However, when I send a call to my provider, and I look at the SIP
> trace I see that FS is sending another codec, not G729 as I specified
> in the global_codec_prefs / outbound_codec_prefs parameters.
>
> I'm sending calls like this:
>
>        <action application="bridge" data="sofia/internal/$1 at 38.102.93.70
> "/>
>
> Here is a SIP trace of a call:
>
> http://pastebin.freeswitch.org/15042
>
> I'm not understanding why FS is sending an INVITE with the G7221 codec
> in line 240, if I'm telling it explicitly that I want G729 as the
> priority when possible in the codec prefs options. But I see G729 in
> the 200 OK in line 291.
>
> I've been told to use absolute_codec_string=G729 in my dialplan or
> enable late negotiation, but why if I'm already telling it to use G729
> in the codec prefs?
>
> my softphone IP: 190.23.80.10
> provider IP: 38.102.93.70
> FS IP: 77.92.65.126
>
> calls flow like this:
>
> softphone -> FS -> provider
>
> Any help appreciated.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Mon Jan 17 00:47:08 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 16 Jan 2011 21:47:08 +0000
Subject: [Freeswitch-users] Removing inband DTMF
In-Reply-To: <A7150993-F25F-474D-A6A9-5EFA84C7ACDB@gmail.com>
References: <AANLkTikBZPbeKLO_fX=s4p1pZxPCzwcAet1win8eaUMn@mail.gmail.com>
	<AANLkTimYd8PZumEY1GzkcAj-kZ1auH8Lvr-7OBm7r3fB@mail.gmail.com>
	<AANLkTimgMxuC=FEQ0UFerTLE1wRCKUx5L1Kr-_KMWJ3b@mail.gmail.com>
	<AANLkTi=BQGJJrVgyPKxX2aP3kq_xWRHDzzsf1-qQd9X3@mail.gmail.com>
	<A7150993-F25F-474D-A6A9-5EFA84C7ACDB@gmail.com>
Message-ID: <AANLkTi=RGqABGRvnEwKVug5XFPPfyYhN2nMdrekkdT22@mail.gmail.com>

I didn't mean a significant amount of time. 40ms with a ptime of 20ms means
you've already sent a packet containing inbound dtmf.

-Steve



On 16 January 2011 20:54, Chris Rienzo <cmrienzo at gmail.com> wrote:

> That's not true.  A detector can spot the start of dtmf in less than 40ms.
>  The audio frames can then be replaced with 2833 packets.  This is how it
> works on media gateways.  I'm sure FS could be patched to do this if it uses
> the spandsp dtmf detector, which can detect dtmf duration.
>
> On Jan 16, 2011, at 11:50, Steven Ayre <steveayre at gmail.com> wrote:
>
> First of, AFAIK there's nothing in FS that can currently do it.
>
> I'm not even sure it's possible at all. You can subtract a noise from
> audio, but the noisy nature of an encoded signal would mean it wouldn't work
> well (you'd still hear something), and DTMF can't be detected until some of
> it will have already been sent to the other endpoint so you'd still have the
> start of each keypress heard.  For the same reason at the end until the
> detector spots the dtmf has ended you'll still be subtracting from the audio
> which will actually generate noise. The only way to avoid that may be to
> introduce a delay but that's not something FS can do.
>
> -Steve
>
>
>
> On 16 January 2011 15:33, Tamas Jalsovszky < <jalsot at gmail.com>
> jalsot at gmail.com> wrote:
>
>> Hello,
>>
>> Thanks.
>> Yeah, I know that command, as far as I know, it is only for detection, it
>> does not change the media.
>> Anybody else?
>>
>> Regards,
>>    T.
>>
>>
>> On Fri, Jan 14, 2011 at 11:23 AM, Avi Marcus < <Avi at amarcus.com>
>> Avi at amarcus.com> wrote:
>>
>>> <http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf>
>>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf
>>> start_dtmf does detection from the inband. I'm not sure that it removes
>>> them from the rtp, though...
>>> -Avi
>>>
>>>
>>> On Fri, Jan 14, 2011 at 12:14 PM, Tamas Jalsovszky < <jalsot at gmail.com>
>>> jalsot at gmail.com> wrote:
>>> > Hello,
>>> >
>>> > Is there a way to remove an inband DTMF signal from the RTP stream with
>>> > FreeSWITCH?
>>> > We have a partner with ugly Cirpack which can not remove inband DTMF
>>> when
>>> > the codec is g711. The problem is, that our other party wants RFC2833
>>> only -
>>> > thus we have to remove inband dtmf. Unfortunately Cirpack is not open
>>> to do
>>> > this.
>>> >
>>> > Any advice?
>>> >
>>> > Kind regards,
>>> >    T.
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Mon Jan 17 00:57:50 2011
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 16 Jan 2011 18:57:50 -0300
Subject: [Freeswitch-users] Codecs issue
In-Reply-To: <AANLkTimCkD97JC90ewd8uH03bspt_EGpTC_GZuGgn2cB@mail.gmail.com>
References: <AANLkTin=U-+LOeJ4JdeL2u3gXapaPdnZDMGk7-kAFAYe@mail.gmail.com>
	<AANLkTimCkD97JC90ewd8uH03bspt_EGpTC_GZuGgn2cB@mail.gmail.com>
Message-ID: <AANLkTimA6Y4WG7rLCXYvAdwbT92vMtnmn5YidG=F-8tX@mail.gmail.com>

>From what I can tell the global_codec_prefs variable is used in the
internal profile, which is the profile I'm using

    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>

freeswitch at internal> sofia status profile internal

CODECS IN               G729,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
CODECS OUT              G729,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM

Still I don't know why FS is picking G7221

On Sun, Jan 16, 2011 at 6:45 PM, Steven Ayre <steveayre at gmail.com> wrote:
> Are those variables being used in the outbound-codec-prefs or codec-prefs
> params in the sip profile?
>
> What does the "sofia status profile NAME" output show for "CODECS OUT"?
>
> -Steve
>
>
>
> On 16 January 2011 20:59, Diego Viola <diego.viola at gmail.com> wrote:
>>
>> Hello,
>>
>> I'm experiencing some strange issue with codecs. I have the following
>> in my vars.xml file:
>>
>> ?<X-PRE-PROCESS cmd="set"
>>
>> data="global_codec_prefs=G729,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>> ?<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
>>
>> "inbound-late-negotiation" and "disable-transcoding" are commented in
>> my internal SIP profile. So I guess I'm in Early Negotiation (default
>> behavior) mode.
>>
>> However, when I send a call to my provider, and I look at the SIP
>> trace I see that FS is sending another codec, not G729 as I specified
>> in the global_codec_prefs / outbound_codec_prefs parameters.
>>
>> I'm sending calls like this:
>>
>> ? ? ? ?<action application="bridge"
>> data="sofia/internal/$1 at 38.102.93.70"/>
>>
>> Here is a SIP trace of a call:
>>
>> http://pastebin.freeswitch.org/15042
>>
>> I'm not understanding why FS is sending an INVITE with the G7221 codec
>> in line 240, if I'm telling it explicitly that I want G729 as the
>> priority when possible in the codec prefs options. But I see G729 in
>> the 200 OK in line 291.
>>
>> I've been told to use absolute_codec_string=G729 in my dialplan or
>> enable late negotiation, but why if I'm already telling it to use G729
>> in the codec prefs?
>>
>> my softphone IP: 190.23.80.10
>> provider IP: 38.102.93.70
>> FS IP: 77.92.65.126
>>
>> calls flow like this:
>>
>> softphone -> FS -> provider
>>
>> Any help appreciated.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Diego Viola
Representative of Bridgecom LLC
Phone: +595 971 320 520
GTalk: diego.viola at gmail.com
MSN: diegoev at msn.com
LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609
www.bridgecom.com.py


From tayeb.meftah at gmail.com  Mon Jan 17 00:55:22 2011
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sun, 16 Jan 2011 22:55:22 +0100
Subject: [Freeswitch-users] Codecs issue
In-Reply-To: <AANLkTin=U-+LOeJ4JdeL2u3gXapaPdnZDMGk7-kAFAYe@mail.gmail.com>
References: <AANLkTin=U-+LOeJ4JdeL2u3gXapaPdnZDMGk7-kAFAYe@mail.gmail.com>
Message-ID: <4D33694A.7000703@gmail.com>

diego,
just one thing:
remove widband codecs
your switch is used for wholesale, not for confrancing
that will confuse your vandor
thanks
Le 16/01/2011 21:59, Diego Viola a ?crit :
> Hello,
>
> I'm experiencing some strange issue with codecs. I have the following
> in my vars.xml file:
>
>    <X-PRE-PROCESS cmd="set"
> data="global_codec_prefs=G729,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>    <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
>
> "inbound-late-negotiation" and "disable-transcoding" are commented in
> my internal SIP profile. So I guess I'm in Early Negotiation (default
> behavior) mode.
>
> However, when I send a call to my provider, and I look at the SIP
> trace I see that FS is sending another codec, not G729 as I specified
> in the global_codec_prefs / outbound_codec_prefs parameters.
>
> I'm sending calls like this:
>
>          <action application="bridge" data="sofia/internal/$1 at 38.102.93.70"/>
>
> Here is a SIP trace of a call:
>
> http://pastebin.freeswitch.org/15042
>
> I'm not understanding why FS is sending an INVITE with the G7221 codec
> in line 240, if I'm telling it explicitly that I want G729 as the
> priority when possible in the codec prefs options. But I see G729 in
> the 200 OK in line 291.
>
> I've been told to use absolute_codec_string=G729 in my dialplan or
> enable late negotiation, but why if I'm already telling it to use G729
> in the codec prefs?
>
> my softphone IP: 190.23.80.10
> provider IP: 38.102.93.70
> FS IP: 77.92.65.126
>
> calls flow like this:
>
> softphone ->  FS ->  provider
>
> Any help appreciated.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313



From rupa at rupa.com  Mon Jan 17 01:02:26 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Sun, 16 Jan 2011 16:02:26 -0600
Subject: [Freeswitch-users] nibblebill no bal action
In-Reply-To: <COL122-W14D36811E53F5598D6E855B5F50@phx.gbl>
References: <COL122-W14D36811E53F5598D6E855B5F50@phx.gbl>
Message-ID: <AANLkTike0xWZqFMvOui_FuZCF9U1AF4seoRkGZYSEKnZ@mail.gmail.com>

Try setting a channel var "nobal_target" to diff values in each leg.
Then when the call is transfered to the no balalance amount extension
transfer to the final nobal_target you defined up front.

At least that is worth a shot....

On Sun, Jan 16, 2011 at 3:36 PM, Rafqat . <rafonline at hotmail.com> wrote:
>
>
> Hi,
>
> I would like leg A and leg B transferred to different extensions when the no balance amount is triggered.
>
> I have tried to set a different nobal_action in leg A's session to what I set in the dial string for leg b. But it seems whatever action i put into the dial string gets executed for leg A aswell as leg B.
>
> After having a look at the wiki, it seems that this is not currently supported.? Can any suggest any work arounds please?
>
>
> Cheers
>
> Raf
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa


From diego.viola at gmail.com  Mon Jan 17 01:15:58 2011
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 16 Jan 2011 19:15:58 -0300
Subject: [Freeswitch-users] Codecs issue
In-Reply-To: <4D33694A.7000703@gmail.com>
References: <AANLkTin=U-+LOeJ4JdeL2u3gXapaPdnZDMGk7-kAFAYe@mail.gmail.com>
	<4D33694A.7000703@gmail.com>
Message-ID: <AANLkTikQxx=Yhbb+j7qwTj6Xvqa73e7-JMgYL0-JWBVQ@mail.gmail.com>

Done, thanks.

freeswitch at internal> sofia status profile internal

CODECS IN               G729,PCMU,PCMA,GSM
CODECS OUT              G729,PCMU,PCMA,GSM


On Sun, Jan 16, 2011 at 6:55 PM, Meftah Tayeb <tayeb.meftah at gmail.com> wrote:
> diego,
> just one thing:
> remove widband codecs
> your switch is used for wholesale, not for confrancing
> that will confuse your vandor
> thanks
> Le 16/01/2011 21:59, Diego Viola a ?crit :
>>
>> Hello,
>>
>> I'm experiencing some strange issue with codecs. I have the following
>> in my vars.xml file:
>>
>> ? <X-PRE-PROCESS cmd="set"
>>
>> data="global_codec_prefs=G729,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>> ? <X-PRE-PROCESS cmd="set"
>> data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
>>
>> "inbound-late-negotiation" and "disable-transcoding" are commented in
>> my internal SIP profile. So I guess I'm in Early Negotiation (default
>> behavior) mode.
>>
>> However, when I send a call to my provider, and I look at the SIP
>> trace I see that FS is sending another codec, not G729 as I specified
>> in the global_codec_prefs / outbound_codec_prefs parameters.
>>
>> I'm sending calls like this:
>>
>> ? ? ? ? <action application="bridge"
>> data="sofia/internal/$1 at 38.102.93.70"/>
>>
>> Here is a SIP trace of a call:
>>
>> http://pastebin.freeswitch.org/15042
>>
>> I'm not understanding why FS is sending an INVITE with the G7221 codec
>> in line 240, if I'm telling it explicitly that I want G729 as the
>> priority when possible in the codec prefs options. But I see G729 in
>> the 200 OK in line 291.
>>
>> I've been told to use absolute_codec_string=G729 in my dialplan or
>> enable late negotiation, but why if I'm already telling it to use G729
>> in the codec prefs?
>>
>> my softphone IP: 190.23.80.10
>> provider IP: 38.102.93.70
>> FS IP: 77.92.65.126
>>
>> calls flow like this:
>>
>> softphone -> ?FS -> ?provider
>>
>> Any help appreciated.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> --
> Meftah Tayeb
> inum: +883510001288000
> Phone: +13602276297
> Fax: +12538020313
>
>



-- 
Diego Viola
Representative of Bridgecom LLC
Phone: +595 971 320 520
GTalk: diego.viola at gmail.com
MSN: diegoev at msn.com
LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609
www.bridgecom.com.py


From brian at freeswitch.org  Mon Jan 17 01:16:41 2011
From: brian at freeswitch.org (Brian West)
Date: Sun, 16 Jan 2011 16:16:41 -0600
Subject: [Freeswitch-users] Removing inband DTMF
In-Reply-To: <AANLkTi=RGqABGRvnEwKVug5XFPPfyYhN2nMdrekkdT22@mail.gmail.com>
References: <AANLkTikBZPbeKLO_fX=s4p1pZxPCzwcAet1win8eaUMn@mail.gmail.com>
	<AANLkTimYd8PZumEY1GzkcAj-kZ1auH8Lvr-7OBm7r3fB@mail.gmail.com>
	<AANLkTimgMxuC=FEQ0UFerTLE1wRCKUx5L1Kr-_KMWJ3b@mail.gmail.com>
	<AANLkTi=BQGJJrVgyPKxX2aP3kq_xWRHDzzsf1-qQd9X3@mail.gmail.com>
	<A7150993-F25F-474D-A6A9-5EFA84C7ACDB@gmail.com>
	<AANLkTi=RGqABGRvnEwKVug5XFPPfyYhN2nMdrekkdT22@mail.gmail.com>
Message-ID: <101096F0-3FCF-4099-B622-EB7F315CFBF2@freeswitch.org>

Beat your provider to death for doing something utterly stupid.   If you are only acting on 2833 it won't matter if they send inband and 2833 at the same time..  It won't matter as long as you are not staring a dtmf detector on the target system and only the system converting from inband to 2833 does so.

/b

On Jan 16, 2011, at 3:47 PM, Steven Ayre wrote:

> I didn't mean a significant amount of time. 40ms with a ptime of 20ms means you've already sent a packet containing inbound dtmf.
> 
> -Steve



From brian at freeswitch.org  Mon Jan 17 01:17:37 2011
From: brian at freeswitch.org (Brian West)
Date: Sun, 16 Jan 2011 16:17:37 -0600
Subject: [Freeswitch-users] Codecs issue
In-Reply-To: <AANLkTimA6Y4WG7rLCXYvAdwbT92vMtnmn5YidG=F-8tX@mail.gmail.com>
References: <AANLkTin=U-+LOeJ4JdeL2u3gXapaPdnZDMGk7-kAFAYe@mail.gmail.com>
	<AANLkTimCkD97JC90ewd8uH03bspt_EGpTC_GZuGgn2cB@mail.gmail.com>
	<AANLkTimA6Y4WG7rLCXYvAdwbT92vMtnmn5YidG=F-8tX@mail.gmail.com>
Message-ID: <4F27499B-99CA-49CD-89B0-987B47271383@freeswitch.org>

Why not load mod_com_g729 or mod_g729... the only reason an offer would be removed is you DO NOT have the codec loaded.

/b

On Jan 16, 2011, at 3:57 PM, Diego Viola wrote:

>> From what I can tell the global_codec_prefs variable is used in the
> internal profile, which is the profile I'm using
> 
>    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
>    <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
> 
> freeswitch at internal> sofia status profile internal
> 
> CODECS IN               G729,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
> CODECS OUT              G729,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
> 
> Still I don't know why FS is picking G7221



From brian at freeswitch.org  Mon Jan 17 01:18:13 2011
From: brian at freeswitch.org (Brian West)
Date: Sun, 16 Jan 2011 16:18:13 -0600
Subject: [Freeswitch-users] Codecs issue
In-Reply-To: <AANLkTikQxx=Yhbb+j7qwTj6Xvqa73e7-JMgYL0-JWBVQ@mail.gmail.com>
References: <AANLkTin=U-+LOeJ4JdeL2u3gXapaPdnZDMGk7-kAFAYe@mail.gmail.com>
	<4D33694A.7000703@gmail.com>
	<AANLkTikQxx=Yhbb+j7qwTj6Xvqa73e7-JMgYL0-JWBVQ@mail.gmail.com>
Message-ID: <57B70F8C-7B71-49E7-A7F7-A1070BD05FDC@freeswitch.org>

This can say exactly what ever you wish it to say but if the codec module is NOT loaded it will be removed from the offer.

/b

On Jan 16, 2011, at 4:15 PM, Diego Viola wrote:

> Done, thanks.
> 
> freeswitch at internal> sofia status profile internal
> 
> CODECS IN               G729,PCMU,PCMA,GSM
> CODECS OUT              G729,PCMU,PCMA,GSM



From diego.viola at gmail.com  Mon Jan 17 01:22:56 2011
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 16 Jan 2011 19:22:56 -0300
Subject: [Freeswitch-users] Codecs issue
In-Reply-To: <57B70F8C-7B71-49E7-A7F7-A1070BD05FDC@freeswitch.org>
References: <AANLkTin=U-+LOeJ4JdeL2u3gXapaPdnZDMGk7-kAFAYe@mail.gmail.com>
	<4D33694A.7000703@gmail.com>
	<AANLkTikQxx=Yhbb+j7qwTj6Xvqa73e7-JMgYL0-JWBVQ@mail.gmail.com>
	<57B70F8C-7B71-49E7-A7F7-A1070BD05FDC@freeswitch.org>
Message-ID: <AANLkTikvnJhEREXqBHWBtvkwCepafe6QGej9FoeuFz0b@mail.gmail.com>

I have mod_com_g729 loaded

freeswitch at internal> load mod_com_g729
+OK Reloading XML
-ERR [Module already loaded]

2011-01-16 22:22:49.832772 [WARNING] switch_loadable_module.c:996
Module mod_com_g729 Already Loaded!
freeswitch at internal> 2011-01-16 22:22:49.832772 [INFO] mod_enum.c:808
ENUM Reloaded
2011-01-16 22:22:49.832772 [INFO] switch_time.c:954 Timezone reloaded
530 definitions

freeswitch at internal>








On Sun, Jan 16, 2011 at 7:18 PM, Brian West <brian at freeswitch.org> wrote:
> This can say exactly what ever you wish it to say but if the codec module is NOT loaded it will be removed from the offer.
>
> /b
>
> On Jan 16, 2011, at 4:15 PM, Diego Viola wrote:
>
>> Done, thanks.
>>
>> freeswitch at internal> sofia status profile internal
>>
>> CODECS IN ? ? ? ? ? ? ? G729,PCMU,PCMA,GSM
>> CODECS OUT ? ? ? ? ? ? ?G729,PCMU,PCMA,GSM
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Diego Viola
Representative of Bridgecom LLC
Phone: +595 971 320 520
GTalk: diego.viola at gmail.com
MSN: diegoev at msn.com
LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609
www.bridgecom.com.py


From brian at freeswitch.org  Mon Jan 17 01:29:10 2011
From: brian at freeswitch.org (Brian West)
Date: Sun, 16 Jan 2011 16:29:10 -0600
Subject: [Freeswitch-users] Codecs issue
In-Reply-To: <AANLkTikvnJhEREXqBHWBtvkwCepafe6QGej9FoeuFz0b@mail.gmail.com>
References: <AANLkTin=U-+LOeJ4JdeL2u3gXapaPdnZDMGk7-kAFAYe@mail.gmail.com>
	<4D33694A.7000703@gmail.com>
	<AANLkTikQxx=Yhbb+j7qwTj6Xvqa73e7-JMgYL0-JWBVQ@mail.gmail.com>
	<57B70F8C-7B71-49E7-A7F7-A1070BD05FDC@freeswitch.org>
	<AANLkTikvnJhEREXqBHWBtvkwCepafe6QGej9FoeuFz0b@mail.gmail.com>
Message-ID: <0DE8D0D6-3645-43D8-BA87-10494B98364F@freeswitch.org>

What does g729_info say?

/b

On Jan 16, 2011, at 4:22 PM, Diego Viola wrote:

> I have mod_com_g729 loaded
> 
> freeswitch at internal> load mod_com_g729
> +OK Reloading XML
> -ERR [Module already loaded]
> 
> 2011-01-16 22:22:49.832772 [WARNING] switch_loadable_module.c:996
> Module mod_com_g729 Already Loaded!
> freeswitch at internal> 2011-01-16 22:22:49.832772 [INFO] mod_enum.c:808
> ENUM Reloaded
> 2011-01-16 22:22:49.832772 [INFO] switch_time.c:954 Timezone reloaded
> 530 definitions
> 
> freeswitch at internal>



From brian at freeswitch.org  Mon Jan 17 01:29:48 2011
From: brian at freeswitch.org (Brian West)
Date: Sun, 16 Jan 2011 16:29:48 -0600
Subject: [Freeswitch-users] IP Failover with Sofia Recover - not
	recovering?!
In-Reply-To: <AANLkTi=FOkWq9pRFqUhGN-gUVjvajGH1iSxU9u1nu+X-@mail.gmail.com>
References: <mailman.1927.1295137414.24555.freeswitch-users@lists.freeswitch.org>
	<AANLkTinA9f19XPq6d+TzeXt3iE7PW500yBxPGAS+R3GH@mail.gmail.com>
	<AANLkTi=FOkWq9pRFqUhGN-gUVjvajGH1iSxU9u1nu+X-@mail.gmail.com>
Message-ID: <388C53B0-D1D6-439B-A3B8-1616323766AE@freeswitch.org>

Hostnames would need to be the same on both.

/b

On Jan 16, 2011, at 3:18 AM, Steven Ayre wrote:

> I believe that hostname is the hostname of the server and set on FS startup.
> 
> -Steve



From brian at freeswitch.org  Mon Jan 17 01:30:33 2011
From: brian at freeswitch.org (Brian West)
Date: Sun, 16 Jan 2011 16:30:33 -0600
Subject: [Freeswitch-users] BLIND_TRANFER dialplan
In-Reply-To: <13C9ECCC43834B309130668C170F0D53@e1705>
References: <13C9ECCC43834B309130668C170F0D53@e1705>
Message-ID: <31497377-7C0D-4BA8-9B34-A8BFCBE5C701@freeswitch.org>

What exactly are you trying to accomplish?  A little back story and info as to WHY you want the dialplan to do this.

/b


On Jan 15, 2011, at 10:06 AM, Madovsky wrote:

> is it possible to create a blind transfer manually without deflect ?
>  
> Thanks

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From diego.viola at gmail.com  Mon Jan 17 01:31:32 2011
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 16 Jan 2011 19:31:32 -0300
Subject: [Freeswitch-users] Codecs issue
In-Reply-To: <0DE8D0D6-3645-43D8-BA87-10494B98364F@freeswitch.org>
References: <AANLkTin=U-+LOeJ4JdeL2u3gXapaPdnZDMGk7-kAFAYe@mail.gmail.com>
	<4D33694A.7000703@gmail.com>
	<AANLkTikQxx=Yhbb+j7qwTj6Xvqa73e7-JMgYL0-JWBVQ@mail.gmail.com>
	<57B70F8C-7B71-49E7-A7F7-A1070BD05FDC@freeswitch.org>
	<AANLkTikvnJhEREXqBHWBtvkwCepafe6QGej9FoeuFz0b@mail.gmail.com>
	<0DE8D0D6-3645-43D8-BA87-10494B98364F@freeswitch.org>
Message-ID: <AANLkTinyY+GN=KgazOMvHv0d+4OZ3OVov9Z0e-ovw0JG@mail.gmail.com>

freeswitch at internal> g729_info
Permitted G729 channels: 10
Encoders in use: 0
Decoders in use: 0

freeswitch at internal>


On Sun, Jan 16, 2011 at 7:29 PM, Brian West <brian at freeswitch.org> wrote:
> What does g729_info say?
>
> /b
>
> On Jan 16, 2011, at 4:22 PM, Diego Viola wrote:
>
>> I have mod_com_g729 loaded
>>
>> freeswitch at internal> load mod_com_g729
>> +OK Reloading XML
>> -ERR [Module already loaded]
>>
>> 2011-01-16 22:22:49.832772 [WARNING] switch_loadable_module.c:996
>> Module mod_com_g729 Already Loaded!
>> freeswitch at internal> 2011-01-16 22:22:49.832772 [INFO] mod_enum.c:808
>> ENUM Reloaded
>> 2011-01-16 22:22:49.832772 [INFO] switch_time.c:954 Timezone reloaded
>> 530 definitions
>>
>> freeswitch at internal>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Diego Viola
Representative of Bridgecom LLC
Phone: +595 971 320 520
GTalk: diego.viola at gmail.com
MSN: diegoev at msn.com
LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609
www.bridgecom.com.py


From brian at freeswitch.org  Mon Jan 17 01:32:17 2011
From: brian at freeswitch.org (Brian West)
Date: Sun, 16 Jan 2011 16:32:17 -0600
Subject: [Freeswitch-users] SOFIA_REFER_TO_VARIABLE
In-Reply-To: <FE8BF1788FF04027A7D52B9685F4ACDE@e1705>
References: <FE8BF1788FF04027A7D52B9685F4ACDE@e1705>
Message-ID: <07F635FF-2313-43EF-B587-B45A7F764492@freeswitch.org>

Do you think you can wait 20 min and group your questions into a single email instead of sending more than one with a single sentence question.

Also can you clarify what you mean?  This variable is set when FreeSWITCH itself is sent a refer.  Its set to tell you where you were referred to.

/b

On Jan 14, 2011, at 9:38 PM, Madovsky wrote:

> can anyone what the rul of this var ?
>  
> Thanks

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From brian at freeswitch.org  Mon Jan 17 01:34:33 2011
From: brian at freeswitch.org (Brian West)
Date: Sun, 16 Jan 2011 16:34:33 -0600
Subject: [Freeswitch-users] watchdog
In-Reply-To: <0CFFA71FF3DC475DB0CC476A30C7BBE6@e1705>
References: <0CFFA71FF3DC475DB0CC476A30C7BBE6@e1705>
Message-ID: <92F823F5-2451-44D4-973E-2C46E8E58960@freeswitch.org>

What do the logs say when yous tart up?

/b

On Jan 15, 2011, at 10:57 AM, Madovsky wrote:

> I tried to add this on sip profile
>  
>     <!--<param name="watchdog-enabled" value="yes"/>-->
>     <!--<param name="watchdog-step-timeout" value="30000"/>-->
>     <!--<param name="watchdog-event-timeout" value="30000"/>-->
> but after that freeswitch shutdown saying STACK error.
>  
> Thanks

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From steveayre at gmail.com  Mon Jan 17 01:42:10 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 16 Jan 2011 22:42:10 +0000
Subject: [Freeswitch-users] Codecs issue
In-Reply-To: <AANLkTimA6Y4WG7rLCXYvAdwbT92vMtnmn5YidG=F-8tX@mail.gmail.com>
References: <AANLkTin=U-+LOeJ4JdeL2u3gXapaPdnZDMGk7-kAFAYe@mail.gmail.com>
	<AANLkTimCkD97JC90ewd8uH03bspt_EGpTC_GZuGgn2cB@mail.gmail.com>
	<AANLkTimA6Y4WG7rLCXYvAdwbT92vMtnmn5YidG=F-8tX@mail.gmail.com>
Message-ID: <AANLkTi=7MA_LtsJR8-bPpT388vxA6powoP_NB+_JM3Rp@mail.gmail.com>

That's why then. Unless you have disable-transcoding FS will offer all of
those to the endpoint. The endpoint picks which one it wants to use. FS will
then transcode if necessary.

The way to avoid that would be disable-transcoding (FS will pick the aleg
codec and then only offer that on the bleg), or use absolute_codec_string.

-Steve


On 16 January 2011 21:57, Diego Viola <diego.viola at gmail.com> wrote:

> >From what I can tell the global_codec_prefs variable is used in the
> internal profile, which is the profile I'm using
>
>    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
>    <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
>
> freeswitch at internal> sofia status profile internal
>
> CODECS IN               G729,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
> CODECS OUT              G729,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
>
> Still I don't know why FS is picking G7221
>
> On Sun, Jan 16, 2011 at 6:45 PM, Steven Ayre <steveayre at gmail.com> wrote:
> > Are those variables being used in the outbound-codec-prefs or codec-prefs
> > params in the sip profile?
> >
> > What does the "sofia status profile NAME" output show for "CODECS OUT"?
> >
> > -Steve
> >
> >
> >
> > On 16 January 2011 20:59, Diego Viola <diego.viola at gmail.com> wrote:
> >>
> >> Hello,
> >>
> >> I'm experiencing some strange issue with codecs. I have the following
> >> in my vars.xml file:
> >>
> >>  <X-PRE-PROCESS cmd="set"
> >>
> >> data="global_codec_prefs=G729,G7221 at 32000h,G7221 at 16000h
> ,G722,PCMU,PCMA,GSM"/>
> >>  <X-PRE-PROCESS cmd="set"
> data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
> >>
> >> "inbound-late-negotiation" and "disable-transcoding" are commented in
> >> my internal SIP profile. So I guess I'm in Early Negotiation (default
> >> behavior) mode.
> >>
> >> However, when I send a call to my provider, and I look at the SIP
> >> trace I see that FS is sending another codec, not G729 as I specified
> >> in the global_codec_prefs / outbound_codec_prefs parameters.
> >>
> >> I'm sending calls like this:
> >>
> >>        <action application="bridge"
> >> data="sofia/internal/$1 at 38.102.93.70"/>
> >>
> >> Here is a SIP trace of a call:
> >>
> >> http://pastebin.freeswitch.org/15042
> >>
> >> I'm not understanding why FS is sending an INVITE with the G7221 codec
> >> in line 240, if I'm telling it explicitly that I want G729 as the
> >> priority when possible in the codec prefs options. But I see G729 in
> >> the 200 OK in line 291.
> >>
> >> I've been told to use absolute_codec_string=G729 in my dialplan or
> >> enable late negotiation, but why if I'm already telling it to use G729
> >> in the codec prefs?
> >>
> >> my softphone IP: 190.23.80.10
> >> provider IP: 38.102.93.70
> >> FS IP: 77.92.65.126
> >>
> >> calls flow like this:
> >>
> >> softphone -> FS -> provider
> >>
> >> Any help appreciated.
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Diego Viola
> Representative of Bridgecom LLC
> Phone: +595 971 320 520
> GTalk: diego.viola at gmail.com
> MSN: diegoev at msn.com
> LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609
> www.bridgecom.com.py
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ayhkor at gmail.com  Mon Jan 17 01:45:05 2011
From: ayhkor at gmail.com (deniro)
Date: Sun, 16 Jan 2011 17:45:05 -0500
Subject: [Freeswitch-users] read conference PIN from a file or db
In-Reply-To: <AANLkTi=PFQTZbeNCm7fjE9LrZNFi0V+idB5ypjQeoY=S@mail.gmail.com>
References: <AANLkTik3goWsqBVSjXBSD8HJLPHV4FWUFyVPsJXsycdh@mail.gmail.com>
	<AANLkTi=PFQTZbeNCm7fjE9LrZNFi0V+idB5ypjQeoY=S@mail.gmail.com>
Message-ID: <AANLkTi=oV+hN0E12A=St7faH+65U+nCZBHKKb444LA1D@mail.gmail.com>

how about writing a javascript and reading from mysql database
when a match is hit return a value and  load it into $pin
how can I do this type of javascripting?
thx



On Sun, Jan 16, 2011 at 3:49 PM, Steven Ayre <steveayre at gmail.com> wrote:

> There's several ways, you can write your own module, a script, use xml curl
> etc.
>
> The simplest option is probably to use lua - you can access odbc if you
> install luasql. I'm sure file access is possible too.
>
> -Steve
>
>
>   On 16 January 2011 19:46, deniro <ayhkor at gmail.com> wrote:
>
>>   Hi
>> Regarding providing PIN numbers to a conference
>> how can I read PINs from a file or database and allow them to go to a
>> meeting
>> thx
>> deniro--
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From avi at avimarcus.net  Mon Jan 17 01:45:55 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 17 Jan 2011 00:45:55 +0200
Subject: [Freeswitch-users] IP Failover with Sofia Recover - not
	recovering?!
In-Reply-To: <388C53B0-D1D6-439B-A3B8-1616323766AE@freeswitch.org>
References: <mailman.1927.1295137414.24555.freeswitch-users@lists.freeswitch.org>
	<AANLkTinA9f19XPq6d+TzeXt3iE7PW500yBxPGAS+R3GH@mail.gmail.com>
	<AANLkTi=FOkWq9pRFqUhGN-gUVjvajGH1iSxU9u1nu+X-@mail.gmail.com>
	<388C53B0-D1D6-439B-A3B8-1616323766AE@freeswitch.org>
Message-ID: <AANLkTikARk4teUNpc2JySRbUVsbp+SRbCaQczfT1EnfJ@mail.gmail.com>

I'm supposed to have two machines with the same hostname - it's not just a
variable in the xml files..? Isn't that a bit odd?
And even when I modified the sql table to change the hostname in the DB and
then do a sofia recover, it didn't recover. I'm not sure how to debug this,
I'm just getting "no calls to recover" and I don't hear anything.

-Avi

On Mon, Jan 17, 2011 at 12:29 AM, Brian West <brian at freeswitch.org> wrote:

> Hostnames would need to be the same on both.
>
> /b
>
> On Jan 16, 2011, at 3:18 AM, Steven Ayre wrote:
>
> > I believe that hostname is the hostname of the server and set on FS
> startup.
> >
> > -Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.ponzone at ipeva.fr  Mon Jan 17 01:48:22 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sun, 16 Jan 2011 23:48:22 +0100
Subject: [Freeswitch-users] Codecs issue
In-Reply-To: <AANLkTin=U-+LOeJ4JdeL2u3gXapaPdnZDMGk7-kAFAYe@mail.gmail.com>
References: <AANLkTin=U-+LOeJ4JdeL2u3gXapaPdnZDMGk7-kAFAYe@mail.gmail.com>
Message-ID: <ED953AB7-108D-452B-B42E-C68FEBA8005F@ipeva.fr>

That sounds like a bad reading of the trace.

The important line is the 239th:
 m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13

That means codec 3 is sent first (so GSM), then 18 (so G729).

I think you should read a little bit on early negotiation.
If leg A has GSM in 1st position, if your inbound codec list allows GSM, and your outbound codec list is : G729, PCM, GSM, than your outbound INVITE to leg B will have GSM in 1st position, because that was requested by leg A.

That's in the wiki, it's a part I rewrote myself some weeks ago, in the most readable way I could.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 16/01/2011 ? 21:59, Diego Viola a ?crit :

> Hello,
> 
> I'm experiencing some strange issue with codecs. I have the following
> in my vars.xml file:
> 
>  <X-PRE-PROCESS cmd="set"
> data="global_codec_prefs=G729,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
> 
> "inbound-late-negotiation" and "disable-transcoding" are commented in
> my internal SIP profile. So I guess I'm in Early Negotiation (default
> behavior) mode.
> 
> However, when I send a call to my provider, and I look at the SIP
> trace I see that FS is sending another codec, not G729 as I specified
> in the global_codec_prefs / outbound_codec_prefs parameters.
> 
> I'm sending calls like this:
> 
>        <action application="bridge" data="sofia/internal/$1 at 38.102.93.70"/>
> 
> Here is a SIP trace of a call:
> 
> http://pastebin.freeswitch.org/15042
> 
> I'm not understanding why FS is sending an INVITE with the G7221 codec
> in line 240, if I'm telling it explicitly that I want G729 as the
> priority when possible in the codec prefs options. But I see G729 in
> the 200 OK in line 291.
> 
> I've been told to use absolute_codec_string=G729 in my dialplan or
> enable late negotiation, but why if I'm already telling it to use G729
> in the codec prefs?
> 
> my softphone IP: 190.23.80.10
> provider IP: 38.102.93.70
> FS IP: 77.92.65.126
> 
> calls flow like this:
> 
> softphone -> FS -> provider
> 
> Any help appreciated.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From tayeb.meftah at gmail.com  Mon Jan 17 01:46:11 2011
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sun, 16 Jan 2011 23:46:11 +0100
Subject: [Freeswitch-users] core dump while using Codec2
Message-ID: <4D337533.9090107@gmail.com>

guys,
here is a core dump while testing codec 2 module
http://pastebin.freeswitch.org/15043
thanks

-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313



From steveayre at gmail.com  Mon Jan 17 01:55:12 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 16 Jan 2011 22:55:12 +0000
Subject: [Freeswitch-users] IP Failover with Sofia Recover - not
	recovering?!
In-Reply-To: <AANLkTikARk4teUNpc2JySRbUVsbp+SRbCaQczfT1EnfJ@mail.gmail.com>
References: <mailman.1927.1295137414.24555.freeswitch-users@lists.freeswitch.org>
	<AANLkTinA9f19XPq6d+TzeXt3iE7PW500yBxPGAS+R3GH@mail.gmail.com>
	<AANLkTi=FOkWq9pRFqUhGN-gUVjvajGH1iSxU9u1nu+X-@mail.gmail.com>
	<388C53B0-D1D6-439B-A3B8-1616323766AE@freeswitch.org>
	<AANLkTikARk4teUNpc2JySRbUVsbp+SRbCaQczfT1EnfJ@mail.gmail.com>
Message-ID: <AANLkTikwvmHjmMVZ7OeQiGX_EXh9Kwty97Dc5Q=9j2NB@mail.gmail.com>

Well, partly that hostname variable predates all the sofia recover stuff so
there was no reason for it originally.

-Steve



On 16 January 2011 22:45, Avi Marcus <avi at avimarcus.net> wrote:

> I'm supposed to have two machines with the same hostname - it's not just a
> variable in the xml files..? Isn't that a bit odd?
> And even when I modified the sql table to change the hostname in the DB and
> then do a sofia recover, it didn't recover. I'm not sure how to debug this,
> I'm just getting "no calls to recover" and I don't hear anything.
>
> -Avi
>
>
> On Mon, Jan 17, 2011 at 12:29 AM, Brian West <brian at freeswitch.org> wrote:
>
>> Hostnames would need to be the same on both.
>>
>> /b
>>
>> On Jan 16, 2011, at 3:18 AM, Steven Ayre wrote:
>>
>> > I believe that hostname is the hostname of the server and set on FS
>> startup.
>> >
>> > -Steve
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Mon Jan 17 01:58:36 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 16 Jan 2011 22:58:36 +0000
Subject: [Freeswitch-users] read conference PIN from a file or db
In-Reply-To: <AANLkTi=oV+hN0E12A=St7faH+65U+nCZBHKKb444LA1D@mail.gmail.com>
References: <AANLkTik3goWsqBVSjXBSD8HJLPHV4FWUFyVPsJXsycdh@mail.gmail.com>
	<AANLkTi=PFQTZbeNCm7fjE9LrZNFi0V+idB5ypjQeoY=S@mail.gmail.com>
	<AANLkTi=oV+hN0E12A=St7faH+65U+nCZBHKKb444LA1D@mail.gmail.com>
Message-ID: <AANLkTinMEVy55S1kqyKq5RVxBJdmd3pELr5fnezs7dGR@mail.gmail.com>

Sure you can - mod_spidermonkey gives you javascript, if you configure
spidermonkey.conf.xml to load mod_spidermonkey_odbc you'll get odbc access
so you can access a mysql database.

http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc
http://wiki.freeswitch.org/wiki/Session_setVariable

-Steve



On 16 January 2011 22:45, deniro <ayhkor at gmail.com> wrote:

> how about writing a javascript and reading from mysql database
> when a match is hit return a value and  load it into $pin
> how can I do this type of javascripting?
> thx
>
>
>
> On Sun, Jan 16, 2011 at 3:49 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> There's several ways, you can write your own module, a script, use xml
>> curl etc.
>>
>> The simplest option is probably to use lua - you can access odbc if you
>> install luasql. I'm sure file access is possible too.
>>
>> -Steve
>>
>>
>>   On 16 January 2011 19:46, deniro <ayhkor at gmail.com> wrote:
>>
>>>   Hi
>>> Regarding providing PIN numbers to a conference
>>> how can I read PINs from a file or database and allow them to go to a
>>> meeting
>>> thx
>>> deniro--
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Jan 17 02:03:12 2011
From: brian at freeswitch.org (Brian West)
Date: Sun, 16 Jan 2011 17:03:12 -0600
Subject: [Freeswitch-users] core dump while using Codec2
In-Reply-To: <4D337533.9090107@gmail.com>
References: <4D337533.9090107@gmail.com>
Message-ID: <F22B9D28-F3CA-4D81-A356-C1C5EB28C3DA@freeswitch.org>

Yes it will coredump it still has bugs please report it to the codec2 project.

/b

On Jan 16, 2011, at 4:46 PM, Meftah Tayeb wrote:

> guys,
> here is a core dump while testing codec 2 module
> http://pastebin.freeswitch.org/15043
> thanks

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From brian at freeswitch.org  Mon Jan 17 02:04:15 2011
From: brian at freeswitch.org (Brian West)
Date: Sun, 16 Jan 2011 17:04:15 -0600
Subject: [Freeswitch-users] IP Failover with Sofia Recover - not
	recovering?!
In-Reply-To: <AANLkTikARk4teUNpc2JySRbUVsbp+SRbCaQczfT1EnfJ@mail.gmail.com>
References: <mailman.1927.1295137414.24555.freeswitch-users@lists.freeswitch.org>
	<AANLkTinA9f19XPq6d+TzeXt3iE7PW500yBxPGAS+R3GH@mail.gmail.com>
	<AANLkTi=FOkWq9pRFqUhGN-gUVjvajGH1iSxU9u1nu+X-@mail.gmail.com>
	<388C53B0-D1D6-439B-A3B8-1616323766AE@freeswitch.org>
	<AANLkTikARk4teUNpc2JySRbUVsbp+SRbCaQczfT1EnfJ@mail.gmail.com>
Message-ID: <EA5325B3-A66A-4783-AF4E-EB2359CE13DC@freeswitch.org>

Well since all the sql queries will only look for stuff in the db thats for ITS host.

/b

On Jan 16, 2011, at 4:45 PM, Avi Marcus wrote:

> I'm supposed to have two machines with the same hostname - it's not just a variable in the xml files..? Isn't that a bit odd?
> And even when I modified the sql table to change the hostname in the DB and then do a sofia recover, it didn't recover. I'm not sure how to debug this, I'm just getting "no calls to recover" and I don't hear anything.
> 
> -Avi



From brian at freeswitch.org  Mon Jan 17 02:04:58 2011
From: brian at freeswitch.org (Brian West)
Date: Sun, 16 Jan 2011 17:04:58 -0600
Subject: [Freeswitch-users] IP Failover with Sofia Recover - not
	recovering?!
In-Reply-To: <AANLkTikwvmHjmMVZ7OeQiGX_EXh9Kwty97Dc5Q=9j2NB@mail.gmail.com>
References: <mailman.1927.1295137414.24555.freeswitch-users@lists.freeswitch.org>
	<AANLkTinA9f19XPq6d+TzeXt3iE7PW500yBxPGAS+R3GH@mail.gmail.com>
	<AANLkTi=FOkWq9pRFqUhGN-gUVjvajGH1iSxU9u1nu+X-@mail.gmail.com>
	<388C53B0-D1D6-439B-A3B8-1616323766AE@freeswitch.org>
	<AANLkTikARk4teUNpc2JySRbUVsbp+SRbCaQczfT1EnfJ@mail.gmail.com>
	<AANLkTikwvmHjmMVZ7OeQiGX_EXh9Kwty97Dc5Q=9j2NB@mail.gmail.com>
Message-ID: <FAC8D42E-2047-4682-873C-29D3E5F379FF@freeswitch.org>

Its so you can have multiple boxes talking to the same db and they won't clobber each others data.  But in recover you have to pair them so that the secondary box can recover for the records of the primary.

/b

On Jan 16, 2011, at 4:55 PM, Steven Ayre wrote:

> Well, partly that hostname variable predates all the sofia recover stuff so there was no reason for it originally.
> 
> -Steve
> 
> 
> 
> On 16 January 2011 22:45, Avi Marcus <avi at avimarcus.net> wrote:
> I'm supposed to have two machines with the same hostname - it's not just a variable in the xml files..? Isn't that a bit odd?
> And even when I modified the sql table to change the hostname in the DB and then do a sofia recover, it didn't recover. I'm not sure how to debug this, I'm just getting "no calls to recover" and I don't hear anything.
> 
> -Avi

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From diego.viola at gmail.com  Mon Jan 17 02:18:12 2011
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 16 Jan 2011 20:18:12 -0300
Subject: [Freeswitch-users] Codecs issue
In-Reply-To: <ED953AB7-108D-452B-B42E-C68FEBA8005F@ipeva.fr>
References: <AANLkTin=U-+LOeJ4JdeL2u3gXapaPdnZDMGk7-kAFAYe@mail.gmail.com>
	<ED953AB7-108D-452B-B42E-C68FEBA8005F@ipeva.fr>
Message-ID: <AANLkTimxTdLD=y4eV84aoc0CQE5hkBvKr0vnPuAa+c5v@mail.gmail.com>

How do you understand these RTP/AVP numbers?

 m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13

What does the numbers mean? I don't see that on the wiki.

Any help appreciated.

On Sun, Jan 16, 2011 at 7:48 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
> That sounds like a bad reading of the trace.
> The important line is the 239th:
> ?m=audio?23344?RTP/AVP?3?18?98?99?9?0?8?101?13
> That means codec 3 is sent first (so GSM), then 18 (so G729).
> I think you should read a little bit on early negotiation.
> If leg A has GSM in 1st position, if your inbound codec list allows GSM, and
> your outbound codec list is : G729, PCM, GSM, than your outbound INVITE to
> leg B will have GSM in 1st position, because that was requested by leg A.
> That's in the wiki, it's a part I rewrote myself some weeks ago, in the most
> readable way I could.
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 16/01/2011 ? 21:59, Diego Viola a ?crit :
>
> Hello,
>
> I'm experiencing some strange issue with codecs. I have the following
> in my vars.xml file:
>
> ?<X-PRE-PROCESS cmd="set"
> data="global_codec_prefs=G729,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
> ?<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
>
> "inbound-late-negotiation" and "disable-transcoding" are commented in
> my internal SIP profile. So I guess I'm in Early Negotiation (default
> behavior) mode.
>
> However, when I send a call to my provider, and I look at the SIP
> trace I see that FS is sending another codec, not G729 as I specified
> in the global_codec_prefs / outbound_codec_prefs parameters.
>
> I'm sending calls like this:
>
> ???????<action application="bridge" data="sofia/internal/$1 at 38.102.93.70"/>
>
> Here is a SIP trace of a call:
>
> http://pastebin.freeswitch.org/15042
>
> I'm not understanding why FS is sending an INVITE with the G7221 codec
> in line 240, if I'm telling it explicitly that I want G729 as the
> priority when possible in the codec prefs options. But I see G729 in
> the 200 OK in line 291.
>
> I've been told to use absolute_codec_string=G729 in my dialplan or
> enable late negotiation, but why if I'm already telling it to use G729
> in the codec prefs?
>
> my softphone IP: 190.23.80.10
> provider IP: 38.102.93.70
> FS IP: 77.92.65.126
>
> calls flow like this:
>
> softphone -> FS -> provider
>
> Any help appreciated.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Diego Viola
Representative of Bridgecom LLC
Phone: +595 971 320 520
GTalk: diego.viola at gmail.com
MSN: diegoev at msn.com
LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609
www.bridgecom.com.py


From steveayre at gmail.com  Mon Jan 17 02:21:50 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 16 Jan 2011 23:21:50 +0000
Subject: [Freeswitch-users] Codecs issue
In-Reply-To: <AANLkTimxTdLD=y4eV84aoc0CQE5hkBvKr0vnPuAa+c5v@mail.gmail.com>
References: <AANLkTin=U-+LOeJ4JdeL2u3gXapaPdnZDMGk7-kAFAYe@mail.gmail.com>
	<ED953AB7-108D-452B-B42E-C68FEBA8005F@ipeva.fr>
	<AANLkTimxTdLD=y4eV84aoc0CQE5hkBvKr0vnPuAa+c5v@mail.gmail.com>
Message-ID: <AANLkTinrhW2WiwiYK+WoVjQi=brKihXE=04UsKvU1WFw@mail.gmail.com>

It's part of the RTP/SDP standards. They're either static or dynamic
numbers.

Dynamic ones (96-127) are named by name in a=rtpmap lines.

Static numbers are reserved numbers and the a=rtpmap is allowed but optional
(although some devices incorrectly require it)
The list of reserved static numbers is:
http://www.iana.org/assignments/rtp-parameters

-Steve


On 16 January 2011 23:18, Diego Viola <diego.viola at gmail.com> wrote:

> How do you understand these RTP/AVP numbers?
>
>  m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13
>
> What does the numbers mean? I don't see that on the wiki.
>
> Any help appreciated.
>
> On Sun, Jan 16, 2011 at 7:48 PM, David Ponzone <david.ponzone at ipeva.fr>
> wrote:
> > That sounds like a bad reading of the trace.
> > The important line is the 239th:
> >  m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13
> > That means codec 3 is sent first (so GSM), then 18 (so G729).
> > I think you should read a little bit on early negotiation.
> > If leg A has GSM in 1st position, if your inbound codec list allows GSM,
> and
> > your outbound codec list is : G729, PCM, GSM, than your outbound INVITE
> to
> > leg B will have GSM in 1st position, because that was requested by leg A.
> > That's in the wiki, it's a part I rewrote myself some weeks ago, in the
> most
> > readable way I could.
> > David Ponzone  Direction Technique
> > email: david.ponzone at ipeva.fr
> > tel:      01 74 03 18 97
> > gsm:   06 66 98 76 34
> > Service Client IPeva
> > tel:      0811 46 26 26
> > www.ipeva.fr  -   www.ipeva-studio.com
> > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> > l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion
> > non autoris?e est interdite. Tout message ?lectronique est susceptible
> > d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il
> > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> >
> >
> >
> > Le 16/01/2011 ? 21:59, Diego Viola a ?crit :
> >
> > Hello,
> >
> > I'm experiencing some strange issue with codecs. I have the following
> > in my vars.xml file:
> >
> >  <X-PRE-PROCESS cmd="set"
> > data="global_codec_prefs=G729,G7221 at 32000h,G7221 at 16000h
> ,G722,PCMU,PCMA,GSM"/>
> >  <X-PRE-PROCESS cmd="set"
> data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
> >
> > "inbound-late-negotiation" and "disable-transcoding" are commented in
> > my internal SIP profile. So I guess I'm in Early Negotiation (default
> > behavior) mode.
> >
> > However, when I send a call to my provider, and I look at the SIP
> > trace I see that FS is sending another codec, not G729 as I specified
> > in the global_codec_prefs / outbound_codec_prefs parameters.
> >
> > I'm sending calls like this:
> >
> >        <action application="bridge" data="sofia/internal/$1 at 38.102.93.70
> "/>
> >
> > Here is a SIP trace of a call:
> >
> > http://pastebin.freeswitch.org/15042
> >
> > I'm not understanding why FS is sending an INVITE with the G7221 codec
> > in line 240, if I'm telling it explicitly that I want G729 as the
> > priority when possible in the codec prefs options. But I see G729 in
> > the 200 OK in line 291.
> >
> > I've been told to use absolute_codec_string=G729 in my dialplan or
> > enable late negotiation, but why if I'm already telling it to use G729
> > in the codec prefs?
> >
> > my softphone IP: 190.23.80.10
> > provider IP: 38.102.93.70
> > FS IP: 77.92.65.126
> >
> > calls flow like this:
> >
> > softphone -> FS -> provider
> >
> > Any help appreciated.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Diego Viola
> Representative of Bridgecom LLC
> Phone: +595 971 320 520
> GTalk: diego.viola at gmail.com
> MSN: diegoev at msn.com
> LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609
> www.bridgecom.com.py
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Mon Jan 17 02:23:29 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 16 Jan 2011 23:23:29 +0000
Subject: [Freeswitch-users] Codecs issue
In-Reply-To: <AANLkTinrhW2WiwiYK+WoVjQi=brKihXE=04UsKvU1WFw@mail.gmail.com>
References: <AANLkTin=U-+LOeJ4JdeL2u3gXapaPdnZDMGk7-kAFAYe@mail.gmail.com>
	<ED953AB7-108D-452B-B42E-C68FEBA8005F@ipeva.fr>
	<AANLkTimxTdLD=y4eV84aoc0CQE5hkBvKr0vnPuAa+c5v@mail.gmail.com>
	<AANLkTinrhW2WiwiYK+WoVjQi=brKihXE=04UsKvU1WFw@mail.gmail.com>
Message-ID: <AANLkTi=-EvKsuc0h_6AhKOi_kn++AhqNbLZw==h8iAVG@mail.gmail.com>

Oh, and it is allowed in the standard to override a static number with a
rtpmap line for a dynamic codec if you run out of numbers in the dynamic
range, but the unassigned numbers should be used before any of the assigned
ones.

-Steve



On 16 January 2011 23:21, Steven Ayre <steveayre at gmail.com> wrote:

> It's part of the RTP/SDP standards. They're either static or dynamic
> numbers.
>
> Dynamic ones (96-127) are named by name in a=rtpmap lines.
>
> Static numbers are reserved numbers and the a=rtpmap is allowed but
> optional (although some devices incorrectly require it)
> The list of reserved static numbers is:
> http://www.iana.org/assignments/rtp-parameters
>
> -Steve
>
>
>
> On 16 January 2011 23:18, Diego Viola <diego.viola at gmail.com> wrote:
>
>> How do you understand these RTP/AVP numbers?
>>
>>  m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13
>>
>> What does the numbers mean? I don't see that on the wiki.
>>
>> Any help appreciated.
>>
>> On Sun, Jan 16, 2011 at 7:48 PM, David Ponzone <david.ponzone at ipeva.fr>
>> wrote:
>> > That sounds like a bad reading of the trace.
>> > The important line is the 239th:
>> >  m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13
>> > That means codec 3 is sent first (so GSM), then 18 (so G729).
>> > I think you should read a little bit on early negotiation.
>> > If leg A has GSM in 1st position, if your inbound codec list allows GSM,
>> and
>> > your outbound codec list is : G729, PCM, GSM, than your outbound INVITE
>> to
>> > leg B will have GSM in 1st position, because that was requested by leg
>> A.
>> > That's in the wiki, it's a part I rewrote myself some weeks ago, in the
>> most
>> > readable way I could.
>> > David Ponzone  Direction Technique
>> > email: david.ponzone at ipeva.fr
>> > tel:      01 74 03 18 97
>> > gsm:   06 66 98 76 34
>> > Service Client IPeva
>> > tel:      0811 46 26 26
>> > www.ipeva.fr  -   www.ipeva-studio.com
>> > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> > l'intention exclusive de ses destinataires. Toute utilisation ou
>> diffusion
>> > non autoris?e est interdite. Tout message ?lectronique est susceptible
>> > d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
>> s'il
>> > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> >
>> >
>> >
>> > Le 16/01/2011 ? 21:59, Diego Viola a ?crit :
>> >
>> > Hello,
>> >
>> > I'm experiencing some strange issue with codecs. I have the following
>> > in my vars.xml file:
>> >
>> >  <X-PRE-PROCESS cmd="set"
>> > data="global_codec_prefs=G729,G7221 at 32000h,G7221 at 16000h
>> ,G722,PCMU,PCMA,GSM"/>
>> >  <X-PRE-PROCESS cmd="set"
>> data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
>> >
>> > "inbound-late-negotiation" and "disable-transcoding" are commented in
>> > my internal SIP profile. So I guess I'm in Early Negotiation (default
>> > behavior) mode.
>> >
>> > However, when I send a call to my provider, and I look at the SIP
>> > trace I see that FS is sending another codec, not G729 as I specified
>> > in the global_codec_prefs / outbound_codec_prefs parameters.
>> >
>> > I'm sending calls like this:
>> >
>> >        <action application="bridge" data="sofia/internal/$
>> 1 at 38.102.93.70"/>
>> >
>> > Here is a SIP trace of a call:
>> >
>> > http://pastebin.freeswitch.org/15042
>> >
>> > I'm not understanding why FS is sending an INVITE with the G7221 codec
>> > in line 240, if I'm telling it explicitly that I want G729 as the
>> > priority when possible in the codec prefs options. But I see G729 in
>> > the 200 OK in line 291.
>> >
>> > I've been told to use absolute_codec_string=G729 in my dialplan or
>> > enable late negotiation, but why if I'm already telling it to use G729
>> > in the codec prefs?
>> >
>> > my softphone IP: 190.23.80.10
>> > provider IP: 38.102.93.70
>> > FS IP: 77.92.65.126
>> >
>> > calls flow like this:
>> >
>> > softphone -> FS -> provider
>> >
>> > Any help appreciated.
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Diego Viola
>> Representative of Bridgecom LLC
>> Phone: +595 971 320 520
>> GTalk: diego.viola at gmail.com
>> MSN: diegoev at msn.com
>> LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609
>> www.bridgecom.com.py
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From diego.viola at gmail.com  Mon Jan 17 02:24:21 2011
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 16 Jan 2011 20:24:21 -0300
Subject: [Freeswitch-users] Codecs issue
In-Reply-To: <AANLkTinrhW2WiwiYK+WoVjQi=brKihXE=04UsKvU1WFw@mail.gmail.com>
References: <AANLkTin=U-+LOeJ4JdeL2u3gXapaPdnZDMGk7-kAFAYe@mail.gmail.com>
	<ED953AB7-108D-452B-B42E-C68FEBA8005F@ipeva.fr>
	<AANLkTimxTdLD=y4eV84aoc0CQE5hkBvKr0vnPuAa+c5v@mail.gmail.com>
	<AANLkTinrhW2WiwiYK+WoVjQi=brKihXE=04UsKvU1WFw@mail.gmail.com>
Message-ID: <AANLkTin_oDSut1vt6iB1iHtDwOBRnko39HEMOdM2K7MT@mail.gmail.com>

Thanks, I will read that

On Sun, Jan 16, 2011 at 8:21 PM, Steven Ayre <steveayre at gmail.com> wrote:
> It's part of the RTP/SDP standards. They're either static or dynamic
> numbers.
>
> Dynamic ones (96-127) are named by name in a=rtpmap lines.
>
> Static numbers are reserved numbers and the a=rtpmap is allowed but optional
> (although some devices incorrectly require it)
> The list of reserved static numbers is:
> http://www.iana.org/assignments/rtp-parameters
>
> -Steve
>
>
> On 16 January 2011 23:18, Diego Viola <diego.viola at gmail.com> wrote:
>>
>> How do you understand these RTP/AVP numbers?
>>
>> ?m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13
>>
>> What does the numbers mean? I don't see that on the wiki.
>>
>> Any help appreciated.
>>
>> On Sun, Jan 16, 2011 at 7:48 PM, David Ponzone <david.ponzone at ipeva.fr>
>> wrote:
>> > That sounds like a bad reading of the trace.
>> > The important line is the 239th:
>> > ?m=audio?23344?RTP/AVP?3?18?98?99?9?0?8?101?13
>> > That means codec 3 is sent first (so GSM), then 18 (so G729).
>> > I think you should read a little bit on early negotiation.
>> > If leg A has GSM in 1st position, if your inbound codec list allows GSM,
>> > and
>> > your outbound codec list is : G729, PCM, GSM, than your outbound INVITE
>> > to
>> > leg B will have GSM in 1st position, because that was requested by leg
>> > A.
>> > That's in the wiki, it's a part I rewrote myself some weeks ago, in the
>> > most
>> > readable way I could.
>> > David Ponzone ?Direction Technique
>> > email: david.ponzone at ipeva.fr
>> > tel: ? ? ?01 74 03 18 97
>> > gsm: ? 06 66 98 76 34
>> > Service Client?IPeva
>> > tel: ? ? ?0811 46 26 26
>> > www.ipeva.fr? -? ?www.ipeva-studio.com
>> > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> > l'intention exclusive de ses destinataires. Toute utilisation ou
>> > diffusion
>> > non autoris?e est interdite. Tout message ?lectronique est susceptible
>> > d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message
>> > s'il
>> > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> >
>> >
>> >
>> > Le 16/01/2011 ? 21:59, Diego Viola a ?crit :
>> >
>> > Hello,
>> >
>> > I'm experiencing some strange issue with codecs. I have the following
>> > in my vars.xml file:
>> >
>> > ?<X-PRE-PROCESS cmd="set"
>> >
>> > data="global_codec_prefs=G729,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>> > ?<X-PRE-PROCESS cmd="set"
>> > data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
>> >
>> > "inbound-late-negotiation" and "disable-transcoding" are commented in
>> > my internal SIP profile. So I guess I'm in Early Negotiation (default
>> > behavior) mode.
>> >
>> > However, when I send a call to my provider, and I look at the SIP
>> > trace I see that FS is sending another codec, not G729 as I specified
>> > in the global_codec_prefs / outbound_codec_prefs parameters.
>> >
>> > I'm sending calls like this:
>> >
>> > ???????<action application="bridge"
>> > data="sofia/internal/$1 at 38.102.93.70"/>
>> >
>> > Here is a SIP trace of a call:
>> >
>> > http://pastebin.freeswitch.org/15042
>> >
>> > I'm not understanding why FS is sending an INVITE with the G7221 codec
>> > in line 240, if I'm telling it explicitly that I want G729 as the
>> > priority when possible in the codec prefs options. But I see G729 in
>> > the 200 OK in line 291.
>> >
>> > I've been told to use absolute_codec_string=G729 in my dialplan or
>> > enable late negotiation, but why if I'm already telling it to use G729
>> > in the codec prefs?
>> >
>> > my softphone IP: 190.23.80.10
>> > provider IP: 38.102.93.70
>> > FS IP: 77.92.65.126
>> >
>> > calls flow like this:
>> >
>> > softphone -> FS -> provider
>> >
>> > Any help appreciated.
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Diego Viola
>> Representative of Bridgecom LLC
>> Phone: +595 971 320 520
>> GTalk: diego.viola at gmail.com
>> MSN: diegoev at msn.com
>> LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609
>> www.bridgecom.com.py
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Diego Viola
Representative of Bridgecom LLC
Phone: +595 971 320 520
GTalk: diego.viola at gmail.com
MSN: diegoev at msn.com
LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609
www.bridgecom.com.py


From diego.viola at gmail.com  Mon Jan 17 02:25:20 2011
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 16 Jan 2011 20:25:20 -0300
Subject: [Freeswitch-users] Codecs issue
In-Reply-To: <AANLkTin_oDSut1vt6iB1iHtDwOBRnko39HEMOdM2K7MT@mail.gmail.com>
References: <AANLkTin=U-+LOeJ4JdeL2u3gXapaPdnZDMGk7-kAFAYe@mail.gmail.com>
	<ED953AB7-108D-452B-B42E-C68FEBA8005F@ipeva.fr>
	<AANLkTimxTdLD=y4eV84aoc0CQE5hkBvKr0vnPuAa+c5v@mail.gmail.com>
	<AANLkTinrhW2WiwiYK+WoVjQi=brKihXE=04UsKvU1WFw@mail.gmail.com>
	<AANLkTin_oDSut1vt6iB1iHtDwOBRnko39HEMOdM2K7MT@mail.gmail.com>
Message-ID: <AANLkTimRD3VpMGf6ZsEmxfjx9NaCGxs-D4Ws8=EYwHwp@mail.gmail.com>

Also, I noticed this on the wiki, in the Early Negotiation example.

A -------- GSM/PCMA/G729 --------> FS (allowing G729/PCMA/PCMU)
-------- PCMA/G729/PCMU --------> B

Shouldn't that be:

A -------- GSM/PCMA/G729 --------> FS (allowing G729/PCMA/PCMU)
-------- G729/PCMA/PCMU --------> B

instead?

On Sun, Jan 16, 2011 at 8:24 PM, Diego Viola <diego.viola at gmail.com> wrote:
> Thanks, I will read that
>
> On Sun, Jan 16, 2011 at 8:21 PM, Steven Ayre <steveayre at gmail.com> wrote:
>> It's part of the RTP/SDP standards. They're either static or dynamic
>> numbers.
>>
>> Dynamic ones (96-127) are named by name in a=rtpmap lines.
>>
>> Static numbers are reserved numbers and the a=rtpmap is allowed but optional
>> (although some devices incorrectly require it)
>> The list of reserved static numbers is:
>> http://www.iana.org/assignments/rtp-parameters
>>
>> -Steve
>>
>>
>> On 16 January 2011 23:18, Diego Viola <diego.viola at gmail.com> wrote:
>>>
>>> How do you understand these RTP/AVP numbers?
>>>
>>> ?m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13
>>>
>>> What does the numbers mean? I don't see that on the wiki.
>>>
>>> Any help appreciated.
>>>
>>> On Sun, Jan 16, 2011 at 7:48 PM, David Ponzone <david.ponzone at ipeva.fr>
>>> wrote:
>>> > That sounds like a bad reading of the trace.
>>> > The important line is the 239th:
>>> > ?m=audio?23344?RTP/AVP?3?18?98?99?9?0?8?101?13
>>> > That means codec 3 is sent first (so GSM), then 18 (so G729).
>>> > I think you should read a little bit on early negotiation.
>>> > If leg A has GSM in 1st position, if your inbound codec list allows GSM,
>>> > and
>>> > your outbound codec list is : G729, PCM, GSM, than your outbound INVITE
>>> > to
>>> > leg B will have GSM in 1st position, because that was requested by leg
>>> > A.
>>> > That's in the wiki, it's a part I rewrote myself some weeks ago, in the
>>> > most
>>> > readable way I could.
>>> > David Ponzone ?Direction Technique
>>> > email: david.ponzone at ipeva.fr
>>> > tel: ? ? ?01 74 03 18 97
>>> > gsm: ? 06 66 98 76 34
>>> > Service Client?IPeva
>>> > tel: ? ? ?0811 46 26 26
>>> > www.ipeva.fr? -? ?www.ipeva-studio.com
>>> > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> > l'intention exclusive de ses destinataires. Toute utilisation ou
>>> > diffusion
>>> > non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> > d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message
>>> > s'il
>>> > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>> > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>> >
>>> >
>>> >
>>> > Le 16/01/2011 ? 21:59, Diego Viola a ?crit :
>>> >
>>> > Hello,
>>> >
>>> > I'm experiencing some strange issue with codecs. I have the following
>>> > in my vars.xml file:
>>> >
>>> > ?<X-PRE-PROCESS cmd="set"
>>> >
>>> > data="global_codec_prefs=G729,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>>> > ?<X-PRE-PROCESS cmd="set"
>>> > data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
>>> >
>>> > "inbound-late-negotiation" and "disable-transcoding" are commented in
>>> > my internal SIP profile. So I guess I'm in Early Negotiation (default
>>> > behavior) mode.
>>> >
>>> > However, when I send a call to my provider, and I look at the SIP
>>> > trace I see that FS is sending another codec, not G729 as I specified
>>> > in the global_codec_prefs / outbound_codec_prefs parameters.
>>> >
>>> > I'm sending calls like this:
>>> >
>>> > ???????<action application="bridge"
>>> > data="sofia/internal/$1 at 38.102.93.70"/>
>>> >
>>> > Here is a SIP trace of a call:
>>> >
>>> > http://pastebin.freeswitch.org/15042
>>> >
>>> > I'm not understanding why FS is sending an INVITE with the G7221 codec
>>> > in line 240, if I'm telling it explicitly that I want G729 as the
>>> > priority when possible in the codec prefs options. But I see G729 in
>>> > the 200 OK in line 291.
>>> >
>>> > I've been told to use absolute_codec_string=G729 in my dialplan or
>>> > enable late negotiation, but why if I'm already telling it to use G729
>>> > in the codec prefs?
>>> >
>>> > my softphone IP: 190.23.80.10
>>> > provider IP: 38.102.93.70
>>> > FS IP: 77.92.65.126
>>> >
>>> > calls flow like this:
>>> >
>>> > softphone -> FS -> provider
>>> >
>>> > Any help appreciated.
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Diego Viola
>>> Representative of Bridgecom LLC
>>> Phone: +595 971 320 520
>>> GTalk: diego.viola at gmail.com
>>> MSN: diegoev at msn.com
>>> LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609
>>> www.bridgecom.com.py
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Diego Viola
> Representative of Bridgecom LLC
> Phone: +595 971 320 520
> GTalk: diego.viola at gmail.com
> MSN: diegoev at msn.com
> LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609
> www.bridgecom.com.py
>



-- 
Diego Viola
Representative of Bridgecom LLC
Phone: +595 971 320 520
GTalk: diego.viola at gmail.com
MSN: diegoev at msn.com
LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609
www.bridgecom.com.py


From diego.viola at gmail.com  Mon Jan 17 02:25:32 2011
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 16 Jan 2011 20:25:32 -0300
Subject: [Freeswitch-users] Codecs issue
In-Reply-To: <AANLkTimRD3VpMGf6ZsEmxfjx9NaCGxs-D4Ws8=EYwHwp@mail.gmail.com>
References: <AANLkTin=U-+LOeJ4JdeL2u3gXapaPdnZDMGk7-kAFAYe@mail.gmail.com>
	<ED953AB7-108D-452B-B42E-C68FEBA8005F@ipeva.fr>
	<AANLkTimxTdLD=y4eV84aoc0CQE5hkBvKr0vnPuAa+c5v@mail.gmail.com>
	<AANLkTinrhW2WiwiYK+WoVjQi=brKihXE=04UsKvU1WFw@mail.gmail.com>
	<AANLkTin_oDSut1vt6iB1iHtDwOBRnko39HEMOdM2K7MT@mail.gmail.com>
	<AANLkTimRD3VpMGf6ZsEmxfjx9NaCGxs-D4Ws8=EYwHwp@mail.gmail.com>
Message-ID: <AANLkTimMZN3SPanUbFneBNgjHUNG2yMYd9n3egienbLt@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Codec_Negotiation

On Sun, Jan 16, 2011 at 8:25 PM, Diego Viola <diego.viola at gmail.com> wrote:
> Also, I noticed this on the wiki, in the Early Negotiation example.
>
> A -------- GSM/PCMA/G729 --------> FS (allowing G729/PCMA/PCMU)
> -------- PCMA/G729/PCMU --------> B
>
> Shouldn't that be:
>
> A -------- GSM/PCMA/G729 --------> FS (allowing G729/PCMA/PCMU)
> -------- G729/PCMA/PCMU --------> B
>
> instead?
>
> On Sun, Jan 16, 2011 at 8:24 PM, Diego Viola <diego.viola at gmail.com> wrote:
>> Thanks, I will read that
>>
>> On Sun, Jan 16, 2011 at 8:21 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>> It's part of the RTP/SDP standards. They're either static or dynamic
>>> numbers.
>>>
>>> Dynamic ones (96-127) are named by name in a=rtpmap lines.
>>>
>>> Static numbers are reserved numbers and the a=rtpmap is allowed but optional
>>> (although some devices incorrectly require it)
>>> The list of reserved static numbers is:
>>> http://www.iana.org/assignments/rtp-parameters
>>>
>>> -Steve
>>>
>>>
>>> On 16 January 2011 23:18, Diego Viola <diego.viola at gmail.com> wrote:
>>>>
>>>> How do you understand these RTP/AVP numbers?
>>>>
>>>> ?m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13
>>>>
>>>> What does the numbers mean? I don't see that on the wiki.
>>>>
>>>> Any help appreciated.
>>>>
>>>> On Sun, Jan 16, 2011 at 7:48 PM, David Ponzone <david.ponzone at ipeva.fr>
>>>> wrote:
>>>> > That sounds like a bad reading of the trace.
>>>> > The important line is the 239th:
>>>> > ?m=audio?23344?RTP/AVP?3?18?98?99?9?0?8?101?13
>>>> > That means codec 3 is sent first (so GSM), then 18 (so G729).
>>>> > I think you should read a little bit on early negotiation.
>>>> > If leg A has GSM in 1st position, if your inbound codec list allows GSM,
>>>> > and
>>>> > your outbound codec list is : G729, PCM, GSM, than your outbound INVITE
>>>> > to
>>>> > leg B will have GSM in 1st position, because that was requested by leg
>>>> > A.
>>>> > That's in the wiki, it's a part I rewrote myself some weeks ago, in the
>>>> > most
>>>> > readable way I could.
>>>> > David Ponzone ?Direction Technique
>>>> > email: david.ponzone at ipeva.fr
>>>> > tel: ? ? ?01 74 03 18 97
>>>> > gsm: ? 06 66 98 76 34
>>>> > Service Client?IPeva
>>>> > tel: ? ? ?0811 46 26 26
>>>> > www.ipeva.fr? -? ?www.ipeva-studio.com
>>>> > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>> > l'intention exclusive de ses destinataires. Toute utilisation ou
>>>> > diffusion
>>>> > non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>> > d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message
>>>> > s'il
>>>> > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>>> > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>> >
>>>> >
>>>> >
>>>> > Le 16/01/2011 ? 21:59, Diego Viola a ?crit :
>>>> >
>>>> > Hello,
>>>> >
>>>> > I'm experiencing some strange issue with codecs. I have the following
>>>> > in my vars.xml file:
>>>> >
>>>> > ?<X-PRE-PROCESS cmd="set"
>>>> >
>>>> > data="global_codec_prefs=G729,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>>>> > ?<X-PRE-PROCESS cmd="set"
>>>> > data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
>>>> >
>>>> > "inbound-late-negotiation" and "disable-transcoding" are commented in
>>>> > my internal SIP profile. So I guess I'm in Early Negotiation (default
>>>> > behavior) mode.
>>>> >
>>>> > However, when I send a call to my provider, and I look at the SIP
>>>> > trace I see that FS is sending another codec, not G729 as I specified
>>>> > in the global_codec_prefs / outbound_codec_prefs parameters.
>>>> >
>>>> > I'm sending calls like this:
>>>> >
>>>> > ???????<action application="bridge"
>>>> > data="sofia/internal/$1 at 38.102.93.70"/>
>>>> >
>>>> > Here is a SIP trace of a call:
>>>> >
>>>> > http://pastebin.freeswitch.org/15042
>>>> >
>>>> > I'm not understanding why FS is sending an INVITE with the G7221 codec
>>>> > in line 240, if I'm telling it explicitly that I want G729 as the
>>>> > priority when possible in the codec prefs options. But I see G729 in
>>>> > the 200 OK in line 291.
>>>> >
>>>> > I've been told to use absolute_codec_string=G729 in my dialplan or
>>>> > enable late negotiation, but why if I'm already telling it to use G729
>>>> > in the codec prefs?
>>>> >
>>>> > my softphone IP: 190.23.80.10
>>>> > provider IP: 38.102.93.70
>>>> > FS IP: 77.92.65.126
>>>> >
>>>> > calls flow like this:
>>>> >
>>>> > softphone -> FS -> provider
>>>> >
>>>> > Any help appreciated.
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>>
>>>>
>>>> --
>>>> Diego Viola
>>>> Representative of Bridgecom LLC
>>>> Phone: +595 971 320 520
>>>> GTalk: diego.viola at gmail.com
>>>> MSN: diegoev at msn.com
>>>> LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609
>>>> www.bridgecom.com.py
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> Diego Viola
>> Representative of Bridgecom LLC
>> Phone: +595 971 320 520
>> GTalk: diego.viola at gmail.com
>> MSN: diegoev at msn.com
>> LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609
>> www.bridgecom.com.py
>>
>
>
>
> --
> Diego Viola
> Representative of Bridgecom LLC
> Phone: +595 971 320 520
> GTalk: diego.viola at gmail.com
> MSN: diegoev at msn.com
> LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609
> www.bridgecom.com.py
>



-- 
Diego Viola
Representative of Bridgecom LLC
Phone: +595 971 320 520
GTalk: diego.viola at gmail.com
MSN: diegoev at msn.com
LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609
www.bridgecom.com.py


From cmrienzo at gmail.com  Mon Jan 17 03:12:30 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Sun, 16 Jan 2011 19:12:30 -0500
Subject: [Freeswitch-users] Removing inband DTMF
In-Reply-To: <AANLkTi=RGqABGRvnEwKVug5XFPPfyYhN2nMdrekkdT22@mail.gmail.com>
References: <AANLkTikBZPbeKLO_fX=s4p1pZxPCzwcAet1win8eaUMn@mail.gmail.com>
	<AANLkTimYd8PZumEY1GzkcAj-kZ1auH8Lvr-7OBm7r3fB@mail.gmail.com>
	<AANLkTimgMxuC=FEQ0UFerTLE1wRCKUx5L1Kr-_KMWJ3b@mail.gmail.com>
	<AANLkTi=BQGJJrVgyPKxX2aP3kq_xWRHDzzsf1-qQd9X3@mail.gmail.com>
	<A7150993-F25F-474D-A6A9-5EFA84C7ACDB@gmail.com>
	<AANLkTi=RGqABGRvnEwKVug5XFPPfyYhN2nMdrekkdT22@mail.gmail.com>
Message-ID: <AANLkTin1EPe6ym-52cApR6DR8GmeQynHkKypZjy_UtJ1@mail.gmail.com>

Yes, that's true.  I usually see 10-30ms of dtmf followed by RFC2833
packets.


On Sun, Jan 16, 2011 at 4:47 PM, Steven Ayre <steveayre at gmail.com> wrote:

> I didn't mean a significant amount of time. 40ms with a ptime of 20ms means
> you've already sent a packet containing inbound dtmf.
>
> -Steve
>
>
>
>
> On 16 January 2011 20:54, Chris Rienzo <cmrienzo at gmail.com> wrote:
>
>> That's not true.  A detector can spot the start of dtmf in less than 40ms.
>>  The audio frames can then be replaced with 2833 packets.  This is how it
>> works on media gateways.  I'm sure FS could be patched to do this if it uses
>> the spandsp dtmf detector, which can detect dtmf duration.
>>
>> On Jan 16, 2011, at 11:50, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> First of, AFAIK there's nothing in FS that can currently do it.
>>
>> I'm not even sure it's possible at all. You can subtract a noise from
>> audio, but the noisy nature of an encoded signal would mean it wouldn't work
>> well (you'd still hear something), and DTMF can't be detected until some of
>> it will have already been sent to the other endpoint so you'd still have the
>> start of each keypress heard.  For the same reason at the end until the
>> detector spots the dtmf has ended you'll still be subtracting from the audio
>> which will actually generate noise. The only way to avoid that may be to
>> introduce a delay but that's not something FS can do.
>>
>> -Steve
>>
>>
>>
>> On 16 January 2011 15:33, Tamas Jalsovszky < <jalsot at gmail.com>
>> jalsot at gmail.com> wrote:
>>
>>> Hello,
>>>
>>> Thanks.
>>> Yeah, I know that command, as far as I know, it is only for detection, it
>>> does not change the media.
>>> Anybody else?
>>>
>>> Regards,
>>>    T.
>>>
>>>
>>> On Fri, Jan 14, 2011 at 11:23 AM, Avi Marcus < <Avi at amarcus.com>
>>> Avi at amarcus.com> wrote:
>>>
>>>> <http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf>
>>>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf
>>>> start_dtmf does detection from the inband. I'm not sure that it removes
>>>> them from the rtp, though...
>>>> -Avi
>>>>
>>>>
>>>> On Fri, Jan 14, 2011 at 12:14 PM, Tamas Jalsovszky < <jalsot at gmail.com>
>>>> jalsot at gmail.com> wrote:
>>>> > Hello,
>>>> >
>>>> > Is there a way to remove an inband DTMF signal from the RTP stream
>>>> with
>>>> > FreeSWITCH?
>>>> > We have a partner with ugly Cirpack which can not remove inband DTMF
>>>> when
>>>> > the codec is g711. The problem is, that our other party wants RFC2833
>>>> only -
>>>> > thus we have to remove inband dtmf. Unfortunately Cirpack is not open
>>>> to do
>>>> > this.
>>>> >
>>>> > Any advice?
>>>> >
>>>> > Kind regards,
>>>> >    T.
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From avi at avimarcus.net  Mon Jan 17 04:05:21 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 17 Jan 2011 03:05:21 +0200
Subject: [Freeswitch-users] IP Failover with Sofia Recover - not
	recovering?!
In-Reply-To: <AANLkTikwvmHjmMVZ7OeQiGX_EXh9Kwty97Dc5Q=9j2NB@mail.gmail.com>
References: <mailman.1927.1295137414.24555.freeswitch-users@lists.freeswitch.org>
	<AANLkTinA9f19XPq6d+TzeXt3iE7PW500yBxPGAS+R3GH@mail.gmail.com>
	<AANLkTi=FOkWq9pRFqUhGN-gUVjvajGH1iSxU9u1nu+X-@mail.gmail.com>
	<388C53B0-D1D6-439B-A3B8-1616323766AE@freeswitch.org>
	<AANLkTikARk4teUNpc2JySRbUVsbp+SRbCaQczfT1EnfJ@mail.gmail.com>
	<AANLkTikwvmHjmMVZ7OeQiGX_EXh9Kwty97Dc5Q=9j2NB@mail.gmail.com>
Message-ID: <AANLkTik5a8xQVmq2FTJGY6s1okcM-TKc3-1GqkY7LT1M@mail.gmail.com>

Ah my problem was:
I didn't specify the odbc on internal/external profile page, so there was no
sip_recovery table. I thought specifying it for the core db was enough.

-Avi

On Mon, Jan 17, 2011 at 12:55 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Well, partly that hostname variable predates all the sofia recover stuff so
> there was no reason for it originally.
>
> -Steve
>
>
>
>
> On 16 January 2011 22:45, Avi Marcus <avi at avimarcus.net> wrote:
>
>> I'm supposed to have two machines with the same hostname - it's not just a
>> variable in the xml files..? Isn't that a bit odd?
>> And even when I modified the sql table to change the hostname in the DB
>> and then do a sofia recover, it didn't recover. I'm not sure how to debug
>> this, I'm just getting "no calls to recover" and I don't hear anything.
>>
>> -Avi
>>
>>
>> On Mon, Jan 17, 2011 at 12:29 AM, Brian West <brian at freeswitch.org>wrote:
>>
>>> Hostnames would need to be the same on both.
>>>
>>> /b
>>>
>>> On Jan 16, 2011, at 3:18 AM, Steven Ayre wrote:
>>>
>>> > I believe that hostname is the hostname of the server and set on FS
>>> startup.
>>> >
>>> > -Steve
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Mon Jan 17 04:31:08 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 17 Jan 2011 02:31:08 +0100
Subject: [Freeswitch-users] Codecs issue
In-Reply-To: <AANLkTimRD3VpMGf6ZsEmxfjx9NaCGxs-D4Ws8=EYwHwp@mail.gmail.com>
References: <AANLkTin=U-+LOeJ4JdeL2u3gXapaPdnZDMGk7-kAFAYe@mail.gmail.com>
	<ED953AB7-108D-452B-B42E-C68FEBA8005F@ipeva.fr>
	<AANLkTimxTdLD=y4eV84aoc0CQE5hkBvKr0vnPuAa+c5v@mail.gmail.com>
	<AANLkTinrhW2WiwiYK+WoVjQi=brKihXE=04UsKvU1WFw@mail.gmail.com>
	<AANLkTin_oDSut1vt6iB1iHtDwOBRnko39HEMOdM2K7MT@mail.gmail.com>
	<AANLkTimRD3VpMGf6ZsEmxfjx9NaCGxs-D4Ws8=EYwHwp@mail.gmail.com>
Message-ID: <30812159-A039-48F4-AD41-8AFD0CE8DC3F@ipeva.fr>

Again, read the wiki.
After this example, I wrote:
-FS checks the proposed codecs in order of priority (as listed in the SDP) against its list of allowed codecs (as configured in inbound-codec-prefs), and selects PCMA as the first authorized codec, so the codecs list becomes: PCMA/G729/PCMU.

Basically, that means, that the codec negotiated with A will always be put in the first position of the codecs list sent to B.
At least, that's what I observed during my numerous tests at that time.

I really recommend you do your own home-work in order to understand it.
It really helped me to get it.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 17/01/2011 ? 00:25, Diego Viola a ?crit :

> Also, I noticed this on the wiki, in the Early Negotiation example.
> 
> A -------- GSM/PCMA/G729 --------> FS (allowing G729/PCMA/PCMU)
> -------- PCMA/G729/PCMU --------> B
> 
> Shouldn't that be:
> 
> A -------- GSM/PCMA/G729 --------> FS (allowing G729/PCMA/PCMU)
> -------- G729/PCMA/PCMU --------> B
> 
> instead?
> 
> On Sun, Jan 16, 2011 at 8:24 PM, Diego Viola <diego.viola at gmail.com> wrote:
>> Thanks, I will read that
>> 
>> On Sun, Jan 16, 2011 at 8:21 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>> It's part of the RTP/SDP standards. They're either static or dynamic
>>> numbers.
>>> 
>>> Dynamic ones (96-127) are named by name in a=rtpmap lines.
>>> 
>>> Static numbers are reserved numbers and the a=rtpmap is allowed but optional
>>> (although some devices incorrectly require it)
>>> The list of reserved static numbers is:
>>> http://www.iana.org/assignments/rtp-parameters
>>> 
>>> -Steve
>>> 
>>> 
>>> On 16 January 2011 23:18, Diego Viola <diego.viola at gmail.com> wrote:
>>>> 
>>>> How do you understand these RTP/AVP numbers?
>>>> 
>>>>  m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13
>>>> 
>>>> What does the numbers mean? I don't see that on the wiki.
>>>> 
>>>> Any help appreciated.
>>>> 
>>>> On Sun, Jan 16, 2011 at 7:48 PM, David Ponzone <david.ponzone at ipeva.fr>
>>>> wrote:
>>>>> That sounds like a bad reading of the trace.
>>>>> The important line is the 239th:
>>>>>  m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13
>>>>> That means codec 3 is sent first (so GSM), then 18 (so G729).
>>>>> I think you should read a little bit on early negotiation.
>>>>> If leg A has GSM in 1st position, if your inbound codec list allows GSM,
>>>>> and
>>>>> your outbound codec list is : G729, PCM, GSM, than your outbound INVITE
>>>>> to
>>>>> leg B will have GSM in 1st position, because that was requested by leg
>>>>> A.
>>>>> That's in the wiki, it's a part I rewrote myself some weeks ago, in the
>>>>> most
>>>>> readable way I could.
>>>>> David Ponzone  Direction Technique
>>>>> email: david.ponzone at ipeva.fr
>>>>> tel:      01 74 03 18 97
>>>>> gsm:   06 66 98 76 34
>>>>> Service Client IPeva
>>>>> tel:      0811 46 26 26
>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou
>>>>> diffusion
>>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
>>>>> s'il
>>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>> 
>>>>> 
>>>>> 
>>>>> Le 16/01/2011 ? 21:59, Diego Viola a ?crit :
>>>>> 
>>>>> Hello,
>>>>> 
>>>>> I'm experiencing some strange issue with codecs. I have the following
>>>>> in my vars.xml file:
>>>>> 
>>>>>  <X-PRE-PROCESS cmd="set"
>>>>> 
>>>>> data="global_codec_prefs=G729,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>>>>>  <X-PRE-PROCESS cmd="set"
>>>>> data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
>>>>> 
>>>>> "inbound-late-negotiation" and "disable-transcoding" are commented in
>>>>> my internal SIP profile. So I guess I'm in Early Negotiation (default
>>>>> behavior) mode.
>>>>> 
>>>>> However, when I send a call to my provider, and I look at the SIP
>>>>> trace I see that FS is sending another codec, not G729 as I specified
>>>>> in the global_codec_prefs / outbound_codec_prefs parameters.
>>>>> 
>>>>> I'm sending calls like this:
>>>>> 
>>>>>        <action application="bridge"
>>>>> data="sofia/internal/$1 at 38.102.93.70"/>
>>>>> 
>>>>> Here is a SIP trace of a call:
>>>>> 
>>>>> http://pastebin.freeswitch.org/15042
>>>>> 
>>>>> I'm not understanding why FS is sending an INVITE with the G7221 codec
>>>>> in line 240, if I'm telling it explicitly that I want G729 as the
>>>>> priority when possible in the codec prefs options. But I see G729 in
>>>>> the 200 OK in line 291.
>>>>> 
>>>>> I've been told to use absolute_codec_string=G729 in my dialplan or
>>>>> enable late negotiation, but why if I'm already telling it to use G729
>>>>> in the codec prefs?
>>>>> 
>>>>> my softphone IP: 190.23.80.10
>>>>> provider IP: 38.102.93.70
>>>>> FS IP: 77.92.65.126
>>>>> 
>>>>> calls flow like this:
>>>>> 
>>>>> softphone -> FS -> provider
>>>>> 
>>>>> Any help appreciated.
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>> 
>>>> 
>>>> 
>>>> --
>>>> Diego Viola
>>>> Representative of Bridgecom LLC
>>>> Phone: +595 971 320 520
>>>> GTalk: diego.viola at gmail.com
>>>> MSN: diegoev at msn.com
>>>> LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609
>>>> www.bridgecom.com.py
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>> 
>> 
>> 
>> --
>> Diego Viola
>> Representative of Bridgecom LLC
>> Phone: +595 971 320 520
>> GTalk: diego.viola at gmail.com
>> MSN: diegoev at msn.com
>> LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609
>> www.bridgecom.com.py
>> 
> 
> 
> 
> -- 
> Diego Viola
> Representative of Bridgecom LLC
> Phone: +595 971 320 520
> GTalk: diego.viola at gmail.com
> MSN: diegoev at msn.com
> LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609
> www.bridgecom.com.py
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mrene_lists at avgs.ca  Mon Jan 17 06:12:36 2011
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sun, 16 Jan 2011 22:12:36 -0500
Subject: [Freeswitch-users] Processing Live Audio during a call between
	2 extensions
In-Reply-To: <AANLkTikTMazyD_Q9FOZj=Ky+sC+JAK4aTKE+cV6NgEWf@mail.gmail.com>
References: <AANLkTikTMazyD_Q9FOZj=Ky+sC+JAK4aTKE+cV6NgEWf@mail.gmail.com>
Message-ID: <5AD8856E-3FF6-4EE3-9C6C-A68FAA6D4371@avgs.ca>

Hi,

Your best way is to make a freeswitch module and attach a media bug to the channel, you can take a look at mod_soundtouch's source, which sets one up (using switch_core_media_bug_add()) and changes the audio.
 
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2011-01-14, at 11:44 AM, Michel Habib wrote:

> Dear Freeswitch Developers, i am relatively new to freeswitch and i am seeking some advice.
> 
> I am creating an External Application that does some [live] processing on the Audio of the freeswitch call before sending it to the second Call Leg, and vice versa.
> Can you please direct me to the best way to do that and how to access/capture the live audio channel in both legs while it is streaming, and how to resubmit it again to the other leg?
> I read the book about freeswitch, but still couldnt figure the best approach to do that, any hint is appreciated.
> 
> Best Regards,
> Michel Habib.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From infos at madovsky.org  Mon Jan 17 07:43:02 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 16 Jan 2011 23:43:02 -0500
Subject: [Freeswitch-users] get cepstral volume lower
Message-ID: <854F4E8A03274B099B3DD8819DE0721B@e1705>

I'm trying the SSML example from mod_cepstral wiki

<action application="speak" data="cepstral|david|<prosody volume=13>Thank you for holding</prosody>"/>

but the voice doesn't say the sentence but says something like "slash blablabla...".

Any idea why I can't embed SSML tag in the cepstral sentence ?

Thanks
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From ayhkor at gmail.com  Mon Jan 17 08:45:07 2011
From: ayhkor at gmail.com (deniro)
Date: Mon, 17 Jan 2011 00:45:07 -0500
Subject: [Freeswitch-users] read conference PIN from a file or db
In-Reply-To: <AANLkTinMEVy55S1kqyKq5RVxBJdmd3pELr5fnezs7dGR@mail.gmail.com>
References: <AANLkTik3goWsqBVSjXBSD8HJLPHV4FWUFyVPsJXsycdh@mail.gmail.com>
	<AANLkTi=PFQTZbeNCm7fjE9LrZNFi0V+idB5ypjQeoY=S@mail.gmail.com>
	<AANLkTi=oV+hN0E12A=St7faH+65U+nCZBHKKb444LA1D@mail.gmail.com>
	<AANLkTinMEVy55S1kqyKq5RVxBJdmd3pELr5fnezs7dGR@mail.gmail.com>
Message-ID: <AANLkTin2TNtXkGGLYAyOd-Wse+FADKCC5mdStAb=yzFA@mail.gmail.com>

Thanks Steve
I looked at these links

more specifically what I look for is with the dialplan+javascript;
How can I set the logic in a dialplan in such a way that I enter the
conference PIN, and pass it as a parameter to javascript and compare with
the values(set of pin numbers) in database and when there is a match, will
let you go to conference?

I am able to read the pin number and assign it to $pin.
In a dialplan, I wanna set a condition or something, call javascript, pass
$pin as parameter to javascript, compare it with the values in db if there
is a match, allow you to go to conference.
 So, how can I set this logic in dialplan with javascript?
any working example?

thanks



On Sun, Jan 16, 2011 at 5:58 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Sure you can - mod_spidermonkey gives you javascript, if you configure
> spidermonkey.conf.xml to load mod_spidermonkey_odbc you'll get odbc access
> so you can access a mysql database.
>
> http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc
> http://wiki.freeswitch.org/wiki/Session_setVariable
>
> -Steve
>
>
>
>
> On 16 January 2011 22:45, deniro <ayhkor at gmail.com> wrote:
>
>> how about writing a javascript and reading from mysql database
>> when a match is hit return a value and  load it into $pin
>> how can I do this type of javascripting?
>> thx
>>
>>
>>
>> On Sun, Jan 16, 2011 at 3:49 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> There's several ways, you can write your own module, a script, use xml
>>> curl etc.
>>>
>>> The simplest option is probably to use lua - you can access odbc if you
>>> install luasql. I'm sure file access is possible too.
>>>
>>> -Steve
>>>
>>>
>>>   On 16 January 2011 19:46, deniro <ayhkor at gmail.com> wrote:
>>>
>>>>   Hi
>>>> Regarding providing PIN numbers to a conference
>>>> how can I read PINs from a file or database and allow them to go to a
>>>> meeting
>>>> thx
>>>> deniro--
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From hwnorman at hotmail.com  Mon Jan 17 09:14:12 2011
From: hwnorman at hotmail.com (Norman Lam)
Date: Mon, 17 Jan 2011 14:14:12 +0800
Subject: [Freeswitch-users] Iksemel msvs compiling
In-Reply-To: <1295113990176-5925152.post@n2.nabble.com>
References: <SNT109-DS13313A1FA906527438E8CC1090@phx.gbl>	<SNT109-DS1155D401A0B9742CF63E13C10B0@phx.gbl>	<1294375438447-5898181.post@n2.nabble.com>	<SNT109-DS11CB0ECE5BD0F9C0384E03C10C0@phx.gbl>	<1294535404901-5903534.post@n2.nabble.com>	<SNT109-DS14B90C48B33126784F294AC10E0@phx.gbl>	<1294673236806-5907287.post@n2.nabble.com>	<SNT109-DS227CE87422187AA91545CFC1F30@phx.gbl>
	<1295113990176-5925152.post@n2.nabble.com>
Message-ID: <SNT109-DS2124FB8719BEF6F213D36DC1F40@phx.gbl>

Hi Jeff

I may of mis understood of you saying "modify the project", but all I am
doing is following the wiki on mod_dingaling and IKsemel, aren't these 2
module officially supported in the build for win32 without any errors.

I download the latest Jit today, starting it fresh, plain vanilla and

I am following your guidance but still getting the error. 

1>------ Rebuild All started: Project: iksemel, Configuration: Debug Win32
------
1>Deleting intermediate and output files for project 'iksemel',
configuration 'Debug|Win32'
1>Compiling...
1>dom.c
1>filter.c
1>iks.c
1>ikstack.c
1>io-posix.c
1>jabber.c
1>md5.c
1>sax.c
1>sha.c
1>stream.c
1>..\..\iksemel\src\stream.c(19) : fatal error C1083: Cannot open include
file: 'gnutls/gnutls.h': No such file or directory
1>utility.c
1>base64.c
1>Generating Code...
1>Build log was saved at
"file://c:\FS_GIT2\libs\win32\iksemel\Debug\BuildLog.htm"
1>iksemel - 1 error(s), 0 warning(s)
========== Rebuild All: 0 succeeded, 1 failed, 0 skipped ==========

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff
Lenk
Sent: Sunday, January 16, 2011 1:53 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Iksemel msvs compiling


Hi Norman,

I thought you were trying to modify IKSemel to support gnutls - if so how
can you not have modifed the project?

Compare your project includes to the following line.

..\..\iksemel\include;.;..\..\pthreads-w32-2-7-0-release;

the line looks like this when pulled from git :
..\..\iksemel\include;.

Jeff
-- 
View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/Iksemel-msvs-compiling-tp58912
63p5925152.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brokendash at gmail.com  Mon Jan 17 10:24:37 2011
From: brokendash at gmail.com (broken dash)
Date: Mon, 17 Jan 2011 01:24:37 -0600
Subject: [Freeswitch-users] Sharing storage between servers
In-Reply-To: <760758090.3041286611849227.JavaMail.root@mail>
References: <AANLkTi=8kH5-c8pL1K_KRsNBo2gEO4y+24iwCk=AkJ3D@mail.gmail.com>
	<760758090.3041286611849227.JavaMail.root@mail>
Message-ID: <AANLkTimC5h5Q3Rx-2s1k9RvoJ-r9iWVgLVeV0-Oga4K+@mail.gmail.com>

Your question is also incredibly complex depending on the methods you
could use as well. Storing the audio files within the mysql database
would be REALLY slow... :-)

I remember seeing a bounty for some type of Voicemail/IMAP integration
but I'm not sure if anyone has furthered it. that would be a neat way
to handle things since you could use an existing IMAP mail server
setup, or build one and scale/cluster however you wish an basically
use the IMAP mail system as the storage medium for your users voice
mail. So the Voicemail's would also be email's and one could use the
IMAP client on their desktop to also manage their messages. :-)

Also could be done using webdav/fuse etc..

One could also use multiple icecast servers spread around and have
liquidsoap sit in front of them so that when you go to record the
message it hits liquidsoap locally where its configured to feed the
single message out to multiple shoutcast servers spread around the
globe in different data centers or something.

This one is deff worth checking into tho....

http://www.automatthew.com/2007/12/amazon-simpledb-and-couchdb-compared.html


Late 2 cents...
Cheers,
B



On Sat, Oct 9, 2010 at 3:10 AM, Brett Woollum <brett at woollum.com> wrote:
> Is there a way to store voicemail to a MySQL cluster like Asterisk does?
> This is one of the issues I have been trying to find an answer for, but I
> haven't been able to. It would be great if there were a LUA/Javascript
> command that would stream audio to the call directly from the database, or
> the other way around. This would be useful for voicemail recordings, IVR
> prompts, etc, when running several clustered FS servers. I have a MySQL
> Cluster spread across each of my FS servers that should be able to handle
> the additional load. Using MySQL cluster as an on-demand audio storage
> system would make access to the audio files fault-tolerant and distributed
> (even to other servers such as a web server for web voicemail playback).
>
> Any ideas on how this could be achieved?
>
> Brett Woollum
> Brett at Woollum.com
>
>
> ----- Original Message -----
> From: "Yehavi Bourvine" <yehavi.bourvine at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Saturday, October 9, 2010 12:49:51 AM GMT -08:00 US/Canada Pacific
> Subject: Re: [Freeswitch-users] Sharing storage between servers
>
> Hi,
>
> ? We use a NAS server to share the voicemail between two servers (one is FS,
> the other is WEB interface we wrote to handle? voicemail via WEB).
> ? For the database: we use MySQL with replication.
>
> ????????????????????????? Regards, __Yehavi:
>
> 2010/10/9 Jody Rudolph <jody.rudolph at gmail.com>
>>
>> I am curious as to just how far you can take sharing disk storage for the
>> purpose of clustering. Is anyone doing this with the voicemail storage
>> directories? Is it possible to share the SQLite internal databases to avoid
>> resorting to ODBC? I realize that isn't likely, but I have access to some
>> high performance SAN hardware and want to take the most advantage possible.
>>
>>
>> Thanks,
>> Jody Rudolph
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From bernhard.suttner at winet.ch  Mon Jan 17 10:39:04 2011
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Mon, 17 Jan 2011 08:39:04 +0100
Subject: [Freeswitch-users] Sharing storage between servers
In-Reply-To: <AANLkTimC5h5Q3Rx-2s1k9RvoJ-r9iWVgLVeV0-Oga4K+@mail.gmail.com>
References: <AANLkTi=8kH5-c8pL1K_KRsNBo2gEO4y+24iwCk=AkJ3D@mail.gmail.com>
	<760758090.3041286611849227.JavaMail.root@mail>
	<AANLkTimC5h5Q3Rx-2s1k9RvoJ-r9iWVgLVeV0-Oga4K+@mail.gmail.com>
Message-ID: <31789c10-6c4f-4488-9897-f0ebf2ead350@winet.ch>

I had the same problem and did store the voicemail stuff on a NFS share. The NFS server does run on a heartbeat/drbd server. All the FreeSWITCH server does mount the high-available NFS share through the high-available IP address. Works very good - also in failover of the NFS Server.

BR,
Bernhard

-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von broken dash
Gesendet: Montag, 17. Januar 2011 08:25
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Sharing storage between servers

Your question is also incredibly complex depending on the methods you
could use as well. Storing the audio files within the mysql database
would be REALLY slow... :-)

I remember seeing a bounty for some type of Voicemail/IMAP integration
but I'm not sure if anyone has furthered it. that would be a neat way
to handle things since you could use an existing IMAP mail server
setup, or build one and scale/cluster however you wish an basically
use the IMAP mail system as the storage medium for your users voice
mail. So the Voicemail's would also be email's and one could use the
IMAP client on their desktop to also manage their messages. :-)

Also could be done using webdav/fuse etc..

One could also use multiple icecast servers spread around and have
liquidsoap sit in front of them so that when you go to record the
message it hits liquidsoap locally where its configured to feed the
single message out to multiple shoutcast servers spread around the
globe in different data centers or something.

This one is deff worth checking into tho....

http://www.automatthew.com/2007/12/amazon-simpledb-and-couchdb-compared.html


Late 2 cents...
Cheers,
B



On Sat, Oct 9, 2010 at 3:10 AM, Brett Woollum <brett at woollum.com> wrote:
> Is there a way to store voicemail to a MySQL cluster like Asterisk does?
> This is one of the issues I have been trying to find an answer for, but I
> haven't been able to. It would be great if there were a LUA/Javascript
> command that would stream audio to the call directly from the database, or
> the other way around. This would be useful for voicemail recordings, IVR
> prompts, etc, when running several clustered FS servers. I have a MySQL
> Cluster spread across each of my FS servers that should be able to handle
> the additional load. Using MySQL cluster as an on-demand audio storage
> system would make access to the audio files fault-tolerant and distributed
> (even to other servers such as a web server for web voicemail playback).
>
> Any ideas on how this could be achieved?
>
> Brett Woollum
> Brett at Woollum.com
>
>
> ----- Original Message -----
> From: "Yehavi Bourvine" <yehavi.bourvine at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Saturday, October 9, 2010 12:49:51 AM GMT -08:00 US/Canada Pacific
> Subject: Re: [Freeswitch-users] Sharing storage between servers
>
> Hi,
>
> ? We use a NAS server to share the voicemail between two servers (one is FS,
> the other is WEB interface we wrote to handle? voicemail via WEB).
> ? For the database: we use MySQL with replication.
>
> ????????????????????????? Regards, __Yehavi:
>
> 2010/10/9 Jody Rudolph <jody.rudolph at gmail.com>
>>
>> I am curious as to just how far you can take sharing disk storage for the
>> purpose of clustering. Is anyone doing this with the voicemail storage
>> directories? Is it possible to share the SQLite internal databases to avoid
>> resorting to ODBC? I realize that isn't likely, but I have access to some
>> high performance SAN hardware and want to take the most advantage possible.
>>
>>
>> Thanks,
>> Jody Rudolph
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From spencer at 5ninesolutions.com  Mon Jan 17 03:15:43 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 16 Jan 2011 16:15:43 -0800
Subject: [Freeswitch-users] BLF with Valet Parking
Message-ID: <5E3329D2-286C-4F3E-9F04-F11F5CAACF6B@5ninesolutions.com>

Hello all,
Is it possible to use BLFs with valet parking?  Basically I have  
several Linksys SPA 509G phones and after a call is parked, I'd like  
use use a speed dial/BLF key on the phone to pickup the call.  I do  
have presence configured and working with the extensions.

Thanks,
Spencer


From milan.m.masek at gmail.com  Mon Jan 17 08:38:38 2011
From: milan.m.masek at gmail.com (Milan Masek)
Date: Sun, 16 Jan 2011 21:38:38 -0800
Subject: [Freeswitch-users] Bria 3 and Freeswitch TLS configuration
Message-ID: <AANLkTinntjwHEckZeT3N3=LuFwguMgK0bS=vUD_CWQUv@mail.gmail.com>

Hi there,
What do you think I am doing wrong when my connection to Freeswitch over TLS
from Bria softphone ends with:
"could not be enabled. Problem at server (SIP error 503)"

Configuration:
Server: CentOS 5, Freeswitch 1.0.6 (fresh installation from a tarball)
Client: (K)Ubuntu 10.04, Bria 3.1

-firewall is OK, telnet on port 5061 OK, iptables OK, tcpdump shows traffic
-server CA certificate imported on client computer to system-wide keystore
database (dpkg-reconfigure ca-certificates)
-TLS enabled in vars.xml

Also (with no TLS connection) - I can register phone (Bria, Empathy), I can
dial to another user, hear ring tone, pickup the phone call - but I can not
hear person on other side. I can just see that my microphone works.


Thank you for help. Any advice appreciated.
Cheers,
Milan
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From jason at jasonjgw.net  Mon Jan 17 01:48:49 2011
From: jason at jasonjgw.net (Jason White)
Date: Mon, 17 Jan 2011 09:48:49 +1100
Subject: [Freeswitch-users] FreeSWITCH 1.0.7
References: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org>
Message-ID: <87ipxov50u.fsf@jdc.jasonjgw.net>


Brian West <brian at freeswitch.org> writes:

> http://latest.freeswitch.org/
>
> Enjoy!

Thank you for all of the work that found its way into this release.

A minor point: the 1.0.7 release doesn't appear to have been tagged in
the Git repository. Perhaps the tag hasn't been pushed. git push --tags
should do it in that case.



From jalsot at gmail.com  Mon Jan 17 11:46:55 2011
From: jalsot at gmail.com (Tamas Jalsovszky)
Date: Mon, 17 Jan 2011 09:46:55 +0100
Subject: [Freeswitch-users] Removing inband DTMF
In-Reply-To: <101096F0-3FCF-4099-B622-EB7F315CFBF2@freeswitch.org>
References: <AANLkTikBZPbeKLO_fX=s4p1pZxPCzwcAet1win8eaUMn@mail.gmail.com>
	<AANLkTimYd8PZumEY1GzkcAj-kZ1auH8Lvr-7OBm7r3fB@mail.gmail.com>
	<AANLkTimgMxuC=FEQ0UFerTLE1wRCKUx5L1Kr-_KMWJ3b@mail.gmail.com>
	<AANLkTi=BQGJJrVgyPKxX2aP3kq_xWRHDzzsf1-qQd9X3@mail.gmail.com>
	<A7150993-F25F-474D-A6A9-5EFA84C7ACDB@gmail.com>
	<AANLkTi=RGqABGRvnEwKVug5XFPPfyYhN2nMdrekkdT22@mail.gmail.com>
	<101096F0-3FCF-4099-B622-EB7F315CFBF2@freeswitch.org>
Message-ID: <AANLkTi=wbG_LDeYNZe2hnh4p-AKWvwZiTkTUWHa4qH-u@mail.gmail.com>

Yeah, we came to this conclusion pretty quickly. We can ignore inband dtmf
in case there is FS at the end. The problem is with other vendors' other
devices which are not that flexible as FS is.
I don't know why Cirpack is trying to push their shit over and cannot
substract inband dtmf...

Regards,
  T.

On Sun, Jan 16, 2011 at 11:16 PM, Brian West <brian at freeswitch.org> wrote:

> Beat your provider to death for doing something utterly stupid.   If you
> are only acting on 2833 it won't matter if they send inband and 2833 at the
> same time..  It won't matter as long as you are not staring a dtmf detector
> on the target system and only the system converting from inband to 2833 does
> so.
>
> /b
>
> On Jan 16, 2011, at 3:47 PM, Steven Ayre wrote:
>
> > I didn't mean a significant amount of time. 40ms with a ptime of 20ms
> means you've already sent a packet containing inbound dtmf.
> >
> > -Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Mon Jan 17 11:50:02 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 17 Jan 2011 08:50:02 +0000
Subject: [Freeswitch-users] read conference PIN from a file or db
In-Reply-To: <AANLkTin2TNtXkGGLYAyOd-Wse+FADKCC5mdStAb=yzFA@mail.gmail.com>
References: <AANLkTik3goWsqBVSjXBSD8HJLPHV4FWUFyVPsJXsycdh@mail.gmail.com>
	<AANLkTi=PFQTZbeNCm7fjE9LrZNFi0V+idB5ypjQeoY=S@mail.gmail.com>
	<AANLkTi=oV+hN0E12A=St7faH+65U+nCZBHKKb444LA1D@mail.gmail.com>
	<AANLkTinMEVy55S1kqyKq5RVxBJdmd3pELr5fnezs7dGR@mail.gmail.com>
	<AANLkTin2TNtXkGGLYAyOd-Wse+FADKCC5mdStAb=yzFA@mail.gmail.com>
Message-ID: <AANLkTi=X8Axwf9rTDKJsabUORd0Mq9oSA157bd0cRN16@mail.gmail.com>

Two ways...

1. You can then access that variable from JS to get the pin using
getVariable.

2. <action application="javascript" data="/path/to/some/script.js $pin"/>
and read it from the argv[] array

-Steve



On 17 January 2011 05:45, deniro <ayhkor at gmail.com> wrote:

> Thanks Steve
> I looked at these links
>
> more specifically what I look for is with the dialplan+javascript;
> How can I set the logic in a dialplan in such a way that I enter the
> conference PIN, and pass it as a parameter to javascript and compare with
> the values(set of pin numbers) in database and when there is a match, will
> let you go to conference?
>
> I am able to read the pin number and assign it to $pin.
> In a dialplan, I wanna set a condition or something, call javascript, pass
> $pin as parameter to javascript, compare it with the values in db if there
> is a match, allow you to go to conference.
>  So, how can I set this logic in dialplan with javascript?
> any working example?
>
> thanks
>
>
>
> On Sun, Jan 16, 2011 at 5:58 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Sure you can - mod_spidermonkey gives you javascript, if you configure
>> spidermonkey.conf.xml to load mod_spidermonkey_odbc you'll get odbc access
>> so you can access a mysql database.
>>
>> http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc
>> http://wiki.freeswitch.org/wiki/Session_setVariable
>>
>> -Steve
>>
>>
>>
>>
>> On 16 January 2011 22:45, deniro <ayhkor at gmail.com> wrote:
>>
>>> how about writing a javascript and reading from mysql database
>>> when a match is hit return a value and  load it into $pin
>>> how can I do this type of javascripting?
>>> thx
>>>
>>>
>>>
>>> On Sun, Jan 16, 2011 at 3:49 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>>
>>>> There's several ways, you can write your own module, a script, use xml
>>>> curl etc.
>>>>
>>>> The simplest option is probably to use lua - you can access odbc if you
>>>> install luasql. I'm sure file access is possible too.
>>>>
>>>> -Steve
>>>>
>>>>
>>>>   On 16 January 2011 19:46, deniro <ayhkor at gmail.com> wrote:
>>>>
>>>>>   Hi
>>>>> Regarding providing PIN numbers to a conference
>>>>> how can I read PINs from a file or database and allow them to go to a
>>>>> meeting
>>>>> thx
>>>>> deniro--
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jonas.gauffin at gmail.com  Mon Jan 17 12:56:16 2011
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Mon, 17 Jan 2011 10:56:16 +0100
Subject: [Freeswitch-users] Find audio problem
Message-ID: <AANLkTi=s2Sw7yiwVx6Ef9HhjDTmvSPtcLAP6oAk1pB1H@mail.gmail.com>

Hello,

I got two different customers which have problems with that the audio stops
working in one direction after a while. And not for all calls but only some.

person calling ->  My GW provider -> internet -> freeswitch -> internet ->
router -> customer.

It's always the customer who can't hear the person calling.

Any suggestions on how I can find and fix the problem?

Thanks,
  Jonas
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From bernhard.suttner at winet.ch  Mon Jan 17 13:05:11 2011
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Mon, 17 Jan 2011 11:05:11 +0100
Subject: [Freeswitch-users] Find audio problem
In-Reply-To: <AANLkTi=s2Sw7yiwVx6Ef9HhjDTmvSPtcLAP6oAk1pB1H@mail.gmail.com>
References: <AANLkTi=s2Sw7yiwVx6Ef9HhjDTmvSPtcLAP6oAk1pB1H@mail.gmail.com>
Message-ID: <20df03db-8811-4e8f-b0c6-ec250a1a5efe@winet.ch>

-          Check FreeSWITCH Logs
-          Check if SIP ACK was received
-          Check Firewall Settings
-          Check TCPDUMP if RTP media was dropped
 
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Jonas Gauffin
Gesendet: Montag, 17. Januar 2011 10:56
An: FreeSWITCH Users Help
Betreff: [Freeswitch-users] Find audio problem
 
Hello,
 
I got two different customers which have problems with that the audio stops working in one direction after a while. And not for all calls but only some.
 
person calling ->  My GW provider -> internet -> freeswitch -> internet -> router -> customer.
 
It's always the customer who can't hear the person calling. 
 
Any suggestions on how I can find and fix the problem? 
 
Thanks,
  Jonas
 
 
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From u2nsam at gmail.com  Mon Jan 17 14:20:59 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 17 Jan 2011 16:50:59 +0530
Subject: [Freeswitch-users] console
In-Reply-To: <AANLkTikVECT9BAXCTLLkxXbfdi0wcVv1yvjteyigTqh=@mail.gmail.com>
References: <6ECAF1527329364583AB525CF34ABF950B31A4F9@ms.kallback.com>
	<AANLkTinXiaVGr5bneGCfvXUmbEnJ5Ggjcmvim=Wjqa5O@mail.gmail.com>
	<AANLkTinnKUY9xkWAfLGgs+sf41A-nZ0xH9C05ogeBHqW@mail.gmail.com>
	<AANLkTikVECT9BAXCTLLkxXbfdi0wcVv1yvjteyigTqh=@mail.gmail.com>
Message-ID: <AANLkTikSCWjyS7A_Y15zNajAKAhCm+XawVBJ=Fib6_6C@mail.gmail.com>

I reinstalled FS 2 instances and tried doing fs_cli with 2 different ports
and it worked,
thanks all.

Regds
Sam

On Thu, Jan 13, 2011 at 11:18 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Sam,
>
> Run these command as root with both versions of FS started. If FreeSWITCH
> is running you should see 2 lines like below.
>
> $ netstat -anp | grep 8021
> tcp        0      0 127.0.0.1:8021          0.0.0.0:*
> LISTEN      15394/freeswitch
> $ netstat -anp | grep 8022
> tcp        0      0 127.0.0.1:8022          0.0.0.0:*
> LISTEN      15395/freeswitch
>
> -Steve
>
>
>
>
> On 13 January 2011 17:40, Anthony Minessale <anthony.minessale at gmail.com>wrote:
>
>> 8081 8082 vs 8021 8022
>>
>>
>>
>> On Wed, Jan 12, 2011 at 9:16 PM, Sam <u2nsam at gmail.com> wrote:
>> > Yes the netstat shows the ports 8081 & 8082 not used
>> > Justin, as said earlier i have 2 FS instances running on same server and
>> > having below config:-
>> > for
>> >> 192.168.2.1:-
>> >>    <param name="listen-ip" value="127.0.0.1"/>
>> >>    <param name="listen-port" value="8021"/>
>> >> for
>> >> 192.168.2.2:-
>> >>    <param name="listen-ip" value="127.0.0.1"/>
>> >>    <param name="listen-port" value="8022"/>
>> >
>> > Here I tried all the permutation combination changing the IP port but no
>> > sucess.
>> >
>> > Regds
>> > Sam
>> >
>> > On Wed, Jan 12, 2011 at 11:25 PM, JRichey <jrichey at itltd.net> wrote:
>> >>
>> >> Did you ever try using netstat like Steven suggested?  What do you get
>> as
>> >> output for "netstat -tunlp"?
>> >>
>> >> In one of your commands below you show 127.0.0.2 instead of 127.0.0.1
>> so
>> >> if that's not a typo you may want to try it again.
>> >>
>> >> -Justin
>> >>
>> >>
>> >> -----Original Message-----
>> >> From: freeswitch-users-bounces at lists.freeswitch.org
>> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of Sam
>> >> Sent: Tuesday, January 11, 2011 9:27 PM
>> >> To: FreeSWITCH Users Help
>> >> Subject: Re: [Freeswitch-users] console
>> >>
>> >> Tried this below and received :-
>> >>
>> >>
>> >> # /usr/local/fs_2/bin/fs_cli -H 127.0.0.2 -P 8082
>> >> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
>> Connection
>> >> Error]
>> >>
>> >> tried by creating profile:-
>> >> # /usr/local/fs_2/bin/fs_cli profile1
>> >> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
>> Connection
>> >> Error]
>> >>
>> >> By port:
>> >> # /usr/local/fs_2/bin/fs_cli -P 8082
>> >> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
>> Connection
>> >> Error]
>> >>
>> >>
>> >> netstat -nlp | grep ; ports 8081 & 8082 are available
>> >>
>> >> But i could get a console for other server by
>> >> # /usr/local/fs_2/bin/fs_cli
>> >>
>> >> Regards
>> >> Sam
>> >>
>> >>
>> >> On Tue, Jan 11, 2011 at 10:54 PM, Steven Ayre <steveayre at gmail.com>
>> wrote:
>> >>>
>> >>> Actually, correction it does - just not the event_socket.conf.xml one.
>> >>> It'll read .fs_cli_conf in your home directory if it exists, but that
>> isn't
>> >>> created by default - you create it yourself if you want to use it
>> (it's
>> >>> optional). The command line arguments -H and -P -will- override the
>> config
>> >>> file though.
>> >>>
>> >>> Are you using a capital P? -p is password while -P is port. If there's
>> no
>> >>> password on the event socket you'd get no error from using a small p
>> by
>> >>> accident.
>> >>>
>> >>> -Steve
>> >>>
>> >>>
>> >>>
>> >>>
>> >>>
>> >>> On 11 January 2011 16:59, Steven Ayre <steveayre at gmail.com> wrote:
>> >>>>
>> >>>> It does not read any config file.
>> >>>>
>> >>>>
>> >>>>
>> >>>>
>> >>>> On 11 January 2011 16:56, Sam <u2nsam at gmail.com> wrote:
>> >>>>>
>> >>>>> the port and ip donot work for me,
>> >>>>> is it that the fs_cli is not reading the config from 192.168.2.2 but
>> it
>> >>>>> is reading the config only of 192.168.2.1, though its in the
>> different [FS_1
>> >>>>> & FS_2] path where i am executing.
>> >>>>>
>> >>>>> Regds
>> >>>>> Sam
>> >>>>>
>> >>>>>
>> >>>>> On Tue, Jan 11, 2011 at 9:38 PM, Steven Ayre <steveayre at gmail.com>
>> >>>>> wrote:
>> >>>>>>
>> >>>>>> /usr/local/FS_1/bin/fs_cli -P 8021
>> >>>>>> /usr/local/FS_2/bin/fs_cli -P 8022
>> >>>>>>
>> >>>>>> fs_cll doesn't read any config file. It's not part of the FS server
>> at
>> >>>>>> all, you can have it on a different machine that doesn't have FS
>> installed.
>> >>>>>> It entirely relies on the arguments to control where to connect to.
>> >>>>>>
>> >>>>>> -Steve
>> >>>>>>
>> >>>>>>
>> >>>>>> On 11 January 2011 15:04, Sam <u2nsam at gmail.com> wrote:
>> >>>>>>>
>> >>>>>>> Something more here ... i am getting the console for 192.168.2.1
>> >>>>>>> every time i do fs_cli on both instances .
>> >>>>>>>
>> >>>>>>> like
>> >>>>>>> /usr/local/FS_1/bin/fs_cli
>> >>>>>>> /usr/localFS_2/bin/fs_cli
>> >>>>>>> i get the console for the 1st server only
>> >>>>>>>
>> >>>>>>> the 2 server are listing to 2 different ips .
>> >>>>>>>
>> >>>>>>> Regds
>> >>>>>>> Sam
>> >>>>>>>
>> >>>>>>> On Tue, Jan 11, 2011 at 8:07 PM, Steven Ayre <steveayre at gmail.com
>> >
>> >>>>>>> wrote:
>> >>>>>>>>
>> >>>>>>>> It should work. Is there anything already listening on port 8022?
>> >>>>>>>>
>> >>>>>>>> $ netstat -a -n -p | grep 8022
>> >>>>>>>>
>> >>>>>>>> Are you also sure that they're not both loading the same config
>> >>>>>>>> file?
>> >>>>>>>>
>> >>>>>>>> Regards,
>> >>>>>>>> -Steve
>> >>>>>>>>
>> >>>>>>>>
>> >>>>>>>>
>> >>>>>>>> On 11 January 2011 14:32, Sam <u2nsam at gmail.com> wrote:
>> >>>>>>>> > the scenario is i have 2 ips on 1 server for 2 FS instances;
>> >>>>>>>> >
>> >>>>>>>> > 192.168.2.1
>> >>>>>>>> > 192.168.2.2
>> >>>>>>>> >
>> >>>>>>>> > and the parameters i have set is:-
>> >>>>>>>> >
>> >>>>>>>> > for
>> >>>>>>>> > 192.168.2.1:-
>> >>>>>>>> >    <param name="listen-ip" value="127.0.0.1"/>
>> >>>>>>>> >    <param name="listen-port" value="8021"/>
>> >>>>>>>> > for
>> >>>>>>>> > 192.168.2.2:-
>> >>>>>>>> >    <param name="listen-ip" value="127.0.0.1"/>
>> >>>>>>>> >    <param name="listen-port" value="8022"/>
>> >>>>>>>> >
>> >>>>>>>> > Ideally it should work  but i am getting console for only
>> >>>>>>>> > 192.168.2.1 FS .
>> >>>>>>>> >
>> >>>>>>>> >
>> >>>>>>>> > Regards
>> >>>>>>>> > Sam
>> >>>>>>>> >
>> >>>>>>>> > On Tue, Jan 11, 2011 at 4:20 PM, Steven Ayre <
>> steveayre at gmail.com>
>> >>>>>>>> > wrote:
>> >>>>>>>> >>
>> >>>>>>>> >>    <param name="listen-ip" value="127.0.0.1"/>
>> >>>>>>>> >>    <param name="listen-port" value="8021"/>
>> >>>>>>>> >>
>> >>>>>>>> >> You can bind both to port 8021 on their individual IPs, or
>> >>>>>>>> >> different
>> >>>>>>>> >> ports on the same IP.
>> >>>>>>>> >>
>> >>>>>>>> >> A listen IP of 0.0.0.0 will mean any IP.
>> >>>>>>>> >>
>> >>>>>>>> >> -Steve
>> >>>>>>>> >>
>> >>>>>>>> >> On 11 January 2011 10:44, Sam <u2nsam at gmail.com> wrote:
>> >>>>>>>> >> > A query,
>> >>>>>>>> >> >
>> >>>>>>>> >> > I have 2 FS running on one server on 2 different ips,
>> >>>>>>>> >> > so when i do fs_cli going to respective bins , i see console
>> of
>> >>>>>>>> >> > only the
>> >>>>>>>> >> > first server.
>> >>>>>>>> >> >
>> >>>>>>>> >> > Is there any way to get the console of both the FS on the
>> same
>> >>>>>>>> >> > server .
>> >>>>>>>> >> > I tried changing the port of event socket to 8022 but it
>> donot
>> >>>>>>>> >> > works.
>> >>>>>>>> >> > <param name="listen-port" value="8022"/>
>> >>>>>>>> >> >
>> >>>>>>>> >> > Is there some method to start the console of both the
>> >>>>>>>> >> > instances.
>> >>>>>>>> >> >
>> >>>>>>>> >> > Regds
>> >>>>>>>> >> > Sam
>> >>>>>>>> >> >
>> >>>>>>>> >> > _______________________________________________
>> >>>>>>>> >> > FreeSWITCH-users mailing list
>> >>>>>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >>>>>>>> >> >
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>>>>> >> >
>> >>>>>>>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>>>>> >> > http://www.freeswitch.org
>> >>>>>>>> >> >
>> >>>>>>>> >> >
>> >>>>>>>> >>
>> >>>>>>>> >> _______________________________________________
>> >>>>>>>> >> FreeSWITCH-users mailing list
>> >>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>> >>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>>>>> >>
>> >>>>>>>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>>>>> >> http://www.freeswitch.org
>> >>>>>>>> >
>> >>>>>>>> >
>> >>>>>>>> > _______________________________________________
>> >>>>>>>> > FreeSWITCH-users mailing list
>> >>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>> >>>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>>>>> >
>> >>>>>>>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>>>>> > http://www.freeswitch.org
>> >>>>>>>> >
>> >>>>>>>> >
>> >>>>>>>>
>> >>>>>>>> _______________________________________________
>> >>>>>>>> FreeSWITCH-users mailing list
>> >>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>>>>>
>> >>>>>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>>>>> http://www.freeswitch.org
>> >>>>>>>
>> >>>>>>>
>> >>>>>>> _______________________________________________
>> >>>>>>> FreeSWITCH-users mailing list
>> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>>>>
>> >>>>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>>>> http://www.freeswitch.org
>> >>>>>>>
>> >>>>>>
>> >>>>>>
>> >>>>>> _______________________________________________
>> >>>>>> FreeSWITCH-users mailing list
>> >>>>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>>>
>> >>>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>>> http://www.freeswitch.org
>> >>>>>>
>> >>>>>
>> >>>>>
>> >>>>> _______________________________________________
>> >>>>> FreeSWITCH-users mailing list
>> >>>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>>
>> >>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>> http://www.freeswitch.org
>> >>>>>
>> >>>>
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From grsingh750 at gmail.com  Mon Jan 17 14:56:12 2011
From: grsingh750 at gmail.com (guru singh)
Date: Mon, 17 Jan 2011 17:26:12 +0530
Subject: [Freeswitch-users] Caller-id on incoming FXO with freetdm
Message-ID: <AANLkTi=TujU1976wLCABzsfG2w=SvDQW1p2XEXfRniew@mail.gmail.com>

Hi,

I am unable to get caller-id for incoming FXO calls.
It is always being set to 0000000000
I'm using FreeTDM and am on the latest git. I've set enable_callerid
param to true in freetdm.conf.xml
I can see fs pickup the callerid but then after each digit it says
(debug=0), is this normal?
Here is call debug log in fs for answering fxo and running the info
app. http://pastebin.freeswitch.org/15037
What am I missing?

thanks
guru


From thomas at chaschperli.ch  Mon Jan 17 13:36:19 2011
From: thomas at chaschperli.ch (Thomas Mueller)
Date: Mon, 17 Jan 2011 11:36:19 +0100
Subject: [Freeswitch-users] Find audio problem
In-Reply-To: <AANLkTi=s2Sw7yiwVx6Ef9HhjDTmvSPtcLAP6oAk1pB1H@mail.gmail.com>
References: <AANLkTi=s2Sw7yiwVx6Ef9HhjDTmvSPtcLAP6oAk1pB1H@mail.gmail.com>
Message-ID: <4D341BA3.40407@chaschperli.ch>

On 17.01.2011 10:56, Jonas Gauffin wrote:
> Hello,
> I got two different customers which have problems with that the audio 
> stops working in one direction after a while. And not for all calls 
> but only some.
> person calling ->  My GW provider -> internet -> freeswitch -> 
> internet -> router -> customer.
> It's always the customer who can't hear the person calling. 

If the "after a while" is the more or less the same for the problematic 
calls I would check "router". Maybe it's some NAT session timeout.

- Thomas



From melkybes at mail.ru  Mon Jan 17 16:49:50 2011
From: melkybes at mail.ru (=?koi8-r?Q?=ED=C9=C8=C1=C9=CC_=F3=C1=CC=D4=C1=CE=CF=D7?=)
Date: Mon, 17 Jan 2011 16:49:50 +0300
Subject: [Freeswitch-users] =?koi8-r?b?NC13YXkgYW5kIG1vcmUgY29uZmVyZW5j?=
	=?koi8-r?b?ZSAmIGF0dF94ZmVy?=
Message-ID: <E1PepSw-0005HO-00.melkybes-mail-ru@f64.mail.ru>

Hello  all

I use att_xfer to create 3-way conference  from usual call/ How can I add more users? Can I change some variables in source or   in config only?

Thanks


Michail Saltanov



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From brokendash at gmail.com  Mon Jan 17 16:51:43 2011
From: brokendash at gmail.com (broken dash)
Date: Mon, 17 Jan 2011 07:51:43 -0600
Subject: [Freeswitch-users] Sharing storage between servers
In-Reply-To: <31789c10-6c4f-4488-9897-f0ebf2ead350@winet.ch>
References: <AANLkTi=8kH5-c8pL1K_KRsNBo2gEO4y+24iwCk=AkJ3D@mail.gmail.com>
	<760758090.3041286611849227.JavaMail.root@mail>
	<AANLkTimC5h5Q3Rx-2s1k9RvoJ-r9iWVgLVeV0-Oga4K+@mail.gmail.com>
	<31789c10-6c4f-4488-9897-f0ebf2ead350@winet.ch>
Message-ID: <AANLkTikTPSeb+Mmu17M6t1wq4aEKm_xX0OoeQu62kOwu@mail.gmail.com>

What are you using for your NFS HA setup? I have previously used
Openfiler to get the job done a few years back but always ran into
problems whenever I hit about 15ish clients attached.  :-)  Have you
ever messed around with iSCSI & Multipath IO?

Cheers,
B

On Mon, Jan 17, 2011 at 1:39 AM, Bernhard Suttner
<bernhard.suttner at winet.ch> wrote:
> I had the same problem and did store the voicemail stuff on a NFS share. The NFS server does run on a heartbeat/drbd server. All the FreeSWITCH server does mount the high-available NFS share through the high-available IP address. Works very good - also in failover of the NFS Server.
>
> BR,
> Bernhard
>
> -----Urspr?ngliche Nachricht-----
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von broken dash
> Gesendet: Montag, 17. Januar 2011 08:25
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] Sharing storage between servers
>
> Your question is also incredibly complex depending on the methods you
> could use as well. Storing the audio files within the mysql database
> would be REALLY slow... :-)
>
> I remember seeing a bounty for some type of Voicemail/IMAP integration
> but I'm not sure if anyone has furthered it. that would be a neat way
> to handle things since you could use an existing IMAP mail server
> setup, or build one and scale/cluster however you wish an basically
> use the IMAP mail system as the storage medium for your users voice
> mail. So the Voicemail's would also be email's and one could use the
> IMAP client on their desktop to also manage their messages. :-)
>
> Also could be done using webdav/fuse etc..
>
> One could also use multiple icecast servers spread around and have
> liquidsoap sit in front of them so that when you go to record the
> message it hits liquidsoap locally where its configured to feed the
> single message out to multiple shoutcast servers spread around the
> globe in different data centers or something.
>
> This one is deff worth checking into tho....
>
> http://www.automatthew.com/2007/12/amazon-simpledb-and-couchdb-compared.html
>
>
> Late 2 cents...
> Cheers,
> B
>
>
>
> On Sat, Oct 9, 2010 at 3:10 AM, Brett Woollum <brett at woollum.com> wrote:
>> Is there a way to store voicemail to a MySQL cluster like Asterisk does?
>> This is one of the issues I have been trying to find an answer for, but I
>> haven't been able to. It would be great if there were a LUA/Javascript
>> command that would stream audio to the call directly from the database, or
>> the other way around. This would be useful for voicemail recordings, IVR
>> prompts, etc, when running several clustered FS servers. I have a MySQL
>> Cluster spread across each of my FS servers that should be able to handle
>> the additional load. Using MySQL cluster as an on-demand audio storage
>> system would make access to the audio files fault-tolerant and distributed
>> (even to other servers such as a web server for web voicemail playback).
>>
>> Any ideas on how this could be achieved?
>>
>> Brett Woollum
>> Brett at Woollum.com
>>
>>
>> ----- Original Message -----
>> From: "Yehavi Bourvine" <yehavi.bourvine at gmail.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Saturday, October 9, 2010 12:49:51 AM GMT -08:00 US/Canada Pacific
>> Subject: Re: [Freeswitch-users] Sharing storage between servers
>>
>> Hi,
>>
>> ? We use a NAS server to share the voicemail between two servers (one is FS,
>> the other is WEB interface we wrote to handle? voicemail via WEB).
>> ? For the database: we use MySQL with replication.
>>
>> ????????????????????????? Regards, __Yehavi:
>>
>> 2010/10/9 Jody Rudolph <jody.rudolph at gmail.com>
>>>
>>> I am curious as to just how far you can take sharing disk storage for the
>>> purpose of clustering. Is anyone doing this with the voicemail storage
>>> directories? Is it possible to share the SQLite internal databases to avoid
>>> resorting to ODBC? I realize that isn't likely, but I have access to some
>>> high performance SAN hardware and want to take the most advantage possible.
>>>
>>>
>>> Thanks,
>>> Jody Rudolph
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________ FreeSWITCH-users mailing
>> list FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From kemen04 at gmail.com  Mon Jan 17 18:01:15 2011
From: kemen04 at gmail.com (Travis Kemen)
Date: Mon, 17 Jan 2011 09:01:15 -0600
Subject: [Freeswitch-users] BLF with Valet Parking
In-Reply-To: <5E3329D2-286C-4F3E-9F04-F11F5CAACF6B@5ninesolutions.com>
References: <5E3329D2-286C-4F3E-9F04-F11F5CAACF6B@5ninesolutions.com>
Message-ID: <AANLkTimkN-bdsRxxR8RhaKams=apiHLeRt9CPf42wvXJ@mail.gmail.com>

BLF works fine here with valet parking using polycom/snom phones.

Travis

On Sun, Jan 16, 2011 at 6:15 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> Hello all,
> Is it possible to use BLFs with valet parking?  Basically I have
> several Linksys SPA 509G phones and after a call is parked, I'd like
> use use a speed dial/BLF key on the phone to pickup the call.  I do
> have presence configured and working with the extensions.
>
> Thanks,
> Spencer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jeff at jefflenk.com  Mon Jan 17 18:05:46 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 17 Jan 2011 07:05:46 -0800 (PST)
Subject: [Freeswitch-users] Iksemel msvs compiling
In-Reply-To: <SNT109-DS2124FB8719BEF6F213D36DC1F40@phx.gbl>
References: <SNT109-DS13313A1FA906527438E8CC1090@phx.gbl>
	<SNT109-DS1155D401A0B9742CF63E13C10B0@phx.gbl>
	<1294375438447-5898181.post@n2.nabble.com>
	<SNT109-DS11CB0ECE5BD0F9C0384E03C10C0@phx.gbl>
	<1294535404901-5903534.post@n2.nabble.com>
	<SNT109-DS14B90C48B33126784F294AC10E0@phx.gbl>
	<1294673236806-5907287.post@n2.nabble.com>
	<SNT109-DS227CE87422187AA91545CFC1F30@phx.gbl>
	<1295113990176-5925152.post@n2.nabble.com>
	<SNT109-DS2124FB8719BEF6F213D36DC1F40@phx.gbl>
Message-ID: <1295276746914-5932097.post@n2.nabble.com>


They are experimental. A couple of users have reported success but the wiki
documentation on that change is sparse. If you are successful please help by
adding more specific instructions to those pages.

It looks like you also need to add wherever gnutls.h is located to the
include path as well.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Iksemel-msvs-compiling-tp5891263p5932097.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brian at freeswitch.org  Mon Jan 17 18:23:51 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Jan 2011 09:23:51 -0600
Subject: [Freeswitch-users] FreeSWITCH 1.0.7
In-Reply-To: <87ipxov50u.fsf@jdc.jasonjgw.net>
References: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org>
	<87ipxov50u.fsf@jdc.jasonjgw.net>
Message-ID: <2013F94B-1AD7-44BD-85FE-6F6DF7753D3A@freeswitch.org>

the odd numbers are not tagged in GIT.  Even numbers are.  Expect 1.0.8 in a month or so.

/b

On Jan 16, 2011, at 4:48 PM, Jason White wrote:

> A minor point: the 1.0.7 release doesn't appear to have been tagged in
> the Git repository. Perhaps the tag hasn't been pushed. git push --tags
> should do it in that case.



From brian at freeswitch.org  Mon Jan 17 18:29:08 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Jan 2011 09:29:08 -0600
Subject: [Freeswitch-users] BLF with Valet Parking
In-Reply-To: <AANLkTimkN-bdsRxxR8RhaKams=apiHLeRt9CPf42wvXJ@mail.gmail.com>
References: <5E3329D2-286C-4F3E-9F04-F11F5CAACF6B@5ninesolutions.com>
	<AANLkTimkN-bdsRxxR8RhaKams=apiHLeRt9CPf42wvXJ@mail.gmail.com>
Message-ID: <31AC9159-35FD-450F-9E9A-E648AA99AF8B@freeswitch.org>

I think he wants BLF status of a lot on a key.  That I don't think is there.

/b

On Jan 17, 2011, at 9:01 AM, Travis Kemen wrote:

> BLF works fine here with valet parking using polycom/snom phones.
> 
> Travis
> 
> On Sun, Jan 16, 2011 at 6:15 PM, Spencer Thomason <spencer at 5ninesolutions.com> wrote:
> Hello all,
> Is it possible to use BLFs with valet parking?  Basically I have
> several Linksys SPA 509G phones and after a call is parked, I'd like
> use use a speed dial/BLF key on the phone to pickup the call.  I do
> have presence configured and working with the extensions.
> 
> Thanks,
> Spencer

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From kris at kriskinc.com  Mon Jan 17 18:39:07 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 17 Jan 2011 10:39:07 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.0.7
In-Reply-To: <AANLkTikNCfjHWhQWB9C1THUNdUWhRdP0rcs1JNQOqdmO@mail.gmail.com>
References: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org>
	<AANLkTik+hPwuvDtDOQ-6pPg-wskP_BTCn2=gm9vtV688@mail.gmail.com>
	<AANLkTikNCfjHWhQWB9C1THUNdUWhRdP0rcs1JNQOqdmO@mail.gmail.com>
Message-ID: <AANLkTi=C6LRTtDX87KLs4z+ZENqf_AT38hePvq9TE-qZ@mail.gmail.com>

Yep...

version
FreeSWITCH Version 1.0.head (git-49a524b 2010-12-27 21-31-40 -0300)

freeswitch at 127.0.0.1@internal> status
UP 0 years, 20 days, 9 hours, 34 minutes, 32 seconds, 384
milliseconds, 250 microseconds
3837450 session(s) since startup
778 session(s) 2/200
3000 session(s) max
min idle cpu 0.00/97.00


On Fri, Jan 14, 2011 at 10:46 PM, dome at tel.co.th <dome at tel.co.th> wrote:
> I use (from git) for my production about 2 month ago :)
> I can confirm very stable :)
>
> freeswitch at internal> version
> FreeSWITCH Version 1.0.head (git-6faa4c9 2010-12-02 17-11-04 -0600)
>
> freeswitch at internal> status
> UP 0 years, 42 days, 13 hours, 26 minutes, 11 seconds, 371
> milliseconds, 850 microseconds
> 365465 session(s) since startup
> 13 session(s) 0/30
> 1000 session(s) max
> min idle cpu 0.00/98.00
>
>
>
> Dome C.

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


From jmesquita at freeswitch.org  Mon Jan 17 18:51:42 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 17 Jan 2011 12:51:42 -0300
Subject: [Freeswitch-users] 4-way and more conference & att_xfer
In-Reply-To: <E1PepSw-0005HO-00.melkybes-mail-ru@f64.mail.ru>
References: <E1PepSw-0005HO-00.melkybes-mail-ru@f64.mail.ru>
Message-ID: <AANLkTi=X+q=hJoBxoW52rpaLo9MhKFOKEKfGiHxJ50L_@mail.gmail.com>

You'd have to transfer all the legs to a conference as this is not possible
with att_xfer.

Jo?o Mesquita


2011/1/17 ?????? ???????? <melkybes at mail.ru>

> Hello  all
>
> I use att_xfer to create 3-way conference  from usual call/ How can I add
> more users? Can I change some variables in source or   in config only?
>
> Thanks
>
>
> Michail Saltanov
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Mon Jan 17 19:15:44 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 17 Jan 2011 11:15:44 -0500
Subject: [Freeswitch-users] Sharing storage between servers
References: <AANLkTi=8kH5-c8pL1K_KRsNBo2gEO4y+24iwCk=AkJ3D@mail.gmail.com><760758090.3041286611849227.JavaMail.root@mail>
	<AANLkTimC5h5Q3Rx-2s1k9RvoJ-r9iWVgLVeV0-Oga4K+@mail.gmail.com>
Message-ID: <14AEC4AAA32849AC8E9A4F9CE874651A@e1705>

you can also simply use files in folder and replicate it
to all of your nodes every 3mn...
store binary in DB is more slow as in a folder

----- Original Message ----- 
From: "broken dash" <brokendash at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, January 17, 2011 2:24 AM
Subject: Re: [Freeswitch-users] Sharing storage between servers


Your question is also incredibly complex depending on the methods you
could use as well. Storing the audio files within the mysql database
would be REALLY slow... :-)

I remember seeing a bounty for some type of Voicemail/IMAP integration
but I'm not sure if anyone has furthered it. that would be a neat way
to handle things since you could use an existing IMAP mail server
setup, or build one and scale/cluster however you wish an basically
use the IMAP mail system as the storage medium for your users voice
mail. So the Voicemail's would also be email's and one could use the
IMAP client on their desktop to also manage their messages. :-)

Also could be done using webdav/fuse etc..

One could also use multiple icecast servers spread around and have
liquidsoap sit in front of them so that when you go to record the
message it hits liquidsoap locally where its configured to feed the
single message out to multiple shoutcast servers spread around the
globe in different data centers or something.

This one is deff worth checking into tho....

http://www.automatthew.com/2007/12/amazon-simpledb-and-couchdb-compared.html


Late 2 cents...
Cheers,
B



On Sat, Oct 9, 2010 at 3:10 AM, Brett Woollum <brett at woollum.com> wrote:
> Is there a way to store voicemail to a MySQL cluster like Asterisk does?
> This is one of the issues I have been trying to find an answer for, but I
> haven't been able to. It would be great if there were a LUA/Javascript
> command that would stream audio to the call directly from the database, or
> the other way around. This would be useful for voicemail recordings, IVR
> prompts, etc, when running several clustered FS servers. I have a MySQL
> Cluster spread across each of my FS servers that should be able to handle
> the additional load. Using MySQL cluster as an on-demand audio storage
> system would make access to the audio files fault-tolerant and distributed
> (even to other servers such as a web server for web voicemail playback).
>
> Any ideas on how this could be achieved?
>
> Brett Woollum
> Brett at Woollum.com
>
>
> ----- Original Message -----
> From: "Yehavi Bourvine" <yehavi.bourvine at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Saturday, October 9, 2010 12:49:51 AM GMT -08:00 US/Canada Pacific
> Subject: Re: [Freeswitch-users] Sharing storage between servers
>
> Hi,
>
> We use a NAS server to share the voicemail between two servers (one is FS,
> the other is WEB interface we wrote to handle voicemail via WEB).
> For the database: we use MySQL with replication.
>
> Regards, __Yehavi:
>
> 2010/10/9 Jody Rudolph <jody.rudolph at gmail.com>
>>
>> I am curious as to just how far you can take sharing disk storage for the
>> purpose of clustering. Is anyone doing this with the voicemail storage
>> directories? Is it possible to share the SQLite internal databases to 
>> avoid
>> resorting to ODBC? I realize that isn't likely, but I have access to some
>> high performance SAN hardware and want to take the most advantage 
>> possible.
>>
>>
>> Thanks,
>> Jody Rudolph
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 



From melkybes at mail.ru  Mon Jan 17 19:20:09 2011
From: melkybes at mail.ru (=?koi8-r?Q?=ED=C9=C8=C1=C9=CC_=F3=C1=CC=D4=C1=CE=CF=D7?=)
Date: Mon, 17 Jan 2011 19:20:09 +0300
Subject: [Freeswitch-users] =?koi8-r?b?NC13YXkgYW5kIG1vcmUgY29uZmVyZW5j?=
	=?koi8-r?b?ZSAmIGF0dF94ZmVy?=
In-Reply-To: <AANLkTiX+q=hJoBxoW52rpaLo9MhKFOKEKfGiHxJ50L_@mail.gmail.com>
References: <E1PepSw-0005HO-00.melkybes-mail-ru@f64.mail.ru>
	<AANLkTiX+q=hJoBxoW52rpaLo9MhKFOKEKfGiHxJ50L_@mail.gmail.com>
Message-ID: <E1PeroP-0003iV-00.melkybes-mail-ru@f79.mail.ru>

Can I modify att_xfer limits ?


From saeedahmad1981 at gmail.com  Mon Jan 17 19:25:06 2011
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Mon, 17 Jan 2011 17:25:06 +0100
Subject: [Freeswitch-users] Another: xml_curl vs Lua
In-Reply-To: <AANLkTikGqn_oXe56BZuqZK==aF=mSJQS7sfgrQJc2Gv_@mail.gmail.com>
References: <AANLkTik8RRmhN+-LYAG6o-ssuf4RYEYrine8Jj9s21PK@mail.gmail.com>
	<AANLkTinq=Sh2wQO+0YTF9V5mM51mYFVArd+LE5Dgi2Q-@mail.gmail.com>
	<AANLkTikGqn_oXe56BZuqZK==aF=mSJQS7sfgrQJc2Gv_@mail.gmail.com>
Message-ID: <AANLkTinXR4MCzX9t4jthjH9h0zo2ZKw1NvzRGQavvmR4@mail.gmail.com>

Thanks Dome for a prompt reply.

can you describe a bit about your hardware and structure? what number of cps
you are able to achieve?

Thanks

On Sun, Jan 16, 2011 at 3:16 PM, dome at tel.co.th <dome at tel.co.th> wrote:

> xml_curl -> app server
> my app server use nginx + luajit + tokyotylant
>
>
> Dome C.
>
>
> 2011/1/16 Saeed Ahmed <saeedahmad1981 at gmail.com>:
> > i want to handle 2000 concurrent calls and minimum 100 cps.
> > possible with lua? or should i stay with xml_curl?
> >
> > On Sun, Jan 16, 2011 at 1:24 PM, Saeed Ahmed <saeedahmad1981 at gmail.com>
> > wrote:
> >>
> >> Dear all,
> >>
> >> i know that lua is preferable by FS devs and community, but here i want
> to
> >> ask questions particular to my use case
> >> 1. Current Setup
> >> 1.1 currently i am using xml_curl - for dialplan only
> >> 1.2 xml_curl.conf has two bindings so i am safe if first one dies
> >> 1.3 i am using apache + mono (.net) + mysql on another server (other
> than
> >> FS server)
> >> 1.4 Intel(R) Xeon(R) CPU  E5420  @ 2.50GHz 8 core with 8 gm ram (for
> both
> >> FS and apache+mysql server)
> >> 2. Call Life Cycle
> >> 2.1. call comes on internal profile
> >> 2.2. xml_curl ask xml from webserver
> >> 2.3 i do database query for each call and, return back possible
> >> supplier(s) based on dialed number and customer id
> >> 2.4 i return xml to bridge the call on external profile
> >> Questiosn & Concerns:
> >>
> >> 1. So with above setup i am not able to reach more than 60 cps (most
> >> probably issue with my xml_curl backend), so i am thinking to try LUA
> >> 2. but one point is clicking my mind that, in case of xml_curl the
> >> webserver can be on external server, and xml_curl conf could have
> primary
> >> and backup binding, can i achieve that in LUA too?
> >> 3. Is LUA still be usefull as i am not doing any ivr etc.. its just xml
> >> conf which i have to return.
> >> 4. xml_curl based setup has good feature that i can serve  more FS
> >> servers, can i also do it in LUA?
> >>
> >> Please ask me if i miss some information
> >>
> >> Thanks
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jmesquita at freeswitch.org  Mon Jan 17 19:28:02 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 17 Jan 2011 13:28:02 -0300
Subject: [Freeswitch-users] 4-way and more conference & att_xfer
In-Reply-To: <E1PeroP-0003iV-00.melkybes-mail-ru@f79.mail.ru>
References: <E1PepSw-0005HO-00.melkybes-mail-ru@f64.mail.ru>
	<AANLkTiX+q=hJoBxoW52rpaLo9MhKFOKEKfGiHxJ50L_@mail.gmail.com>
	<E1PeroP-0003iV-00.melkybes-mail-ru@f79.mail.ru>
Message-ID: <AANLkTinij0Mh66WM1VH4X3x9VK3WXkTLtHpKEgVtO=0v@mail.gmail.com>

This is opensource, you can do whatever you want with the source code. The
thing is that if you modify att_xfer to mix more audio streams, won't it
become what mod_conference was made for?

Regards,
Jo?o Mesquita


On Mon, Jan 17, 2011 at 1:20 PM, ?????? ???????? <melkybes at mail.ru> wrote:

> Can I modify att_xfer limits ?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From jmesquita at freeswitch.org  Mon Jan 17 19:26:52 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 17 Jan 2011 13:26:52 -0300
Subject: [Freeswitch-users] Sharing storage between servers
In-Reply-To: <14AEC4AAA32849AC8E9A4F9CE874651A@e1705>
References: <AANLkTi=8kH5-c8pL1K_KRsNBo2gEO4y+24iwCk=AkJ3D@mail.gmail.com>
	<760758090.3041286611849227.JavaMail.root@mail>
	<AANLkTimC5h5Q3Rx-2s1k9RvoJ-r9iWVgLVeV0-Oga4K+@mail.gmail.com>
	<14AEC4AAA32849AC8E9A4F9CE874651A@e1705>
Message-ID: <AANLkTi=eMHm37HSoorJ1J08MPUH=XUEnDFYPq2aL-4NG@mail.gmail.com>

I've used OCFS2 in the past and even though there wasn't a production system
and there wasn't a lot of volume, it was pretty stable and easy to setup.


Jo?o Mesquita


On Mon, Jan 17, 2011 at 1:15 PM, Madovsky <infos at madovsky.org> wrote:

> you can also simply use files in folder and replicate it
> to all of your nodes every 3mn...
> store binary in DB is more slow as in a folder
>
> ----- Original Message -----
> From: "broken dash" <brokendash at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, January 17, 2011 2:24 AM
> Subject: Re: [Freeswitch-users] Sharing storage between servers
>
>
> Your question is also incredibly complex depending on the methods you
> could use as well. Storing the audio files within the mysql database
> would be REALLY slow... :-)
>
> I remember seeing a bounty for some type of Voicemail/IMAP integration
> but I'm not sure if anyone has furthered it. that would be a neat way
> to handle things since you could use an existing IMAP mail server
> setup, or build one and scale/cluster however you wish an basically
> use the IMAP mail system as the storage medium for your users voice
> mail. So the Voicemail's would also be email's and one could use the
> IMAP client on their desktop to also manage their messages. :-)
>
> Also could be done using webdav/fuse etc..
>
> One could also use multiple icecast servers spread around and have
> liquidsoap sit in front of them so that when you go to record the
> message it hits liquidsoap locally where its configured to feed the
> single message out to multiple shoutcast servers spread around the
> globe in different data centers or something.
>
> This one is deff worth checking into tho....
>
>
> http://www.automatthew.com/2007/12/amazon-simpledb-and-couchdb-compared.html
>
>
> Late 2 cents...
> Cheers,
> B
>
>
>
> On Sat, Oct 9, 2010 at 3:10 AM, Brett Woollum <brett at woollum.com> wrote:
> > Is there a way to store voicemail to a MySQL cluster like Asterisk does?
> > This is one of the issues I have been trying to find an answer for, but I
> > haven't been able to. It would be great if there were a LUA/Javascript
> > command that would stream audio to the call directly from the database,
> or
> > the other way around. This would be useful for voicemail recordings, IVR
> > prompts, etc, when running several clustered FS servers. I have a MySQL
> > Cluster spread across each of my FS servers that should be able to handle
> > the additional load. Using MySQL cluster as an on-demand audio storage
> > system would make access to the audio files fault-tolerant and
> distributed
> > (even to other servers such as a web server for web voicemail playback).
> >
> > Any ideas on how this could be achieved?
> >
> > Brett Woollum
> > Brett at Woollum.com
> >
> >
> > ----- Original Message -----
> > From: "Yehavi Bourvine" <yehavi.bourvine at gmail.com>
> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > Sent: Saturday, October 9, 2010 12:49:51 AM GMT -08:00 US/Canada Pacific
> > Subject: Re: [Freeswitch-users] Sharing storage between servers
> >
> > Hi,
> >
> > We use a NAS server to share the voicemail between two servers (one is
> FS,
> > the other is WEB interface we wrote to handle voicemail via WEB).
> > For the database: we use MySQL with replication.
> >
> > Regards, __Yehavi:
> >
> > 2010/10/9 Jody Rudolph <jody.rudolph at gmail.com>
> >>
> >> I am curious as to just how far you can take sharing disk storage for
> the
> >> purpose of clustering. Is anyone doing this with the voicemail storage
> >> directories? Is it possible to share the SQLite internal databases to
> >> avoid
> >> resorting to ODBC? I realize that isn't likely, but I have access to
> some
> >> high performance SAN hardware and want to take the most advantage
> >> possible.
> >>
> >>
> >> Thanks,
> >> Jody Rudolph
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________ FreeSWITCH-users mailing
> > list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From moises.silva at gmail.com  Mon Jan 17 19:40:27 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 17 Jan 2011 11:40:27 -0500
Subject: [Freeswitch-users] Caller-id on incoming FXO with freetdm
In-Reply-To: <AANLkTi=TujU1976wLCABzsfG2w=SvDQW1p2XEXfRniew@mail.gmail.com>
References: <AANLkTi=TujU1976wLCABzsfG2w=SvDQW1p2XEXfRniew@mail.gmail.com>
Message-ID: <AANLkTimh1Ow=0S1DP3fyep=UDcuFakA_fPJ4QYxipGvu@mail.gmail.com>

On Mon, Jan 17, 2011 at 6:56 AM, guru singh <grsingh750 at gmail.com> wrote:

> Hi,
>
> I am unable to get caller-id for incoming FXO calls.
> It is always being set to 0000000000
> I'm using FreeTDM and am on the latest git. I've set enable_callerid
> param to true in freetdm.conf.xml
> I can see fs pickup the callerid but then after each digit it says
> (debug=0), is this normal?
> Here is call debug log in fs for answering fxo and running the info
> app. http://pastebin.freeswitch.org/15037
> What am I missing?
>

Hello,

The debug=0 line is normal when DTMF detection is enabled but DTMF debugging
is not.

However, since you are getting caller id, it should not be enqueuing DTMF
(since currently we do not support DTMF caller id).

Are the DTMF digits that you see detected the correct caller id?

If they are correct, then this looks like an interesting coincidence where a
bug actually mostly got your DTMF caller id to work, however some
adjustments would be required to completely support caller id by DTMF.

Which country is this line from?

Is your caller id sent from the telco (or switch?) via DTMF?

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com




> thanks
> guru
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From infos at madovsky.org  Mon Jan 17 19:50:16 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 17 Jan 2011 11:50:16 -0500
Subject: [Freeswitch-users] Sharing storage between servers
References: <AANLkTi=8kH5-c8pL1K_KRsNBo2gEO4y+24iwCk=AkJ3D@mail.gmail.com><760758090.3041286611849227.JavaMail.root@mail><AANLkTimC5h5Q3Rx-2s1k9RvoJ-r9iWVgLVeV0-Oga4K+@mail.gmail.com><14AEC4AAA32849AC8E9A4F9CE874651A@e1705>
	<AANLkTi=eMHm37HSoorJ1J08MPUH=XUEnDFYPq2aL-4NG@mail.gmail.com>
Message-ID: <200589E71B5546CE82AB1B128CE6AD3F@e1705>

I used too in experimental state,
but I noticed in case of network node problem OCFS2 locked
with big delay on other nodes. However it's stable.
  ----- Original Message ----- 
  From: Jo?o Mesquita 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 17, 2011 11:26 AM
  Subject: Re: [Freeswitch-users] Sharing storage between servers


  I've used OCFS2 in the past and even though there wasn't a production system and there wasn't a lot of volume, it was pretty stable and easy to setup.



  Jo?o Mesquita



  On Mon, Jan 17, 2011 at 1:15 PM, Madovsky <infos at madovsky.org> wrote:

    you can also simply use files in folder and replicate it
    to all of your nodes every 3mn...
    store binary in DB is more slow as in a folder


    ----- Original Message -----
    From: "broken dash" <brokendash at gmail.com>
    To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

    Sent: Monday, January 17, 2011 2:24 AM
    Subject: Re: [Freeswitch-users] Sharing storage between servers


    Your question is also incredibly complex depending on the methods you
    could use as well. Storing the audio files within the mysql database
    would be REALLY slow... :-)

    I remember seeing a bounty for some type of Voicemail/IMAP integration
    but I'm not sure if anyone has furthered it. that would be a neat way
    to handle things since you could use an existing IMAP mail server
    setup, or build one and scale/cluster however you wish an basically
    use the IMAP mail system as the storage medium for your users voice
    mail. So the Voicemail's would also be email's and one could use the
    IMAP client on their desktop to also manage their messages. :-)

    Also could be done using webdav/fuse etc..

    One could also use multiple icecast servers spread around and have
    liquidsoap sit in front of them so that when you go to record the
    message it hits liquidsoap locally where its configured to feed the
    single message out to multiple shoutcast servers spread around the
    globe in different data centers or something.

    This one is deff worth checking into tho....

    http://www.automatthew.com/2007/12/amazon-simpledb-and-couchdb-compared.html


    Late 2 cents...
    Cheers,
    B



    On Sat, Oct 9, 2010 at 3:10 AM, Brett Woollum <brett at woollum.com> wrote:
    > Is there a way to store voicemail to a MySQL cluster like Asterisk does?
    > This is one of the issues I have been trying to find an answer for, but I
    > haven't been able to. It would be great if there were a LUA/Javascript
    > command that would stream audio to the call directly from the database, or
    > the other way around. This would be useful for voicemail recordings, IVR
    > prompts, etc, when running several clustered FS servers. I have a MySQL
    > Cluster spread across each of my FS servers that should be able to handle
    > the additional load. Using MySQL cluster as an on-demand audio storage
    > system would make access to the audio files fault-tolerant and distributed
    > (even to other servers such as a web server for web voicemail playback).
    >
    > Any ideas on how this could be achieved?
    >
    > Brett Woollum
    > Brett at Woollum.com
    >
    >
    > ----- Original Message -----
    > From: "Yehavi Bourvine" <yehavi.bourvine at gmail.com>
    > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
    > Sent: Saturday, October 9, 2010 12:49:51 AM GMT -08:00 US/Canada Pacific
    > Subject: Re: [Freeswitch-users] Sharing storage between servers
    >
    > Hi,
    >
    > We use a NAS server to share the voicemail between two servers (one is FS,
    > the other is WEB interface we wrote to handle voicemail via WEB).
    > For the database: we use MySQL with replication.
    >
    > Regards, __Yehavi:
    >
    > 2010/10/9 Jody Rudolph <jody.rudolph at gmail.com>
    >>
    >> I am curious as to just how far you can take sharing disk storage for the
    >> purpose of clustering. Is anyone doing this with the voicemail storage
    >> directories? Is it possible to share the SQLite internal databases to
    >> avoid
    >> resorting to ODBC? I realize that isn't likely, but I have access to some
    >> high performance SAN hardware and want to take the most advantage
    >> possible.
    >>
    >>
    >> Thanks,
    >> Jody Rudolph
    >>
    >> _______________________________________________
    >> FreeSWITCH-users mailing list
    >> FreeSWITCH-users at lists.freeswitch.org
    >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    >> http://www.freeswitch.org
    >>
    >
    >
    > _______________________________________________ FreeSWITCH-users mailing
    > list FreeSWITCH-users at lists.freeswitch.org
    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > http://www.freeswitch.org
    > _______________________________________________
    > FreeSWITCH-users mailing list
    > FreeSWITCH-users at lists.freeswitch.org
    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > http://www.freeswitch.org
    >
    >

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





------------------------------------------------------------------------------


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From steve.d.ward at gmail.com  Mon Jan 17 20:00:13 2011
From: steve.d.ward at gmail.com (S W)
Date: Mon, 17 Jan 2011 12:00:13 -0500
Subject: [Freeswitch-users] BLF with Valet Parking
In-Reply-To: <31AC9159-35FD-450F-9E9A-E648AA99AF8B@freeswitch.org>
References: <5E3329D2-286C-4F3E-9F04-F11F5CAACF6B@5ninesolutions.com>
	<AANLkTimkN-bdsRxxR8RhaKams=apiHLeRt9CPf42wvXJ@mail.gmail.com>
	<31AC9159-35FD-450F-9E9A-E648AA99AF8B@freeswitch.org>
Message-ID: <AANLkTim6JVF6Db8PYMH0dVtOAa4Lg=XAUQ8py-vUP9Uq@mail.gmail.com>

Travis:

Here is something that might interest you.  I use 509gs, and I have
some of them set up with BLF/speeddials set up on "parking slots".  If
no call is in the slot, you just press the button to park the call.
If you want to retrieve a call from the slot, press that same button.
Following is the config I use to be able to park calls and have
BLF/speed dials.

In autoconf/fifo.conf.xml :
-------------------------------------
<fifo name="ParkingSlot1@$${domain}" importance="0"/>
-------------------------------------

In dialplan, e.g. 100_parking_slots.xml :

? <extension name="parking_slots">
?? <condition field="destination_number"
expression="^(ParkingSlot\d+)$" break="on-false">
???? <!-- Get count of callers in the specified parking slot (fifo) -->
???? <action inline="true" application="set"
data="slot_count=${fifo(count $1@$${domain})}"/>
???? <action inline="true" application="set"
data="slot_count=${slot_count:-3:2}"/>
?? </condition>
?? <condition field="${slot_count}" expression="^\:0$" break="always">
???? <!-- For FIFO count of 0: place the caller into the FIFO (actions
are??? -->
???? <!-- executed).
                     ??????? -->
???? <!-- fifo settings for parking fifos -->
???? <action application="unset" data="fifo_chime_list"/>
???? <action application="set" data="fifo_chime_freq=0"/>
     <!-- Parked callers listen to MOH -->
???? <action application="fifo" data="${destination_number}@$${domain}
in undef local_stream://moh"/>
     <!-- This hangup should never happen. -->
???? <action application="hangup"/>
???? <!-- For non-zero FIFO count: connect the caller to the parked
channel?? -->
???? <!-- (remove the channel from the FIFO and bridge it to the
caller using -->
???? <!-- fifo ... out)
                  ??????????????? -->
???? <anti-action application="fifo"
data="${destination_number}@$${domain} out nowait"/>
     <!-- This hangup should never happen -->
???? <anti-action application="hangup"/>
?? </condition>
</extension>

And here is an example SPA509g config (I actually use this):

fnc=blf+sd;sub=ParkingSlot1@$PROXY;ext=ParkingSlot1@$PROXY

Note that, above, my FreeSWITCH box that runs the fifo code above is
the $PROXY to which my 509g is also registered, etc.

Let me know if you have any questions on that; I hope it helps.


On Mon, Jan 17, 2011 at 10:29 AM, Brian West <brian at freeswitch.org> wrote:
>
> I think he wants BLF status of a lot on a key. ?That I don't think is there.
> /b
> On Jan 17, 2011, at 9:01 AM, Travis Kemen wrote:
>
> BLF works fine here with valet parking using polycom/snom phones.
>
> Travis
>
> On Sun, Jan 16, 2011 at 6:15 PM, Spencer Thomason?<spencer at 5ninesolutions.com>?wrote:
>>
>> Hello all,
>> Is it possible to use BLFs with valet parking? ?Basically I have
>> several Linksys SPA 509G phones and after a call is parked, I'd like
>> use use a speed dial/BLF key on the phone to pickup the call. ?I do
>> have presence configured and working with the extensions.
>>
>> Thanks,
>> Spencer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From anthony.minessale at gmail.com  Mon Jan 17 20:07:42 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 17 Jan 2011 11:07:42 -0600
Subject: [Freeswitch-users] BLF with Valet Parking
In-Reply-To: <AANLkTim6JVF6Db8PYMH0dVtOAa4Lg=XAUQ8py-vUP9Uq@mail.gmail.com>
References: <5E3329D2-286C-4F3E-9F04-F11F5CAACF6B@5ninesolutions.com>
	<AANLkTimkN-bdsRxxR8RhaKams=apiHLeRt9CPf42wvXJ@mail.gmail.com>
	<31AC9159-35FD-450F-9E9A-E648AA99AF8B@freeswitch.org>
	<AANLkTim6JVF6Db8PYMH0dVtOAa4Lg=XAUQ8py-vUP9Uq@mail.gmail.com>
Message-ID: <AANLkTi=BXXPovnFGfETomdwc3gwQwuNZ==ieVdycPme4@mail.gmail.com>

yes using fifo for parking is the more elegant solution.
the valet_parking is more for people who miss the asterisk flavored parking.


On Mon, Jan 17, 2011 at 11:00 AM, S W <steve.d.ward at gmail.com> wrote:
> Travis:
>
> Here is something that might interest you. ?I use 509gs, and I have
> some of them set up with BLF/speeddials set up on "parking slots". ?If
> no call is in the slot, you just press the button to park the call.
> If you want to retrieve a call from the slot, press that same button.
> Following is the config I use to be able to park calls and have
> BLF/speed dials.
>
> In autoconf/fifo.conf.xml :
> -------------------------------------
> <fifo name="ParkingSlot1@$${domain}" importance="0"/>
> -------------------------------------
>
> In dialplan, e.g. 100_parking_slots.xml :
>
> ? <extension name="parking_slots">
> ?? <condition field="destination_number"
> expression="^(ParkingSlot\d+)$" break="on-false">
> ???? <!-- Get count of callers in the specified parking slot (fifo) -->
> ???? <action inline="true" application="set"
> data="slot_count=${fifo(count $1@$${domain})}"/>
> ???? <action inline="true" application="set"
> data="slot_count=${slot_count:-3:2}"/>
> ?? </condition>
> ?? <condition field="${slot_count}" expression="^\:0$" break="always">
> ???? <!-- For FIFO count of 0: place the caller into the FIFO (actions
> are??? -->
> ???? <!-- executed).
> ? ? ? ? ? ? ? ? ? ? ??????? -->
> ???? <!-- fifo settings for parking fifos -->
> ???? <action application="unset" data="fifo_chime_list"/>
> ???? <action application="set" data="fifo_chime_freq=0"/>
> ? ? <!-- Parked callers listen to MOH -->
> ???? <action application="fifo" data="${destination_number}@$${domain}
> in undef local_stream://moh"/>
> ? ? <!-- This hangup should never happen. -->
> ???? <action application="hangup"/>
> ???? <!-- For non-zero FIFO count: connect the caller to the parked
> channel?? -->
> ???? <!-- (remove the channel from the FIFO and bridge it to the
> caller using -->
> ???? <!-- fifo ... out)
> ? ? ? ? ? ? ? ? ???????????????? -->
> ???? <anti-action application="fifo"
> data="${destination_number}@$${domain} out nowait"/>
> ? ? <!-- This hangup should never happen -->
> ???? <anti-action application="hangup"/>
> ?? </condition>
> </extension>
>
> And here is an example SPA509g config (I actually use this):
>
> fnc=blf+sd;sub=ParkingSlot1@$PROXY;ext=ParkingSlot1@$PROXY
>
> Note that, above, my FreeSWITCH box that runs the fifo code above is
> the $PROXY to which my 509g is also registered, etc.
>
> Let me know if you have any questions on that; I hope it helps.
>
>
> On Mon, Jan 17, 2011 at 10:29 AM, Brian West <brian at freeswitch.org> wrote:
>>
>> I think he wants BLF status of a lot on a key. ?That I don't think is there.
>> /b
>> On Jan 17, 2011, at 9:01 AM, Travis Kemen wrote:
>>
>> BLF works fine here with valet parking using polycom/snom phones.
>>
>> Travis
>>
>> On Sun, Jan 16, 2011 at 6:15 PM, Spencer Thomason?<spencer at 5ninesolutions.com>?wrote:
>>>
>>> Hello all,
>>> Is it possible to use BLFs with valet parking? ?Basically I have
>>> several Linksys SPA 509G phones and after a call is parked, I'd like
>>> use use a speed dial/BLF key on the phone to pickup the call. ?I do
>>> have presence configured and working with the extensions.
>>>
>>> Thanks,
>>> Spencer
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From infos at madovsky.org  Mon Jan 17 20:20:50 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 17 Jan 2011 12:20:50 -0500
Subject: [Freeswitch-users] bridge with port
Message-ID: <9A03A8F0F5634BF5A2AB0FA8026C2EF5@e1705>

is it possible to provide the port for a bridge like

<action application="bridge" data="sofia/internal/user123 at domain:5080"/> ???

for now the answer is "MANDATORY_IE_MISSING".

Thanks
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From kbdfck at gmail.com  Mon Jan 17 15:19:48 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Mon, 17 Jan 2011 15:19:48 +0300
Subject: [Freeswitch-users] MOH while running bind_digit_action or
	bind_meta_app
Message-ID: <AANLkTimMemLGM9KZ4q8-MP0neeff0WQqpf_7SvgSp8Zn@mail.gmail.com>

Hi!

I faced strange problem. After running bind_meta_app or
bind_digit_action with att_xfer transferee listens to MOH, but when
transferer cancels xfer with #, there are no music on hold for
transferee on second run of att_xfer . It doesn't even try to launch
playback of MOH.

I use local channel in att_xfer to allow attended transfer to
arbitrary extensions, so I had to add
loopback_bowout/bowout_on_execute=false variables, app_xfer did'nt
work as expected at all without them. Could this cause such a strange
MOH behavior?

first try:

2011-01-17 15:09:34.044500 [DEBUG] switch_rtp.c:3098 RTP RECV DTMF *:884
2011-01-17 15:09:34.264674 [DEBUG] switch_rtp.c:3098 RTP RECV DTMF 7:804
2011-01-17 15:09:34.264674 [DEBUG] switch_ivr_async.c:2866
sofia/local/sytchev2 at 85.114.2.200 Processing meta digit '7'
[execute_extension::att_xfer XML features]
2011-01-17 15:09:34.264674 [DEBUG] switch_core_session.c:954 Send
signal sofia/local/sytchev2 at 85.114.2.200 [BREAK]
2011-01-17 15:09:34.264674 [DEBUG] switch_core_session.c:709 Send
signal sofia/local/sytchev2 at 85.114.2.200 [BREAK]
2011-01-17 15:09:34.404746 [DEBUG] switch_core_session.c:954 Send
signal sofia/local/sip:sytchev3 at 192.168.4.130 [BREAK]
2011-01-17 15:09:34.424718 [DEBUG] switch_core_session.c:709 Send
signal sofia/local/sip:sytchev3 at 192.168.4.130 [BREAK]
2011-01-17 15:09:34.564684 [DEBUG] switch_ivr.c:563
sofia/local/sip:sytchev3 at 192.168.4.130 Command Execute
playback(local_stream://moh)
EXECUTE sofia/local/sip:sytchev3 at 192.168.4.130 playback(local_stream://moh)
2011-01-17 15:09:34.564684 [WARNING] mod_local_stream.c:393 Unknown
source moh, trying 'default'
2011-01-17 15:09:34.564684 [DEBUG] mod_local_stream.c:421 Opening
Stream [default] 8000hz
2011-01-17 15:09:34.564684 [DEBUG] switch_ivr_play_say.c:1236 Codec
Activated L16 at 8000hz 1 channels 20ms
2011-01-17 15:09:34.564684 [DEBUG] switch_ivr.c:563
sofia/local/sytchev2 at 85.114.2.200 Command Execute
execute_extension(att_xfer XML features)
EXECUTE sofia/local/sytchev2 at 85.114.2.200 execute_extension(att_xfer
XML features)
2011-01-17 15:09:34.565718 [INFO] mod_dialplan_xml.c:331 Processing
8126778008 <sytchev2>->att_xfer in context features
Dialplan: sofia/local/sytchev2 at 85.114.2.200 parsing [features->dx]
continue=false
Dialplan: sofia/local/sytchev2 at 85.114.2.200 Regex (FAIL) [dx]
destination_number(att_xfer) =~ /^dx$/ break=on-false
Dialplan: sofia/local/sytchev2 at 85.114.2.200 parsing
[features->att_xfer] continue=false
Dialplan: sofia/local/sytchev2 at 85.114.2.200 Regex (PASS) [att_xfer]
destination_number(att_xfer) =~ /^att_xfer$/ break=on-false
Dialplan: sofia/local/sytchev2 at 85.114.2.200 Action read(3 11
'tone_stream://%(10000,0,350,440)' digits 30000 #)
Dialplan: sofia/local/sytchev2 at 85.114.2.200 Action
set(origination_cancel_key=#)
Dialplan: sofia/local/sytchev2 at 85.114.2.200 Action
att_xfer(loopback/${digits}/route_customer_global)
2011-01-17 15:09:34.565718 [NOTICE] switch_core_session.c:2137 Execute
read(3 11 'tone_stream://%(10000,0,350,440)' digits 30000 #)
EXECUTE sofia/local/sytchev2 at 85.114.2.200 read(3 11
'tone_stream://%(10000,0,350,440)' digits 30000 #)
2011-01-17 15:09:34.567713 [DEBUG] switch_ivr_play_say.c:1236 Codec
Activated L16 at 8000hz 1 channels 20ms
2011-01-17 15:09:44.564756 [DEBUG] switch_ivr_play_say.c:1573 done playing file


Second try:

2011-01-17 15:10:10.643923 [DEBUG] switch_rtp.c:3098 RTP RECV DTMF *:804
2011-01-17 15:10:10.964291 [DEBUG] switch_rtp.c:3098 RTP RECV DTMF 7:804
2011-01-17 15:10:10.964291 [DEBUG] switch_ivr_async.c:2866
sofia/local/sytchev2 at 85.114.2.200 Processing meta digit '7'
[execute_extension::att_xfer XML features]
2011-01-17 15:10:10.964291 [DEBUG] switch_core_session.c:954 Send
signal sofia/local/sytchev2 at 85.114.2.200 [BREAK]
2011-01-17 15:10:10.964291 [DEBUG] switch_core_session.c:709 Send
signal sofia/local/sytchev2 at 85.114.2.200 [BREAK]
2011-01-17 15:10:11.104279 [DEBUG] switch_ivr.c:563
sofia/local/sytchev2 at 85.114.2.200 Command Execute
execute_extension(att_xfer XML features)
EXECUTE sofia/local/sytchev2 at 85.114.2.200 execute_extension(att_xfer
XML features)
2011-01-17 15:10:11.104279 [INFO] mod_dialplan_xml.c:331 Processing
8126778008 <sytchev2>->att_xfer in context features
Dialplan: sofia/local/sytchev2 at 85.114.2.200 parsing [features->dx]
continue=false
Dialplan: sofia/local/sytchev2 at 85.114.2.200 Regex (FAIL) [dx]
destination_number(att_xfer) =~ /^dx$/ break=on-false
Dialplan: sofia/local/sytchev2 at 85.114.2.200 parsing
[features->att_xfer] continue=false
Dialplan: sofia/local/sytchev2 at 85.114.2.200 Regex (PASS) [att_xfer]
destination_number(att_xfer) =~ /^att_xfer$/ break=on-false
Dialplan: sofia/local/sytchev2 at 85.114.2.200 Action read(3 11
'tone_stream://%(10000,0,350,440)' digits 30000 #)
Dialplan: sofia/local/sytchev2 at 85.114.2.200 Action
set(origination_cancel_key=#)
Dialplan: sofia/local/sytchev2 at 85.114.2.200 Action
att_xfer(loopback/${digits}/route_customer_global)
2011-01-17 15:10:11.104279 [NOTICE] switch_core_session.c:2137 Execute
read(3 11 'tone_stream://%(10000,0,350,440)' digits 30000 #)
EXECUTE sofia/local/sytchev2 at 85.114.2.200 read(3 11
'tone_stream://%(10000,0,350,440)' digits 30000 #)
2011-01-17 15:10:11.106278 [DEBUG] switch_ivr_play_say.c:1236 Codec
Activated L16 at 8000hz 1 channels 20ms
2011-01-17 15:10:21.104543 [DEBUG] switch_ivr_play_say.c:1573 done playing file


-- 
Best regards,

Dmitry Sytchev,
IT Engineer


From jerry.richards at teotech.com  Mon Jan 17 20:59:23 2011
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 17 Jan 2011 09:59:23 -0800
Subject: [Freeswitch-users] Simultaneous Ringing Of Internal Extension And
	PSTN Number
Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D69606600303C8@VA3DIAXVS351.RED001.local>

Hello All,

I would like a configuration that rings both an internal extension and a PSTN number (via an internal PRI), but the PRI immediately answers with PROGRESS_MEDIA (i.e. ringback), which cancels the call to the internal extension.  Is there a way to prevent Freeswitch from canceling the internal call?  I would like both endpoints to ring until one of them actually answers the call.

Here is an example of my bridge statement:

bridge({presence_id=2002 at 192.168.72.144}sofia/internal/sip:2002 at 192.168.72.22:5060;transport=udp,[lcr_carrier=Carrier / Location 1/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=4257402463]openzap/smg_prid/a/4259438201 at g1)

Thanks,
Jerry
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From david.ponzone at ipeva.fr  Mon Jan 17 21:11:16 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 17 Jan 2011 19:11:16 +0100
Subject: [Freeswitch-users] Simultaneous Ringing Of Internal Extension
	And PSTN Number
In-Reply-To: <2BF7FB90DF25EA4485949F3AF2B9D69606600303C8@VA3DIAXVS351.RED001.local>
References: <2BF7FB90DF25EA4485949F3AF2B9D69606600303C8@VA3DIAXVS351.RED001.local>
Message-ID: <55E90783-7B79-41BE-987D-8FAC17523C55@ipeva.fr>

ignore_early_media=true on the call to the PSTN

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 17/01/2011 ? 18:59, Jerry Richards a ?crit :

> Hello All,
>  
> I would like a configuration that rings both an internal extension and a PSTN number (via an internal PRI), but the PRI immediately answers with PROGRESS_MEDIA (i.e. ringback), which cancels the call to the internal extension.  Is there a way to prevent Freeswitch from canceling the internal call?  I would like both endpoints to ring until one of them actually answers the call.
>  
> Here is an example of my bridge statement:
>  
> bridge({presence_id=2002 at 192.168.72.144}sofia/internal/sip:2002 at 192.168.72.22:5060;transport=udp,[lcr_carrier=Carrier / Location 1/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=4257402463]openzap/smg_prid/a/4259438201 at g1)
>  
> Thanks,
> Jerry
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From avi at avimarcus.net  Mon Jan 17 21:18:16 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 17 Jan 2011 20:18:16 +0200
Subject: [Freeswitch-users] bridge with port
In-Reply-To: <9A03A8F0F5634BF5A2AB0FA8026C2EF5@e1705>
References: <9A03A8F0F5634BF5A2AB0FA8026C2EF5@e1705>
Message-ID: <AANLkTinCi1-SJApo25fspn+uQVAmAeHPaRnwf9MTqgxc@mail.gmail.com>

You're specifying "internal" profile as the endpoint, so no, specifying a
port would be unnecessary and wrong.
You can use the external profile to do that though..
-Avi

On Mon, Jan 17, 2011 at 7:20 PM, Madovsky <infos at madovsky.org> wrote:

>  is it possible to provide the port for a bridge like
>
> <action application="bridge" data="sofia/internal/user123 at domain:5080"/>
> ???
>
> for now the answer is "MANDATORY_IE_MISSING".
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rafonline at hotmail.com  Mon Jan 17 21:23:36 2011
From: rafonline at hotmail.com (Rafqat .)
Date: Mon, 17 Jan 2011 18:23:36 +0000
Subject: [Freeswitch-users] nibblebill no bal action
In-Reply-To: <AANLkTike0xWZqFMvOui_FuZCF9U1AF4seoRkGZYSEKnZ@mail.gmail.com>
References: <COL122-W14D36811E53F5598D6E855B5F50@phx.gbl>,
	<AANLkTike0xWZqFMvOui_FuZCF9U1AF4seoRkGZYSEKnZ@mail.gmail.com>
Message-ID: <COL122-W5926018AB7C33931561142B5F40@phx.gbl>



Thanks Rupa, that worked.

cheers

Raf

----------------------------------------
> Date: Sun, 16 Jan 2011 16:02:26 -0600
> From: rupa at rupa.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] nibblebill no bal action
>
> Try setting a channel var "nobal_target" to diff values in each leg.
> Then when the call is transfered to the no balalance amount extension
> transfer to the final nobal_target you defined up front.
>
> At least that is worth a shot....
>
> On Sun, Jan 16, 2011 at 3:36 PM, Rafqat .  wrote:
> >
> >
> > Hi,
> >
> > I would like leg A and leg B transferred to different extensions when the no balance amount is triggered.
> >
> > I have tried to set a different nobal_action in leg A's session to what I set in the dial string for leg b. But it seems whatever action i put into the dial string gets executed for leg A aswell as leg B.
> >
> > After having a look at the wiki, it seems that this is not currently supported.  Can any suggest any work arounds please?
> >
> >
> > Cheers
> >
> > Raf
> >
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  

From brian at freeswitch.org  Mon Jan 17 21:26:42 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Jan 2011 12:26:42 -0600
Subject: [Freeswitch-users] bridge with port
In-Reply-To: <AANLkTinCi1-SJApo25fspn+uQVAmAeHPaRnwf9MTqgxc@mail.gmail.com>
References: <9A03A8F0F5634BF5A2AB0FA8026C2EF5@e1705>
	<AANLkTinCi1-SJApo25fspn+uQVAmAeHPaRnwf9MTqgxc@mail.gmail.com>
Message-ID: <5F2C5AEB-A348-4DEF-9BB0-A53A6A5C1898@freeswitch.org>

You're getting that because the phone/endpoint is sending a challenge and we lack the knowledge on how to answer it.

/b

On Jan 17, 2011, at 12:18 PM, Avi Marcus wrote:

> You're specifying "internal" profile as the endpoint, so no, specifying a port would be unnecessary and wrong.
> You can use the external profile to do that though..
> -Avi
> 
> On Mon, Jan 17, 2011 at 7:20 PM, Madovsky <infos at madovsky.org> wrote:
> is it possible to provide the port for a bridge like
>  
> <action application="bridge" data="sofia/internal/user123 at domain:5080"/> ???
>  
> for now the answer is "MANDATORY_IE_MISSING".
>  
> Thanks
> 

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From spencer at 5ninesolutions.com  Mon Jan 17 22:02:23 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Mon, 17 Jan 2011 11:02:23 -0800
Subject: [Freeswitch-users] BLF with Valet Parking
In-Reply-To: <AANLkTi=BXXPovnFGfETomdwc3gwQwuNZ==ieVdycPme4@mail.gmail.com>
References: <5E3329D2-286C-4F3E-9F04-F11F5CAACF6B@5ninesolutions.com>
	<AANLkTimkN-bdsRxxR8RhaKams=apiHLeRt9CPf42wvXJ@mail.gmail.com>
	<31AC9159-35FD-450F-9E9A-E648AA99AF8B@freeswitch.org>
	<AANLkTim6JVF6Db8PYMH0dVtOAa4Lg=XAUQ8py-vUP9Uq@mail.gmail.com>
	<AANLkTi=BXXPovnFGfETomdwc3gwQwuNZ==ieVdycPme4@mail.gmail.com>
Message-ID: <45DC17EC-C795-42B0-B31C-CE70434297C9@5ninesolutions.com>

Thanks for the info.  Initially I was trying to mimic the Asterisk  
parking behavior and show the status of a call in a slot.  I'll rework  
this using fifo.

Spencer

On Jan 17, 2011, at 9:07 AM, Anthony Minessale wrote:

> yes using fifo for parking is the more elegant solution.
> the valet_parking is more for people who miss the asterisk flavored  
> parking.
>
>
> On Mon, Jan 17, 2011 at 11:00 AM, S W <steve.d.ward at gmail.com> wrote:
>> Travis:
>>
>> Here is something that might interest you.  I use 509gs, and I have
>> some of them set up with BLF/speeddials set up on "parking slots".   
>> If
>> no call is in the slot, you just press the button to park the call.
>> If you want to retrieve a call from the slot, press that same button.
>> Following is the config I use to be able to park calls and have
>> BLF/speed dials.
>>
>> In autoconf/fifo.conf.xml :
>> -------------------------------------
>> <fifo name="ParkingSlot1@$${domain}" importance="0"/>
>> -------------------------------------
>>
>> In dialplan, e.g. 100_parking_slots.xml :
>>
>>   <extension name="parking_slots">
>>    <condition field="destination_number"
>> expression="^(ParkingSlot\d+)$" break="on-false">
>>      <!-- Get count of callers in the specified parking slot (fifo)  
>> -->
>>      <action inline="true" application="set"
>> data="slot_count=${fifo(count $1@$${domain})}"/>
>>      <action inline="true" application="set"
>> data="slot_count=${slot_count:-3:2}"/>
>>    </condition>
>>    <condition field="${slot_count}" expression="^\:0$"  
>> break="always">
>>      <!-- For FIFO count of 0: place the caller into the FIFO  
>> (actions
>> are    -->
>>      <!-- executed).
>>                             -->
>>      <!-- fifo settings for parking fifos -->
>>      <action application="unset" data="fifo_chime_list"/>
>>      <action application="set" data="fifo_chime_freq=0"/>
>>     <!-- Parked callers listen to MOH -->
>>      <action application="fifo" data="${destination_number}@$$ 
>> {domain}
>> in undef local_stream://moh"/>
>>     <!-- This hangup should never happen. -->
>>      <action application="hangup"/>
>>      <!-- For non-zero FIFO count: connect the caller to the parked
>> channel   -->
>>      <!-- (remove the channel from the FIFO and bridge it to the
>> caller using -->
>>      <!-- fifo ... out)
>>                                  -->
>>      <anti-action application="fifo"
>> data="${destination_number}@$${domain} out nowait"/>
>>     <!-- This hangup should never happen -->
>>      <anti-action application="hangup"/>
>>    </condition>
>> </extension>
>>
>> And here is an example SPA509g config (I actually use this):
>>
>> fnc=blf+sd;sub=ParkingSlot1@$PROXY;ext=ParkingSlot1@$PROXY
>>
>> Note that, above, my FreeSWITCH box that runs the fifo code above is
>> the $PROXY to which my 509g is also registered, etc.
>>
>> Let me know if you have any questions on that; I hope it helps.
>>
>>
>> On Mon, Jan 17, 2011 at 10:29 AM, Brian West <brian at freeswitch.org>  
>> wrote:
>>>
>>> I think he wants BLF status of a lot on a key.  That I don't think  
>>> is there.
>>> /b
>>> On Jan 17, 2011, at 9:01 AM, Travis Kemen wrote:
>>>
>>> BLF works fine here with valet parking using polycom/snom phones.
>>>
>>> Travis
>>>
>>> On Sun, Jan 16, 2011 at 6:15 PM, Spencer Thomason <spencer at 5ninesolutions.com 
>>> > wrote:
>>>>
>>>> Hello all,
>>>> Is it possible to use BLFs with valet parking?  Basically I have
>>>> several Linksys SPA 509G phones and after a call is parked, I'd  
>>>> like
>>>> use use a speed dial/BLF key on the phone to pickup the call.  I do
>>>> have presence configured and working with the extensions.
>>>>
>>>> Thanks,
>>>> Spencer
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From gsaathoff at gmail.com  Mon Jan 17 20:48:21 2011
From: gsaathoff at gmail.com (Graham Saathoff)
Date: Mon, 17 Jan 2011 12:48:21 -0500
Subject: [Freeswitch-users] Cisco Java phones crash when nonce count is
	incremented past 1 -- any ideas?
Message-ID: <AANLkTikg+GY1tZ3n7VNQ8+7JOw=qy3Fkbz3zYFF2v4TF@mail.gmail.com>

Hello all,

I have several Cisco Java powered phones (7941, 7961, 7965) that repeatedly
drop calls.  The issue seems to be that once the phone attempts to update
the nonce count to 2, it crashes and disconnects the call.  Has anyone seen
this behavior before?  Any suggestions for how to fix this would be greatly
appreciated.

Thanks,
Graham
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From steve.d.ward at gmail.com  Mon Jan 17 22:30:44 2011
From: steve.d.ward at gmail.com (S W)
Date: Mon, 17 Jan 2011 14:30:44 -0500
Subject: [Freeswitch-users] BLF with Valet Parking
In-Reply-To: <45DC17EC-C795-42B0-B31C-CE70434297C9@5ninesolutions.com>
References: <5E3329D2-286C-4F3E-9F04-F11F5CAACF6B@5ninesolutions.com>
	<AANLkTimkN-bdsRxxR8RhaKams=apiHLeRt9CPf42wvXJ@mail.gmail.com>
	<31AC9159-35FD-450F-9E9A-E648AA99AF8B@freeswitch.org>
	<AANLkTim6JVF6Db8PYMH0dVtOAa4Lg=XAUQ8py-vUP9Uq@mail.gmail.com>
	<AANLkTi=BXXPovnFGfETomdwc3gwQwuNZ==ieVdycPme4@mail.gmail.com>
	<45DC17EC-C795-42B0-B31C-CE70434297C9@5ninesolutions.com>
Message-ID: <AANLkTi=7z_BtTOjZVCat7T1C70V+H=Roba+pdXs+MuXW@mail.gmail.com>

FYI, on the wiki there is a little further documentation on the config
I wrote up there in the email:

http://wiki.freeswitch.org/wiki/Park_%26_Retrieve

I've been using this code for quite a while.  The fact that fifo just
works solid in freeswitch is wonderful, to say the least.

On Mon, Jan 17, 2011 at 2:02 PM, Spencer Thomason
<spencer at 5ninesolutions.com> wrote:
> Thanks for the info. ?Initially I was trying to mimic the Asterisk
> parking behavior and show the status of a call in a slot. ?I'll rework
> this using fifo.
>
> Spencer
>
> On Jan 17, 2011, at 9:07 AM, Anthony Minessale wrote:
>
>> yes using fifo for parking is the more elegant solution.
>> the valet_parking is more for people who miss the asterisk flavored
>> parking.
>>
>>
>> On Mon, Jan 17, 2011 at 11:00 AM, S W <steve.d.ward at gmail.com> wrote:
>>> Travis:
>>>
>>> Here is something that might interest you. ?I use 509gs, and I have
>>> some of them set up with BLF/speeddials set up on "parking slots".
>>> If
>>> no call is in the slot, you just press the button to park the call.
>>> If you want to retrieve a call from the slot, press that same button.
>>> Following is the config I use to be able to park calls and have
>>> BLF/speed dials.
>>>
>>> In autoconf/fifo.conf.xml :
>>> -------------------------------------
>>> <fifo name="ParkingSlot1@$${domain}" importance="0"/>
>>> -------------------------------------
>>>
>>> In dialplan, e.g. 100_parking_slots.xml :
>>>
>>> ? <extension name="parking_slots">
>>> ? ?<condition field="destination_number"
>>> expression="^(ParkingSlot\d+)$" break="on-false">
>>> ? ? ?<!-- Get count of callers in the specified parking slot (fifo)
>>> -->
>>> ? ? ?<action inline="true" application="set"
>>> data="slot_count=${fifo(count $1@$${domain})}"/>
>>> ? ? ?<action inline="true" application="set"
>>> data="slot_count=${slot_count:-3:2}"/>
>>> ? ?</condition>
>>> ? ?<condition field="${slot_count}" expression="^\:0$"
>>> break="always">
>>> ? ? ?<!-- For FIFO count of 0: place the caller into the FIFO
>>> (actions
>>> are ? ?-->
>>> ? ? ?<!-- executed).
>>> ? ? ? ? ? ? ? ? ? ? ? ? ? ? -->
>>> ? ? ?<!-- fifo settings for parking fifos -->
>>> ? ? ?<action application="unset" data="fifo_chime_list"/>
>>> ? ? ?<action application="set" data="fifo_chime_freq=0"/>
>>> ? ? <!-- Parked callers listen to MOH -->
>>> ? ? ?<action application="fifo" data="${destination_number}@$$
>>> {domain}
>>> in undef local_stream://moh"/>
>>> ? ? <!-- This hangup should never happen. -->
>>> ? ? ?<action application="hangup"/>
>>> ? ? ?<!-- For non-zero FIFO count: connect the caller to the parked
>>> channel ? -->
>>> ? ? ?<!-- (remove the channel from the FIFO and bridge it to the
>>> caller using -->
>>> ? ? ?<!-- fifo ... out)
>>> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?-->
>>> ? ? ?<anti-action application="fifo"
>>> data="${destination_number}@$${domain} out nowait"/>
>>> ? ? <!-- This hangup should never happen -->
>>> ? ? ?<anti-action application="hangup"/>
>>> ? ?</condition>
>>> </extension>
>>>
>>> And here is an example SPA509g config (I actually use this):
>>>
>>> fnc=blf+sd;sub=ParkingSlot1@$PROXY;ext=ParkingSlot1@$PROXY
>>>
>>> Note that, above, my FreeSWITCH box that runs the fifo code above is
>>> the $PROXY to which my 509g is also registered, etc.
>>>
>>> Let me know if you have any questions on that; I hope it helps.
>>>
>>>
>>> On Mon, Jan 17, 2011 at 10:29 AM, Brian West <brian at freeswitch.org>
>>> wrote:
>>>>
>>>> I think he wants BLF status of a lot on a key. ?That I don't think
>>>> is there.
>>>> /b
>>>> On Jan 17, 2011, at 9:01 AM, Travis Kemen wrote:
>>>>
>>>> BLF works fine here with valet parking using polycom/snom phones.
>>>>
>>>> Travis
>>>>
>>>> On Sun, Jan 16, 2011 at 6:15 PM, Spencer Thomason <spencer at 5ninesolutions.com
>>>> > wrote:
>>>>>
>>>>> Hello all,
>>>>> Is it possible to use BLFs with valet parking? ?Basically I have
>>>>> several Linksys SPA 509G phones and after a call is parked, I'd
>>>>> like
>>>>> use use a speed dial/BLF key on the phone to pickup the call. ?I do
>>>>> have presence configured and working with the extensions.
>>>>>
>>>>> Thanks,
>>>>> Spencer
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From msc at freeswitch.org  Mon Jan 17 22:30:47 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Jan 2011 11:30:47 -0800
Subject: [Freeswitch-users] extract channel vars from another channel
In-Reply-To: <9E066B8B7262408F8E3B0381D26BF236@e1705>
References: <9E066B8B7262408F8E3B0381D26BF236@e1705>
Message-ID: <AANLkTik076bKtJKa1B1Z28C8R6SK9iwNNK_7R3fZ=vb6@mail.gmail.com>

If you don't know the uuid of the call then you'll have to make your best
guess. I recommend that you look at the default dialplan's 'hash' entries in
the Local_Extension. There are ways to lookup the uuid with only the dialed
number, but if there is more than one call to that target you'll need to
figure out which call's variables you want to look at.

-MC

On Tue, Jan 11, 2011 at 7:06 PM, Madovsky <infos at madovsky.org> wrote:

>  from xml dialplan I'd like to
> get channel vars from another channel but I know only the number of the
> targetted (internal) user.
> is there a way ?
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Jan 17 22:32:49 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Jan 2011 11:32:49 -0800
Subject: [Freeswitch-users] cepstral and SSML
In-Reply-To: <B619ED95945044D5934B36A8D1AEBEF2@e1705>
References: <B619ED95945044D5934B36A8D1AEBEF2@e1705>
Message-ID: <AANLkTikUgmfGH-4cBpPjuc57Rr6NPan8tQFibSbwvz_j@mail.gmail.com>

Can you pastebin the debug output? That might give us a clue as to where the
escaping needs to occur.
-MC

On Tue, Jan 11, 2011 at 6:08 PM, Madovsky <infos at madovsky.org> wrote:

>  if I use
>
> <action application="speak" data="cepstral|david|<prosody volume='65'>Your
> credit is too low to reach the landline user phone. Transfer to
> voicemail.</prosody>"/>
> it seems that SSML isn't supported like this,
> the voice says "/ prosody" only. the whole sentence is not said.
>
> do I escape the SSML code in the data ?
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Jan 17 22:34:48 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Jan 2011 11:34:48 -0800
Subject: [Freeswitch-users] empty var expression
In-Reply-To: <E2E64EFB13D6470BB46E3F74AE6142DE@e1705>
References: <E2E64EFB13D6470BB46E3F74AE6142DE@e1705>
Message-ID: <AANLkTinawdxU=WGgE=ygvEs60ffWbo+5Cw_XcU4Vfu-3@mail.gmail.com>

Is the var showing up as "UNDEF" when you send the call to the info app?
-MC

On Fri, Jan 14, 2011 at 12:22 PM, Madovsky <infos at madovsky.org> wrote:

>  I tried this for an var marked as "UNDEF" in log
>
> <condition field="${var}" expression="^UNDEF$" break="never">
> <condition field="${var}" expression="UNDEF" break="never">
> <condition field="${var}" expression="undef" break="never">
> <condition field="${var}" expression="^undef$" break="never">
> <condition field="${var}" expression="" break="never">
> <condition field="${var}" expression="^$" break="never">
> <condition field="${var}" expression="[UNDEF]" break="never">
> <condition field="${var}" expression="[]" break="never">
>
> without success.
> which is the right expression for empty var please ?
>
> Thanks
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Jan 17 22:37:16 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Jan 2011 11:37:16 -0800
Subject: [Freeswitch-users] Fwd: Unable to successfully configure icall
 gateway and route inbound DID
In-Reply-To: <AANLkTi=Hs5NMt1xv2OYx=rELyRULtkTWoabeUmgLGfPt@mail.gmail.com>
References: <AANLkTimahoPc=gxMD8NR+=y9jHKSfdz0Bbc_BUuLSLkp@mail.gmail.com>
	<67907271-E9E5-4608-9D3F-41D2D4102BF1@freeswitch.org>
	<AANLkTik=ie4qQiyP4iaB-G_-ekF=mKh14-VjE+UF=y4B@mail.gmail.com>
	<AANLkTi=Hs5NMt1xv2OYx=rELyRULtkTWoabeUmgLGfPt@mail.gmail.com>
Message-ID: <AANLkTim9_bNfyiAix2V68XzJxFe7mAhmNeF+qEVRTrpi@mail.gmail.com>

Were you able to get this one resolved? If you challenge the incoming call
then iCall needs to supply a user name and password, i.e. 'digest auth'. If
you add the iCall IP address to acl.conf.xml with cidr='x.x.x.x/32' in the
'domains' acl then it will let those iCall calls into your dialplan in the
public context.

-MC

On Tue, Jan 11, 2011 at 8:38 AM, Marvin Dillon <marvin.n.dillon at gmail.com>wrote:

> Team,
>
> Can anyone help me with the below issue?
>
> Thanks,
> MD
>
> ---------- Forwarded message ----------
> From: Marvin Dillon <marvin.n.dillon at gmail.com>
> Date: Mon, 10 Jan 2011 21:30:50 -0500
> Subject: Re: [Freeswitch-users] Unable to successfully configure icall
> gateway and route inbound DID
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>
> Hey Brian,
>
> I checked with icall and i am now able to register to the outbound server.
> I
> am still having a problem when i call my DID, it get an error "Rejected by
> acl "domains". Can you say what configuration is missing here?
>
> sofia status
>                     Name
> Type                                       Data      State
>
> =================================================================================================
>                 internal       profile
> sip:mod_sofia at 192.168.1.100:5060      RUNNING (0)
>            internal-ipv6       profile
> sip:mod_sofia@[::1]:5060
> RUNNING (0)
>                 external       profile
> sip:mod_sofia at 192.168.1.100:5080      RUNNING (0)
>    external::example.com       gateway
> sip:joeuser at example.com <sip%3Ajoeuser at example.com> <
> sip%3Ajoeuser at example.com <sip%253Ajoeuser at example.com>>      NOREG
> external::icall_international   gateway
> sip:XXX at gw01-car.dal.us.icall.net <sip%3AXXX at gw01-car.dal.us.icall.net><
> sip%3AXXX at gw01-car.dal.us.icall.net<sip%253AXXX at gw01-car.dal.us.icall.net>
> >
> REGED
>  external::icall_outbound       gateway
> sip:XXX at sbc01-car.dal.us.icall.net <sip%3AXXX at sbc01-car.dal.us.icall.net><
> sip%3AXXX at sbc01-car.dal.us.icall.net<sip%253AXXX at sbc01-car.dal.us.icall.net>
> >
> REGED
>  external::icall_inbound       gateway
> sip:XXX at 72.249.14.242 <sip%3AXXX at 72.249.14.242><sip%3AXXX at 72.249.14.242<sip%253AXXX at 72.249.14.242>
> >
> REGED
>      external::icall.com       gateway
> sip:XXX at 72.249.14.242 <sip%3AXXX at 72.249.14.242><sip%3AXXX at 72.249.14.242<sip%253AXXX at 72.249.14.242>
> >
> REGED
>           208.124.220.35         alias
> internal      ALIASED
>
> =================================================================================================
> 3 profiles 1 alias
> freeswitch at debian> 2011-01-10 21:21:51.819275 [WARNING] sofia.c:6075 IP
> 72.249.14.242 Rejected by acl "domains"
> 2011-01-10 21:21:52.047119 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
> by acl "domains"
> 2011-01-10 21:21:52.335147 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
> by acl "domains"
> 2011-01-10 21:21:52.552260 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
> by acl "domains"
> 2011-01-10 21:21:53.008400 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
> by acl "domains"
> 2011-01-10 21:21:53.239339 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
> by acl "domains"
> 2011-01-10 21:21:53.553315 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
> by acl "domains"
> 2011-01-10 21:21:53.766360 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
> by acl "domains"
> 2011-01-10 21:21:55.245075 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
> by acl "domains"
> 2011-01-10 21:21:55.485528 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
> by acl "domains"
> 2011-01-10 21:21:55.806639 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
> by acl "domains"
> 2011-01-10 21:21:56.060993 [WARNING] sofia.c:6075 IP 72.249.14.242 Rejected
> by acl "domains"
>
>
> On Mon, Jan 10, 2011 at 10:14 AM, Brian West <brian at freeswitch.org> wrote:
>
> > Um I don't think you register to the outbound server.
> >
> >
> > /b
> >
> >  On Jan 10, 2011, at 12:29 AM, Marvin Dillon wrote:
> >
> >  Hello Team,
> >
> > I am a rookie running Freeswitch 1.0.6 on Debian Lenny and need some
> urgent
> > help. I have been facing a challenge getting my icall gateways configured
> > and being able to route my inbound DID back to my Freeswitch platform. My
> > sofia status output is this right now:
> >
> >  sofia status
> >                      Name
> > Type                                       Data      State
> >
> >
> =================================================================================================
> >                  internal       profile
> > sip:mod_sofia at 192.168.1.100:5060      RUNNING (0)
> >                  external       profile
> > sip:mod_sofia at 192.168.1.100:5080      RUNNING (0)
> >     external::example.com       gateway
> > sip:joeuser at example.com <sip%3Ajoeuser at example.com> <
> sip%3Ajoeuser at example.com <sip%253Ajoeuser at example.com>>      NOREG
> > external::icall_international   gateway
> > sip:cust_mdillon at gw01-car.dal.us.icall.net<sip%3Acust_mdillon at gw01-car.dal.us.icall.net>
> <sip%3Acust_mdillon at gw01-car.dal.us.icall.net<sip%253Acust_mdillon at gw01-car.dal.us.icall.net>
> >
> > REGED
> >  external::icall_outbound       gateway
> > sip:cust_mdillon at sbc01-car.dal.us.icall.net<sip%3Acust_mdillon at sbc01-car.dal.us.icall.net>
> <sip%3Acust_mdillon at sbc01-car.dal.us.icall.net<sip%253Acust_mdillon at sbc01-car.dal.us.icall.net>
> >
> > FAIL_WAIT
> >   external::icall_inbound       gateway
> > sip:cust_mdillon at 72.249.14.242 <sip%3Acust_mdillon at 72.249.14.242> <
> sip%3Acust_mdillon at 72.249.14.242 <sip%253Acust_mdillon at 72.249.14.242>>
> > REGED
> >       external::icall.com       gateway
> > sip:cust_mdillon at 72.249.14.242 <sip%3Acust_mdillon at 72.249.14.242> <
> sip%3Acust_mdillon at 72.249.14.242 <sip%253Acust_mdillon at 72.249.14.242>>
> > REGED
> >            208.124.220.35         alias
> > internal      ALIASED
> >             internal-ipv6       profile
> > sip:mod_sofia@[::1]:5060      RUNNING (0)
> >
> >
> =================================================================================================
> > 3 profiles 1 alias
> > but I have no clue why I am getting a busy tone whenever I call my
> inbound
> > DID as the sofia output indicates my inbound gateway is registered. Can
> > someone please help me with this.
> >
> > Thanks,
> > MD
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> --
> Sent from my mobile device
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Jan 17 22:39:17 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Jan 2011 11:39:17 -0800
Subject: [Freeswitch-users] DTMF events issue
In-Reply-To: <COL122-W49E961E8ADD843FE0C0E0BB5F20@phx.gbl>
References: <COL122-W49E961E8ADD843FE0C0E0BB5F20@phx.gbl>
Message-ID: <AANLkTinYzAQ4Yb1a1C18GcCcVUzKiXU___kOPjoOCSbm@mail.gmail.com>

Can you pastebin the dialplan you're using and the debug log of the issue
while it's happening? Also, just curious, what DTMF-based actions are needed
while waiting for the other party to answer? There might be alternatives to
what you are doing.

Thanks,
MC

On Sat, Jan 15, 2011 at 6:44 AM, Rafqat . <rafonline at hotmail.com> wrote:

>
>
> Hi,
>
> I am using ESL to listen to DTMF events.  In most part I can listen to
> these events and respond to them fine.
>
> However, when leg A attempts to bridge to leg B, and presses some keys
> whilst waiting for leg b to answer the call, no DTMF events are received.
> When leg b answers the call all the DTMF events that leg A generated whilst
> waiting for leg B to answer are received in one batch.
>
> What am i doing wrong here? Is this normal behavour? surely not.
>
> Cheers
>
> Raf
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Jan 17 22:40:26 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Jan 2011 11:40:26 -0800
Subject: [Freeswitch-users] ESL mod_socket mod_commands
In-Reply-To: <AANLkTinKqQf12VkM3-kBmLeJ+JdQM5WHFgycqcpd1OND@mail.gmail.com>
References: <AANLkTikXOEhiN4zyRkmbTdPnWfBwuyAt9VguUNDCgzQc@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57ECF05AB1@cooper>
	<AANLkTinKqQf12VkM3-kBmLeJ+JdQM5WHFgycqcpd1OND@mail.gmail.com>
Message-ID: <AANLkTinfjKcpK3G7iG1roFvh8h=7dGr-ZgOto0Fp3nb8@mail.gmail.com>

Don't forget Darren's excellent write-up in chapter 9 of the FreeSWITCH
book. It covers events and ESL very well.
-MC

On Tue, Jan 11, 2011 at 7:30 AM, Holger Esser <hesser4900 at gmail.com> wrote:

> Many thanks Peter. Sometimes I do not see the forest for the trees.
>
> On Tue, Jan 11, 2011 at 9:19 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
>> To play a sound file in ESL I simply use (when intended for one leg only,
>> for instance in an IVR app).
>>
>>
>>
>> SendMsg <uuid>
>>
>> call-command: execute
>>
>> execute-app-name: playback
>>
>> execute-app-arg: /path/to/file.wav
>>
>>
>>
>> /Peter
>>
>>
>>
>> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Holger Esser
>> *Skickat:* den 11 januari 2011 16:10
>> *Till:* FreeSWITCH-users at lists.freeswitch.org
>> *?mne:* [Freeswitch-users] ESL mod_socket mod_commands
>>
>>
>>
>> Hi guys,
>>
>>
>>
>> What would be the best way to play an audio file with the ESL/mod_socket?
>>
>> I assume it is uuid_broadcast. Would it be possible to use mod_shout with
>> it like it does in uuid_displace?
>>
>> Or should I just use uuid_displace for any audio file plays?
>>
>>
>>
>> Kind regards,
>>
>> holger
>>
>> !DSPAM:4d2c73e032765465720928!
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Jan 17 22:44:59 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Jan 2011 11:44:59 -0800
Subject: [Freeswitch-users] mod_conference member-flags
In-Reply-To: <69D8E2E35ADF4A9EB28E9EE4BF4970F7@e1705>
References: <69D8E2E35ADF4A9EB28E9EE4BF4970F7@e1705>
Message-ID: <AANLkTikYiAOGtRnaehtwk_Ag9o_Egh0Dm9nvNfLGcXjz@mail.gmail.com>

Please confirm: you want the conference moderator also to have the flag
where if he/she leaves the conference that the conference ends?

-MC

On Sat, Jan 15, 2011 at 3:36 PM, Madovsky <infos at madovsky.org> wrote:

>  How a "moderator" can be
> also "endconf" in same time ?
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From andy at fabulous4.co.uk  Mon Jan 17 22:52:13 2011
From: andy at fabulous4.co.uk (Andy Ayers)
Date: Mon, 17 Jan 2011 19:52:13 -0000
Subject: [Freeswitch-users] RTCP Keep Alive issue - hangup after 60
	seconds of silence
In-Reply-To: <5CE50C9B-6F53-4A1A-8695-88B6E4073FDC@freeswitch.org>
References: <006001cbb3d4$11487130$33d95390$@fabulous4.co.uk>
	<AANLkTinAKvgAdUsFOci-ORR8t5KFGEWNZ95b9hcJHpH3@mail.gmail.com>
	<5CE50C9B-6F53-4A1A-8695-88B6E4073FDC@freeswitch.org>
Message-ID: <017b01cbb680$0f120760$2d361620$@fabulous4.co.uk>

Many thanks for all your help. Should I be able to see the RTCP messages
going out in the log file?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: 14 January 2011 15:00
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] RTCP Keep Alive issue - hangup after 60
seconds of silence

 

While this will turn on RTCP your provider needs to be beaten for requiring
such a resource wasting process.

 

/b

 

On Jan 14, 2011, at 6:49 AM, Christopher Rienzo wrote:





I don't know what RTCP keep alive is, but if they just mean to turn on RTCP,
you can do it with the following params in your sofia configuration:

 <!-- enable rtcp on every channel also can be done per leg basis 


      with rtcp_audio_interval_msec variable set to passthru to pass 
      it across a call -->
 <param name="rtcp-audio-interval-msec" value="5000"/>

or, set the rtcp_audio_interval_msec channel variable.

See http://wiki.freeswitch.org/wiki/RTCP



 

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From null at invalid.name  Mon Jan 17 23:09:11 2011
From: null at invalid.name (Dan Lane)
Date: Mon, 17 Jan 2011 20:09:11 +0000
Subject: [Freeswitch-users] Nibblebill Lowbal_action
Message-ID: <AANLkTikg2SQkPW1SbXaffT58sqk2LpobNR0vrcKW2mPR@mail.gmail.com>

Am I correct in thinking that lowbal_action hasn't been implemented
yet even though it's present in the default config file?


From msc at freeswitch.org  Mon Jan 17 23:16:20 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Jan 2011 12:16:20 -0800
Subject: [Freeswitch-users] Nibblebill Lowbal_action
In-Reply-To: <AANLkTikg2SQkPW1SbXaffT58sqk2LpobNR0vrcKW2mPR@mail.gmail.com>
References: <AANLkTikg2SQkPW1SbXaffT58sqk2LpobNR0vrcKW2mPR@mail.gmail.com>
Message-ID: <AANLkTi=X3besMNc9zxQiBZqvJ+HR=TsVQuLwZcUD1ZE7@mail.gmail.com>

I'm going to have to say that it is not implemented as there is no reference
to lowbal_action in the source code. There is a lowbal_amt but it does not
look like the trigger has been implemented. I can't tell you why...

-MC

On Mon, Jan 17, 2011 at 12:09 PM, Dan Lane <null at invalid.name> wrote:

> Am I correct in thinking that lowbal_action hasn't been implemented
> yet even though it's present in the default config file?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rafonline at hotmail.com  Mon Jan 17 23:55:13 2011
From: rafonline at hotmail.com (Rafqat .)
Date: Mon, 17 Jan 2011 20:55:13 +0000
Subject: [Freeswitch-users] DTMF events issue
In-Reply-To: <AANLkTinYzAQ4Yb1a1C18GcCcVUzKiXU___kOPjoOCSbm@mail.gmail.com>
References: <COL122-W49E961E8ADD843FE0C0E0BB5F20@phx.gbl>,
	<AANLkTinYzAQ4Yb1a1C18GcCcVUzKiXU___kOPjoOCSbm@mail.gmail.com>
Message-ID: <COL122-W34C09F4B5EFAC73F35F278B5F40@phx.gbl>


Hi,

I have pastebinned (Raf) my debug logs.? My diaplan is trivial (simply calls a lua script) so I have not pasted that.? 

Basically during early media no DTMF events generated by Leg A are passed onto my inbound socket until leg B answers the call (at which point they are all sent as a batch).? 

I even took my own inbound socket out of the equation and used the '/event DTMF' command in the fs cli.? That does not receive the events either.? As you can see from the logs I pasted, I pressed ## at 20:39:58, but freeswitch 
sent the events at 20:40:02 (exactly the time when leg B answers the call).

The application is very simple, all I want to do is if Leg A presses ## then hangup leg b.? I have tried using bind_meta_app (does not seem to support ##) aswell as bind_digit_action (only got it to work if leg presses ##). 

Any help will be much appreciated.

Cheers

Raf



________________________________
> Date: Mon, 17 Jan 2011 11:39:17 -0800
> From: msc at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] DTMF events issue
>
> Can you pastebin the dialplan you're using and the debug log of the
> issue while it's happening? Also, just curious, what DTMF-based actions
> are needed while waiting for the other party to answer? There might be
> alternatives to what you are doing.
>
> Thanks,
> MC
>
> On Sat, Jan 15, 2011 at 6:44 AM, Rafqat .
> > wrote:
>
>
> Hi,
>
> I am using ESL to listen to DTMF events. In most part I can listen to
> these events and respond to them fine.
>
> However, when leg A attempts to bridge to leg B, and presses some keys
> whilst waiting for leg b to answer the call, no DTMF events are
> received. When leg b answers the call all the DTMF events that leg A
> generated whilst waiting for leg B to answer are received in one batch.
>
> What am i doing wrong here? Is this normal behavour? surely not.
>
> Cheers
>
> Raf
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________ FreeSWITCH-users
> mailing list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  

From anthony.minessale at gmail.com  Tue Jan 18 00:08:37 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 17 Jan 2011 15:08:37 -0600
Subject: [Freeswitch-users] DTMF events issue
In-Reply-To: <COL122-W49E961E8ADD843FE0C0E0BB5F20@phx.gbl>
References: <COL122-W49E961E8ADD843FE0C0E0BB5F20@phx.gbl>
Message-ID: <AANLkTikdhabwcPL1uzXMhyf9m7DDmMXSw+pQqR1ViTDS@mail.gmail.com>

yes it's normal, nothing is trying to parse dtmf at that stage.

if you didn't say "certainly not" I probably would have added a patch for you.


On Sat, Jan 15, 2011 at 8:44 AM, Rafqat . <rafonline at hotmail.com> wrote:
>
>
> Hi,
>
> I am using ESL to listen to DTMF events.? In most part I can listen to these events and respond to them fine.
>
> However, when leg A attempts to bridge to leg B, and presses some keys whilst waiting for leg b to answer the call, no DTMF events are received.? When leg b answers the call all the DTMF events that leg A generated whilst waiting for leg B to answer are received in one batch.
>
> What am i doing wrong here? Is this normal behavour? surely not.
>
> Cheers
>
> Raf
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From brian at freeswitch.org  Tue Jan 18 00:08:50 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Jan 2011 15:08:50 -0600
Subject: [Freeswitch-users] DTMF events issue
In-Reply-To: <COL122-W34C09F4B5EFAC73F35F278B5F40@phx.gbl>
References: <COL122-W49E961E8ADD843FE0C0E0BB5F20@phx.gbl>,
	<AANLkTinYzAQ4Yb1a1C18GcCcVUzKiXU___kOPjoOCSbm@mail.gmail.com>
	<COL122-W34C09F4B5EFAC73F35F278B5F40@phx.gbl>
Message-ID: <CE50FAB2-2C0F-49FB-863F-3C556A373BB9@freeswitch.org>

Well to be honest is this your provider?  They may not pass DTMF during early media.

/b

On Jan 17, 2011, at 2:55 PM, Rafqat . wrote:

> 
> Hi,
> 
> I have pastebinned (Raf) my debug logs.  My diaplan is trivial (simply calls a lua script) so I have not pasted that.  
> 
> Basically during early media no DTMF events generated by Leg A are passed onto my inbound socket until leg B answers the call (at which point they are all sent as a batch).  
> 
> I even took my own inbound socket out of the equation and used the '/event DTMF' command in the fs cli.  That does not receive the events either.  As you can see from the logs I pasted, I pressed ## at 20:39:58, but freeswitch 
> sent the events at 20:40:02 (exactly the time when leg B answers the call).
> 
> The application is very simple, all I want to do is if Leg A presses ## then hangup leg b.  I have tried using bind_meta_app (does not seem to support ##) aswell as bind_digit_action (only got it to work if leg presses ##). 
> 
> Any help will be much appreciated.
> 
> Cheers
> 
> Raf



From anthony.minessale at gmail.com  Tue Jan 18 00:11:59 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 17 Jan 2011 15:11:59 -0600
Subject: [Freeswitch-users] RTCP Keep Alive issue - hangup after 60
 seconds of silence
In-Reply-To: <017b01cbb680$0f120760$2d361620$@fabulous4.co.uk>
References: <006001cbb3d4$11487130$33d95390$@fabulous4.co.uk>
	<AANLkTinAKvgAdUsFOci-ORR8t5KFGEWNZ95b9hcJHpH3@mail.gmail.com>
	<5CE50C9B-6F53-4A1A-8695-88B6E4073FDC@freeswitch.org>
	<017b01cbb680$0f120760$2d361620$@fabulous4.co.uk>
Message-ID: <AANLkTi=pv2ed_b7Y_2Gekg3UodfOMkue0JacWFnxOdJh@mail.gmail.com>

if you want the easy way out

set

record_waste_resources=true

before you run the record app.  Then you will send rtp the whole time.


On Mon, Jan 17, 2011 at 1:52 PM, Andy Ayers <andy at fabulous4.co.uk> wrote:
> Many thanks for all your help. Should I be able to see the RTCP messages
> going out in the log file?
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
> West
> Sent: 14 January 2011 15:00
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] RTCP Keep Alive issue - hangup after 60
> seconds of silence
>
>
>
> While this will turn on RTCP your provider needs to be beaten for requiring
> such a resource wasting process.
>
>
>
> /b
>
>
>
> On Jan 14, 2011, at 6:49 AM, Christopher Rienzo wrote:
>
> I don't know what RTCP keep alive is, but if they just mean to turn on RTCP,
> you can do it with the following params in your sofia configuration:
>
>  <!-- enable rtcp on every channel also can be done per leg basis
>
> ??????with rtcp_audio_interval_msec variable set to passthru to pass
> ??????it across a call -->
>  <param name="rtcp-audio-interval-msec" value="5000"/>
>
> or, set the rtcp_audio_interval_msec channel variable.
>
> See?http://wiki.freeswitch.org/wiki/RTCP
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From rafonline at hotmail.com  Tue Jan 18 00:15:38 2011
From: rafonline at hotmail.com (Rafqat .)
Date: Mon, 17 Jan 2011 21:15:38 +0000
Subject: [Freeswitch-users] DTMF events issue
In-Reply-To: <AANLkTikdhabwcPL1uzXMhyf9m7DDmMXSw+pQqR1ViTDS@mail.gmail.com>
References: <COL122-W49E961E8ADD843FE0C0E0BB5F20@phx.gbl>,
	<AANLkTikdhabwcPL1uzXMhyf9m7DDmMXSw+pQqR1ViTDS@mail.gmail.com>
Message-ID: <COL122-W288E1F5D0D0B90071CCC26B5F40@phx.gbl>



Sorry if i caused any offence.

I guess my emotions got the better of me. I blame it on code stress!

Again sorry.

Raf

----------------------------------------
> Date: Mon, 17 Jan 2011 15:08:37 -0600
> From: anthony.minessale at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] DTMF events issue
>
> yes it's normal, nothing is trying to parse dtmf at that stage.
>
> if you didn't say "certainly not" I probably would have added a patch for you.
>
>
> On Sat, Jan 15, 2011 at 8:44 AM, Rafqat .  wrote:
> >
> >
> > Hi,
> >
> > I am using ESL to listen to DTMF events.  In most part I can listen to these events and respond to them fine.
> >
> > However, when leg A attempts to bridge to leg B, and presses some keys whilst waiting for leg b to answer the call, no DTMF events are received.  When leg b answers the call all the DTMF events that leg A generated whilst waiting for leg B to answer are received in one batch.
> >
> > What am i doing wrong here? Is this normal behavour? surely not.
> >
> > Cheers
> >
> > Raf
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  

From vetali100 at gmail.com  Tue Jan 18 00:48:26 2011
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Mon, 17 Jan 2011 23:48:26 +0200
Subject: [Freeswitch-users] [freeswitch-users] mod_cdr_csv - rotate on hup
	is not working reliable
Message-ID: <AANLkTimHO15CvUNBrbMETFgSxqvRSKNDpii-FzV7JzV0@mail.gmail.com>

Hi,

I am using cron job which executes: killall -HUP freeswitch

Usually new cdr file is created every one minute:

2011-01-17 16:55:01.583043 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
/usr/local/freeswitch/log/cdr-csv/Master.csv
2011-01-17 16:56:01.861371 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
/usr/local/freeswitch/log/cdr-csv/Master.csv
2011-01-17 16:57:44.338974 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
/usr/local/freeswitch/log/cdr-csv/Master.csv


Today I got some problems with network DB connection, as this also can be
seen by few lines (I am using xml_curl to serve registrations):

2011-01-17 16:59:04.382035 [ERR] switch_xml.c:1621 Error[[error near line
1]: root tag missing]
2011-01-17 17:00:24.443196 [ERR] switch_xml.c:1621 Error[[error near line
1]: root tag missing]
2011-01-17 17:00:30.997069 [ERR] switch_xml.c:1621 Error[[error near line
1]: root tag missing]
2011-01-17 17:00:30.997069 [WARNING] sofia_reg.c:2171 Can't find user
[1000 at xxx]
You must define a domain called 'xxx' in your directory and add a user with
the id="1000" attribute
and you must configure your device to use the proper domain in it's
authentication credentials.
...

And I did not get any CDR file between 16:57 and 17:01.


CDR rotation was restored after 4 minutes - 17:01 - BUT 3 times in 1 second!

2011-01-17 17:01:*44*.540795 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
/usr/local/freeswitch/log/cdr-csv/Master.csv
2011-01-17 17:01:*44*.540795 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
/usr/local/freeswitch/log/cdr-csv/Master.csv
2011-01-17 17:01:*44*.542071 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
/usr/local/freeswitch/log/cdr-csv/Master.csv


The same repeated again in few minutes:

2011-01-17 17:05:*09*.701507 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
/usr/local/freeswitch/log/cdr-csv/Master.csv
2011-01-17 17:05:*09*.701507 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
/usr/local/freeswitch/log/cdr-csv/Master.csv
2011-01-17 17:05:*09*.853446 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
/usr/local/freeswitch/log/cdr-csv/Master.csv

2011-01-17 17:05:*14*.949815 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
/usr/local/freeswitch/log/cdr-csv/Master.csv
2011-01-17 17:05:*14*.952841 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
/usr/local/freeswitch/log/cdr-csv/Master.csv


Because of this, I lost few call records (it is confirmed).

File at 17:05:*14*.949815 (which contained 1 call record - as per logs) was
overwritten by empty file at 17:05:*14*.952841.



Basically my question is - is it expected behavior that such [rarely
expected] errors make it behave this way?


It is Freeswitch 1.0.7 on Ubuntu Server 10.4.

Thank you,
Vitalie
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From vetali100 at gmail.com  Tue Jan 18 00:54:01 2011
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Mon, 17 Jan 2011 23:54:01 +0200
Subject: [Freeswitch-users] [freeswitch-users] sofia_presence.c:404
 Cannotfind profile [my.domain.com]
In-Reply-To: <2560F9AC02E244CEA258689845671F27@e1705>
References: <AANLkTi=edzD3Q=omSnVwqji-kzY0_Zo7mYdmicgc=3Dq@mail.gmail.com>
	<2560F9AC02E244CEA258689845671F27@e1705>
Message-ID: <AANLkTinsgp4tPa_5izng-vkJb7WQj7-_+5xzwN_qe0tY@mail.gmail.com>

I am using vars.xml and internal.xml from previous installation (1.0.6).

So it is not default.



2011/1/16 Madovsky <infos at madovsky.org>

>  maybe some file settings changed so you have now
> default config
>
> ----- Original Message -----
> *From:* Vitalii Colosov <vetali100 at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Sunday, January 16, 2011 4:13 AM
> *Subject:* [Freeswitch-users] [freeswitch-users] sofia_presence.c:404
> Cannotfind profile [my.domain.com]
>
> Hi,
>
> After installing 1.0.7 I observed in the log the following message in error
> verbosity:
>
>      sofia_presence.c:404 Cannot find profile [my.domain.com]
>
> It looks like everything works good, however I never saw this error in
> 1.0.6.
>
> Could you please hint what this error means and what should be configured
> in order to fix it?
>
> Thank you,
> Vitalie
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Tue Jan 18 01:02:56 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Jan 2011 16:02:56 -0600
Subject: [Freeswitch-users] [freeswitch-users] sofia_presence.c:404
	Cannot find profile [my.domain.com]
In-Reply-To: <AANLkTi=edzD3Q=omSnVwqji-kzY0_Zo7mYdmicgc=3Dq@mail.gmail.com>
References: <AANLkTi=edzD3Q=omSnVwqji-kzY0_Zo7mYdmicgc=3Dq@mail.gmail.com>
Message-ID: <EBB3B87E-D4FA-4A4E-ADA1-F7D1F1711A62@freeswitch.org>

You're missing an alias for my.domain.com on the appropriate profile.

/b

On Jan 16, 2011, at 3:13 AM, Vitalii Colosov wrote:

> Hi,
> 
> After installing 1.0.7 I observed in the log the following message in error verbosity:
> 
>      sofia_presence.c:404 Cannot find profile [my.domain.com]
> 
> It looks like everything works good, however I never saw this error in 1.0.6.
> 
> Could you please hint what this error means and what should be configured in order to fix it?
> 
> Thank you,
> Vitalie

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From grsingh750 at gmail.com  Tue Jan 18 01:31:06 2011
From: grsingh750 at gmail.com (guru singh)
Date: Tue, 18 Jan 2011 04:01:06 +0530
Subject: [Freeswitch-users] Caller-id on incoming FXO with freetdm
In-Reply-To: <AANLkTimh1Ow=0S1DP3fyep=UDcuFakA_fPJ4QYxipGvu@mail.gmail.com>
References: <AANLkTi=TujU1976wLCABzsfG2w=SvDQW1p2XEXfRniew@mail.gmail.com>
	<AANLkTimh1Ow=0S1DP3fyep=UDcuFakA_fPJ4QYxipGvu@mail.gmail.com>
Message-ID: <AANLkTi=uJ-buEkFYUdz_CQdH6FVegRkFVbkbTnzQ1g1T@mail.gmail.com>

Hi,

Yes, the DTMF digits detected are the correct caller id.
This line is in India from a telco called Airtel.
I think caller id is being sent via DTMF. Based on what I've read on
various forums, from people trying to get it to work on sipura devices
or dahdi with asterisk, Airtel uses polarity reversal to first alert
the other end to pick up caller id and then sends it via DTMF.
I'll try and confirm this soon.

guru

On Mon, Jan 17, 2011 at 10:10 PM, Moises Silva <moises.silva at gmail.com> wrote:
> On Mon, Jan 17, 2011 at 6:56 AM, guru singh <grsingh750 at gmail.com> wrote:
>>
>> Hi,
>>
>> I am unable to get caller-id for incoming FXO calls.
>> It is always being set to 0000000000
>> I'm using FreeTDM and am on the latest git. I've set enable_callerid
>> param to true in freetdm.conf.xml
>> I can see fs pickup the callerid but then after each digit it says
>> (debug=0), is this normal?
>> Here is call debug log in fs for answering fxo and running the info
>> app. http://pastebin.freeswitch.org/15037
>> What am I missing?
>
> Hello,
> The debug=0 line is normal when DTMF detection is enabled but DTMF debugging
> is not.
> However, since you are getting caller id, it should not be enqueuing DTMF
> (since currently we do not support DTMF caller id).
> Are the DTMF digits that you see detected the correct caller id?
> If they are correct, then this looks like an interesting coincidence where a
> bug actually mostly got your DTMF caller id to work, however some
> adjustments would be required to completely support caller id by DTMF.
> Which country is this line from?
> Is your caller id sent from the telco (or switch?) via DTMF?
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
> Canada
> t. 1 905 474 1990 x128 | e.?moy at sangoma.com
>
>
>
>>
>> thanks
>> guru
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mranga at gmail.com  Tue Jan 18 02:15:27 2011
From: mranga at gmail.com (M. Ranganathan)
Date: Mon, 17 Jan 2011 18:15:27 -0500
Subject: [Freeswitch-users] sendonly attribute ignored?
Message-ID: <AANLkTin4hFOEYo7s7odh79b=EZORg3fYMA5ypjVBzyKE@mail.gmail.com>

Hello,

I am sending INVITEs to a FS conference from two parties, both of whom
have sendonly in their respective SDPs. I am noticing that FreeSWITCH
sends RTP to each participant ( which it should not because the
participants have marked their streams sendonly).

Is this expected behavior?

Thanks

Ranga

-- 
M. Ranganathan


From moises.silva at gmail.com  Tue Jan 18 02:30:04 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 17 Jan 2011 18:30:04 -0500
Subject: [Freeswitch-users] Caller-id on incoming FXO with freetdm
In-Reply-To: <AANLkTi=uJ-buEkFYUdz_CQdH6FVegRkFVbkbTnzQ1g1T@mail.gmail.com>
References: <AANLkTi=TujU1976wLCABzsfG2w=SvDQW1p2XEXfRniew@mail.gmail.com>
	<AANLkTimh1Ow=0S1DP3fyep=UDcuFakA_fPJ4QYxipGvu@mail.gmail.com>
	<AANLkTi=uJ-buEkFYUdz_CQdH6FVegRkFVbkbTnzQ1g1T@mail.gmail.com>
Message-ID: <AANLkTi=S0JaVWozL1Nrc52ivK5WF8VO6mxgGWvDoZwx4@mail.gmail.com>

On Mon, Jan 17, 2011 at 5:31 PM, guru singh <grsingh750 at gmail.com> wrote:

> Hi,
>
> Yes, the DTMF digits detected are the correct caller id.
> This line is in India from a telco called Airtel.
> I think caller id is being sent via DTMF. Based on what I've read on
> various forums, from people trying to get it to work on sipura devices
> or dahdi with asterisk, Airtel uses polarity reversal to first alert
> the other end to pick up caller id and then sends it via DTMF.
> I'll try and confirm this soon.
>

Ok, so this is a good opportunity for us to add support for DTMF caller id.

If you can give me ssh to your box I can log in to code and test this
feature.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com
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From infos at madovsky.org  Tue Jan 18 02:54:13 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 17 Jan 2011 18:54:13 -0500
Subject: [Freeswitch-users] mod_conference member-flags
References: <69D8E2E35ADF4A9EB28E9EE4BF4970F7@e1705>
	<AANLkTikYiAOGtRnaehtwk_Ag9o_Egh0Dm9nvNfLGcXjz@mail.gmail.com>
Message-ID: <69B27340DF63444AB2001FB610FE9E6C@e1705>

yes, but if I understand together it's not possible ?
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 17, 2011 2:44 PM
  Subject: Re: [Freeswitch-users] mod_conference member-flags


  Please confirm: you want the conference moderator also to have the flag where if he/she leaves the conference that the conference ends?


  -MC


  On Sat, Jan 15, 2011 at 3:36 PM, Madovsky <infos at madovsky.org> wrote:

    How a "moderator" can be
    also "endconf" in same time ?

    Thanks

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org






------------------------------------------------------------------------------


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Tue Jan 18 02:55:32 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 17 Jan 2011 18:55:32 -0500
Subject: [Freeswitch-users] empty var expression
References: <E2E64EFB13D6470BB46E3F74AE6142DE@e1705>
	<AANLkTinawdxU=WGgE=ygvEs60ffWbo+5Cw_XcU4Vfu-3@mail.gmail.com>
Message-ID: <82AF2D8789E54716A79B97D16AF1B84B@e1705>

yes, I can see UNDEF
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 17, 2011 2:34 PM
  Subject: Re: [Freeswitch-users] empty var expression


  Is the var showing up as "UNDEF" when you send the call to the info app?
  -MC


  On Fri, Jan 14, 2011 at 12:22 PM, Madovsky <infos at madovsky.org> wrote:

    I tried this for an var marked as "UNDEF" in log

    <condition field="${var}" expression="^UNDEF$" break="never">
    <condition field="${var}" expression="UNDEF" break="never">
    <condition field="${var}" expression="undef" break="never">
    <condition field="${var}" expression="^undef$" break="never">
    <condition field="${var}" expression="" break="never">
    <condition field="${var}" expression="^$" break="never">
    <condition field="${var}" expression="[UNDEF]" break="never">
    <condition field="${var}" expression="[]" break="never">

    without success.
    which is the right expression for empty var please ?

    Thanks





    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org






------------------------------------------------------------------------------


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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From anthony.minessale at gmail.com  Tue Jan 18 02:56:42 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 17 Jan 2011 17:56:42 -0600
Subject: [Freeswitch-users] sendonly attribute ignored?
In-Reply-To: <AANLkTin4hFOEYo7s7odh79b=EZORg3fYMA5ypjVBzyKE@mail.gmail.com>
References: <AANLkTin4hFOEYo7s7odh79b=EZORg3fYMA5ypjVBzyKE@mail.gmail.com>
Message-ID: <AANLkTinHeVcmO_skXoQAr3w7_VjbPTHX2f1=qGQv8oRH@mail.gmail.com>

we don't support this.
We have not encountered a need to implement this as we have not
encountered anyone doing this.
It would require a patch.


On Mon, Jan 17, 2011 at 5:15 PM, M. Ranganathan <mranga at gmail.com> wrote:
> Hello,
>
> I am sending INVITEs to a FS conference from two parties, both of whom
> have sendonly in their respective SDPs. I am noticing that FreeSWITCH
> sends RTP to each participant ( which it should not because the
> participants have marked their streams sendonly).
>
> Is this expected behavior?
>
> Thanks
>
> Ranga
>
> --
> M. Ranganathan
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From msc at freeswitch.org  Tue Jan 18 03:03:20 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Jan 2011 16:03:20 -0800
Subject: [Freeswitch-users] RTCP Keep Alive issue - hangup after 60
 seconds of silence
In-Reply-To: <AANLkTi=pv2ed_b7Y_2Gekg3UodfOMkue0JacWFnxOdJh@mail.gmail.com>
References: <006001cbb3d4$11487130$33d95390$@fabulous4.co.uk>
	<AANLkTinAKvgAdUsFOci-ORR8t5KFGEWNZ95b9hcJHpH3@mail.gmail.com>
	<5CE50C9B-6F53-4A1A-8695-88B6E4073FDC@freeswitch.org>
	<017b01cbb680$0f120760$2d361620$@fabulous4.co.uk>
	<AANLkTi=pv2ed_b7Y_2Gekg3UodfOMkue0JacWFnxOdJh@mail.gmail.com>
Message-ID: <AANLkTi=0rHr6a5HrSdCBbdyfj7=fgoueTzQop=zJf+W4@mail.gmail.com>

FYI,

This variable:

http://wiki.freeswitch.org/wiki/Variable_record_waste_resources

Wasn't listed on the main chan vars page. I just rectified that by adding it
to the call recording related variables section.

-MC

On Mon, Jan 17, 2011 at 1:11 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> if you want the easy way out
>
> set
>
> record_waste_resources=true
>
> before you run the record app.  Then you will send rtp the whole time.
>
>
> On Mon, Jan 17, 2011 at 1:52 PM, Andy Ayers <andy at fabulous4.co.uk> wrote:
> > Many thanks for all your help. Should I be able to see the RTCP messages
> > going out in the log file?
> >
> >
> >
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Brian
> > West
> > Sent: 14 January 2011 15:00
> > To: FreeSWITCH Users Help
> > Subject: Re: [Freeswitch-users] RTCP Keep Alive issue - hangup after 60
> > seconds of silence
> >
> >
> >
> > While this will turn on RTCP your provider needs to be beaten for
> requiring
> > such a resource wasting process.
> >
> >
> >
> > /b
> >
> >
> >
> > On Jan 14, 2011, at 6:49 AM, Christopher Rienzo wrote:
> >
> > I don't know what RTCP keep alive is, but if they just mean to turn on
> RTCP,
> > you can do it with the following params in your sofia configuration:
> >
> >  <!-- enable rtcp on every channel also can be done per leg basis
> >
> >       with rtcp_audio_interval_msec variable set to passthru to pass
> >       it across a call -->
> >  <param name="rtcp-audio-interval-msec" value="5000"/>
> >
> > or, set the rtcp_audio_interval_msec channel variable.
> >
> > See http://wiki.freeswitch.org/wiki/RTCP
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Jan 18 03:06:11 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Jan 2011 16:06:11 -0800
Subject: [Freeswitch-users] mod_conference member-flags
In-Reply-To: <69B27340DF63444AB2001FB610FE9E6C@e1705>
References: <69D8E2E35ADF4A9EB28E9EE4BF4970F7@e1705>
	<AANLkTikYiAOGtRnaehtwk_Ag9o_Egh0Dm9nvNfLGcXjz@mail.gmail.com>
	<69B27340DF63444AB2001FB610FE9E6C@e1705>
Message-ID: <AANLkTi=2PpkY-RxRi_iyzvBAKm8NPC-sgUq=6EJXS83z@mail.gmail.com>

Where did you read that? I've not tried it but I didn't see any indication
that the two items are mutually exclusive.

-MC

On Mon, Jan 17, 2011 at 3:54 PM, Madovsky <infos at madovsky.org> wrote:

>  yes, but if I understand together it's not possible ?
>
> ----- Original Message -----
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, January 17, 2011 2:44 PM
> *Subject:* Re: [Freeswitch-users] mod_conference member-flags
>
> Please confirm: you want the conference moderator also to have the flag
> where if he/she leaves the conference that the conference ends?
>
> -MC
>
> On Sat, Jan 15, 2011 at 3:36 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  How a "moderator" can be
>> also "endconf" in same time ?
>>
>> Thanks
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Jan 18 03:10:14 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Jan 2011 16:10:14 -0800
Subject: [Freeswitch-users] [freeswitch-users] mod_cdr_csv - rotate on
 hup is not working reliable
In-Reply-To: <AANLkTimHO15CvUNBrbMETFgSxqvRSKNDpii-FzV7JzV0@mail.gmail.com>
References: <AANLkTimHO15CvUNBrbMETFgSxqvRSKNDpii-FzV7JzV0@mail.gmail.com>
Message-ID: <AANLkTinL2QsbQ=PsxfyPFXHp1kLKQFLzCOeAn61hf5yi@mail.gmail.com>

Do you have any information about what was happening during that four-minute
period? I would have to wonder if there was something goofy happening with
disk i/o or some other weirdness. The hard part is going to be reproducing
it. If you are able to reproduce the symptoms then getting a gcore while the
system is in the weird state might be useful for debugging purposes.

What kind of load is on this system? Also, what are the system specs?

-MC

On Mon, Jan 17, 2011 at 1:48 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:

> Hi,
>
> I am using cron job which executes: killall -HUP freeswitch
>
> Usually new cdr file is created every one minute:
>
> 2011-01-17 16:55:01.583043 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
> /usr/local/freeswitch/log/cdr-csv/Master.csv
> 2011-01-17 16:56:01.861371 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
> /usr/local/freeswitch/log/cdr-csv/Master.csv
> 2011-01-17 16:57:44.338974 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
> /usr/local/freeswitch/log/cdr-csv/Master.csv
>
>
> Today I got some problems with network DB connection, as this also can be
> seen by few lines (I am using xml_curl to serve registrations):
>
> 2011-01-17 16:59:04.382035 [ERR] switch_xml.c:1621 Error[[error near line
> 1]: root tag missing]
> 2011-01-17 17:00:24.443196 [ERR] switch_xml.c:1621 Error[[error near line
> 1]: root tag missing]
> 2011-01-17 17:00:30.997069 [ERR] switch_xml.c:1621 Error[[error near line
> 1]: root tag missing]
> 2011-01-17 17:00:30.997069 [WARNING] sofia_reg.c:2171 Can't find user
> [1000 at xxx]
> You must define a domain called 'xxx' in your directory and add a user with
> the id="1000" attribute
> and you must configure your device to use the proper domain in it's
> authentication credentials.
> ...
>
> And I did not get any CDR file between 16:57 and 17:01.
>
>
> CDR rotation was restored after 4 minutes - 17:01 - BUT 3 times in 1
> second!
>
> 2011-01-17 17:01:*44*.540795 [NOTICE] mod_cdr_csv.c:123 Rotated CDR
> logfile /usr/local/freeswitch/log/cdr-csv/Master.csv
> 2011-01-17 17:01:*44*.540795 [NOTICE] mod_cdr_csv.c:123 Rotated CDR
> logfile /usr/local/freeswitch/log/cdr-csv/Master.csv
> 2011-01-17 17:01:*44*.542071 [NOTICE] mod_cdr_csv.c:123 Rotated CDR
> logfile /usr/local/freeswitch/log/cdr-csv/Master.csv
>
>
> The same repeated again in few minutes:
>
> 2011-01-17 17:05:*09*.701507 [NOTICE] mod_cdr_csv.c:123 Rotated CDR
> logfile /usr/local/freeswitch/log/cdr-csv/Master.csv
> 2011-01-17 17:05:*09*.701507 [NOTICE] mod_cdr_csv.c:123 Rotated CDR
> logfile /usr/local/freeswitch/log/cdr-csv/Master.csv
> 2011-01-17 17:05:*09*.853446 [NOTICE] mod_cdr_csv.c:123 Rotated CDR
> logfile /usr/local/freeswitch/log/cdr-csv/Master.csv
>
> 2011-01-17 17:05:*14*.949815 [NOTICE] mod_cdr_csv.c:123 Rotated CDR
> logfile /usr/local/freeswitch/log/cdr-csv/Master.csv
> 2011-01-17 17:05:*14*.952841 [NOTICE] mod_cdr_csv.c:123 Rotated CDR
> logfile /usr/local/freeswitch/log/cdr-csv/Master.csv
>
>
> Because of this, I lost few call records (it is confirmed).
>
> File at 17:05:*14*.949815 (which contained 1 call record - as per logs)
> was overwritten by empty file at 17:05:*14*.952841.
>
>
>
> Basically my question is - is it expected behavior that such [rarely
> expected] errors make it behave this way?
>
>
> It is Freeswitch 1.0.7 on Ubuntu Server 10.4.
>
> Thank you,
> Vitalie
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Tue Jan 18 03:19:22 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 17 Jan 2011 19:19:22 -0500
Subject: [Freeswitch-users] mod_conference member-flags
References: <69D8E2E35ADF4A9EB28E9EE4BF4970F7@e1705><AANLkTikYiAOGtRnaehtwk_Ag9o_Egh0Dm9nvNfLGcXjz@mail.gmail.com><69B27340DF63444AB2001FB610FE9E6C@e1705>
	<AANLkTi=2PpkY-RxRi_iyzvBAKm8NPC-sgUq=6EJXS83z@mail.gmail.com>
Message-ID: <339DD15147E24681AC1A175628B1D2A4@e1705>

so can I use membe-flags=moderator,endconf ?
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 17, 2011 7:06 PM
  Subject: Re: [Freeswitch-users] mod_conference member-flags


  Where did you read that? I've not tried it but I didn't see any indication that the two items are mutually exclusive.


  -MC


  On Mon, Jan 17, 2011 at 3:54 PM, Madovsky <infos at madovsky.org> wrote:

    yes, but if I understand together it's not possible ?
      ----- Original Message ----- 
      From: Michael Collins 
      To: FreeSWITCH Users Help 
      Sent: Monday, January 17, 2011 2:44 PM
      Subject: Re: [Freeswitch-users] mod_conference member-flags


      Please confirm: you want the conference moderator also to have the flag where if he/she leaves the conference that the conference ends? 


      -MC


      On Sat, Jan 15, 2011 at 3:36 PM, Madovsky <infos at madovsky.org> wrote:

        How a "moderator" can be
        also "endconf" in same time ?

        Thanks

        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org






--------------------------------------------------------------------------


      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org



    _______________________________________________
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    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org






------------------------------------------------------------------------------


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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From msc at freeswitch.org  Tue Jan 18 03:21:26 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Jan 2011 16:21:26 -0800
Subject: [Freeswitch-users] empty var expression
In-Reply-To: <82AF2D8789E54716A79B97D16AF1B84B@e1705>
References: <E2E64EFB13D6470BB46E3F74AE6142DE@e1705>
	<AANLkTinawdxU=WGgE=ygvEs60ffWbo+5Cw_XcU4Vfu-3@mail.gmail.com>
	<82AF2D8789E54716A79B97D16AF1B84B@e1705>
Message-ID: <AANLkTinxWBQFM=1kGz821O4ijr+fDE-gMXrtQXPOHJ-6@mail.gmail.com>

Where is this variable getting or losing the "undef" value? Also, add this
line to info and see what gets printed:

<action application="log" data="WARNING $var contains '${var}'"/>

That will print it out in purple letters to make it easier to see. FWIW, I
tried this with various permutations of 'undefined' channel vars and
expression="^$" always worked for me.

-MC

On Mon, Jan 17, 2011 at 3:55 PM, Madovsky <infos at madovsky.org> wrote:

>  yes, I can see UNDEF
>
> ----- Original Message -----
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, January 17, 2011 2:34 PM
> *Subject:* Re: [Freeswitch-users] empty var expression
>
> Is the var showing up as "UNDEF" when you send the call to the info app?
> -MC
>
> On Fri, Jan 14, 2011 at 12:22 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  I tried this for an var marked as "UNDEF" in log
>>
>> <condition field="${var}" expression="^UNDEF$" break="never">
>> <condition field="${var}" expression="UNDEF" break="never">
>> <condition field="${var}" expression="undef" break="never">
>> <condition field="${var}" expression="^undef$" break="never">
>> <condition field="${var}" expression="" break="never">
>> <condition field="${var}" expression="^$" break="never">
>> <condition field="${var}" expression="[UNDEF]" break="never">
>> <condition field="${var}" expression="[]" break="never">
>>
>> without success.
>> which is the right expression for empty var please ?
>>
>> Thanks
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Jan 18 03:22:30 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Jan 2011 16:22:30 -0800
Subject: [Freeswitch-users] mod_conference member-flags
In-Reply-To: <339DD15147E24681AC1A175628B1D2A4@e1705>
References: <69D8E2E35ADF4A9EB28E9EE4BF4970F7@e1705>
	<AANLkTikYiAOGtRnaehtwk_Ag9o_Egh0Dm9nvNfLGcXjz@mail.gmail.com>
	<69B27340DF63444AB2001FB610FE9E6C@e1705>
	<AANLkTi=2PpkY-RxRi_iyzvBAKm8NPC-sgUq=6EJXS83z@mail.gmail.com>
	<339DD15147E24681AC1A175628B1D2A4@e1705>
Message-ID: <AANLkTikbFiQ3xGBS_wheD_2-3Tt9akUojdCoaZy+KoKA@mail.gmail.com>

>From the wiki:

"Can be any combination of: *deaf, waste, mute-detect, dist-dtmf, moderator,
endconf, mintwo*."

Give it a try and see what happens. Let us know.
-MC

On Mon, Jan 17, 2011 at 4:19 PM, Madovsky <infos at madovsky.org> wrote:

>  so can I use membe-flags=moderator,endconf ?
>
> ----- Original Message -----
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, January 17, 2011 7:06 PM
> *Subject:* Re: [Freeswitch-users] mod_conference member-flags
>
> Where did you read that? I've not tried it but I didn't see any indication
> that the two items are mutually exclusive.
>
> -MC
>
> On Mon, Jan 17, 2011 at 3:54 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  yes, but if I understand together it's not possible ?
>>
>>   ----- Original Message -----
>> *From:* Michael Collins <msc at freeswitch.org>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Sent:* Monday, January 17, 2011 2:44 PM
>> *Subject:* Re: [Freeswitch-users] mod_conference member-flags
>>
>> Please confirm: you want the conference moderator also to have the flag
>> where if he/she leaves the conference that the conference ends?
>>
>> -MC
>>
>> On Sat, Jan 15, 2011 at 3:36 PM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  How a "moderator" can be
>>> also "endconf" in same time ?
>>>
>>> Thanks
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>  ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From johnrose at comtex.net  Tue Jan 18 03:36:07 2011
From: johnrose at comtex.net (John Rose)
Date: Mon, 17 Jan 2011 17:36:07 -0700
Subject: [Freeswitch-users] C# new ESLevent(
Message-ID: <006801cbb6a7$b2b65ac0$18231040$@comtex.net>

Does someone have an example of the C# EslConnection method SendEvent with a
new ESLevent ?

 

Like how to fill in any info on the ESLevent object before sending it?

 

Thanks,

John

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From infos at madovsky.org  Tue Jan 18 03:41:59 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 17 Jan 2011 19:41:59 -0500
Subject: [Freeswitch-users] mod_conference member-flags
References: <69D8E2E35ADF4A9EB28E9EE4BF4970F7@e1705><AANLkTikYiAOGtRnaehtwk_Ag9o_Egh0Dm9nvNfLGcXjz@mail.gmail.com><69B27340DF63444AB2001FB610FE9E6C@e1705><AANLkTi=2PpkY-RxRi_iyzvBAKm8NPC-sgUq=6EJXS83z@mail.gmail.com><339DD15147E24681AC1A175628B1D2A4@e1705>
	<AANLkTikbFiQ3xGBS_wheD_2-3Tt9akUojdCoaZy+KoKA@mail.gmail.com>
Message-ID: <F9255D7FDC404C9FB30025062227E8BE@e1705>

Ok I though it was only one flag possible.
I will test thanks for your help
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 17, 2011 7:22 PM
  Subject: Re: [Freeswitch-users] mod_conference member-flags


  >From the wiki:


  "Can be any combination of: deaf, waste, mute-detect, dist-dtmf, moderator, endconf, mintwo."


  Give it a try and see what happens. Let us know.
  -MC


  On Mon, Jan 17, 2011 at 4:19 PM, Madovsky <infos at madovsky.org> wrote:

    so can I use membe-flags=moderator,endconf ?
      ----- Original Message ----- 
      From: Michael Collins 
      To: FreeSWITCH Users Help 
      Sent: Monday, January 17, 2011 7:06 PM
      Subject: Re: [Freeswitch-users] mod_conference member-flags


      Where did you read that? I've not tried it but I didn't see any indication that the two items are mutually exclusive. 


      -MC


      On Mon, Jan 17, 2011 at 3:54 PM, Madovsky <infos at madovsky.org> wrote:

        yes, but if I understand together it's not possible ?
          ----- Original Message ----- 
          From: Michael Collins 
          To: FreeSWITCH Users Help 
          Sent: Monday, January 17, 2011 2:44 PM
          Subject: Re: [Freeswitch-users] mod_conference member-flags


          Please confirm: you want the conference moderator also to have the flag where if he/she leaves the conference that the conference ends? 


          -MC


          On Sat, Jan 15, 2011 at 3:36 PM, Madovsky <infos at madovsky.org> wrote:

            How a "moderator" can be
            also "endconf" in same time ?

            Thanks

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From brian at freeswitch.org  Tue Jan 18 03:46:25 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Jan 2011 18:46:25 -0600
Subject: [Freeswitch-users] empty var expression
In-Reply-To: <AANLkTinxWBQFM=1kGz821O4ijr+fDE-gMXrtQXPOHJ-6@mail.gmail.com>
References: <E2E64EFB13D6470BB46E3F74AE6142DE@e1705>
	<AANLkTinawdxU=WGgE=ygvEs60ffWbo+5Cw_XcU4Vfu-3@mail.gmail.com>
	<82AF2D8789E54716A79B97D16AF1B84B@e1705>
	<AANLkTinxWBQFM=1kGz821O4ijr+fDE-gMXrtQXPOHJ-6@mail.gmail.com>
Message-ID: <EFEDAD13-AFC7-4F01-B0E5-8ECA9B94C610@freeswitch.org>

be careful... you shouldn't have anything if the variable doesn't exist but are you using lua or js?

/b

On Jan 17, 2011, at 6:21 PM, Michael Collins wrote:

> Where is this variable getting or losing the "undef" value? Also, add this line to info and see what gets printed:
> 
> <action application="log" data="WARNING $var contains '${var}'"/>
> 
> That will print it out in purple letters to make it easier to see. FWIW, I tried this with various permutations of 'undefined' channel vars and expression="^$" always worked for me.
> 
> -MC



From infos at madovsky.org  Tue Jan 18 04:17:23 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 17 Jan 2011 20:17:23 -0500
Subject: [Freeswitch-users] empty var expression
References: <E2E64EFB13D6470BB46E3F74AE6142DE@e1705><AANLkTinawdxU=WGgE=ygvEs60ffWbo+5Cw_XcU4Vfu-3@mail.gmail.com><82AF2D8789E54716A79B97D16AF1B84B@e1705><AANLkTinxWBQFM=1kGz821O4ijr+fDE-gMXrtQXPOHJ-6@mail.gmail.com>
	<EFEDAD13-AFC7-4F01-B0E5-8ECA9B94C610@freeswitch.org>
Message-ID: <886EA95F839543CDA23BF4DD29C4F135@e1705>

no js nor lua,
I just set a var from db select/bla/bla
in the dialplan


----- Original Message ----- 
From: "Brian West" <brian at freeswitch.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, January 17, 2011 7:46 PM
Subject: Re: [Freeswitch-users] empty var expression


> be careful... you shouldn't have anything if the variable doesn't exist 
> but are you using lua or js?
>
> /b
>
> On Jan 17, 2011, at 6:21 PM, Michael Collins wrote:
>
>> Where is this variable getting or losing the "undef" value? Also, add 
>> this line to info and see what gets printed:
>>
>> <action application="log" data="WARNING $var contains '${var}'"/>
>>
>> That will print it out in purple letters to make it easier to see. FWIW, 
>> I tried this with various permutations of 'undefined' channel vars and 
>> expression="^$" always worked for me.
>>
>> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From infos at madovsky.org  Tue Jan 18 05:35:10 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 17 Jan 2011 21:35:10 -0500
Subject: [Freeswitch-users] watchdog
References: <0CFFA71FF3DC475DB0CC476A30C7BBE6@e1705>
	<92F823F5-2451-44D4-973E-2C46E8E58960@freeswitch.org>
Message-ID: <771C947EF2804C8F87EBE5570023C66C@e1705>

2011-01-17 21:34:47.330954 [CRIT] sofia.c:1361 Profile internal: SIP STACK FAILURE DETECTED!
GOODBYE CRUEL WORLD, I'M LEAVING YOU TODAY....GOODBYE, GOODBYE, GOOD BYE
Aborted (core dumped)

:P
  ----- Original Message ----- 
  From: Brian West 
  To: FreeSWITCH Users Help 
  Sent: Sunday, January 16, 2011 5:34 PM
  Subject: Re: [Freeswitch-users] watchdog


  What do the logs say when yous tart up?


  /b


  On Jan 15, 2011, at 10:57 AM, Madovsky wrote:


    I tried to add this on sip profile

        <!--<param name="watchdog-enabled" value="yes"/>-->
        <!--<param name="watchdog-step-timeout" value="30000"/>-->
        <!--<param name="watchdog-event-timeout" value="30000"/>-->

    but after that freeswitch shutdown saying STACK error.

    Thanks




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From hwnorman at hotmail.com  Tue Jan 18 05:36:51 2011
From: hwnorman at hotmail.com (Norman Lam)
Date: Tue, 18 Jan 2011 10:36:51 +0800
Subject: [Freeswitch-users] Iksemel msvs compiling
In-Reply-To: <1295276746914-5932097.post@n2.nabble.com>
References: <SNT109-DS13313A1FA906527438E8CC1090@phx.gbl>	<SNT109-DS1155D401A0B9742CF63E13C10B0@phx.gbl>	<1294375438447-5898181.post@n2.nabble.com>	<SNT109-DS11CB0ECE5BD0F9C0384E03C10C0@phx.gbl>	<1294535404901-5903534.post@n2.nabble.com>	<SNT109-DS14B90C48B33126784F294AC10E0@phx.gbl>	<1294673236806-5907287.post@n2.nabble.com>	<SNT109-DS227CE87422187AA91545CFC1F30@phx.gbl>	<1295113990176-5925152.post@n2.nabble.com>	<SNT109-DS2124FB8719BEF6F213D36DC1F40@phx.gbl>
	<1295276746914-5932097.post@n2.nabble.com>
Message-ID: <SNT109-DS6447C073B028EC7D60228C1F70@phx.gbl>

Hi Jeff 

I have tried adding the ..\..\..\gnutls-2.9.9\include\gnutls,
or..\..\..\gnutls-2.9.9\include  to the include directories, but it still
cause the same error. 

Also tried copying the gnutls sub directory to where stream.c is located or
iksemel\include is located but still no luck, 

Norman Lam 

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff
Lenk
Sent: Monday, January 17, 2011 11:06 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Iksemel msvs compiling


They are experimental. A couple of users have reported success but the wiki
documentation on that change is sparse. If you are successful please help by
adding more specific instructions to those pages.

It looks like you also need to add wherever gnutls.h is located to the
include path as well.
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From infos at madovsky.org  Tue Jan 18 05:40:17 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 17 Jan 2011 21:40:17 -0500
Subject: [Freeswitch-users] watchdog
Message-ID: <224D99A7750A4738833DE219CB9AE836@e1705>

if I comment out these two lines

    <!--<param name="watchdog-step-timeout" value="30000"/>-->
    <!--<param name="watchdog-event-timeout" value="30000"/>-->

FS runs fine
  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 17, 2011 9:35 PM
  Subject: Re: [Freeswitch-users] watchdog


  2011-01-17 21:34:47.330954 [CRIT] sofia.c:1361 Profile internal: SIP STACK FAILURE DETECTED!
  GOODBYE CRUEL WORLD, I'M LEAVING YOU TODAY....GOODBYE, GOODBYE, GOOD BYE
  Aborted (core dumped)

  :P
    ----- Original Message ----- 
    From: Brian West 
    To: FreeSWITCH Users Help 
    Sent: Sunday, January 16, 2011 5:34 PM
    Subject: Re: [Freeswitch-users] watchdog


    What do the logs say when yous tart up? 


    /b


    On Jan 15, 2011, at 10:57 AM, Madovsky wrote:


      I tried to add this on sip profile

          <!--<param name="watchdog-enabled" value="yes"/>-->
          <!--<param name="watchdog-step-timeout" value="30000"/>-->
          <!--<param name="watchdog-event-timeout" value="30000"/>-->

      but after that freeswitch shutdown saying STACK error.

      Thanks




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From infos at madovsky.org  Tue Jan 18 05:46:42 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 17 Jan 2011 21:46:42 -0500
Subject: [Freeswitch-users] empty var expression
References: <E2E64EFB13D6470BB46E3F74AE6142DE@e1705><AANLkTinawdxU=WGgE=ygvEs60ffWbo+5Cw_XcU4Vfu-3@mail.gmail.com><82AF2D8789E54716A79B97D16AF1B84B@e1705>
	<AANLkTinxWBQFM=1kGz821O4ijr+fDE-gMXrtQXPOHJ-6@mail.gmail.com>
Message-ID: <586AF477B5E446D989A522BD54A75032@e1705>

ok I understoo what's happened.
weirdly with db app apparently if the result of a select is empty
so it will be in fact never empty but a space will be.
so

<condition field="${redir_conf}" expression="^ $" break="never"> 

with a space between ^ and $ works
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 17, 2011 7:21 PM
  Subject: Re: [Freeswitch-users] empty var expression


  Where is this variable getting or losing the "undef" value? Also, add this line to info and see what gets printed:


  <action application="log" data="WARNING $var contains '${var}'"/>


  That will print it out in purple letters to make it easier to see. FWIW, I tried this with various permutations of 'undefined' channel vars and expression="^$" always worked for me.


  -MC


  On Mon, Jan 17, 2011 at 3:55 PM, Madovsky <infos at madovsky.org> wrote:

    yes, I can see UNDEF
      ----- Original Message ----- 
      From: Michael Collins 
      To: FreeSWITCH Users Help 
      Sent: Monday, January 17, 2011 2:34 PM
      Subject: Re: [Freeswitch-users] empty var expression


      Is the var showing up as "UNDEF" when you send the call to the info app? 
      -MC


      On Fri, Jan 14, 2011 at 12:22 PM, Madovsky <infos at madovsky.org> wrote:

        I tried this for an var marked as "UNDEF" in log

        <condition field="${var}" expression="^UNDEF$" break="never">
        <condition field="${var}" expression="UNDEF" break="never">
        <condition field="${var}" expression="undef" break="never">
        <condition field="${var}" expression="^undef$" break="never">
        <condition field="${var}" expression="" break="never">
        <condition field="${var}" expression="^$" break="never">
        <condition field="${var}" expression="[UNDEF]" break="never">
        <condition field="${var}" expression="[]" break="never">

        without success.
        which is the right expression for empty var please ?

        Thanks





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From infos at madovsky.org  Tue Jan 18 06:12:53 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 17 Jan 2011 22:12:53 -0500
Subject: [Freeswitch-users] empty var expression
Message-ID: <F1D996C172564F8F9CC4896F9AE297F3@e1705>

Correction, my fault
it works with "^$"

thanks
  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 17, 2011 9:46 PM
  Subject: Re: [Freeswitch-users] empty var expression


  ok I understoo what's happened.
  weirdly with db app apparently if the result of a select is empty
  so it will be in fact never empty but a space will be.
  so

  <condition field="${redir_conf}" expression="^ $" break="never"> 

  with a space between ^ and $ works
    ----- Original Message ----- 
    From: Michael Collins 
    To: FreeSWITCH Users Help 
    Sent: Monday, January 17, 2011 7:21 PM
    Subject: Re: [Freeswitch-users] empty var expression


    Where is this variable getting or losing the "undef" value? Also, add this line to info and see what gets printed: 


    <action application="log" data="WARNING $var contains '${var}'"/>


    That will print it out in purple letters to make it easier to see. FWIW, I tried this with various permutations of 'undefined' channel vars and expression="^$" always worked for me.


    -MC


    On Mon, Jan 17, 2011 at 3:55 PM, Madovsky <infos at madovsky.org> wrote:

      yes, I can see UNDEF
        ----- Original Message ----- 
        From: Michael Collins 
        To: FreeSWITCH Users Help 
        Sent: Monday, January 17, 2011 2:34 PM
        Subject: Re: [Freeswitch-users] empty var expression


        Is the var showing up as "UNDEF" when you send the call to the info app? 
        -MC


        On Fri, Jan 14, 2011 at 12:22 PM, Madovsky <infos at madovsky.org> wrote:

          I tried this for an var marked as "UNDEF" in log

          <condition field="${var}" expression="^UNDEF$" break="never">
          <condition field="${var}" expression="UNDEF" break="never">
          <condition field="${var}" expression="undef" break="never">
          <condition field="${var}" expression="^undef$" break="never">
          <condition field="${var}" expression="" break="never">
          <condition field="${var}" expression="^$" break="never">
          <condition field="${var}" expression="[UNDEF]" break="never">
          <condition field="${var}" expression="[]" break="never">

          without success.
          which is the right expression for empty var please ?

          Thanks





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          http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From jeff at jefflenk.com  Tue Jan 18 06:14:42 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 17 Jan 2011 19:14:42 -0800 (PST)
Subject: [Freeswitch-users] Iksemel msvs compiling
In-Reply-To: <SNT109-DS6447C073B028EC7D60228C1F70@phx.gbl>
References: <1294375438447-5898181.post@n2.nabble.com>
	<SNT109-DS11CB0ECE5BD0F9C0384E03C10C0@phx.gbl>
	<1294535404901-5903534.post@n2.nabble.com>
	<SNT109-DS14B90C48B33126784F294AC10E0@phx.gbl>
	<1294673236806-5907287.post@n2.nabble.com>
	<SNT109-DS227CE87422187AA91545CFC1F30@phx.gbl>
	<1295113990176-5925152.post@n2.nabble.com>
	<SNT109-DS2124FB8719BEF6F213D36DC1F40@phx.gbl>
	<1295276746914-5932097.post@n2.nabble.com>
	<SNT109-DS6447C073B028EC7D60228C1F70@phx.gbl>
Message-ID: <1295320482333-5934373.post@n2.nabble.com>


use an absolute path to find the required includes and libs ex.
c:\gnutls-2.9.9\include

make sure you take into account whether the code is looking for
"path\file.h" or whatever.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Iksemel-msvs-compiling-tp5891263p5934373.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From dome at tel.co.th  Tue Jan 18 06:23:55 2011
From: dome at tel.co.th (dome at tel.co.th)
Date: Tue, 18 Jan 2011 10:23:55 +0700
Subject: [Freeswitch-users] Another: xml_curl vs Lua
In-Reply-To: <AANLkTinXR4MCzX9t4jthjH9h0zo2ZKw1NvzRGQavvmR4@mail.gmail.com>
References: <AANLkTik8RRmhN+-LYAG6o-ssuf4RYEYrine8Jj9s21PK@mail.gmail.com>
	<AANLkTinq=Sh2wQO+0YTF9V5mM51mYFVArd+LE5Dgi2Q-@mail.gmail.com>
	<AANLkTikGqn_oXe56BZuqZK==aF=mSJQS7sfgrQJc2Gv_@mail.gmail.com>
	<AANLkTinXR4MCzX9t4jthjH9h0zo2ZKw1NvzRGQavvmR4@mail.gmail.com>
Message-ID: <AANLkTi=ti5cGTPG+HkJpWBMTUU4xaac754sG+x-cw5UD@mail.gmail.com>

2011/1/17 Saeed Ahmed <saeedahmad1981 at gmail.com>:
> Thanks Dome for a prompt reply.
>
> can you describe a bit about your hardware and structure? what number of cps
> you are able to achieve?
i use openvz (good for live migration).
my system setting 300 cps buts maximum about 120 cps

Dome C.

>
> Thanks
>
> On Sun, Jan 16, 2011 at 3:16 PM, dome at tel.co.th <dome at tel.co.th> wrote:
>>
>> xml_curl -> app server
>> my app server use nginx + luajit + tokyotylant
>>
>>
>> Dome C.
>>
>>
>> 2011/1/16 Saeed Ahmed <saeedahmad1981 at gmail.com>:
>> > i want to handle 2000 concurrent calls and minimum 100 cps.
>> > possible with lua? or should i stay with xml_curl?
>> >
>> > On Sun, Jan 16, 2011 at 1:24 PM, Saeed Ahmed <saeedahmad1981 at gmail.com>
>> > wrote:
>> >>
>> >> Dear all,
>> >>
>> >> i know that lua is preferable by FS devs and community, but here i want
>> >> to
>> >> ask questions particular to my use case
>> >> 1. Current Setup
>> >> 1.1 currently i am using xml_curl - for dialplan only
>> >> 1.2 xml_curl.conf has two bindings so i am safe if first one dies
>> >> 1.3 i am using apache + mono (.net) + mysql on another server (other
>> >> than
>> >> FS server)
>> >> 1.4?Intel(R) Xeon(R) CPU ?E5420 ?@ 2.50GHz 8 core with 8 gm ram (for
>> >> both
>> >> FS and apache+mysql server)
>> >> 2. Call Life Cycle
>> >> 2.1. call comes on internal profile
>> >> 2.2. xml_curl ask xml from webserver
>> >> 2.3 i do database query for each call and, return back possible
>> >> supplier(s) based on dialed number and customer id
>> >> 2.4 i return xml to bridge the call on external profile
>> >> Questiosn & Concerns:
>> >>
>> >> 1. So with above setup i am not able to reach more than 60 cps (most
>> >> probably issue with my xml_curl backend), so i am thinking to try LUA
>> >> 2. but one point is clicking my mind that, in case of xml_curl the
>> >> webserver can be on external server, and xml_curl conf could have
>> >> primary
>> >> and backup binding, can i achieve that in LUA too?
>> >> 3. Is LUA still be usefull as i am not doing any ivr etc.. its just xml
>> >> conf which i have to return.
>> >> 4. xml_curl based setup has good feature that i can serve ?more FS
>> >> servers, can i also do it in LUA?
>> >>
>> >> Please ask me if i miss some information
>> >>
>> >> Thanks
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From infos at madovsky.org  Tue Jan 18 06:24:40 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 17 Jan 2011 22:24:40 -0500
Subject: [Freeswitch-users] bridge a call to remote conference
Message-ID: <B92791F73CF243D79BEA67CE9951ECE8@e1705>

I did more tests,
I can say that if the user bridged to the conference
has the same HZ rate of the conference (even if the codec is different)
so the sound is broken. 
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, January 16, 2011 2:33 PM
  Subject: Re: [Freeswitch-users] bridge a call to remote conference


  more weird,

  if I choose a codec from the sip phone based on 8000hz it doesn't work.
  it's like a rate conversion has to be forced to make the audio work on the conference bridge...
    ----- Original Message ----- 
    From: Madovsky 
    To: FreeSWITCH Users Help 
    Sent: Sunday, January 16, 2011 12:10 PM
    Subject: Re: [Freeswitch-users] bridge a call to remote conference


    after some test I confirm that it should force to offer a codec
    at the same rate of the conference, example

    <action application="bridge" data="{absolute_codec_string=GSM,ignore_early_media=true}sofia/internal/${conference}"/>

    as GSM is a 8000hz code so it works now...


    thanks
      ----- Original Message ----- 
      From: Steven Ayre 
      To: FreeSWITCH Users Help 
      Sent: Sunday, January 16, 2011 4:20 AM
      Subject: Re: [Freeswitch-users] bridge a call to remote conference


      Try answering the call before you enter the conference, possibly with a small sleep inbetween too. That might give the media time to startup before the conference IVR starts.

      http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_answer
      http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sleep

      -Steve



      On 16 January 2011 03:54, Madovsky <infos at madovsky.org> wrote:

        everyhting is ok unless that conference ivr can't be heard (like fraction of sec of sound every 2/3 sec)
        but the hold music (in 8000hz) works well.
        should I set any special codec in the legB conference bridge ?

        Thanks

        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org






--------------------------------------------------------------------------


      _______________________________________________
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From hwnorman at hotmail.com  Tue Jan 18 06:40:34 2011
From: hwnorman at hotmail.com (Norman Lam)
Date: Tue, 18 Jan 2011 11:40:34 +0800
Subject: [Freeswitch-users] Iksemel msvs compiling
In-Reply-To: <1295320482333-5934373.post@n2.nabble.com>
References: <1294375438447-5898181.post@n2.nabble.com>	<SNT109-DS11CB0ECE5BD0F9C0384E03C10C0@phx.gbl>	<1294535404901-5903534.post@n2.nabble.com>	<SNT109-DS14B90C48B33126784F294AC10E0@phx.gbl>	<1294673236806-5907287.post@n2.nabble.com>	<SNT109-DS227CE87422187AA91545CFC1F30@phx.gbl>	<1295113990176-5925152.post@n2.nabble.com>	<SNT109-DS2124FB8719BEF6F213D36DC1F40@phx.gbl>	<1295276746914-5932097.post@n2.nabble.com>	<SNT109-DS6447C073B028EC7D60228C1F70@phx.gbl>
	<1295320482333-5934373.post@n2.nabble.com>
Message-ID: <SNT109-DS70F892DEEB91A55B85103C1F70@phx.gbl>

Tried with absolute path , no luck 

The stream.c define is  

#include <gnutls/gnutls.h>

Norman 

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff
Lenk
Sent: Tuesday, January 18, 2011 11:15 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Iksemel msvs compiling


use an absolute path to find the required includes and libs ex.
c:\gnutls-2.9.9\include

make sure you take into account whether the code is looking for
"path\file.h" or whatever.
-- 
View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/Iksemel-msvs-compiling-tp58912
63p5934373.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From kris at livecall.com  Tue Jan 18 06:46:53 2011
From: kris at livecall.com (Kris)
Date: Mon, 17 Jan 2011 19:46:53 -0800
Subject: [Freeswitch-users] C# new ESLevent(
References: <006801cbb6a7$b2b65ac0$18231040$@comtex.net>
Message-ID: <A3C3DC4F188A4327885EB87CBF218235@stor1>

I don't know about ESL-still very new to FS. I should look at that. If you 
figure it out, send some of the code here.


I used this to notify my running DLL that a new one is being loaded, then 
start a new thread to pop events. It works and pulls events fine. I still 
have a problem where freeswitch crashes toward the end of it's shutdown so 
something is not being freed or is corrupted.(switch_core_memory.c:587 
Stopping memory pool queue) is the last written on the screen.
If someone knows why, let me know.
Kris

    public class Loading : FreeSWITCH.ILoadNotificationPlugin
    {
        public bool Load()
        {
           Event loading_event = new Event("CUSTOM", 
"livematch::maintenance");
           loading_event.AddHeader("Action", "loading");
           loading_event.Fire();
           freeswitch.msleep(500);

           ManagedApplicationBase.LoadSettings(); //from the Database

            ManagedApplicationBase.StartEventThread();// this does the pop 
in a thread, terminates itself on the next loading event.
}
    abstract public class ManagedApplicationBase : IAppPlugin, IApiPlugin
    {
        //common Parameters
        public static bool ParmUsingWatchdogTimer=true;

        public static bool LoadingNewDLL = false;
        public static bool FreeSwitchShutdown = false;
        public static bool EventThreadStarted = false;
        public static System.Threading.Thread EventThread = new 
System.Threading.Thread(new System.Threading.ThreadStart(CheckEvents));
.......
        static private void CheckEvents()//runs only in the new thread
        {
            BaseLog.WriteLine(BaseLogLevel.Info, "Starting CheckEvents");
            string EventName;
            ODBCData Data = new 
ODBCData(global::LiveMatch.Properties.Settings.Default.ODBCConnectionMain);
            ODBCData CallsData = new 
ODBCData(global::LiveMatch.Properties.Settings.Default.LMCallDetailsConnectionMain);
            FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api();
            //        EventConsumer con_CUSTOM = new EventConsumer("CUSTOM", 
"Conference::Maintenance");
                EventConsumer con_all = new EventConsumer("ALL", null);
            while (!FreeSwitchShutdown && !LoadingNewDLL) //            {
                Event ev_all = con_all.pop(0);
                if (ev_all != null)
                {
                    EventName = ev_all.GetHeader("Event-Name");
                    switch (EventName)
                    {
                        case "HEARTBEAT":
                            Log.WriteLine(LogLevel.Notice, "Got event " + 
EventName);
                            BaseLog.WriteLine(BaseLogLevel.FSEVENT, 
ev_all.Serialize("xml"));
                            if(ParmUsingWatchdogTimer)
                            Data.SaveParameter(SystemInformation.ComputerName, 
Parameters.SPARAMETER_WATCHDOG_TIME, DateTime.Now.ToString("MM/dd/yyyy 
HH:mm:ss"));//12/11/2010 03:53:02
                            break;
                        case "CHANNEL_HANGUP_COMPLETE":
                            Log.WriteLine(LogLevel.Notice, "Got event " + 
EventName);
                            BaseLog.WriteLine(BaseLogLevel.FSEVENT, 
ev_all.Serialize("xml"));
                            string SQL = String.Format("UPDATE Calls SET 
StopTime='{0}' WHERE CallGUID='{1}'",
                                         DateTime.UtcNow.ToString("MM/dd/yyyy 
HH:mm:ss.fff"), ev_all.GetHeader("Caller-Unique-ID"));
                            CallsData.ExecuteSQL(SQL);
                            break;
                        case "SHUTDOWN":
                            Log.WriteLine(LogLevel.Notice, "Got event " + 
EventName);
                            BaseLog.WriteLine(BaseLogLevel.FSEVENT, 
ev_all.Serialize("xml"));
                            FreeSwitchShutdown = true;

                            BaseLog.WriteLine(BaseLogLevel.FSEVENT, "Exiting 
CheckEvents");
                            BaseLog.SwitchShutdown = true; //to prevent 
further writing

                            goto EXIT;
                        case "CUSTOM":
                            Log.WriteLine(LogLevel.Notice, "Got event   " + 
EventName);
                            //                            //ManagedSession 
Session = new ManagedSession(ev_all.GetHeader("Caller-Unique-ID"));
                            switch (ev_all.GetHeader("Event-Subclass"))
                            {
                                case "conference::maintenance":
                                    Log.WriteLine(LogLevel.Notice, 
"Event-Subclass  " + ev_all.GetHeader("Event-Subclass"));
                                    switch (ev_all.GetHeader("Action"))
                                    {
                                        case "del-member":
                                            Log.WriteLine(LogLevel.Notice, 
"Action  " + ev_all.GetHeader("Action"));
                                            //member is leaving or hung up?
                                            //update Conferences SET
                                            break;
                                        default:
                                            Log.WriteLine(LogLevel.Notice, 
"Action  " + ev_all.GetHeader("Action"));
                                            break;
                                    }
                                    break;
                                case "livematch::maintenance":
                                    Log.WriteLine(LogLevel.Notice, 
"Event-Subclass  " + ev_all.GetHeader("Event-Subclass"));
                                    switch (ev_all.GetHeader("Action"))
                                    {
                                        case "loading": //a new version is 
loading, terminate this one
                                            Log.WriteLine(LogLevel.Notice, 
"Action  " + ev_all.GetHeader("Action"));
                                            FreeSwitchShutdown = true;
                                            LoadingNewDLL = true;
                                            break;
                                        default:
                                            Log.WriteLine(LogLevel.Notice, 
"Action  " + ev_all.GetHeader("Action"));
                                            break;
                                    }
                                    break;
                                default:
                                    Log.WriteLine(LogLevel.Notice, 
"Event-Subclass  " + ev_all.GetHeader("Event-Subclass"));
                                    break;
                            }



----- Original Message ----- 
From: "John Rose" <johnrose at comtex.net>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Monday, January 17, 2011 4:36 PM
Subject: [Freeswitch-users] C# new ESLevent(


> Does someone have an example of the C# EslConnection method SendEvent with 
> a
> new ESLevent ?
>
>
>
> Like how to fill in any info on the ESLevent object before sending it?
>
>
>
> Thanks,
>
> John
>
> 



From infos at madovsky.org  Tue Jan 18 08:46:43 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 18 Jan 2011 00:46:43 -0500
Subject: [Freeswitch-users] user hangup in a conference
Message-ID: <E633254F6B744A5FB54DBB406C69D01E@e1705>

is there a way to know whe a conference doesn't exist anymore from the dialplan ?

like after this line

<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="continue_on_fail=true"/>
<action application="conference" data="conf_name}@default+${conf_pin}"/>

i'd like to know if there are members yet in the conference

<action application="set" data="myVar=${conference(conf_name list)}"/>

it works if the user send a dtmf key lke # (from conference default control keys)
but if the user hangup as normal so the dialplan doesn't continue after the conference XML line.

Any idea ?

Thanks
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From erik.dekkers at wvds.nl  Tue Jan 18 11:13:20 2011
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Tue, 18 Jan 2011 09:13:20 +0100
Subject: [Freeswitch-users] Cisco Java phones crash when nonce count
	is	incremented past 1 -- any ideas?
In-Reply-To: <AANLkTikg+GY1tZ3n7VNQ8+7JOw=qy3Fkbz3zYFF2v4TF@mail.gmail.com>
References: <AANLkTikg+GY1tZ3n7VNQ8+7JOw=qy3Fkbz3zYFF2v4TF@mail.gmail.com>
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE0782B278B4@FS1.wvds.local>

Graham,

Without more information we only can guess. Please give us more information on what freeswitch version you at and the firmware versions of the cisco phones?

Regards,

Erik

Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Graham Saathoff
Verzonden: maandag 17 januari 2011 18:48
Aan: freeswitch-users at lists.freeswitch.org
Onderwerp: [Freeswitch-users] Cisco Java phones crash when nonce count is incremented past 1 -- any ideas?

Hello all,

I have several Cisco Java powered phones (7941, 7961, 7965) that repeatedly drop calls.  The issue seems to be that once the phone attempts to update the nonce count to 2, it crashes and disconnects the call.  Has anyone seen this behavior before?  Any suggestions for how to fix this would be greatly appreciated.

Thanks,
Graham
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From Stefan.Weigel at allianz-warranty.com  Tue Jan 18 11:56:39 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Tue, 18 Jan 2011 09:56:39 +0100
Subject: [Freeswitch-users] mod_callcenter related questions
Message-ID: <5003D7D3E06F514E8C682F18D223265C046D61E5B1@AZWSMS03.azwarranty.int>

Hi all,

I'm playing around with mod_callcenter and hitting some questions:


*         what means position and level in tier configuration ? Is it only related to queue parameter 'strategy' ?

*         having more than one agent in a queue logged in only the first agents phone rings for tier_rule_wait_seconds but the call is not directed to the next tier/agent logged in to the queue

*         is it possible to configure overflow destinations means if a call isn't answered in a certain queue it's overflowing to another queue ?

Any help would be nice.


Thanks in advance and best regards

Stefan

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From andy at fabulous4.co.uk  Tue Jan 18 14:49:31 2011
From: andy at fabulous4.co.uk (Andy Ayers)
Date: Tue, 18 Jan 2011 11:49:31 -0000
Subject: [Freeswitch-users] RTCP Keep Alive issue - hangup after 60
	seconds of silence
In-Reply-To: <AANLkTi=0rHr6a5HrSdCBbdyfj7=fgoueTzQop=zJf+W4@mail.gmail.com>
References: <006001cbb3d4$11487130$33d95390$@fabulous4.co.uk>
	<AANLkTinAKvgAdUsFOci-ORR8t5KFGEWNZ95b9hcJHpH3@mail.gmail.com>
	<5CE50C9B-6F53-4A1A-8695-88B6E4073FDC@freeswitch.org>
	<017b01cbb680$0f120760$2d361620$@fabulous4.co.uk>
	<AANLkTi=pv2ed_b7Y_2Gekg3UodfOMkue0JacWFnxOdJh@mail.gmail.com>
	<AANLkTi=0rHr6a5HrSdCBbdyfj7=fgoueTzQop=zJf+W4@mail.gmail.com>
Message-ID: <012f01cbb705$c5e882e0$51b988a0$@fabulous4.co.uk>

All sorted, Many thanks!

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: 18 January 2011 00:03
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] RTCP Keep Alive issue - hangup after 60
seconds of silence

 

FYI,

 

This variable:

 

http://wiki.freeswitch.org/wiki/Variable_record_waste_resources

 

Wasn't listed on the main chan vars page. I just rectified that by adding it
to the call recording related variables section.

 

-MC

On Mon, Jan 17, 2011 at 1:11 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:

if you want the easy way out

set

record_waste_resources=true

before you run the record app.  Then you will send rtp the whole time.


On Mon, Jan 17, 2011 at 1:52 PM, Andy Ayers <andy at fabulous4.co.uk> wrote:
> Many thanks for all your help. Should I be able to see the RTCP messages
> going out in the log file?
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
> West
> Sent: 14 January 2011 15:00
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] RTCP Keep Alive issue - hangup after 60
> seconds of silence
>
>
>
> While this will turn on RTCP your provider needs to be beaten for
requiring
> such a resource wasting process.
>
>
>
> /b
>
>
>
> On Jan 14, 2011, at 6:49 AM, Christopher Rienzo wrote:
>
> I don't know what RTCP keep alive is, but if they just mean to turn on
RTCP,
> you can do it with the following params in your sofia configuration:
>
>  <!-- enable rtcp on every channel also can be done per leg basis
>
>       with rtcp_audio_interval_msec variable set to passthru to pass
>       it across a call -->
>  <param name="rtcp-audio-interval-msec" value="5000"/>
>
> or, set the rtcp_audio_interval_msec channel variable.
>
> See http://wiki.freeswitch.org/wiki/RTCP
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From anthony.minessale at gmail.com  Tue Jan 18 18:28:50 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 18 Jan 2011 09:28:50 -0600
Subject: [Freeswitch-users] [freeswitch-users] mod_cdr_csv - rotate on
 hup is not working reliable
In-Reply-To: <AANLkTinL2QsbQ=PsxfyPFXHp1kLKQFLzCOeAn61hf5yi@mail.gmail.com>
References: <AANLkTimHO15CvUNBrbMETFgSxqvRSKNDpii-FzV7JzV0@mail.gmail.com>
	<AANLkTinL2QsbQ=PsxfyPFXHp1kLKQFLzCOeAn61hf5yi@mail.gmail.com>
Message-ID: <AANLkTinNcbq-vuLXhvKxrzdYEEEOjyXYUfP1ty5mrT4c@mail.gmail.com>

you probably did not have your timeout set on the xml_curl
or are you using odbc as well? when networks time out during odbc you
were probably blocking the whole FS waiting for it to return.
It's best not to rely on public networks for critical data.


On Mon, Jan 17, 2011 at 6:10 PM, Michael Collins <msc at freeswitch.org> wrote:
> Do you have any information about what was happening during that four-minute
> period? I would have to wonder if there was something goofy happening with
> disk i/o or some other weirdness. The hard part is going to be reproducing
> it. If you are able to reproduce the symptoms then getting a gcore while the
> system is in the weird state might be useful for debugging purposes.
> What kind of load is on this system? Also, what are the system specs?
> -MC
>
> On Mon, Jan 17, 2011 at 1:48 PM, Vitalii Colosov <vetali100 at gmail.com>
> wrote:
>>
>> Hi,
>> I am using cron job which executes:?killall -HUP freeswitch
>> Usually new cdr file is created every one minute:
>> 2011-01-17 16:55:01.583043 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
>> /usr/local/freeswitch/log/cdr-csv/Master.csv
>> 2011-01-17 16:56:01.861371 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
>> /usr/local/freeswitch/log/cdr-csv/Master.csv
>> 2011-01-17 16:57:44.338974 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
>> /usr/local/freeswitch/log/cdr-csv/Master.csv
>>
>> Today I got some problems with network DB connection, as this also can be
>> seen by few lines (I am using xml_curl to serve registrations):
>>
>> 2011-01-17 16:59:04.382035 [ERR] switch_xml.c:1621 Error[[error near line
>> 1]: root tag missing]
>> 2011-01-17 17:00:24.443196 [ERR] switch_xml.c:1621 Error[[error near line
>> 1]: root tag missing]
>> 2011-01-17 17:00:30.997069 [ERR] switch_xml.c:1621 Error[[error near line
>> 1]: root tag missing]
>> 2011-01-17 17:00:30.997069 [WARNING] sofia_reg.c:2171 Can't find user
>> [1000 at xxx]
>> You must define a domain called 'xxx' in your directory and add a user
>> with the id="1000" attribute
>> and you must configure your device to use the proper domain in it's
>> authentication credentials.
>> ...
>> And I did not get any CDR file between 16:57 and 17:01.
>>
>> CDR rotation was restored after 4 minutes - 17:01 - BUT 3 times in 1
>> second!
>> 2011-01-17 17:01:44.540795 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
>> /usr/local/freeswitch/log/cdr-csv/Master.csv
>> 2011-01-17 17:01:44.540795 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
>> /usr/local/freeswitch/log/cdr-csv/Master.csv
>> 2011-01-17 17:01:44.542071 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
>> /usr/local/freeswitch/log/cdr-csv/Master.csv
>>
>> The same repeated again in few minutes:
>> 2011-01-17 17:05:09.701507 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
>> /usr/local/freeswitch/log/cdr-csv/Master.csv
>> 2011-01-17 17:05:09.701507 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
>> /usr/local/freeswitch/log/cdr-csv/Master.csv
>> 2011-01-17 17:05:09.853446 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
>> /usr/local/freeswitch/log/cdr-csv/Master.csv
>> 2011-01-17 17:05:14.949815 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
>> /usr/local/freeswitch/log/cdr-csv/Master.csv
>> 2011-01-17 17:05:14.952841 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
>> /usr/local/freeswitch/log/cdr-csv/Master.csv
>>
>> Because of this, I?lost few call records (it is confirmed).
>> File at?17:05:14.949815?(which contained 1 call record - as per logs) was
>> overwritten by empty file at?17:05:14.952841.
>>
>>
>> Basically my question is - is it expected behavior that such [rarely
>> expected] errors make it behave this way?
>>
>> It is Freeswitch 1.0.7 on Ubuntu Server 10.4.
>> Thank you,
>> Vitalie
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From johnrose at comtex.net  Tue Jan 18 19:39:42 2011
From: johnrose at comtex.net (John Rose)
Date: Tue, 18 Jan 2011 09:39:42 -0700
Subject: [Freeswitch-users] dp+ prefixed on From URI
Message-ID: <009401cbb72e$4f3b0e00$edb12a00$@comtex.net>

Why does the chat API command prefix a "dp+" onto the From URI when I call
the chat API?  Here is an argument that I am using: 

"sip|+15186819448 at 65.41.13.124|external/sip:+12062990047 at 65.41.13.125|Test
message."

Then the From header:

From: \"+15186819448\"
<sip:dp++15186819448 at 65.41.13.124>;tag=FUetK564c4egm\\r\\n

John



From jeff at jefflenk.com  Tue Jan 18 21:02:37 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 18 Jan 2011 10:02:37 -0800 (PST)
Subject: [Freeswitch-users] C# new ESLevent(
In-Reply-To: <006801cbb6a7$b2b65ac0$18231040$@comtex.net>
References: <006801cbb6a7$b2b65ac0$18231040$@comtex.net>
Message-ID: <1295373757332-5936549.post@n2.nabble.com>


have you seen this code? maybe it will help.

libs\esl\managed\ManagedEslTest
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/C-new-ESLevent-tp5934057p5936549.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From johnrose at comtex.net  Tue Jan 18 21:16:38 2011
From: johnrose at comtex.net (John Rose)
Date: Tue, 18 Jan 2011 11:16:38 -0700
Subject: [Freeswitch-users] C# new ESLevent(
In-Reply-To: <1295373757332-5936549.post@n2.nabble.com>
References: <006801cbb6a7$b2b65ac0$18231040$@comtex.net>
	<1295373757332-5936549.post@n2.nabble.com>
Message-ID: <00a701cbb73b$d9997f20$8ccc7d60$@comtex.net>

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> 
> 
> have you seen this code? maybe it will help.
> 
> libs\esl\managed\ManagedEslTest
> --


Yes, thanks but no example there to construct an ESLevent to send.

John




From acichocki at supermedia.pl  Tue Jan 18 16:13:56 2011
From: acichocki at supermedia.pl (Artur Cichocki)
Date: Tue, 18 Jan 2011 14:13:56 +0100
Subject: [Freeswitch-users] Usecure identification of Gateway (incoming
	call).
Message-ID: <4D359214.8020009@supermedia.pl>

Hi.

What is the proper way of gateway identification in incoming call?

The variable_sip_gateway looks like directly taken from the sip_req_user 
(username from INVITE).

Examples:
I am using two FS, FS2 as a gateway for FS1. Gateway name "testgw", 
username "testuser", and a lua script dumping variables.

Case 1:
=======
Simple configuration according to FS wiki. Everything looks fine, 
incoming call from FS2:

FS2: Registred as: Contact: "user" 
<sip:gw+testgw at 212.180.x.x:5080;transport=udp;gw=testgw>
FS1: variable_sip_gateway: testgw

Case 2:
=======
Configuration with added:
<param name="extension-in-contact" value="true"/>

FS2: Registered as :Contact: "user" 
<sip:testuser at 212.180.x.x:5080;transport=udp;gw=testgw>
FS1: variable_sip_gateway does not exist (!)

Case 3:
=======
Configuration from Case 1.
I am using Linksys SPA adapter, calling FS1 directly by ip, using 
"gw+testgw" as a destination user/number.

FS1: variable_sip_gateway: testgw
(looks like serious security flaw)

-- 
Artur Cichocki


From manieq at wp.eu  Tue Jan 18 20:33:23 2011
From: manieq at wp.eu (Mariusz Czulada)
Date: Tue, 18 Jan 2011 18:33:23 +0100
Subject: [Freeswitch-users] bind_meta_app blocks DTMFs?
In-Reply-To: <4d307a65a43472.49337517@wp.pl>
	<AANLkTinAx-cxv_a5b8+VA89duWY=xHjOOmMcZ604Z8R8@mail.gmail.com>
References: <4d307a65a43472.49337517@wp.pl>
	<AANLkTinAx-cxv_a5b8+VA89duWY=xHjOOmMcZ604Z8R8@mail.gmail.com>
Message-ID: <4d35cee3110bc5.51489825@wp.pl>

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From noyami at hotmail.com  Tue Jan 18 11:50:44 2011
From: noyami at hotmail.com (ami noy)
Date: Tue, 18 Jan 2011 08:50:44 +0000
Subject: [Freeswitch-users] load skypopen fail
Message-ID: <SNT105-W100A50A3E4E446E65A02E2D2F70@phx.gbl>


Hi
 
I installed last FreeSwitch version on windows (15.1.11)
I want to work with skype but when I use the command "load mod_skypopen" I get error message:
 
2011-01-18 10:16:44.348536 [CRIT] switch_loadable_module.c:928 Error Loading mod
ule C:\Program Files\FreeSWITCH\mod\mod_skypopen.dll
**dll open error [126l]
**
I couldn't fine mod_skypopen.dll anywhere/
What shall I do?
 
Thanks
Ami 		 	   		  
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From tlstutsman at gmail.com  Tue Jan 18 19:46:34 2011
From: tlstutsman at gmail.com (Travis Stutsman)
Date: Tue, 18 Jan 2011 11:46:34 -0500
Subject: [Freeswitch-users] Yet another DTMF question
Message-ID: <AANLkTik030ecFw2bvKe74pOs+NzfqHknX8keDO3mQ=cH@mail.gmail.com>

I've looked through a lot of threads regarding DTMF troubles, but none of
them seem to relate to me.  I've just pulled a fresh copy of the freeswitch
source and compiled it and I cannot get my digits recognized - neither
in-band nor out-of-band.  I haven't messed with INFO yet as my phones don't
support it.  I've done packet captures on my local interface and on the
server's interface and I can see the RTP rfc2833 digits showing up, but
freeswitch completely ignores them.  I feel like I must be overlooking
something horribly obvious.  How can this be?  Any hints guys?  Thanks in
advance!
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From steveayre at gmail.com  Tue Jan 18 22:02:00 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 18 Jan 2011 19:02:00 +0000
Subject: [Freeswitch-users] Yet another DTMF question
In-Reply-To: <AANLkTik030ecFw2bvKe74pOs+NzfqHknX8keDO3mQ=cH@mail.gmail.com>
References: <AANLkTik030ecFw2bvKe74pOs+NzfqHknX8keDO3mQ=cH@mail.gmail.com>
Message-ID: <AANLkTikqQodjsMT_VB1xeQdtY-2DEgL68ndEnHTBTvkb@mail.gmail.com>

Has the call been answered? (DTMF is ignored before the call is answered)

What payload type are the RFC2833 packets using?
What does the SDP look like?
Does the payload type in the packets match the telephone-event in the SDP?

-Steve


On 18 January 2011 16:46, Travis Stutsman <tlstutsman at gmail.com> wrote:

> I've looked through a lot of threads regarding DTMF troubles, but none of
> them seem to relate to me.  I've just pulled a fresh copy of the freeswitch
> source and compiled it and I cannot get my digits recognized - neither
> in-band nor out-of-band.  I haven't messed with INFO yet as my phones don't
> support it.  I've done packet captures on my local interface and on the
> server's interface and I can see the RTP rfc2833 digits showing up, but
> freeswitch completely ignores them.  I feel like I must be overlooking
> something horribly obvious.  How can this be?  Any hints guys?  Thanks in
> advance!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kbdfck at gmail.com  Tue Jan 18 22:41:30 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Tue, 18 Jan 2011 22:41:30 +0300
Subject: [Freeswitch-users] Music on hold runs only once with bind_meta_app
	or bind_digit_action
Message-ID: <AANLkTi=ZoVVM5ikk7dWU8f+Oe-tO6hFCgtMcVvrsruFZ@mail.gmail.com>

Hi all,
Sorry for duplicating this issue
(http://lists.freeswitch.org/pipermail/freeswitch-users/2011-January/067460.html)
but I run some tests and have additional info.

A calls B
B answers A
A launches meta app by using *7, application calls read() and collects
extension digits.
B listens to MOH
A calls att_xfer to entered extension via local channel or direct
sofia bridge to C. (no difference for the issue).
We are testing situation when bridge fails - in this case A trying to
repeat *7 and re-enter extension, so:

A is still connected to B
A launches meta app by using *7, application calls read() and collects digits.
B DOESN'T hear MOH, just silence
A calls att_xfer to entered extension via local channel or direct
sofia bridge to C.

Att_xfer works fine by itself except for second and following retries
there is no MOH at all. There are no difference for local channel
proxy or direct sofia bridge. What can be done to track down and fix
this issue?
Also, to make att_xfer work with local_channel I had to set
loopback_bowout=false and loopback_bowout_on_execute=false, but I
can't find any details on their behavior. Which one is really needed,
and what they exactly do?

Thanks in advance

-- 
Best regards,

Dmitry Sytchev,
IT Engineer


From phone.bytes at gmail.com  Tue Jan 18 23:16:50 2011
From: phone.bytes at gmail.com (Phone)
Date: Tue, 18 Jan 2011 13:16:50 -0700
Subject: [Freeswitch-users] Caller ID Number not going out on PRI
Message-ID: <4D35F532.2080802@gmail.com>

A few upgrades ago, the Caller ID Number on outbound calls stopped 
working.  Now it always displays the extension number, where before it 
was using the value specified in the directory entry.  Seems like maybe 
the variables are not passing from the directory entry to the dial plan.

Any ideas?


From hesser4900 at gmail.com  Tue Jan 18 23:26:49 2011
From: hesser4900 at gmail.com (Holger Esser)
Date: Tue, 18 Jan 2011 14:26:49 -0600
Subject: [Freeswitch-users] TDM line maximum
Message-ID: <AANLkTim4Mhm+KTAs-5vbNYLtijkziZfbzcrWzmLPj5Bw@mail.gmail.com>

All,

Does anybody have experience with implementing 4 Sangoma  octal cards and
bridge these channels to SIP sessions on a single server for a total of 1536
sessions (TDM+SIP)

Regards,
Holger
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From wstephen80 at gmail.com  Tue Jan 18 23:42:50 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 18 Jan 2011 21:42:50 +0100
Subject: [Freeswitch-users] TDM line maximum
In-Reply-To: <AANLkTim4Mhm+KTAs-5vbNYLtijkziZfbzcrWzmLPj5Bw@mail.gmail.com>
References: <AANLkTim4Mhm+KTAs-5vbNYLtijkziZfbzcrWzmLPj5Bw@mail.gmail.com>
Message-ID: <AANLkTi=sCw9KmY=_fyQjv9jjaMSLfnB6r_onFFh_ZoXP@mail.gmail.com>

Yes, we have a single server installation with 5 Sangoma A108 board that
regularly handles more than 1500 contemporary sessions.

Regards,
Stephen

On Tue, Jan 18, 2011 at 9:26 PM, Holger Esser <hesser4900 at gmail.com> wrote:

> All,
>
> Does anybody have experience with implementing 4 Sangoma  octal cards and
> bridge these channels to SIP sessions on a single server for a total of 1536
> sessions (TDM+SIP)
>
> Regards,
> Holger
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From hesser4900 at gmail.com  Tue Jan 18 23:47:27 2011
From: hesser4900 at gmail.com (Holger Esser)
Date: Tue, 18 Jan 2011 14:47:27 -0600
Subject: [Freeswitch-users] TDM line maximum
In-Reply-To: <AANLkTi=sCw9KmY=_fyQjv9jjaMSLfnB6r_onFFh_ZoXP@mail.gmail.com>
References: <AANLkTim4Mhm+KTAs-5vbNYLtijkziZfbzcrWzmLPj5Bw@mail.gmail.com>
	<AANLkTi=sCw9KmY=_fyQjv9jjaMSLfnB6r_onFFh_ZoXP@mail.gmail.com>
Message-ID: <AANLkTinPyGGSrXGD5zDQgq7kQEyeoj_Lpq6gc+aytyc5@mail.gmail.com>

May I ask on what hardware, I am planing on using an HP g7.

Many thanks,
Holger

On Tue, Jan 18, 2011 at 2:42 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> Yes, we have a single server installation with 5 Sangoma A108 board that
> regularly handles more than 1500 contemporary sessions.
>
> Regards,
> Stephen
>
> On Tue, Jan 18, 2011 at 9:26 PM, Holger Esser <hesser4900 at gmail.com>wrote:
>
>> All,
>>
>> Does anybody have experience with implementing 4 Sangoma  octal cards and
>> bridge these channels to SIP sessions on a single server for a total of 1536
>> sessions (TDM+SIP)
>>
>> Regards,
>> Holger
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tlstutsman at gmail.com  Tue Jan 18 22:17:15 2011
From: tlstutsman at gmail.com (Travis Stutsman)
Date: Tue, 18 Jan 2011 14:17:15 -0500
Subject: [Freeswitch-users] Yet another DTMF question
In-Reply-To: <AANLkTikqQodjsMT_VB1xeQdtY-2DEgL68ndEnHTBTvkb@mail.gmail.com>
References: <AANLkTik030ecFw2bvKe74pOs+NzfqHknX8keDO3mQ=cH@mail.gmail.com>
	<AANLkTikqQodjsMT_VB1xeQdtY-2DEgL68ndEnHTBTvkb@mail.gmail.com>
Message-ID: <AANLkTiksK3T24TUDO-7aq_tbkFCkj_4BRNpLRt4=GtBj@mail.gmail.com>

Call has been answered - I can hear the IVR.  Payload type is 101 - packets
on both ends verify this.  SDP example follows:


2011-01-18 19:10:13.579728 [DEBUG] mod_sofia.c:681 Local SDP
sofia/internal/1000 at x.x.x.x:
v=0
o=FreeSWITCH 1295351291 1295351292 IN IP4 x.x.x.x
s=FreeSWITCH
c=IN IP4 x.x.x.x
t=0 0
m=audio 26522 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv






On Tue, Jan 18, 2011 at 2:02 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Has the call been answered? (DTMF is ignored before the call is answered)
>
> What payload type are the RFC2833 packets using?
> What does the SDP look like?
> Does the payload type in the packets match the telephone-event in the SDP?
>
> -Steve
>
>
> On 18 January 2011 16:46, Travis Stutsman <tlstutsman at gmail.com> wrote:
>
>> I've looked through a lot of threads regarding DTMF troubles, but none of
>> them seem to relate to me.  I've just pulled a fresh copy of the freeswitch
>> source and compiled it and I cannot get my digits recognized - neither
>> in-band nor out-of-band.  I haven't messed with INFO yet as my phones don't
>> support it.  I've done packet captures on my local interface and on the
>> server's interface and I can see the RTP rfc2833 digits showing up, but
>> freeswitch completely ignores them.  I feel like I must be overlooking
>> something horribly obvious.  How can this be?  Any hints guys?  Thanks in
>> advance!
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From wstephen80 at gmail.com  Wed Jan 19 00:32:05 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 18 Jan 2011 22:32:05 +0100
Subject: [Freeswitch-users] TDM line maximum
In-Reply-To: <AANLkTinPyGGSrXGD5zDQgq7kQEyeoj_Lpq6gc+aytyc5@mail.gmail.com>
References: <AANLkTim4Mhm+KTAs-5vbNYLtijkziZfbzcrWzmLPj5Bw@mail.gmail.com>
	<AANLkTi=sCw9KmY=_fyQjv9jjaMSLfnB6r_onFFh_ZoXP@mail.gmail.com>
	<AANLkTinPyGGSrXGD5zDQgq7kQEyeoj_Lpq6gc+aytyc5@mail.gmail.com>
Message-ID: <AANLkTikCruXsaL1Jo6rarbCHmbPjLDfT2Hoa4UGaVM5p@mail.gmail.com>

HP DL380 G6 with two xeon 5520 8Gb ram.

Stephen

On Tue, Jan 18, 2011 at 9:47 PM, Holger Esser <hesser4900 at gmail.com> wrote:

> May I ask on what hardware, I am planing on using an HP g7.
>
> Many thanks,
> Holger
>
>
> On Tue, Jan 18, 2011 at 2:42 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> Yes, we have a single server installation with 5 Sangoma A108 board that
>> regularly handles more than 1500 contemporary sessions.
>>
>> Regards,
>> Stephen
>>
>> On Tue, Jan 18, 2011 at 9:26 PM, Holger Esser <hesser4900 at gmail.com>wrote:
>>
>>> All,
>>>
>>> Does anybody have experience with implementing 4 Sangoma  octal cards and
>>> bridge these channels to SIP sessions on a single server for a total of 1536
>>> sessions (TDM+SIP)
>>>
>>> Regards,
>>> Holger
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From avi at avimarcus.net  Wed Jan 19 02:17:24 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 19 Jan 2011 01:17:24 +0200
Subject: [Freeswitch-users] Sharing a VM Mailbox?
Message-ID: <AANLkTi=1OMJ93+vrF0bQvytJAD1TMTRs0qBm-obOdDGV@mail.gmail.com>

How do I set users to share a VM box?
I'm referring mostly to 1) receiving MWI for the other account, and 2)
dialing *98 to get the shared messages, without funny stuff in the dialplan.
(Or a not too convoluted workaround...)

I see <user id="1010" *mailbox*="1010"> on
http://wiki.freeswitch.org/wiki/Variable_voicemail_domain
but that didn't seem to work.

Suggestions?

Thanks,
Avi Marcus
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From msc at freeswitch.org  Wed Jan 19 03:02:40 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 Jan 2011 16:02:40 -0800
Subject: [Freeswitch-users] user hangup in a conference
In-Reply-To: <E633254F6B744A5FB54DBB406C69D01E@e1705>
References: <E633254F6B744A5FB54DBB406C69D01E@e1705>
Message-ID: <AANLkTin4tmLDDAAh-tPtF-7-sLx2R2YTwmfGYJ5tveOq@mail.gmail.com>

Your best bet would be to launch a lua or perl script and do the checking
there, then route based upon what you find.
-MC

On Mon, Jan 17, 2011 at 9:46 PM, Madovsky <infos at madovsky.org> wrote:

>  is there a way to know whe a conference doesn't exist anymore from the
> dialplan ?
>
> like after this line
>
> <action application="set" data="hangup_after_bridge=true"/>
> <action application="set" data="continue_on_fail=true"/>
> <action application="conference" data="conf_name}@default+${conf_pin}"/<conf_name%7D at default+$%7Bconf_pin%7D%22/>
> >
>
> i'd like to know if there are members yet in the conference
>
> <action application="set" data="myVar=${conference(conf_name list)}"/>
>
> it works if the user send a dtmf key lke # (from conference default control
> keys)
> but if the user hangup as normal so the dialplan doesn't continue after the
> conference XML line.
>
> Any idea ?
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From wasim at convergence.pk  Wed Jan 19 03:05:59 2011
From: wasim at convergence.pk (Wasim Baig)
Date: Wed, 19 Jan 2011 05:05:59 +0500
Subject: [Freeswitch-users] TDM line maximum
In-Reply-To: <AANLkTikCruXsaL1Jo6rarbCHmbPjLDfT2Hoa4UGaVM5p@mail.gmail.com>
References: <AANLkTim4Mhm+KTAs-5vbNYLtijkziZfbzcrWzmLPj5Bw@mail.gmail.com>
	<AANLkTi=sCw9KmY=_fyQjv9jjaMSLfnB6r_onFFh_ZoXP@mail.gmail.com>
	<AANLkTinPyGGSrXGD5zDQgq7kQEyeoj_Lpq6gc+aytyc5@mail.gmail.com>
	<AANLkTikCruXsaL1Jo6rarbCHmbPjLDfT2Hoa4UGaVM5p@mail.gmail.com>
Message-ID: <AANLkTintVezYNGzpsqe4cfL4Rsn1ituw=hOeYqwfjj=Q@mail.gmail.com>

Stephen:

Can you post the CPU, memory and IO footprints at these call volumes?

a dstat line or two would do ...

thanks

-wasim

On Wed, Jan 19, 2011 at 02:32, Stephen Wilde <wstephen80 at gmail.com> wrote:

> HP DL380 G6 with two xeon 5520 8Gb ram.
>
> Stephen
>
>
> On Tue, Jan 18, 2011 at 9:47 PM, Holger Esser <hesser4900 at gmail.com>wrote:
>
>> May I ask on what hardware, I am planing on using an HP g7.
>>
>> Many thanks,
>> Holger
>>
>>
>> On Tue, Jan 18, 2011 at 2:42 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>
>>> Yes, we have a single server installation with 5 Sangoma A108 board that
>>> regularly handles more than 1500 contemporary sessions.
>>>
>>> Regards,
>>> Stephen
>>>
>>> On Tue, Jan 18, 2011 at 9:26 PM, Holger Esser <hesser4900 at gmail.com>wrote:
>>>
>>>> All,
>>>>
>>>> Does anybody have experience with implementing 4 Sangoma  octal cards
>>>> and bridge these channels to SIP sessions on a single server for a total of
>>>> 1536 sessions (TDM+SIP)
>>>>
>>>> Regards,
>>>> Holger
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
peace be upon you ...
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From hesser4900 at gmail.com  Wed Jan 19 03:07:53 2011
From: hesser4900 at gmail.com (Holger Esser)
Date: Tue, 18 Jan 2011 18:07:53 -0600
Subject: [Freeswitch-users] TDM line maximum
In-Reply-To: <AANLkTikCruXsaL1Jo6rarbCHmbPjLDfT2Hoa4UGaVM5p@mail.gmail.com>
References: <AANLkTim4Mhm+KTAs-5vbNYLtijkziZfbzcrWzmLPj5Bw@mail.gmail.com>
	<AANLkTi=sCw9KmY=_fyQjv9jjaMSLfnB6r_onFFh_ZoXP@mail.gmail.com>
	<AANLkTinPyGGSrXGD5zDQgq7kQEyeoj_Lpq6gc+aytyc5@mail.gmail.com>
	<AANLkTikCruXsaL1Jo6rarbCHmbPjLDfT2Hoa4UGaVM5p@mail.gmail.com>
Message-ID: <AANLkTing1_9B=k+0tYq0qj3nVg4Lp1cc0YL8uN-R-8GN@mail.gmail.com>

Thank you. Would you have CPU data under load.

Holger

On 1/18/11, Stephen Wilde <wstephen80 at gmail.com> wrote:
> HP DL380 G6 with two xeon 5520 8Gb ram.
>
> Stephen
>
> On Tue, Jan 18, 2011 at 9:47 PM, Holger Esser <hesser4900 at gmail.com> wrote:
>
>> May I ask on what hardware, I am planing on using an HP g7.
>>
>> Many thanks,
>> Holger
>>
>>
>> On Tue, Jan 18, 2011 at 2:42 PM, Stephen Wilde
>> <wstephen80 at gmail.com>wrote:
>>
>>> Yes, we have a single server installation with 5 Sangoma A108 board that
>>> regularly handles more than 1500 contemporary sessions.
>>>
>>> Regards,
>>> Stephen
>>>
>>> On Tue, Jan 18, 2011 at 9:26 PM, Holger Esser
>>> <hesser4900 at gmail.com>wrote:
>>>
>>>> All,
>>>>
>>>> Does anybody have experience with implementing 4 Sangoma  octal cards
>>>> and
>>>> bridge these channels to SIP sessions on a single server for a total of
>>>> 1536
>>>> sessions (TDM+SIP)
>>>>
>>>> Regards,
>>>> Holger
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>

-- 
Sent from my mobile device


From cjbujold at accra.ca  Wed Jan 19 03:04:19 2011
From: cjbujold at accra.ca (Charles Bujold)
Date: Tue, 18 Jan 2011 20:04:19 -0400
Subject: [Freeswitch-users] Dial Plan timing error
Message-ID: <000f01cbb76c$6bd2f710$4378e530$@accra.ca>

 

Need help configuring the following dial plan.  To make a PSTN call I have
to call extension 510 or 520, then bridge and answer the call to get a dial
tone.  Then dial the outgoing number.  My problem is that the send_dtmf
command does not wait 1 second (W command) after the answer.  It seems to
delay the answer command by 1 second.   How do I get the call to answer and
then wait 1 second and then send /dial the number?

 

Thanks

CJB

 

 

<extension name="Local 7 digit calls" >

   <condition field="destination_number" expression="^(\d{7})$">   

       <action application="set"
data="effective_caller_id_name=${outbound_caller_id_name}"/>

       <action application="set"
data="effective_caller_id_number=${outbound_caller_id_number}"/>

       <action application="set" data="hangup_after_bridge=true"/>

       <action application="send_dtmf" data="W$1 at 100"/> 

       <action application="bridge" data="sofia/${domain}/510"/> 

       <action application="bridge" data="sofia/${domain}/520"/> 

       <action application="bridge" data="error/user_busy" />

   </condition>

</extension>

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From infos at madovsky.org  Wed Jan 19 03:10:57 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 18 Jan 2011 19:10:57 -0500
Subject: [Freeswitch-users] user hangup in a conference
References: <E633254F6B744A5FB54DBB406C69D01E@e1705>
	<AANLkTin4tmLDDAAh-tPtF-7-sLx2R2YTwmfGYJ5tveOq@mail.gmail.com>
Message-ID: <688E55E68E474689B057715B4D918EC6@e1705>

yes,
I found also

<action application="set" data="session_in_hangup_hook=true"/>
<action application="set" data="api_hangup_hook=system /usr/local/bin/php /usr/local/freeswitch/scripts/php/myPhpScript.php 'blablah' 'bililibli'"/>

and in my php script I do what needed

Thx


  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, January 18, 2011 7:02 PM
  Subject: Re: [Freeswitch-users] user hangup in a conference


  Your best bet would be to launch a lua or perl script and do the checking there, then route based upon what you find.
  -MC


  On Mon, Jan 17, 2011 at 9:46 PM, Madovsky <infos at madovsky.org> wrote:

    is there a way to know whe a conference doesn't exist anymore from the dialplan ?

    like after this line

    <action application="set" data="hangup_after_bridge=true"/>
    <action application="set" data="continue_on_fail=true"/>
    <action application="conference" data="conf_name}@default+${conf_pin}"/>

    i'd like to know if there are members yet in the conference

    <action application="set" data="myVar=${conference(conf_name list)}"/>

    it works if the user send a dtmf key lke # (from conference default control keys)
    but if the user hangup as normal so the dialplan doesn't continue after the conference XML line.

    Any idea ?

    Thanks

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org






------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Wed Jan 19 03:37:56 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 18 Jan 2011 18:37:56 -0600
Subject: [Freeswitch-users] Dial Plan timing error
In-Reply-To: <000f01cbb76c$6bd2f710$4378e530$@accra.ca>
References: <000f01cbb76c$6bd2f710$4378e530$@accra.ca>
Message-ID: <4A4582DA-68D5-4F37-B1F8-35AFED2DA12B@freeswitch.org>

you use queue_dtmf and not send_dtmf in this case.

/b

On Jan 18, 2011, at 6:04 PM, Charles Bujold wrote:

>  
> Need help configuring the following dial plan.  To make a PSTN call I have to call extension 510 or 520, then bridge and answer the call to get a dial tone.  Then dial the outgoing number.  My problem is that the send_dtmf  command does not wait 1 second (W command) after the answer.  It seems to delay the answer command by 1 second.   How do I get the call to answer and then wait 1 second and then send /dial the number?
>  
> Thanks
> CJB
>  

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From brian at freeswitch.org  Wed Jan 19 03:55:19 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 18 Jan 2011 18:55:19 -0600
Subject: [Freeswitch-users] TDM line maximum
In-Reply-To: <AANLkTim4Mhm+KTAs-5vbNYLtijkziZfbzcrWzmLPj5Bw@mail.gmail.com>
References: <AANLkTim4Mhm+KTAs-5vbNYLtijkziZfbzcrWzmLPj5Bw@mail.gmail.com>
Message-ID: <61E86548-7627-45AF-9955-B7A53796735D@freeswitch.org>

Sangoma always tests with 32 E1's in their labs... so I know it can do it.

/b

On Jan 18, 2011, at 2:26 PM, Holger Esser wrote:

> All,
> 
> Does anybody have experience with implementing 4 Sangoma  octal cards and bridge these channels to SIP sessions on a single server for a total of 1536 sessions (TDM+SIP)
> 
> Regards,
> Holger



From hesser4900 at gmail.com  Wed Jan 19 04:41:36 2011
From: hesser4900 at gmail.com (Holger Esser)
Date: Tue, 18 Jan 2011 19:41:36 -0600
Subject: [Freeswitch-users] TDM line maximum
In-Reply-To: <61E86548-7627-45AF-9955-B7A53796735D@freeswitch.org>
References: <AANLkTim4Mhm+KTAs-5vbNYLtijkziZfbzcrWzmLPj5Bw@mail.gmail.com>
	<61E86548-7627-45AF-9955-B7A53796735D@freeswitch.org>
Message-ID: <AANLkTinB956aekSxnzH_1LoRRa0+FzwRAEZB=H7t+tTU@mail.gmail.com>

Many thanks for all the pointers.

On 1/18/11, Brian West <brian at freeswitch.org> wrote:
> Sangoma always tests with 32 E1's in their labs... so I know it can do it.
>
> /b
>
> On Jan 18, 2011, at 2:26 PM, Holger Esser wrote:
>
>> All,
>>
>> Does anybody have experience with implementing 4 Sangoma  octal cards and
>> bridge these channels to SIP sessions on a single server for a total of
>> 1536 sessions (TDM+SIP)
>>
>> Regards,
>> Holger
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device


From chris at cloudtel.com  Wed Jan 19 09:59:35 2011
From: chris at cloudtel.com (Chris Burns)
Date: Wed, 19 Jan 2011 01:59:35 -0500
Subject: [Freeswitch-users] Sharing a VM Mailbox?
In-Reply-To: <AANLkTi=1OMJ93+vrF0bQvytJAD1TMTRs0qBm-obOdDGV@mail.gmail.com>
References: <AANLkTi=1OMJ93+vrF0bQvytJAD1TMTRs0qBm-obOdDGV@mail.gmail.com>
Message-ID: <AANLkTi=0Q=5ky+FXpNRSk-cHTqUj8D=k6uu4s1Rmf8Ff@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Mod_voicemail#Check_Voice_Mail

Following that example, you could add ${mailbox} to data when you call
voicemail app in the dialplan (eg. "check default ${domain_name}
${mailbox}"). You are setting the mailbox variable in the directory but you
need a bit of "funny stuff in the dialplan" to use it :)

On Tue, Jan 18, 2011 at 6:17 PM, Avi Marcus <avi at avimarcus.net> wrote:

> How do I set users to share a VM box?
> I'm referring mostly to 1) receiving MWI for the other account, and 2)
> dialing *98 to get the shared messages, without funny stuff in the dialplan.
> (Or a not too convoluted workaround...)
>
> I see <user id="1010" *mailbox*="1010"> on
> http://wiki.freeswitch.org/wiki/Variable_voicemail_domain
> but that didn't seem to work.
>
> Suggestions?
>
> Thanks,
> Avi Marcus
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Wed Jan 19 10:55:46 2011
From: u2nsam at gmail.com (Sam)
Date: Wed, 19 Jan 2011 13:25:46 +0530
Subject: [Freeswitch-users] call disconnects on call forwarding from sangoma
Message-ID: <AANLkTinQWTD=NuGE77qAjJHt4o-WLBHc=Dz+bEdROj5T@mail.gmail.com>

Hi,

When i make a call from extension to extension, after timeout it gets
forwarded properly on failure(i.e no answer) to my mobile number,

but when i call from outside phone to my extension it do not get forwarded
properly giving me the sangoma error  ,

which i have pasted it in pastebin link ( check line 244 ) : :
http://pastebin.freeswitch.org/15064

What could be the reason ?

Regards
Sam
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From steveayre at gmail.com  Wed Jan 19 11:28:32 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 19 Jan 2011 08:28:32 +0000
Subject: [Freeswitch-users] Dial Plan timing error
In-Reply-To: <000f01cbb76c$6bd2f710$4378e530$@accra.ca>
References: <000f01cbb76c$6bd2f710$4378e530$@accra.ca>
Message-ID: <A667782C-EB9A-437A-8C37-510148B2394D@gmail.com>

Try doing the send_dtmf
in the execute_on_answer variable

Steve on iPhone

On 19 Jan 2011, at 00:04, "Charles Bujold" <cjbujold at accra.ca> wrote:

>  
> 
> Need help configuring the following dial plan.  To make a PSTN call I have to call extension 510 or 520, then bridge and answer the call to get a dial tone.  Then dial the outgoing number.  My problem is that the send_dtmf  command does not wait 1 second (W command) after the answer.  It seems to delay the answer command by 1 second.   How do I get the call to answer and then wait 1 second and then send /dial the number?
> 
>  
> 
> Thanks
> 
> CJB
> 
>  
> 
>  
> 
> <extension name="Local 7 digit calls" >
> 
>    <condition field="destination_number" expression="^(\d{7})$">  
> 
>        <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>
> 
>        <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
> 
>        <action application="set" data="hangup_after_bridge=true"/>
> 
>        <action application="send_dtmf" data="W$1 at 100"/>
> 
>        <action application="bridge" data="sofia/${domain}/510"/>
> 
>        <action application="bridge" data="sofia/${domain}/520"/>
> 
>        <action application="bridge" data="error/user_busy" />
> 
>    </condition>
> 
> </extension>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From steveayre at gmail.com  Wed Jan 19 11:30:51 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 19 Jan 2011 08:30:51 +0000
Subject: [Freeswitch-users] Yet another DTMF question
In-Reply-To: <AANLkTiksK3T24TUDO-7aq_tbkFCkj_4BRNpLRt4=GtBj@mail.gmail.com>
References: <AANLkTik030ecFw2bvKe74pOs+NzfqHknX8keDO3mQ=cH@mail.gmail.com>
	<AANLkTikqQodjsMT_VB1xeQdtY-2DEgL68ndEnHTBTvkb@mail.gmail.com>
	<AANLkTiksK3T24TUDO-7aq_tbkFCkj_4BRNpLRt4=GtBj@mail.gmail.com>
Message-ID: <62C5671A-8657-4F99-8184-272EEBEC8E77@gmail.com>

What about the remote sdp? Just checking fs is expecting 101.

Steve on iPhone

On 18 Jan 2011, at 19:17, Travis Stutsman <tlstutsman at gmail.com> wrote:

> Call has been answered - I can hear the IVR.  Payload type is 101 - packets on both ends verify this.  SDP example follows:
> 
> 
> 2011-01-18 19:10:13.579728 [DEBUG] mod_sofia.c:681 Local SDP sofia/internal/1000 at x.x.x.x:
> v=0
> o=FreeSWITCH 1295351291 1295351292 IN IP4 x.x.x.x
> s=FreeSWITCH
> c=IN IP4 x.x.x.x
> t=0 0
> m=audio 26522 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> 
> 
> 
> 
> 
> 
> On Tue, Jan 18, 2011 at 2:02 PM, Steven Ayre <steveayre at gmail.com> wrote:
> Has the call been answered? (DTMF is ignored before the call is answered)
> 
> What payload type are the RFC2833 packets using?
> What does the SDP look like?
> Does the payload type in the packets match the telephone-event in the SDP?
> 
> -Steve
> 
> 
> On 18 January 2011 16:46, Travis Stutsman <tlstutsman at gmail.com> wrote:
> I've looked through a lot of threads regarding DTMF troubles, but none of them seem to relate to me.  I've just pulled a fresh copy of the freeswitch source and compiled it and I cannot get my digits recognized - neither in-band nor out-of-band.  I haven't messed with INFO yet as my phones don't support it.  I've done packet captures on my local interface and on the server's interface and I can see the RTP rfc2833 digits showing up, but freeswitch completely ignores them.  I feel like I must be overlooking something horribly obvious.  How can this be?  Any hints guys?  Thanks in advance!
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From michal.bielicki at seventhsignal.de  Wed Jan 19 12:11:46 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Wed, 19 Jan 2011 10:11:46 +0100
Subject: [Freeswitch-users] Ideas for FreeSWITCH Marketing Material ?
Message-ID: <59B81A5D-5555-4BF0-82C1-ED44375046EA@seventhsignal.de>

We are doing a FreeSWITCH booth at Fosdem in 2 weeks and I am wondering if anybody has ideas for a flyer we could give to people there. I will cover the printing costs but I overestimated time for creativity I'd have before fosdem so looking for ideas :). Format would be A4 handouts and a A2 poster.
We would traanslate them into getrman as well for the Linux Tage in Chemnitz in March where we are running another FreeSWITCH booth.
cheers

Michal

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de

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From wstephen80 at gmail.com  Wed Jan 19 12:26:01 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Wed, 19 Jan 2011 10:26:01 +0100
Subject: [Freeswitch-users] TDM line maximum
In-Reply-To: <AANLkTintVezYNGzpsqe4cfL4Rsn1ituw=hOeYqwfjj=Q@mail.gmail.com>
References: <AANLkTim4Mhm+KTAs-5vbNYLtijkziZfbzcrWzmLPj5Bw@mail.gmail.com>
	<AANLkTi=sCw9KmY=_fyQjv9jjaMSLfnB6r_onFFh_ZoXP@mail.gmail.com>
	<AANLkTinPyGGSrXGD5zDQgq7kQEyeoj_Lpq6gc+aytyc5@mail.gmail.com>
	<AANLkTikCruXsaL1Jo6rarbCHmbPjLDfT2Hoa4UGaVM5p@mail.gmail.com>
	<AANLkTintVezYNGzpsqe4cfL4Rsn1ituw=hOeYqwfjj=Q@mail.gmail.com>
Message-ID: <AANLkTikzjwYv5gw=m70NhG_DOfQuCB9dv+V=RP_EcgNy@mail.gmail.com>

[root at localhost bin]# dstat
----total-cpu-usage---- -dsk/total- -net/total- ---paging-- ---system--
usr sys idl wai hiq siq| read  writ| recv  send|  in   out | int   csw
 11   6  80   0   1   2|1940B  314k|   0     0 |   0   0.1 |  12k   80k
 26  10  59   0   1   4|   0  2680k|3288k 3472k|   0     0 |  26k  183k
 26  11  58   0   1   4|   0   400k|3292k 3463k|   0     0 |  25k  190k
 24  11  60   0   1   4|   0     0 |3287k 3443k|   0     0 |  26k  192k
 25  11  59   0   1   4|   0     0 |3328k 3482k|   0     0 |  26k  192k
 24  11  60   0   1   4|   0     0 |3282k 3441k|   0     0 |  27k  187k
 26  12  57   0   1   4|   0     0 |3329k 3455k|   0     0 |  27k  180k
 26  12  57   0   1   4|   0   152k|3320k 3506k|   0     0 |  27k  178k
 25  12  58   0   1   3|   0     0 |3310k 3478k|   0     0 |  26k  181k


[root at localhost bin]# ./fs_cli -x "status"
UP 0 years, 0 days, 8 hours, 40 minutes, 32 seconds, 50 milliseconds, 274
microseconds
175400 session(s) since startup
1454 session(s) 25/100
2000 session(s) max
min idle cpu 0.00/57.00



[root at localhost bin]# wanrouter hwprobe
-------------------------------
| Wanpipe Hardware Probe Info |
-------------------------------
1 . AFT-A108-SH : SLOT=4 : BUS=17 : IRQ=169 : CPU=A : PORT=1 : HWEC=0 : V=41
2 . AFT-A108-SH : SLOT=4 : BUS=17 : IRQ=169 : CPU=A : PORT=2 : HWEC=0 : V=41
3 . AFT-A108-SH : SLOT=4 : BUS=17 : IRQ=169 : CPU=A : PORT=3 : HWEC=0 : V=41
4 . AFT-A108-SH : SLOT=4 : BUS=17 : IRQ=169 : CPU=A : PORT=4 : HWEC=0 : V=41
5 . AFT-A108-SH : SLOT=4 : BUS=17 : IRQ=169 : CPU=A : PORT=5 : HWEC=0 : V=41
6 . AFT-A108-SH : SLOT=4 : BUS=17 : IRQ=169 : CPU=A : PORT=6 : HWEC=0 : V=41
7 . AFT-A108-SH : SLOT=4 : BUS=17 : IRQ=169 : CPU=A : PORT=7 : HWEC=0 : V=41
8 . AFT-A108-SH : SLOT=4 : BUS=17 : IRQ=169 : CPU=A : PORT=8 : HWEC=0 : V=41
9 . AFT-A108-SH : SLOT=4 : BUS=21 : IRQ=177 : CPU=A : PORT=1 : HWEC=0 : V=41
10. AFT-A108-SH : SLOT=4 : BUS=21 : IRQ=177 : CPU=A : PORT=2 : HWEC=0 : V=41
11. AFT-A108-SH : SLOT=4 : BUS=21 : IRQ=177 : CPU=A : PORT=3 : HWEC=0 : V=41
12. AFT-A108-SH : SLOT=4 : BUS=21 : IRQ=177 : CPU=A : PORT=4 : HWEC=0 : V=41
13. AFT-A108-SH : SLOT=4 : BUS=21 : IRQ=177 : CPU=A : PORT=5 : HWEC=0 : V=41
14. AFT-A108-SH : SLOT=4 : BUS=21 : IRQ=177 : CPU=A : PORT=6 : HWEC=0 : V=41
15. AFT-A108-SH : SLOT=4 : BUS=21 : IRQ=177 : CPU=A : PORT=7 : HWEC=0 : V=41
16. AFT-A108-SH : SLOT=4 : BUS=21 : IRQ=177 : CPU=A : PORT=8 : HWEC=0 : V=41
17. AFT-A108-SH : SLOT=4 : BUS=24 : IRQ=185 : CPU=A : PORT=1 : HWEC=0 : V=41
18. AFT-A108-SH : SLOT=4 : BUS=24 : IRQ=185 : CPU=A : PORT=2 : HWEC=0 : V=41
19. AFT-A108-SH : SLOT=4 : BUS=24 : IRQ=185 : CPU=A : PORT=3 : HWEC=0 : V=41
20. AFT-A108-SH : SLOT=4 : BUS=24 : IRQ=185 : CPU=A : PORT=4 : HWEC=0 : V=41
21. AFT-A108-SH : SLOT=4 : BUS=24 : IRQ=185 : CPU=A : PORT=5 : HWEC=0 : V=41
22. AFT-A108-SH : SLOT=4 : BUS=24 : IRQ=185 : CPU=A : PORT=6 : HWEC=0 : V=41
23. AFT-A108-SH : SLOT=4 : BUS=24 : IRQ=185 : CPU=A : PORT=7 : HWEC=0 : V=41
24. AFT-A108-SH : SLOT=4 : BUS=24 : IRQ=185 : CPU=A : PORT=8 : HWEC=0 : V=41
25. AFT-A108-SH : SLOT=4 : BUS=14 : IRQ=193 : CPU=A : PORT=1 : HWEC=0 : V=41
26. AFT-A108-SH : SLOT=4 : BUS=14 : IRQ=193 : CPU=A : PORT=2 : HWEC=0 : V=41
27. AFT-A108-SH : SLOT=4 : BUS=14 : IRQ=193 : CPU=A : PORT=3 : HWEC=0 : V=41
28. AFT-A108-SH : SLOT=4 : BUS=14 : IRQ=193 : CPU=A : PORT=4 : HWEC=0 : V=41
29. AFT-A108-SH : SLOT=4 : BUS=14 : IRQ=193 : CPU=A : PORT=5 : HWEC=0 : V=41
30. AFT-A108-SH : SLOT=4 : BUS=14 : IRQ=193 : CPU=A : PORT=6 : HWEC=0 : V=41
31. AFT-A108-SH : SLOT=4 : BUS=14 : IRQ=193 : CPU=A : PORT=7 : HWEC=0 : V=41
32. AFT-A108-SH : SLOT=4 : BUS=14 : IRQ=193 : CPU=A : PORT=8 : HWEC=0 : V=41
33. AFT-A108-SH : SLOT=4 : BUS=11 : IRQ=201 : CPU=A : PORT=1 : HWEC=0 : V=41
34. AFT-A108-SH : SLOT=4 : BUS=11 : IRQ=201 : CPU=A : PORT=2 : HWEC=0 : V=41
35. AFT-A108-SH : SLOT=4 : BUS=11 : IRQ=201 : CPU=A : PORT=3 : HWEC=0 : V=41
36. AFT-A108-SH : SLOT=4 : BUS=11 : IRQ=201 : CPU=A : PORT=4 : HWEC=0 : V=41
37. AFT-A108-SH : SLOT=4 : BUS=11 : IRQ=201 : CPU=A : PORT=5 : HWEC=0 : V=41
38. AFT-A108-SH : SLOT=4 : BUS=11 : IRQ=201 : CPU=A : PORT=6 : HWEC=0 : V=41
39. AFT-A108-SH : SLOT=4 : BUS=11 : IRQ=201 : CPU=A : PORT=7 : HWEC=0 : V=41
40. AFT-A108-SH : SLOT=4 : BUS=11 : IRQ=201 : CPU=A : PORT=8 : HWEC=0 : V=41
41. AFT-A108-SH : SLOT=4 : BUS=8 : IRQ=209 : CPU=A : PORT=1 : HWEC=0 : V=41
42. AFT-A108-SH : SLOT=4 : BUS=8 : IRQ=209 : CPU=A : PORT=2 : HWEC=0 : V=41
43. AFT-A108-SH : SLOT=4 : BUS=8 : IRQ=209 : CPU=A : PORT=3 : HWEC=0 : V=41
44. AFT-A108-SH : SLOT=4 : BUS=8 : IRQ=209 : CPU=A : PORT=4 : HWEC=0 : V=41
45. AFT-A108-SH : SLOT=4 : BUS=8 : IRQ=209 : CPU=A : PORT=5 : HWEC=0 : V=41
46. AFT-A108-SH : SLOT=4 : BUS=8 : IRQ=209 : CPU=A : PORT=6 : HWEC=0 : V=41
47. AFT-A108-SH : SLOT=4 : BUS=8 : IRQ=209 : CPU=A : PORT=7 : HWEC=0 : V=41
48. AFT-A108-SH : SLOT=4 : BUS=8 : IRQ=209 : CPU=A : PORT=8 : HWEC=0 : V=41

Card Cnt: A108=6



On Wed, Jan 19, 2011 at 1:05 AM, Wasim Baig <wasim at convergence.pk> wrote:

> Stephen:
>
> Can you post the CPU, memory and IO footprints at these call volumes?
>
> a dstat line or two would do ...
>
> thanks
>
> -wasim
>
>
> On Wed, Jan 19, 2011 at 02:32, Stephen Wilde <wstephen80 at gmail.com> wrote:
>
>> HP DL380 G6 with two xeon 5520 8Gb ram.
>>
>> Stephen
>>
>>
>> On Tue, Jan 18, 2011 at 9:47 PM, Holger Esser <hesser4900 at gmail.com>wrote:
>>
>>> May I ask on what hardware, I am planing on using an HP g7.
>>>
>>> Many thanks,
>>> Holger
>>>
>>>
>>> On Tue, Jan 18, 2011 at 2:42 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>>
>>>> Yes, we have a single server installation with 5 Sangoma A108 board that
>>>> regularly handles more than 1500 contemporary sessions.
>>>>
>>>> Regards,
>>>> Stephen
>>>>
>>>> On Tue, Jan 18, 2011 at 9:26 PM, Holger Esser <hesser4900 at gmail.com>wrote:
>>>>
>>>>> All,
>>>>>
>>>>> Does anybody have experience with implementing 4 Sangoma  octal cards
>>>>> and bridge these channels to SIP sessions on a single server for a total of
>>>>> 1536 sessions (TDM+SIP)
>>>>>
>>>>> Regards,
>>>>> Holger
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
> peace be upon you ...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From avi at avimarcus.net  Wed Jan 19 12:40:41 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 19 Jan 2011 11:40:41 +0200
Subject: [Freeswitch-users] Sharing a VM Mailbox?
In-Reply-To: <AANLkTi=0Q=5ky+FXpNRSk-cHTqUj8D=k6uu4s1Rmf8Ff@mail.gmail.com>
References: <AANLkTi=1OMJ93+vrF0bQvytJAD1TMTRs0qBm-obOdDGV@mail.gmail.com>
	<AANLkTi=0Q=5ky+FXpNRSk-cHTqUj8D=k6uu4s1Rmf8Ff@mail.gmail.com>
Message-ID: <AANLkTikvgVSjNnbzzxYbNfc+iBt8ROQL52E3JYyZW-nj@mail.gmail.com>

I suppose that's possible, but what about the MWI subscribe?
On Jan 19, 2011 9:00 AM, "Chris Burns" <chris at cloudtel.com> wrote:
> http://wiki.freeswitch.org/wiki/Mod_voicemail#Check_Voice_Mail
>
> Following that example, you could add ${mailbox} to data when you call
> voicemail app in the dialplan (eg. "check default ${domain_name}
> ${mailbox}"). You are setting the mailbox variable in the directory but
you
> need a bit of "funny stuff in the dialplan" to use it :)
>
> On Tue, Jan 18, 2011 at 6:17 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> How do I set users to share a VM box?
>> I'm referring mostly to 1) receiving MWI for the other account, and 2)
>> dialing *98 to get the shared messages, without funny stuff in the
dialplan.
>> (Or a not too convoluted workaround...)
>>
>> I see <user id="1010" *mailbox*="1010"> on
>> http://wiki.freeswitch.org/wiki/Variable_voicemail_domain
>> but that didn't seem to work.
>>
>> Suggestions?
>>
>> Thanks,
>> Avi Marcus
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
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From yurazilot1 at list.ru  Wed Jan 19 12:41:28 2011
From: yurazilot1 at list.ru (ZILOT)
Date: Wed, 19 Jan 2011 12:41:28 +0300
Subject: [Freeswitch-users] my FS doesn't answer on OK from the provider
Message-ID: <E1PfUXg-0000tt-00.yurazilot1-list-ru@f117.mail.ru>

I have one problem - my FS doesn't answer on OK from the provider.
Probably the problem consists in an incorrectness incoming OK.
Please help.

  
U 2011/01/19 11:04:05.243062 194.190.211.190:5060 -> 77.72.17.23:5060
INVITE sip:74742515541 at 77.72.17.23 SIP/2.0.
Via: SIP/2.0/UDP 194.190.211.190;rport;branch=z9hG4bK6rQjK6B04ZeNN.
Max-Forwards: 69.
From: "DEN78141_120" <sip:TCAPI_User at 77.72.17.23;transport=udp>;tag=vZcKXKa0Q7B1e.
To: <sip:74742515541 at 77.72.17.23>.
Call-ID: 86959393-9e45-122e-1782-33002db2a994.
CSeq: 7376058 INVITE.
Contact: <sip:gw+trunk_8 at 194.190.211.190:5060;transport=udp;gw=trunk_8>.
User-Agent: Configured by 2600hz.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: precondition, path, replaces.
Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 298.
X-FS-Support: update_display.
Remote-Party-ID: "DEN78141_120" <sip:DEN78141_120 at 77.72.17.23>;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1295404175 1295404176 IN IP4 194.190.211.190.
s=FreeSWITCH.
c=IN IP4 194.190.211.190.
t=0 0.
m=audio 20070 RTP/AVP 0 9 8 101 13.
a=rtpmap:0 PCMU/8000.
a=rtpmap:9 G722/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:30.

U 2011/01/19 11:04:05.288882 77.72.17.23:5060 -> 194.190.211.190:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 194.190.211.190:5060;rport;branch=z9hG4bK6rQjK6B04ZeNN.
From: "DEN78141_120" <sip:TCAPI_User at 77.72.17.23;transport=udp>;tag=vZcKXKa0Q7B1e.
To: <sip:74742515541 at 77.72.17.23>;tag=0eff5d003cff3610ff00001372630472.
Call-ID: 86959393-9e45-122e-1782-33002db2a994.
CSeq: 7376058 INVITE.
Contact: <sip:74742515541 at 77.72.17.23;user=phone>.
Server: MERA MSIP v.1.0.2.
Content-Length: 0.
.

U 2011/01/19 11:04:06.609863 77.72.17.23:5060 -> 194.190.211.190:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 194.190.211.190:5060;rport;branch=z9hG4bK6rQjK6B04ZeNN.
From: "DEN78141_120" <sip:TCAPI_User at 77.72.17.23;transport=udp>;tag=vZcKXKa0Q7B1e.
To: <sip:74742515541 at 77.72.17.23>;tag=0eff5d003cff3610ff00001372630472.
Call-ID: 86959393-9e45-122e-1782-33002db2a994.
CSeq: 7376058 INVITE.
Contact: <sip:74742515541 at 77.72.17.23;user=phone>.
Server: MERA MSIP v.1.0.2.
Content-Type: application/sdp.
Content-Length:   212.
.
v=0.
o=- 1295424317 1295424317 IN IP4 77.72.17.23.
s=-.
c=IN IP4 77.72.17.23.
t=0 0.
m=audio 18360 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

U 2011/01/19 11:04:14.238660 77.72.17.23:5060 -> 194.190.211.190:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 194.190.211.190:5060;rport;branch=z9hG4bK6rQjK6B04ZeNN.
From: "DEN78141_120" <sip:TCAPI_User at 77.72.17.23;transport=udp>;tag=vZcKXKa0Q7B1e.
To: <sip:74742515541 at 77.72.17.23>;tag=0eff5d003cff3610ff00001372630472.
Call-ID: 86959393-9e45-122e-1782-33002db2a994.
CSeq: 7376058 INVITE.
Contact: <sip:74742515541 at 77.72.17.23;user=phone>.
Server: MERA MSIP v.1.0.2.
Content-Type: application/sdp.
Content-Length:   212.
.
v=0.
o=- 1295424325 1295424325 IN IP4 77.72.17.23.
s=-.
c=IN IP4 77.72.17.23.
t=0 0.
m=audio 18360 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

U 2011/01/19 11:04:14.246657 77.72.17.23:5060 -> 194.190.211.190:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 194.190.211.190:5060;rport;branch=z9hG4bK6rQjK6B04ZeNN.
From: "DEN78141_120" <sip:TCAPI_User at 77.72.17.23;transport=udp>;tag=vZcKXKa0Q7B1e.
To: <sip:74742515541 at 77.72.17.23>;tag=0eff5d003cff3610ff00001372630472.
Call-ID: 86959393-9e45-122e-1782-33002db2a994.
CSeq: 7376058 INVITE.
Contact: <sip:74742515541 at 77.72.17.23;user=phone>.
Server: MERA MSIP v.1.0.2.
Content-Type: application/sdp.
Content-Length:   212.
.
v=0.
o=- 1295424325 1295424325 IN IP4 77.72.17.23.
s=-.
c=IN IP4 77.72.17.23.
t=0 0.
m=audio 18360 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

U 2011/01/19 11:04:14.747863 77.72.17.23:5060 -> 194.190.211.190:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 194.190.211.190:5060;rport;branch=z9hG4bK6rQjK6B04ZeNN.
From: "DEN78141_120" <sip:TCAPI_User at 77.72.17.23;transport=udp>;tag=vZcKXKa0Q7B1e.
To: <sip:74742515541 at 77.72.17.23>;tag=0eff5d003cff3610ff00001372630472.
Call-ID: 86959393-9e45-122e-1782-33002db2a994.
CSeq: 7376058 INVITE.
Contact: <sip:74742515541 at 77.72.17.23;user=phone>.
Server: MERA MSIP v.1.0.2.
Content-Type: application/sdp.
Content-Length:   212.
.
v=0.
o=- 1295424325 1295424325 IN IP4 77.72.17.23.
s=-.
c=IN IP4 77.72.17.23.
t=0 0.
m=audio 18360 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.




--
?????@Mail.Ru ? ????? ?????????. 
?????? ????? ? ???????? ?? m.mail.ru

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From steveayre at gmail.com  Wed Jan 19 14:05:18 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 19 Jan 2011 11:05:18 +0000
Subject: [Freeswitch-users] my FS doesn't answer on OK from the provider
In-Reply-To: <E1PfUXg-0000tt-00.yurazilot1-list-ru@f117.mail.ru>
References: <E1PfUXg-0000tt-00.yurazilot1-list-ru@f117.mail.ru>
Message-ID: <AANLkTi=XQYd6tgjvq5-1SjacE5R6pfqhU9fJPQEX83Ww@mail.gmail.com>

What does debugging log and siptrace show?
> sofia global siptrace on

You can also look at the Sofia SIP stack debugging information to see if
there's an error occuring in the stack that's ignoring the packet.
> sofia loglevel all 9

-Steve



2011/1/19 ZILOT <yurazilot1 at list.ru>

> I have one problem - my FS doesn't answer on OK from the provider.
> Probably the problem consists in an incorrectness incoming OK.
> Please help.
>
>
> U 2011/01/19 11:04:05.243062 194.190.211.190:5060 -> 77.72.17.23:5060
> INVITE sip:74742515541 at 77.72.17.23 <sip%3A74742515541 at 77.72.17.23>SIP/2.0.
> Via: SIP/2.0/UDP 194.190.211.190;rport;branch=z9hG4bK6rQjK6B04ZeNN.
> Max-Forwards: 69.
> From: "DEN78141_120" <sip:TCAPI_User at 77.72.17.23<sip%3ATCAPI_User at 77.72.17.23>
> ;transport=udp>;tag=vZcKXKa0Q7B1e.
> To: <sip:74742515541 at 77.72.17.23 <sip%3A74742515541 at 77.72.17.23>>.
> Call-ID: 86959393-9e45-122e-1782-33002db2a994.
> CSeq: 7376058 INVITE.
> Contact: <sip:gw+trunk_8 at 194.190.211.190:5060;transport=udp;gw=trunk_8>.
> User-Agent: Configured by 2600hz.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE.
> Supported: precondition, path, replaces.
> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 298.
> X-FS-Support: update_display.
> Remote-Party-ID: "DEN78141_120" <sip:DEN78141_120 at 77.72.17.23<sip%3ADEN78141_120 at 77.72.17.23>
> >;party=calling;screen=yes;privacy=off.
> .
> v=0.
> o=FreeSWITCH 1295404175 1295404176 IN IP4 194.190.211.190.
> s=FreeSWITCH.
> c=IN IP4 194.190.211.190.
> t=0 0.
> m=audio 20070 RTP/AVP 0 9 8 101 13.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:9 G722/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=rtpmap:13 CN/8000.
> a=ptime:30.
>
> U 2011/01/19 11:04:05.288882 77.72.17.23:5060 -> 194.190.211.190:5060
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP 194.190.211.190:5060;rport;branch=z9hG4bK6rQjK6B04ZeNN.
> From: "DEN78141_120" <sip:TCAPI_User at 77.72.17.23<sip%3ATCAPI_User at 77.72.17.23>
> ;transport=udp>;tag=vZcKXKa0Q7B1e.
> To: <sip:74742515541 at 77.72.17.23 <sip%3A74742515541 at 77.72.17.23>
> >;tag=0eff5d003cff3610ff00001372630472.
> Call-ID: 86959393-9e45-122e-1782-33002db2a994.
> CSeq: 7376058 INVITE.
> Contact: <sip:74742515541 at 77.72.17.23 <sip%3A74742515541 at 77.72.17.23>
> ;user=phone>.
> Server: MERA MSIP v.1.0.2.
> Content-Length: 0.
> .
>
> U 2011/01/19 11:04:06.609863 77.72.17.23:5060 -> 194.190.211.190:5060
> SIP/2.0 183 Session Progress.
> Via: SIP/2.0/UDP 194.190.211.190:5060;rport;branch=z9hG4bK6rQjK6B04ZeNN.
> From: "DEN78141_120" <sip:TCAPI_User at 77.72.17.23<sip%3ATCAPI_User at 77.72.17.23>
> ;transport=udp>;tag=vZcKXKa0Q7B1e.
> To: <sip:74742515541 at 77.72.17.23 <sip%3A74742515541 at 77.72.17.23>
> >;tag=0eff5d003cff3610ff00001372630472.
> Call-ID: 86959393-9e45-122e-1782-33002db2a994.
> CSeq: 7376058 INVITE.
> Contact: <sip:74742515541 at 77.72.17.23 <sip%3A74742515541 at 77.72.17.23>
> ;user=phone>.
> Server: MERA MSIP v.1.0.2.
> Content-Type: application/sdp.
> Content-Length:   212.
> .
> v=0.
> o=- 1295424317 1295424317 IN IP4 77.72.17.23.
> s=-.
> c=IN IP4 77.72.17.23.
> t=0 0.
> m=audio 18360 RTP/AVP 0 8 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
>
> U 2011/01/19 11:04:14.238660 77.72.17.23:5060 -> 194.190.211.190:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 194.190.211.190:5060;rport;branch=z9hG4bK6rQjK6B04ZeNN.
> From: "DEN78141_120" <sip:TCAPI_User at 77.72.17.23<sip%3ATCAPI_User at 77.72.17.23>
> ;transport=udp>;tag=vZcKXKa0Q7B1e.
> To: <sip:74742515541 at 77.72.17.23 <sip%3A74742515541 at 77.72.17.23>
> >;tag=0eff5d003cff3610ff00001372630472.
> Call-ID: 86959393-9e45-122e-1782-33002db2a994.
> CSeq: 7376058 INVITE.
> Contact: <sip:74742515541 at 77.72.17.23 <sip%3A74742515541 at 77.72.17.23>
> ;user=phone>.
> Server: MERA MSIP v.1.0.2.
> Content-Type: application/sdp.
> Content-Length:   212.
> .
> v=0.
> o=- 1295424325 1295424325 IN IP4 77.72.17.23.
> s=-.
> c=IN IP4 77.72.17.23.
> t=0 0.
> m=audio 18360 RTP/AVP 0 8 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
>
> U 2011/01/19 11:04:14.246657 77.72.17.23:5060 -> 194.190.211.190:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 194.190.211.190:5060;rport;branch=z9hG4bK6rQjK6B04ZeNN.
> From: "DEN78141_120" <sip:TCAPI_User at 77.72.17.23<sip%3ATCAPI_User at 77.72.17.23>
> ;transport=udp>;tag=vZcKXKa0Q7B1e.
> To: <sip:74742515541 at 77.72.17.23 <sip%3A74742515541 at 77.72.17.23>
> >;tag=0eff5d003cff3610ff00001372630472.
> Call-ID: 86959393-9e45-122e-1782-33002db2a994.
> CSeq: 7376058 INVITE.
> Contact: <sip:74742515541 at 77.72.17.23 <sip%3A74742515541 at 77.72.17.23>
> ;user=phone>.
> Server: MERA MSIP v.1.0.2.
> Content-Type: application/sdp.
> Content-Length:   212.
> .
> v=0.
> o=- 1295424325 1295424325 IN IP4 77.72.17.23.
> s=-.
> c=IN IP4 77.72.17.23.
> t=0 0.
> m=audio 18360 RTP/AVP 0 8 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
>
> U 2011/01/19 11:04:14.747863 77.72.17.23:5060 -> 194.190.211.190:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 194.190.211.190:5060;rport;branch=z9hG4bK6rQjK6B04ZeNN.
> From: "DEN78141_120" <sip:TCAPI_User at 77.72.17.23<sip%3ATCAPI_User at 77.72.17.23>
> ;transport=udp>;tag=vZcKXKa0Q7B1e.
> To: <sip:74742515541 at 77.72.17.23 <sip%3A74742515541 at 77.72.17.23>
> >;tag=0eff5d003cff3610ff00001372630472.
> Call-ID: 86959393-9e45-122e-1782-33002db2a994.
> CSeq: 7376058 INVITE.
> Contact: <sip:74742515541 at 77.72.17.23 <sip%3A74742515541 at 77.72.17.23>
> ;user=phone>.
> Server: MERA MSIP v.1.0.2.
> Content-Type: application/sdp.
> Content-Length:   212.
> .
> v=0.
> o=- 1295424325 1295424325 IN IP4 77.72.17.23.
> s=-.
> c=IN IP4 77.72.17.23.
> t=0 0.
> m=audio 18360 RTP/AVP 0 8 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
>
>
>
>
> --
> ?????@Mail.Ru ? ????? ?????????.
> ?????? ????? ? ???????? ?? m.mail.ru
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ross at ossiantelecom.co.uk  Wed Jan 19 14:11:49 2011
From: ross at ossiantelecom.co.uk (Ross McKillop)
Date: Wed, 19 Jan 2011 11:11:49 +0000
Subject: [Freeswitch-users] Using "Reason" from hangup
In-Reply-To: <026C72DF-B796-4BCC-B502-269221A80430@freeswitch.org>
References: <D522FF78-D722-43DD-9D4A-5382472457E7@ossiantelecom.co.uk>
	<AANLkTim8ZoTJBXUJu59cWLCQqr99kWaX9V+8utMpKQ0g@mail.gmail.com>
	<2FD902BB-6288-4FAF-9AD7-9AB92CC57865@ossiantelecom.co.uk>
	<026C72DF-B796-4BCC-B502-269221A80430@freeswitch.org>
Message-ID: <6E0459C4-FF6B-441A-B9A6-B3C5E0F446AC@ossiantelecom.co.uk>

I've been away for a while and recently had a chance to come back to look at this issue (which a colleague has also attempted to resolve without success)

On 7 Jan 2011, at 15:12, Brian West wrote:

> don't call hangup at all...  Just let it pass it back.

I've tried this also with and without hangup_after_bridge=true

Still returned 16 "NORMAL_CLEARING" even when the B party ended with cause 31.

Tried this in both 1.0.6 and git-head from December 8th.

Could this be related to the (older) thread "FreeSWITCH overrides/dose no accept hangup cause"?

> On Jan 7, 2011, at 5:04 AM, Ross McKillop wrote:
> 
>> (have also tried <action application="hangup" data="${originate_disposition}" /> as well
>> as not setting the sip_ignore_remote_cause variable)


From thomas at chaschperli.ch  Wed Jan 19 14:42:17 2011
From: thomas at chaschperli.ch (Thomas Mueller)
Date: Wed, 19 Jan 2011 12:42:17 +0100
Subject: [Freeswitch-users] Next version of Gemeinschaft based on FreeSWITCH
Message-ID: <4D36CE19.2070503@chaschperli.ch>

just read at GOLEM*  that next verion of "Gemeinschaft"** is backed by 
FreeSWITCH and not Asterisk anymore.

Article states that the relase is expected Q3 2011.

IMHO a nice frontend .

- Thomas

* http://www.golem.de/1101/80818.html
** http://amooma.de/


From tlstutsman at gmail.com  Wed Jan 19 14:45:04 2011
From: tlstutsman at gmail.com (Travis Stutsman)
Date: Wed, 19 Jan 2011 06:45:04 -0500
Subject: [Freeswitch-users] Yet another DTMF question
In-Reply-To: <62C5671A-8657-4F99-8184-272EEBEC8E77@gmail.com>
References: <AANLkTik030ecFw2bvKe74pOs+NzfqHknX8keDO3mQ=cH@mail.gmail.com>
	<AANLkTikqQodjsMT_VB1xeQdtY-2DEgL68ndEnHTBTvkb@mail.gmail.com>
	<AANLkTiksK3T24TUDO-7aq_tbkFCkj_4BRNpLRt4=GtBj@mail.gmail.com>
	<62C5671A-8657-4F99-8184-272EEBEC8E77@gmail.com>
Message-ID: <AANLkTimFMwJ1EkP_+1uCyiKUCKocgh-mSYVEd39navS7@mail.gmail.com>

First I want to thank you for taking the time, Steve.  I really appreciate
it.

Secondly, here is the SDP from the client end.  It appears to be 101 as
expected.

v=0
o=- 12939910547141841 1 IN IP4 x.x.x.x
s=CounterPath X-Lite 4.0
c=IN IP4 x.x.x.x
t=0 0
a=ice-ufrag:8ef9c3
a=ice-pwd:768def2c0edb34ec3cd4c49ed6b1279b
m=audio 55982 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 x.x.x.x 55982 typ host
a=candidate:1 2 UDP 659134 x.x.x.x 55983 typ host




On Wed, Jan 19, 2011 at 3:30 AM, Steven Ayre <steveayre at gmail.com> wrote:

> What about the remote sdp? Just checking fs is expecting 101.
>
> Steve on iPhone
>
> On 18 Jan 2011, at 19:17, Travis Stutsman <tlstutsman at gmail.com> wrote:
>
> Call has been answered - I can hear the IVR.  Payload type is 101 - packets
> on both ends verify this.  SDP example follows:
>
>
> 2011-01-18 19:10:13.579728 [DEBUG] mod_sofia.c:681 Local SDP
> sofia/internal/1000 at x.x.x.x:
> v=0
> o=FreeSWITCH 1295351291 1295351292 IN IP4 x.x.x.x
> s=FreeSWITCH
> c=IN IP4 x.x.x.x
> t=0 0
> m=audio 26522 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
>
>
>
>
>
> On Tue, Jan 18, 2011 at 2:02 PM, Steven Ayre < <steveayre at gmail.com>
> steveayre at gmail.com> wrote:
>
>> Has the call been answered? (DTMF is ignored before the call is answered)
>>
>> What payload type are the RFC2833 packets using?
>> What does the SDP look like?
>> Does the payload type in the packets match the telephone-event in the SDP?
>>
>> -Steve
>>
>>
>> On 18 January 2011 16:46, Travis Stutsman < <tlstutsman at gmail.com>
>> tlstutsman at gmail.com> wrote:
>>
>>> I've looked through a lot of threads regarding DTMF troubles, but none of
>>> them seem to relate to me.  I've just pulled a fresh copy of the freeswitch
>>> source and compiled it and I cannot get my digits recognized - neither
>>> in-band nor out-of-band.  I haven't messed with INFO yet as my phones don't
>>> support it.  I've done packet captures on my local interface and on the
>>> server's interface and I can see the RTP rfc2833 digits showing up, but
>>> freeswitch completely ignores them.  I feel like I must be overlooking
>>> something horribly obvious.  How can this be?  Any hints guys?  Thanks in
>>> advance!
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From avi at avimarcus.net  Wed Jan 19 14:57:13 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 19 Jan 2011 13:57:13 +0200
Subject: [Freeswitch-users] Next version of Gemeinschaft based on
	FreeSWITCH
In-Reply-To: <4D36CE19.2070503@chaschperli.ch>
References: <4D36CE19.2070503@chaschperli.ch>
Message-ID: <AANLkTi=GdpLZvBBkY4EgxdL-h3axDRJDzjG676mH0mEt@mail.gmail.com>

It's free and the interface is mostly in english? Seems that will be the 4th
major gui for FS. Cool.
-Avi

On Wed, Jan 19, 2011 at 1:42 PM, Thomas Mueller <thomas at chaschperli.ch>wrote:

> just read at GOLEM*  that next verion of "Gemeinschaft"** is backed by
> FreeSWITCH and not Asterisk anymore.
>
> Article states that the relase is expected Q3 2011.
>
> IMHO a nice frontend .
>
> - Thomas
>
> * http://www.golem.de/1101/80818.html
> ** http://amooma.de/
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From hesser4900 at gmail.com  Wed Jan 19 15:21:43 2011
From: hesser4900 at gmail.com (Holger Esser)
Date: Wed, 19 Jan 2011 06:21:43 -0600
Subject: [Freeswitch-users] TDM line maximum
In-Reply-To: <AANLkTikzjwYv5gw=m70NhG_DOfQuCB9dv+V=RP_EcgNy@mail.gmail.com>
References: <AANLkTim4Mhm+KTAs-5vbNYLtijkziZfbzcrWzmLPj5Bw@mail.gmail.com>
	<AANLkTi=sCw9KmY=_fyQjv9jjaMSLfnB6r_onFFh_ZoXP@mail.gmail.com>
	<AANLkTinPyGGSrXGD5zDQgq7kQEyeoj_Lpq6gc+aytyc5@mail.gmail.com>
	<AANLkTikCruXsaL1Jo6rarbCHmbPjLDfT2Hoa4UGaVM5p@mail.gmail.com>
	<AANLkTintVezYNGzpsqe4cfL4Rsn1ituw=hOeYqwfjj=Q@mail.gmail.com>
	<AANLkTikzjwYv5gw=m70NhG_DOfQuCB9dv+V=RP_EcgNy@mail.gmail.com>
Message-ID: <AANLkTiko8WA9nZTv_6Mn4VeR0D6iNkPdJv=4anbbQSr7@mail.gmail.com>

Stephen,

again, many thanks for the output. It will help me greatly.

Holger

On Wed, Jan 19, 2011 at 3:26 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> [root at localhost bin]# dstat
> ----total-cpu-usage---- -dsk/total- -net/total- ---paging-- ---system--
> usr sys idl wai hiq siq| read  writ| recv  send|  in   out | int   csw
>  11   6  80   0   1   2|1940B  314k|   0     0 |   0   0.1 |  12k   80k
>  26  10  59   0   1   4|   0  2680k|3288k 3472k|   0     0 |  26k  183k
>  26  11  58   0   1   4|   0   400k|3292k 3463k|   0     0 |  25k  190k
>  24  11  60   0   1   4|   0     0 |3287k 3443k|   0     0 |  26k  192k
>  25  11  59   0   1   4|   0     0 |3328k 3482k|   0     0 |  26k  192k
>  24  11  60   0   1   4|   0     0 |3282k 3441k|   0     0 |  27k  187k
>  26  12  57   0   1   4|   0     0 |3329k 3455k|   0     0 |  27k  180k
>  26  12  57   0   1   4|   0   152k|3320k 3506k|   0     0 |  27k  178k
>  25  12  58   0   1   3|   0     0 |3310k 3478k|   0     0 |  26k  181k
>
>
> [root at localhost bin]# ./fs_cli -x "status"
> UP 0 years, 0 days, 8 hours, 40 minutes, 32 seconds, 50 milliseconds, 274
> microseconds
> 175400 session(s) since startup
> 1454 session(s) 25/100
> 2000 session(s) max
> min idle cpu 0.00/57.00
>
>
>
> [root at localhost bin]# wanrouter hwprobe
> -------------------------------
> | Wanpipe Hardware Probe Info |
> -------------------------------
> 1 . AFT-A108-SH : SLOT=4 : BUS=17 : IRQ=169 : CPU=A : PORT=1 : HWEC=0 :
> V=41
> 2 . AFT-A108-SH : SLOT=4 : BUS=17 : IRQ=169 : CPU=A : PORT=2 : HWEC=0 :
> V=41
> 3 . AFT-A108-SH : SLOT=4 : BUS=17 : IRQ=169 : CPU=A : PORT=3 : HWEC=0 :
> V=41
> 4 . AFT-A108-SH : SLOT=4 : BUS=17 : IRQ=169 : CPU=A : PORT=4 : HWEC=0 :
> V=41
> 5 . AFT-A108-SH : SLOT=4 : BUS=17 : IRQ=169 : CPU=A : PORT=5 : HWEC=0 :
> V=41
> 6 . AFT-A108-SH : SLOT=4 : BUS=17 : IRQ=169 : CPU=A : PORT=6 : HWEC=0 :
> V=41
> 7 . AFT-A108-SH : SLOT=4 : BUS=17 : IRQ=169 : CPU=A : PORT=7 : HWEC=0 :
> V=41
> 8 . AFT-A108-SH : SLOT=4 : BUS=17 : IRQ=169 : CPU=A : PORT=8 : HWEC=0 :
> V=41
> 9 . AFT-A108-SH : SLOT=4 : BUS=21 : IRQ=177 : CPU=A : PORT=1 : HWEC=0 :
> V=41
> 10. AFT-A108-SH : SLOT=4 : BUS=21 : IRQ=177 : CPU=A : PORT=2 : HWEC=0 :
> V=41
> 11. AFT-A108-SH : SLOT=4 : BUS=21 : IRQ=177 : CPU=A : PORT=3 : HWEC=0 :
> V=41
> 12. AFT-A108-SH : SLOT=4 : BUS=21 : IRQ=177 : CPU=A : PORT=4 : HWEC=0 :
> V=41
> 13. AFT-A108-SH : SLOT=4 : BUS=21 : IRQ=177 : CPU=A : PORT=5 : HWEC=0 :
> V=41
> 14. AFT-A108-SH : SLOT=4 : BUS=21 : IRQ=177 : CPU=A : PORT=6 : HWEC=0 :
> V=41
> 15. AFT-A108-SH : SLOT=4 : BUS=21 : IRQ=177 : CPU=A : PORT=7 : HWEC=0 :
> V=41
> 16. AFT-A108-SH : SLOT=4 : BUS=21 : IRQ=177 : CPU=A : PORT=8 : HWEC=0 :
> V=41
> 17. AFT-A108-SH : SLOT=4 : BUS=24 : IRQ=185 : CPU=A : PORT=1 : HWEC=0 :
> V=41
> 18. AFT-A108-SH : SLOT=4 : BUS=24 : IRQ=185 : CPU=A : PORT=2 : HWEC=0 :
> V=41
> 19. AFT-A108-SH : SLOT=4 : BUS=24 : IRQ=185 : CPU=A : PORT=3 : HWEC=0 :
> V=41
> 20. AFT-A108-SH : SLOT=4 : BUS=24 : IRQ=185 : CPU=A : PORT=4 : HWEC=0 :
> V=41
> 21. AFT-A108-SH : SLOT=4 : BUS=24 : IRQ=185 : CPU=A : PORT=5 : HWEC=0 :
> V=41
> 22. AFT-A108-SH : SLOT=4 : BUS=24 : IRQ=185 : CPU=A : PORT=6 : HWEC=0 :
> V=41
> 23. AFT-A108-SH : SLOT=4 : BUS=24 : IRQ=185 : CPU=A : PORT=7 : HWEC=0 :
> V=41
> 24. AFT-A108-SH : SLOT=4 : BUS=24 : IRQ=185 : CPU=A : PORT=8 : HWEC=0 :
> V=41
> 25. AFT-A108-SH : SLOT=4 : BUS=14 : IRQ=193 : CPU=A : PORT=1 : HWEC=0 :
> V=41
> 26. AFT-A108-SH : SLOT=4 : BUS=14 : IRQ=193 : CPU=A : PORT=2 : HWEC=0 :
> V=41
> 27. AFT-A108-SH : SLOT=4 : BUS=14 : IRQ=193 : CPU=A : PORT=3 : HWEC=0 :
> V=41
> 28. AFT-A108-SH : SLOT=4 : BUS=14 : IRQ=193 : CPU=A : PORT=4 : HWEC=0 :
> V=41
> 29. AFT-A108-SH : SLOT=4 : BUS=14 : IRQ=193 : CPU=A : PORT=5 : HWEC=0 :
> V=41
> 30. AFT-A108-SH : SLOT=4 : BUS=14 : IRQ=193 : CPU=A : PORT=6 : HWEC=0 :
> V=41
> 31. AFT-A108-SH : SLOT=4 : BUS=14 : IRQ=193 : CPU=A : PORT=7 : HWEC=0 :
> V=41
> 32. AFT-A108-SH : SLOT=4 : BUS=14 : IRQ=193 : CPU=A : PORT=8 : HWEC=0 :
> V=41
> 33. AFT-A108-SH : SLOT=4 : BUS=11 : IRQ=201 : CPU=A : PORT=1 : HWEC=0 :
> V=41
> 34. AFT-A108-SH : SLOT=4 : BUS=11 : IRQ=201 : CPU=A : PORT=2 : HWEC=0 :
> V=41
> 35. AFT-A108-SH : SLOT=4 : BUS=11 : IRQ=201 : CPU=A : PORT=3 : HWEC=0 :
> V=41
> 36. AFT-A108-SH : SLOT=4 : BUS=11 : IRQ=201 : CPU=A : PORT=4 : HWEC=0 :
> V=41
> 37. AFT-A108-SH : SLOT=4 : BUS=11 : IRQ=201 : CPU=A : PORT=5 : HWEC=0 :
> V=41
> 38. AFT-A108-SH : SLOT=4 : BUS=11 : IRQ=201 : CPU=A : PORT=6 : HWEC=0 :
> V=41
> 39. AFT-A108-SH : SLOT=4 : BUS=11 : IRQ=201 : CPU=A : PORT=7 : HWEC=0 :
> V=41
> 40. AFT-A108-SH : SLOT=4 : BUS=11 : IRQ=201 : CPU=A : PORT=8 : HWEC=0 :
> V=41
> 41. AFT-A108-SH : SLOT=4 : BUS=8 : IRQ=209 : CPU=A : PORT=1 : HWEC=0 : V=41
> 42. AFT-A108-SH : SLOT=4 : BUS=8 : IRQ=209 : CPU=A : PORT=2 : HWEC=0 : V=41
> 43. AFT-A108-SH : SLOT=4 : BUS=8 : IRQ=209 : CPU=A : PORT=3 : HWEC=0 : V=41
> 44. AFT-A108-SH : SLOT=4 : BUS=8 : IRQ=209 : CPU=A : PORT=4 : HWEC=0 : V=41
> 45. AFT-A108-SH : SLOT=4 : BUS=8 : IRQ=209 : CPU=A : PORT=5 : HWEC=0 : V=41
> 46. AFT-A108-SH : SLOT=4 : BUS=8 : IRQ=209 : CPU=A : PORT=6 : HWEC=0 : V=41
> 47. AFT-A108-SH : SLOT=4 : BUS=8 : IRQ=209 : CPU=A : PORT=7 : HWEC=0 : V=41
> 48. AFT-A108-SH : SLOT=4 : BUS=8 : IRQ=209 : CPU=A : PORT=8 : HWEC=0 : V=41
>
> Card Cnt: A108=6
>
>
>
> On Wed, Jan 19, 2011 at 1:05 AM, Wasim Baig <wasim at convergence.pk> wrote:
>
>> Stephen:
>>
>> Can you post the CPU, memory and IO footprints at these call volumes?
>>
>> a dstat line or two would do ...
>>
>> thanks
>>
>> -wasim
>>
>>
>> On Wed, Jan 19, 2011 at 02:32, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>
>>> HP DL380 G6 with two xeon 5520 8Gb ram.
>>>
>>> Stephen
>>>
>>>
>>> On Tue, Jan 18, 2011 at 9:47 PM, Holger Esser <hesser4900 at gmail.com>wrote:
>>>
>>>> May I ask on what hardware, I am planing on using an HP g7.
>>>>
>>>> Many thanks,
>>>> Holger
>>>>
>>>>
>>>> On Tue, Jan 18, 2011 at 2:42 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>>>
>>>>> Yes, we have a single server installation with 5 Sangoma A108 board
>>>>> that regularly handles more than 1500 contemporary sessions.
>>>>>
>>>>> Regards,
>>>>> Stephen
>>>>>
>>>>> On Tue, Jan 18, 2011 at 9:26 PM, Holger Esser <hesser4900 at gmail.com>wrote:
>>>>>
>>>>>> All,
>>>>>>
>>>>>> Does anybody have experience with implementing 4 Sangoma  octal cards
>>>>>> and bridge these channels to SIP sessions on a single server for a total of
>>>>>> 1536 sessions (TDM+SIP)
>>>>>>
>>>>>> Regards,
>>>>>> Holger
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
>> peace be upon you ...
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From steveayre at gmail.com  Wed Jan 19 15:27:35 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 19 Jan 2011 12:27:35 +0000
Subject: [Freeswitch-users] Yet another DTMF question
In-Reply-To: <AANLkTimFMwJ1EkP_+1uCyiKUCKocgh-mSYVEd39navS7@mail.gmail.com>
References: <AANLkTik030ecFw2bvKe74pOs+NzfqHknX8keDO3mQ=cH@mail.gmail.com>
	<AANLkTikqQodjsMT_VB1xeQdtY-2DEgL68ndEnHTBTvkb@mail.gmail.com>
	<AANLkTiksK3T24TUDO-7aq_tbkFCkj_4BRNpLRt4=GtBj@mail.gmail.com>
	<62C5671A-8657-4F99-8184-272EEBEC8E77@gmail.com>
	<AANLkTimFMwJ1EkP_+1uCyiKUCKocgh-mSYVEd39navS7@mail.gmail.com>
Message-ID: <AANLkTikaerx0b+A7EhFLXxuBKBBY+RBog8Lpgyd1ZZmR@mail.gmail.com>

Looks ok then...

At debug level in the logs DTMF digits showing up?

It'll look something similar to:
2011-01-19 12:02:54.855830 [DEBUG] switch_rtp.c:2925 RTP RECV DTMF 1:400

-Steve


On 19 January 2011 11:45, Travis Stutsman <tlstutsman at gmail.com> wrote:

> First I want to thank you for taking the time, Steve.  I really appreciate
> it.
>
> Secondly, here is the SDP from the client end.  It appears to be 101 as
> expected.
>
> v=0
> o=- 12939910547141841 1 IN IP4 x.x.x.x
> s=CounterPath X-Lite 4.0
>
> c=IN IP4 x.x.x.x
> t=0 0
> a=ice-ufrag:8ef9c3
> a=ice-pwd:768def2c0edb34ec3cd4c49ed6b1279b
> m=audio 55982 RTP/AVP 8 101
>
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
> a=candidate:1 1 UDP 659136 x.x.x.x 55982 typ host
> a=candidate:1 2 UDP 659134 x.x.x.x 55983 typ host
>
>
>
>
>
> On Wed, Jan 19, 2011 at 3:30 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> What about the remote sdp? Just checking fs is expecting 101.
>>
>> Steve on iPhone
>>
>> On 18 Jan 2011, at 19:17, Travis Stutsman <tlstutsman at gmail.com> wrote:
>>
>> Call has been answered - I can hear the IVR.  Payload type is 101 -
>> packets on both ends verify this.  SDP example follows:
>>
>>
>> 2011-01-18 19:10:13.579728 [DEBUG] mod_sofia.c:681 Local SDP
>> sofia/internal/1000 at x.x.x.x:
>> v=0
>> o=FreeSWITCH 1295351291 1295351292 IN IP4 x.x.x.x
>> s=FreeSWITCH
>> c=IN IP4 x.x.x.x
>> t=0 0
>> m=audio 26522 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>>
>>
>>
>>
>>
>>
>> On Tue, Jan 18, 2011 at 2:02 PM, Steven Ayre < <steveayre at gmail.com>
>> steveayre at gmail.com> wrote:
>>
>>> Has the call been answered? (DTMF is ignored before the call is answered)
>>>
>>> What payload type are the RFC2833 packets using?
>>> What does the SDP look like?
>>> Does the payload type in the packets match the telephone-event in the
>>> SDP?
>>>
>>> -Steve
>>>
>>>
>>> On 18 January 2011 16:46, Travis Stutsman < <tlstutsman at gmail.com>
>>> tlstutsman at gmail.com> wrote:
>>>
>>>> I've looked through a lot of threads regarding DTMF troubles, but none
>>>> of them seem to relate to me.  I've just pulled a fresh copy of the
>>>> freeswitch source and compiled it and I cannot get my digits recognized -
>>>> neither in-band nor out-of-band.  I haven't messed with INFO yet as my
>>>> phones don't support it.  I've done packet captures on my local interface
>>>> and on the server's interface and I can see the RTP rfc2833 digits showing
>>>> up, but freeswitch completely ignores them.  I feel like I must be
>>>> overlooking something horribly obvious.  How can this be?  Any hints guys?
>>>> Thanks in advance!
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tlstutsman at gmail.com  Wed Jan 19 15:34:02 2011
From: tlstutsman at gmail.com (Travis Stutsman)
Date: Wed, 19 Jan 2011 07:34:02 -0500
Subject: [Freeswitch-users] Yet another DTMF question
In-Reply-To: <AANLkTikaerx0b+A7EhFLXxuBKBBY+RBog8Lpgyd1ZZmR@mail.gmail.com>
References: <AANLkTik030ecFw2bvKe74pOs+NzfqHknX8keDO3mQ=cH@mail.gmail.com>
	<AANLkTikqQodjsMT_VB1xeQdtY-2DEgL68ndEnHTBTvkb@mail.gmail.com>
	<AANLkTiksK3T24TUDO-7aq_tbkFCkj_4BRNpLRt4=GtBj@mail.gmail.com>
	<62C5671A-8657-4F99-8184-272EEBEC8E77@gmail.com>
	<AANLkTimFMwJ1EkP_+1uCyiKUCKocgh-mSYVEd39navS7@mail.gmail.com>
	<AANLkTikaerx0b+A7EhFLXxuBKBBY+RBog8Lpgyd1ZZmR@mail.gmail.com>
Message-ID: <AANLkTinL1NxOjMJjShOwXEo5CJERLbY4N7Z7re=QRp8z@mail.gmail.com>

Yeah, I've been scouring the debug output and it's definitely not showing
up.



On Wed, Jan 19, 2011 at 7:27 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Looks ok then...
>
> At debug level in the logs DTMF digits showing up?
>
> It'll look something similar to:
> 2011-01-19 12:02:54.855830 [DEBUG] switch_rtp.c:2925 RTP RECV DTMF 1:400
>
> -Steve
>
>
>
> On 19 January 2011 11:45, Travis Stutsman <tlstutsman at gmail.com> wrote:
>
>> First I want to thank you for taking the time, Steve.  I really appreciate
>> it.
>>
>> Secondly, here is the SDP from the client end.  It appears to be 101 as
>> expected.
>>
>> v=0
>> o=- 12939910547141841 1 IN IP4 x.x.x.x
>> s=CounterPath X-Lite 4.0
>>
>> c=IN IP4 x.x.x.x
>> t=0 0
>> a=ice-ufrag:8ef9c3
>> a=ice-pwd:768def2c0edb34ec3cd4c49ed6b1279b
>> m=audio 55982 RTP/AVP 8 101
>>
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=sendrecv
>> a=candidate:1 1 UDP 659136 x.x.x.x 55982 typ host
>> a=candidate:1 2 UDP 659134 x.x.x.x 55983 typ host
>>
>>
>>
>>
>>
>> On Wed, Jan 19, 2011 at 3:30 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> What about the remote sdp? Just checking fs is expecting 101.
>>>
>>> Steve on iPhone
>>>
>>> On 18 Jan 2011, at 19:17, Travis Stutsman <tlstutsman at gmail.com> wrote:
>>>
>>> Call has been answered - I can hear the IVR.  Payload type is 101 -
>>> packets on both ends verify this.  SDP example follows:
>>>
>>>
>>> 2011-01-18 19:10:13.579728 [DEBUG] mod_sofia.c:681 Local SDP
>>> sofia/internal/1000 at x.x.x.x:
>>> v=0
>>> o=FreeSWITCH 1295351291 1295351292 IN IP4 x.x.x.x
>>> s=FreeSWITCH
>>> c=IN IP4 x.x.x.x
>>> t=0 0
>>> m=audio 26522 RTP/AVP 8 101
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=silenceSupp:off - - - -
>>> a=ptime:20
>>> a=sendrecv
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Tue, Jan 18, 2011 at 2:02 PM, Steven Ayre < <steveayre at gmail.com>
>>> steveayre at gmail.com> wrote:
>>>
>>>> Has the call been answered? (DTMF is ignored before the call is
>>>> answered)
>>>>
>>>> What payload type are the RFC2833 packets using?
>>>> What does the SDP look like?
>>>> Does the payload type in the packets match the telephone-event in the
>>>> SDP?
>>>>
>>>> -Steve
>>>>
>>>>
>>>> On 18 January 2011 16:46, Travis Stutsman < <tlstutsman at gmail.com>
>>>> tlstutsman at gmail.com> wrote:
>>>>
>>>>> I've looked through a lot of threads regarding DTMF troubles, but none
>>>>> of them seem to relate to me.  I've just pulled a fresh copy of the
>>>>> freeswitch source and compiled it and I cannot get my digits recognized -
>>>>> neither in-band nor out-of-band.  I haven't messed with INFO yet as my
>>>>> phones don't support it.  I've done packet captures on my local interface
>>>>> and on the server's interface and I can see the RTP rfc2833 digits showing
>>>>> up, but freeswitch completely ignores them.  I feel like I must be
>>>>> overlooking something horribly obvious.  How can this be?  Any hints guys?
>>>>> Thanks in advance!
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From thomas at chaschperli.ch  Wed Jan 19 15:37:37 2011
From: thomas at chaschperli.ch (Thomas Mueller)
Date: Wed, 19 Jan 2011 13:37:37 +0100
Subject: [Freeswitch-users] Next version of Gemeinschaft based
	on	FreeSWITCH
In-Reply-To: <AANLkTi=GdpLZvBBkY4EgxdL-h3axDRJDzjG676mH0mEt@mail.gmail.com>
References: <4D36CE19.2070503@chaschperli.ch>
	<AANLkTi=GdpLZvBBkY4EgxdL-h3axDRJDzjG676mH0mEt@mail.gmail.com>
Message-ID: <4D36DB11.2090500@chaschperli.ch>

On 19.01.2011 12:57, Avi Marcus wrote:
> It's free and the interface is mostly in english? Seems that will be
> the 4th major gui for FS. Cool.

it's GPLv2. IMHO Initially developped for the german market, but seems 
all is localized to english. I'm more interested in the german part. :)

code of the 3.0 (aserisk based-) branch sits here: 
https://github.com/amooma/GemeinschaftPBX

- Thomas


From jgalaz at yx.cl  Wed Jan 19 15:19:15 2011
From: jgalaz at yx.cl (Javier Galaz Jeria)
Date: Wed, 19 Jan 2011 09:19:15 -0300
Subject: [Freeswitch-users] Different api calls from Dialplan
Message-ID: <20110119121915.GL17734@jgalaz-desktop>

Hello, freeswitch-users, I've a question regarding dialplan.

Is there a difference between this two function calls (made from the dialplan)
for FreeSWITCH?

a) <action application="set" data="api_result=${ilx_read $1}" inline="true"/>

b) <action application="set" data="api_result=${ilx_read ($1)}" inline="true"/>
							 ^  ^
The thing is on one of my boxes both calls work, and on the other only b) works

Regards

j


From yurazilot1 at list.ru  Wed Jan 19 15:43:27 2011
From: yurazilot1 at list.ru (ZILOT)
Date: Wed, 19 Jan 2011 15:43:27 +0300
Subject: [Freeswitch-users] =?koi8-r?b?bXkgRlMgZG9lc24ndCBhbnN3ZXIgb24g?=
	=?koi8-r?b?T0sgZnJvbSB0aGUgcHJvdmlkZXI=?=
In-Reply-To: <AANLkTiXQYd6tgjvq5-1SjacE5R6pfqhU9fJPQEX83Ww@mail.gmail.com>
References: <E1PfUXg-0000tt-00.yurazilot1-list-ru@f117.mail.ru>
	<AANLkTiXQYd6tgjvq5-1SjacE5R6pfqhU9fJPQEX83Ww@mail.gmail.com>
Message-ID: <E1PfXNn-0006Jz-00.yurazilot1-list-ru@f44.mail.ru>


I have entered "sofia profile sipinterface_1 siptrace off" and have seen Trying from my provider, but i can't see "OK"
If I enter "sofia loglevel all 9" it I don't manage to disassemble with a stream of messages.
freeswitch> sofia loglevel <all|default|tport|iptsec|nea|nta|nth_client|nth_server|nua|soa|sresolv|stun> [0-9]
That from this it is better to look?


 


recv 416 bytes from udp/[77.72.17.23]:5060 at 12:25:57.236897:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 194.190.211.190:5060;rport;branch=z9hG4bKy0U8cX4FFvDQr
   From: "DEN78141_140" <sip:TCAPI_User at 77.72.17.23;transport=udp>;tag=52egjQQgr1Kmr
   To: <sip:79213777763 at 77.72.17.23>;tag=7aff7900ffff3610ff00001372630472
   Call-ID: 1ba1a990-9e6a-122e-1782-33002db2a994
   CSeq: 7383914 INVITE
   Contact: <sip:79213777763 at 77.72.17.23;user=phone>
   Server: MERA MSIP v.1.0.2
   Content-Length: 0





Wed, 19 Jan 2011 11:05:18 +0000 ?????? ?? Steven Ayre <steveayre at gmail.com>:

What does debugging log and siptrace show?
> sofia global siptrace on

You can also look at the Sofia SIP stack debugging information to see if there's an error occuring in the stack that's ignoring the packet.
> sofia loglevel all 9

-Steve



2011/1/19 ZILOT < yurazilot1 at list.ru (http://win.mail.ru/cgi-bin/sentmsg?compose&amp;To=yurazilot1 at list.ru) >
I have one problem - my FS doesn't answer on OK from the provider.
Probably the problem consists in an incorrectness incoming OK.
Please help.

  
U 2011/01/19 11:04:05.243062  194.190.211.190:5060 (http://194.190.211.190:5060/)  ->  77.72.17.23:5060 (http://77.72.17.23:5060/) 
INVITE  sip:74742515541 at 77.72.17.23 (http://win.mail.ru/cgi-bin/sentmsg?compose&amp;To=sip%3A74742515541 at 77.72.17.23)  SIP/2.0.
Via: SIP/2.0/UDP 194.190.211.190;rport;branch=z9hG4bK6rQjK6B04ZeNN.
Max-Forwards: 69.
From: "DEN78141_120" < sip:TCAPI_User at 77.72.17.23 (http://win.mail.ru/cgi-bin/sentmsg?compose&amp;To=sip%3ATCAPI_User at 77.72.17.23) ;transport=udp>;tag=vZcKXKa0Q7B1e.
To: < sip:74742515541 at 77.72.17.23 (http://win.mail.ru/cgi-bin/sentmsg?compose&amp;To=sip%3A74742515541 at 77.72.17.23) >.
Call-ID: 86959393-9e45-122e-1782-33002db2a994.
CSeq: 7376058 INVITE.
Contact: <sip:gw+trunk_8 at 194.190.211.190:5060;transport=udp;gw=trunk_8>.
User-Agent: Configured by 2600hz.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: precondition, path, replaces.
Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 298.
X-FS-Support: update_display.
Remote-Party-ID: "DEN78141_120" < sip:DEN78141_120 at 77.72.17.23 (http://win.mail.ru/cgi-bin/sentmsg?compose&amp;To=sip%3ADEN78141_120 at 77.72.17.23) >;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1295404175 1295404176 IN IP4 194.190.211.190.
s=FreeSWITCH.
c=IN IP4 194.190.211.190.
t=0 0.
m=audio 20070 RTP/AVP 0 9 8 101 13.
a=rtpmap:0 PCMU/8000.
a=rtpmap:9 G722/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:30.

U 2011/01/19 11:04:05.288882  77.72.17.23:5060 (http://77.72.17.23:5060/)  ->  194.190.211.190:5060 (http://194.190.211.190:5060/) 
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 194.190.211.190:5060;rport;branch=z9hG4bK6rQjK6B04ZeNN.
From: "DEN78141_120" < sip:TCAPI_User at 77.72.17.23 (http://win.mail.ru/cgi-bin/sentmsg?compose&amp;To=sip%3ATCAPI_User at 77.72.17.23) ;transport=udp>;tag=vZcKXKa0Q7B1e.
To: < sip:74742515541 at 77.72.17.23 (http://win.mail.ru/cgi-bin/sentmsg?compose&amp;To=sip%3A74742515541 at 77.72.17.23) >;tag=0eff5d003cff3610ff00001372630472.
Call-ID: 86959393-9e45-122e-1782-33002db2a994.
CSeq: 7376058 INVITE.
Contact: < sip:74742515541 at 77.72.17.23 (http://win.mail.ru/cgi-bin/sentmsg?compose&amp;To=sip%3A74742515541 at 77.72.17.23) ;user=phone>.
Server: MERA MSIP v.1.0.2.
Content-Length: 0.
.

U 2011/01/19 11:04:06.609863  77.72.17.23:5060 (http://77.72.17.23:5060/)  ->  194.190.211.190:5060 (http://194.190.211.190:5060/) 
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 194.190.211.190:5060;rport;branch=z9hG4bK6rQjK6B04ZeNN.
From: "DEN78141_120" < sip:TCAPI_User at 77.72.17.23 (http://win.mail.ru/cgi-bin/sentmsg?compose&amp;To=sip%3ATCAPI_User at 77.72.17.23) ;transport=udp>;tag=vZcKXKa0Q7B1e.
To: < sip:74742515541 at 77.72.17.23 (http://win.mail.ru/cgi-bin/sentmsg?compose&amp;To=sip%3A74742515541 at 77.72.17.23) >;tag=0eff5d003cff3610ff00001372630472.
Call-ID: 86959393-9e45-122e-1782-33002db2a994.
CSeq: 7376058 INVITE.
Contact: < sip:74742515541 at 77.72.17.23 (http://win.mail.ru/cgi-bin/sentmsg?compose&amp;To=sip%3A74742515541 at 77.72.17.23) ;user=phone>.
Server: MERA MSIP v.1.0.2.
Content-Type: application/sdp.
Content-Length:   212.
.
v=0.
o=- 1295424317 1295424317 IN IP4 77.72.17.23.
s=-.
c=IN IP4 77.72.17.23.
t=0 0.
m=audio 18360 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

U 2011/01/19 11:04:14.238660  77.72.17.23:5060 (http://77.72.17.23:5060/)  ->  194.190.211.190:5060 (http://194.190.211.190:5060/) 
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 194.190.211.190:5060;rport;branch=z9hG4bK6rQjK6B04ZeNN.
From: "DEN78141_120" < sip:TCAPI_User at 77.72.17.23 (http://win.mail.ru/cgi-bin/sentmsg?compose&amp;To=sip%3ATCAPI_User at 77.72.17.23) ;transport=udp>;tag=vZcKXKa0Q7B1e.
To: < sip:74742515541 at 77.72.17.23 (http://win.mail.ru/cgi-bin/sentmsg?compose&amp;To=sip%3A74742515541 at 77.72.17.23) >;tag=0eff5d003cff3610ff00001372630472.
Call-ID: 86959393-9e45-122e-1782-33002db2a994.
CSeq: 7376058 INVITE.
Contact: < sip:74742515541 at 77.72.17.23 (http://win.mail.ru/cgi-bin/sentmsg?compose&amp;To=sip%3A74742515541 at 77.72.17.23) ;user=phone>.
Server: MERA MSIP v.1.0.2.
Content-Type: application/sdp.
Content-Length:   212.
.
v=0.
o=- 1295424325 1295424325 IN IP4 77.72.17.23.
s=-.
c=IN IP4 77.72.17.23.
t=0 0.
m=audio 18360 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

U 2011/01/19 11:04:14.246657  77.72.17.23:5060 (http://77.72.17.23:5060/)  ->  194.190.211.190:5060 (http://194.190.211.190:5060/) 
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 194.190.211.190:5060;rport;branch=z9hG4bK6rQjK6B04ZeNN.
From: "DEN78141_120" < sip:TCAPI_User at 77.72.17.23 (http://win.mail.ru/cgi-bin/sentmsg?compose&amp;To=sip%3ATCAPI_User at 77.72.17.23) ;transport=udp>;tag=vZcKXKa0Q7B1e.
To: < sip:74742515541 at 77.72.17.23 (http://win.mail.ru/cgi-bin/sentmsg?compose&amp;To=sip%3A74742515541 at 77.72.17.23) >;tag=0eff5d003cff3610ff00001372630472.
Call-ID: 86959393-9e45-122e-1782-33002db2a994.
CSeq: 7376058 INVITE.
Contact: < sip:74742515541 at 77.72.17.23 (http://win.mail.ru/cgi-bin/sentmsg?compose&amp;To=sip%3A74742515541 at 77.72.17.23) ;user=phone>.
Server: MERA MSIP v.1.0.2.
Content-Type: application/sdp.
Content-Length:   212.
.
v=0.
o=- 1295424325 1295424325 IN IP4 77.72.17.23.
s=-.
c=IN IP4 77.72.17.23.
t=0 0.
m=audio 18360 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

U 2011/01/19 11:04:14.747863  77.72.17.23:5060 (http://77.72.17.23:5060/)  ->  194.190.211.190:5060 (http://194.190.211.190:5060/) 
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 194.190.211.190:5060;rport;branch=z9hG4bK6rQjK6B04ZeNN.
From: "DEN78141_120" < sip:TCAPI_User at 77.72.17.23 (http://win.mail.ru/cgi-bin/sentmsg?compose&amp;To=sip%3ATCAPI_User at 77.72.17.23) ;transport=udp>;tag=vZcKXKa0Q7B1e.
To: < sip:74742515541 at 77.72.17.23 (http://win.mail.ru/cgi-bin/sentmsg?compose&amp;To=sip%3A74742515541 at 77.72.17.23) >;tag=0eff5d003cff3610ff00001372630472.
Call-ID: 86959393-9e45-122e-1782-33002db2a994.
CSeq: 7376058 INVITE.
Contact: < sip:74742515541 at 77.72.17.23 (http://win.mail.ru/cgi-bin/sentmsg?compose&amp;To=sip%3A74742515541 at 77.72.17.23) ;user=phone>.
Server: MERA MSIP v.1.0.2.
Content-Type: application/sdp.
Content-Length:   212.
.
v=0.
o=- 1295424325 1295424325 IN IP4 77.72.17.23.
s=-.
c=IN IP4 77.72.17.23.
t=0 0.
m=audio 18360 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.




--
?????@Mail.Ru ? ????? ?????????. 
?????? ????? ? ???????? ??  m.mail.ru (http://m.mail.ru/) 

_______________________________________________
FreeSWITCH-users mailing list
 FreeSWITCH-users at lists.freeswitch.org (http://win.mail.ru/cgi-bin/sentmsg?compose&amp;To=FreeSWITCH-users at lists.freeswitch.org) 
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users (http://lists.freeswitch.org/mailman/listinfo/freeswitch-users) 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users (http://lists.freeswitch.org/mailman/options/freeswitch-users) 
 http://www.freeswitch.org (http://www.freeswitch.org/) 


_______________________________________________
FreeSWITCH-users mailing list
 FreeSWITCH-users at lists.freeswitch.org (http://win.mail.ru/cgi-bin/sentmsg?compose&amp;To=FreeSWITCH%2dusers at lists.freeswitch.org) 
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users (http://lists.freeswitch.org/mailman/listinfo/freeswitch-users) 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users (http://lists.freeswitch.org/mailman/options/freeswitch-users) 
 http://www.freeswitch.org (http://www.freeswitch.org/) 



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From steveayre at gmail.com  Wed Jan 19 15:53:18 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 19 Jan 2011 12:53:18 +0000
Subject: [Freeswitch-users] Different api calls from Dialplan
In-Reply-To: <20110119121915.GL17734@jgalaz-desktop>
References: <20110119121915.GL17734@jgalaz-desktop>
Message-ID: <AANLkTi=chejRCBeEkA4Ht8vQb3xORKVyZvwmS8wvu1aE@mail.gmail.com>

Are they both running the same FS version?

The first syntax is a newer syntax than the other and wouldn't be in some
older versions.

-Steve


On 19 January 2011 12:19, Javier Galaz Jeria <jgalaz at yx.cl> wrote:

> Hello, freeswitch-users, I've a question regarding dialplan.
>
> Is there a difference between this two function calls (made from the
> dialplan)
> for FreeSWITCH?
>
> a) <action application="set" data="api_result=${ilx_read $1}"
> inline="true"/>
>
> b) <action application="set" data="api_result=${ilx_read ($1)}"
> inline="true"/>
>                                                         ^  ^
> The thing is on one of my boxes both calls work, and on the other only b)
> works
>
> Regards
>
> j
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From stkn at freeswitch.org  Wed Jan 19 15:54:37 2011
From: stkn at freeswitch.org (Stefan Knoblich)
Date: Wed, 19 Jan 2011 13:54:37 +0100
Subject: [Freeswitch-users] Next version of Gemeinschaft based on
	FreeSWITCH
In-Reply-To: <4D36CE19.2070503@chaschperli.ch>
References: <4D36CE19.2070503@chaschperli.ch>
Message-ID: <201101191354.37235.stkn@freeswitch.org>

There's a FAQ posted on their ML (german):

http://groups.google.com/group/gemeinschaft-users/browse_thread/thread/7e4feebab5ef59e0

-- 
-------------------------------------------------------------------------------
Stefan Knoblich                        | Web:   http://www.axsentis.de/
axsentis GmbH                          |        http://oss.axsentis.de/
Eupener Str. 74, 50933 Koeln, Germany  |
Amtsgericht Koeln: HR B 56238          | Email: s.knoblich at axsentis.de
UST-ID: DE244977565                    | JID:   s.knoblich at jabber.axsentis.de
-------------------------------------------------------------------------------
Web:   http://stkn.techmage.de/
Email: stkn at freeswitch.org
IRC:   #freeswitch-de @ irc.freenode.net


From edpimentl at gmail.com  Wed Jan 19 16:02:48 2011
From: edpimentl at gmail.com (EdPimentl)
Date: Wed, 19 Jan 2011 08:02:48 -0500
Subject: [Freeswitch-users] Ideas for FreeSWITCH Marketing Material ?
In-Reply-To: <59B81A5D-5555-4BF0-82C1-ED44375046EA@seventhsignal.de>
References: <59B81A5D-5555-4BF0-82C1-ED44375046EA@seventhsignal.de>
Message-ID: <AANLkTimj-=36JMFpGGY2LLfY1nmVnjd8XqzHQkHL9vQ8@mail.gmail.com>

Michal,

I will send you a professionally designed flyer and brochure.

-E
vCardCloud.com
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From kjv at ken-ton.com  Wed Jan 19 16:09:40 2011
From: kjv at ken-ton.com (Karl Vesterling)
Date: Wed, 19 Jan 2011 08:09:40 -0500
Subject: [Freeswitch-users] Ideas for FreeSWITCH Marketing Material ?
In-Reply-To: <AANLkTimj-=36JMFpGGY2LLfY1nmVnjd8XqzHQkHL9vQ8@mail.gmail.com>
References: <59B81A5D-5555-4BF0-82C1-ED44375046EA@seventhsignal.de>
	<AANLkTimj-=36JMFpGGY2LLfY1nmVnjd8XqzHQkHL9vQ8@mail.gmail.com>
Message-ID: <B097DE9E-3D39-4A76-BDB6-C15747C4D386@ken-ton.com>

Hey, this video is pretty good... ;-)

http://www.youtube.com/watch?v=9Katqjx5RJ4

Actually it was my first attempt at Apple Motion.  I'm much better at it now...


Best Regards,
Karl J. Vesterling
kjv at ken-ton.com
202-461-3231 x0

"Failure is not an option!  It comes bundled with your Microsoft Operating System..."

On Jan 19, 2011, at 8:02 AM, EdPimentl wrote:

> Michal,
> 
> I will send you a professionally designed flyer and brochure.
> 
> -E
> vCardCloud.com
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From edpimentl at gmail.com  Wed Jan 19 16:26:14 2011
From: edpimentl at gmail.com (EdPimentl)
Date: Wed, 19 Jan 2011 08:26:14 -0500
Subject: [Freeswitch-users] Ideas for FreeSWITCH Marketing Material ?
In-Reply-To: <B097DE9E-3D39-4A76-BDB6-C15747C4D386@ken-ton.com>
References: <59B81A5D-5555-4BF0-82C1-ED44375046EA@seventhsignal.de>
	<AANLkTimj-=36JMFpGGY2LLfY1nmVnjd8XqzHQkHL9vQ8@mail.gmail.com>
	<B097DE9E-3D39-4A76-BDB6-C15747C4D386@ken-ton.com>
Message-ID: <AANLkTikzPH35X-2-VUGtaa2tGT2OiPm9GDD2xNV93V7E@mail.gmail.com>

Michal,

If you are also looking for a Cinematic Intro, do you have the Images to
include in it?
See here http://wammawards.org/wammintro/index.html

-E
vCardCloud.com
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From jonas.gauffin at gmail.com  Wed Jan 19 16:29:29 2011
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Wed, 19 Jan 2011 14:29:29 +0100
Subject: [Freeswitch-users] SpanDSP: Timed out waiting for initial
	communication
Message-ID: <AANLkTimMBauUj5WdKfOofDjtT+VnDjc1H4NQq4sdr08p@mail.gmail.com>

Hello,

I've got a fax service used to send an outbound faxes to lot of receivers.

Sometimes I get "Timed out waiting for initial communication". What does it
mean?
Have freeswitch connected to the other end, but no fax tone is detected? Or
something else?

Should I retry again or remove the receiver from the list?

Thanks,
  Jonas
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From jgalaz at yx.cl  Wed Jan 19 16:42:02 2011
From: jgalaz at yx.cl (Javier Galaz Jeria)
Date: Wed, 19 Jan 2011 10:42:02 -0300
Subject: [Freeswitch-users] Different api calls from Dialplan
In-Reply-To: <AANLkTi=chejRCBeEkA4Ht8vQb3xORKVyZvwmS8wvu1aE@mail.gmail.com>
References: <20110119121915.GL17734@jgalaz-desktop>
	<AANLkTi=chejRCBeEkA4Ht8vQb3xORKVyZvwmS8wvu1aE@mail.gmail.com>
Message-ID: <20110119134202.GN17734@jgalaz-desktop>

Nope, they aren't first box running git, 2nd running 1.0.6
Thanks for your answer

j

On Wed, Jan 19, 2011 at 12:53:18PM +0000, Steven Ayre wrote:
>    Are they both running the same FS version?
> 
>    The first syntax is a newer syntax than the other and wouldn't be in some
>    older versions.
> 
>    -Steve
> 
>    On 19 January 2011 12:19, Javier Galaz Jeria <jgalaz at yx.cl> wrote:
> 
>      Hello, freeswitch-users, I've a question regarding dialplan.
> 
>      Is there a difference between this two function calls (made from the
>      dialplan)
>      for FreeSWITCH?
> 
>      a) <action application="set" data="api_result=${ilx_read $1}"
>      inline="true"/>
> 
>      b) <action application="set" data="api_result=${ilx_read ($1)}"
>      inline="true"/>
>                                                              ^  ^
>      The thing is on one of my boxes both calls work, and on the other only
>      b) works
> 
>      Regards
> 
>      j
> 
>      _______________________________________________
>      FreeSWITCH-users mailing list
>      FreeSWITCH-users at lists.freeswitch.org
>      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>      http://www.freeswitch.org

> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From avi at avimarcus.net  Wed Jan 19 17:10:07 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 19 Jan 2011 16:10:07 +0200
Subject: [Freeswitch-users] Ideas for FreeSWITCH Marketing Material ?
In-Reply-To: <AANLkTimj-=36JMFpGGY2LLfY1nmVnjd8XqzHQkHL9vQ8@mail.gmail.com>
References: <59B81A5D-5555-4BF0-82C1-ED44375046EA@seventhsignal.de>
	<AANLkTimj-=36JMFpGGY2LLfY1nmVnjd8XqzHQkHL9vQ8@mail.gmail.com>
Message-ID: <AANLkTi=9VRo0_nWD1GDxwUQh-xFJ1Tvh=RVmKthd7exh@mail.gmail.com>

I'm curious to see it, too.
-Avi

On Wed, Jan 19, 2011 at 3:02 PM, EdPimentl <edpimentl at gmail.com> wrote:

> Michal,
>
> I will send you a professionally designed flyer and brochure.
>
> -E
> vCardCloud.com
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Wed Jan 19 17:32:33 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 19 Jan 2011 14:32:33 +0000
Subject: [Freeswitch-users] Different api calls from Dialplan
In-Reply-To: <20110119134202.GN17734@jgalaz-desktop>
References: <20110119121915.GL17734@jgalaz-desktop>
	<AANLkTi=chejRCBeEkA4Ht8vQb3xORKVyZvwmS8wvu1aE@mail.gmail.com>
	<20110119134202.GN17734@jgalaz-desktop>
Message-ID: <AANLkTimyCJDta82=0RpDEQEUsDsq_1tD5rsLTHZyt4ps@mail.gmail.com>

I believe the change was made since 1.0.6, so that will be the reason.

-Steve



On 19 January 2011 13:42, Javier Galaz Jeria <jgalaz at yx.cl> wrote:

> Nope, they aren't first box running git, 2nd running 1.0.6
> Thanks for your answer
>
> j
>
> On Wed, Jan 19, 2011 at 12:53:18PM +0000, Steven Ayre wrote:
> >    Are they both running the same FS version?
> >
> >    The first syntax is a newer syntax than the other and wouldn't be in
> some
> >    older versions.
> >
> >    -Steve
> >
> >    On 19 January 2011 12:19, Javier Galaz Jeria <jgalaz at yx.cl> wrote:
> >
> >      Hello, freeswitch-users, I've a question regarding dialplan.
> >
> >      Is there a difference between this two function calls (made from the
> >      dialplan)
> >      for FreeSWITCH?
> >
> >      a) <action application="set" data="api_result=${ilx_read $1}"
> >      inline="true"/>
> >
> >      b) <action application="set" data="api_result=${ilx_read ($1)}"
> >      inline="true"/>
> >                                                              ^  ^
> >      The thing is on one of my boxes both calls work, and on the other
> only
> >      b) works
> >
> >      Regards
> >
> >      j
> >
> >      _______________________________________________
> >      FreeSWITCH-users mailing list
> >      FreeSWITCH-users at lists.freeswitch.org
> >      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >      UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >      http://www.freeswitch.org
>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From thomas at chaschperli.ch  Wed Jan 19 18:10:20 2011
From: thomas at chaschperli.ch (Thomas Mueller)
Date: Wed, 19 Jan 2011 16:10:20 +0100
Subject: [Freeswitch-users] [freeswitch-users] SIP auth challenge
 (REGISTER) on sofia profile 'internal' for [1000@my.domain.com] from [some
 ip]
In-Reply-To: <AANLkTinX91gdzrPPp0wxhcUtWLU7YSQStuXVy3RtoM1h@mail.gmail.com>
References: <AANLkTik7LbV5ZJxB14yp7hsR6-UwK9nYxECv6grDbwmL@mail.gmail.com>
	<AANLkTinX91gdzrPPp0wxhcUtWLU7YSQStuXVy3RtoM1h@mail.gmail.com>
Message-ID: <4D36FEDC.9070004@chaschperli.ch>

On 16.01.2011 10:28, Steven Ayre wrote:
> It's logging that a register request is occuring. It's neither 
> succeeded nor failed at this point.
>
> It's there to provide enough logging for fail2ban to work with.
>
I've also stumbled upon this. So this is more a INFO than a WARNING? why 
not chaning it to INFO?

- Thomas



From steveayre at gmail.com  Wed Jan 19 19:53:05 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 19 Jan 2011 16:53:05 +0000
Subject: [Freeswitch-users] [freeswitch-users] SIP auth challenge
 (REGISTER) on sofia profile 'internal' for [1000@my.domain.com] from [some
 ip]
In-Reply-To: <4D36FEDC.9070004@chaschperli.ch>
References: <AANLkTik7LbV5ZJxB14yp7hsR6-UwK9nYxECv6grDbwmL@mail.gmail.com>
	<AANLkTinX91gdzrPPp0wxhcUtWLU7YSQStuXVy3RtoM1h@mail.gmail.com>
	<4D36FEDC.9070004@chaschperli.ch>
Message-ID: <AANLkTikwon8s1_63rtdsh-qRkuTReUZo-HjfuPUHCUoY@mail.gmail.com>

I think that's so that the logging level can be set to warning and fail2ban
will still work. It might be nice to have a parameter that enables/disables
fail2ban's extra logging though.

-Steve


On 19 January 2011 15:10, Thomas Mueller <thomas at chaschperli.ch> wrote:

> On 16.01.2011 10:28, Steven Ayre wrote:
> > It's logging that a register request is occuring. It's neither
> > succeeded nor failed at this point.
> >
> > It's there to provide enough logging for fail2ban to work with.
> >
> I've also stumbled upon this. So this is more a INFO than a WARNING? why
> not chaning it to INFO?
>
> - Thomas
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From peter.olsson at visionutveckling.se  Wed Jan 19 20:02:47 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 19 Jan 2011 18:02:47 +0100
Subject: [Freeswitch-users] RFC 2833 DTMF detection - has something changed?
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57ED00ADDA@cooper>

Hi,

I have some problems with DTMF detection, which I've never seen earlier versions. I know that there was a bug in the rtp code a few days (or week(s)?) back, but I'm running on latest git (as of today). I'm also running on the same machine as before, and more or less no config changes (except for a few dialplan changes, but nothing that changes DTMF detection).

Anyway, it seems to me that the 2833-detection is not as accurate as it was before. When using a couple of months old FS version I rarely missed any DTMF's (I can't remember I ever did..:)), but now it seems to happen once in a while. Also, I decided today to get wireshark up and running, and after 5 DTMF's, FS missed the last one. I looked inside my wireshark dump, and I could clearly see all DTMF packets in there, but FS somehow missed this.

Is there some kind of debugging I could enable, for instance DEBUG_2833 directive?

What's the best way to move this forward? I can send my wireshark dump if that helps, but I guess you will need some more debugging info from FS as well?

Thanks,

Peter Olsson

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From brian at freeswitch.org  Wed Jan 19 20:10:02 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 19 Jan 2011 11:10:02 -0600
Subject: [Freeswitch-users] RFC 2833 DTMF detection - has something
	changed?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57ED00ADDA@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C57ED00ADDA@cooper>
Message-ID: <9CCA8E46-03D7-4F13-B6BD-8AE8D2564006@freeswitch.org>

Update we did have one day where it was messed up.

/b

On Jan 19, 2011, at 11:02 AM, Peter Olsson wrote:

> Hi,
>  
> I have some problems with DTMF detection, which I?ve never seen earlier versions. I know that there was a bug in the rtp code a few days (or week(s)?) back, but I?m running on latest git (as of today). I?m also running on the same machine as before, and more or less no config changes (except for a few dialplan changes, but nothing that changes DTMF detection).
>  
> Anyway, it seems to me that the 2833-detection is not as accurate as it was before. When using a couple of months old FS version I rarely missed any DTMF?s (I can?t remember I ever did..:)), but now it seems to happen once in a while. Also, I decided today to get wireshark up and running, and after 5 DTMF?s, FS missed the last one. I looked inside my wireshark dump, and I could clearly see all DTMF packets in there, but FS somehow missed this.
>  
> Is there some kind of debugging I could enable, for instance DEBUG_2833 directive?
>  
> What?s the best way to move this forward? I can send my wireshark dump if that helps, but I guess you will need some more debugging info from FS as well?
>  
> Thanks,
>  
> Peter Olsson

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From brian at freeswitch.org  Wed Jan 19 20:13:49 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 19 Jan 2011 11:13:49 -0600
Subject: [Freeswitch-users] [freeswitch-users] SIP auth challenge
	(REGISTER) on sofia profile 'internal' for
	[1000@my.domain.com] from [some ip]
In-Reply-To: <AANLkTikwon8s1_63rtdsh-qRkuTReUZo-HjfuPUHCUoY@mail.gmail.com>
References: <AANLkTik7LbV5ZJxB14yp7hsR6-UwK9nYxECv6grDbwmL@mail.gmail.com>
	<AANLkTinX91gdzrPPp0wxhcUtWLU7YSQStuXVy3RtoM1h@mail.gmail.com>
	<4D36FEDC.9070004@chaschperli.ch>
	<AANLkTikwon8s1_63rtdsh-qRkuTReUZo-HjfuPUHCUoY@mail.gmail.com>
Message-ID: <6F6DAEA8-ACA9-4299-A09C-BD240122B225@freeswitch.org>

Its at warning so you can have you loglevel set lower and still catch it.. info will be too high for some people if you have a really busy system.

Warning is level 4 and Info is level 6.

/b

On Jan 19, 2011, at 10:53 AM, Steven Ayre wrote:

> I think that's so that the logging level can be set to warning and fail2ban will still work. It might be nice to have a parameter that enables/disables fail2ban's extra logging though.
> 
> -Steve



From infos at madovsky.org  Wed Jan 19 20:32:12 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 19 Jan 2011 12:32:12 -0500
Subject: [Freeswitch-users] sip_ph_X-var in conference
Message-ID: <09F27B7B578941D688EB8717D621F330@e1705>

is it possilbe to send
sip_ph_X-var to the caller in a conference ?
if yes, how to do it ?

thanks
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From steveayre at gmail.com  Wed Jan 19 20:32:44 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 19 Jan 2011 17:32:44 +0000
Subject: [Freeswitch-users] Yet another DTMF question
In-Reply-To: <AANLkTinL1NxOjMJjShOwXEo5CJERLbY4N7Z7re=QRp8z@mail.gmail.com>
References: <AANLkTik030ecFw2bvKe74pOs+NzfqHknX8keDO3mQ=cH@mail.gmail.com>
	<AANLkTikqQodjsMT_VB1xeQdtY-2DEgL68ndEnHTBTvkb@mail.gmail.com>
	<AANLkTiksK3T24TUDO-7aq_tbkFCkj_4BRNpLRt4=GtBj@mail.gmail.com>
	<62C5671A-8657-4F99-8184-272EEBEC8E77@gmail.com>
	<AANLkTimFMwJ1EkP_+1uCyiKUCKocgh-mSYVEd39navS7@mail.gmail.com>
	<AANLkTikaerx0b+A7EhFLXxuBKBBY+RBog8Lpgyd1ZZmR@mail.gmail.com>
	<AANLkTinL1NxOjMJjShOwXEo5CJERLbY4N7Z7re=QRp8z@mail.gmail.com>
Message-ID: <AANLkTi=Trve2-zDb60FEahz2TLD_=BAvhzisf7gSOoFo@mail.gmail.com>

Try uploading to the latest Git, a post from Brian just reminded me that
there was a day where some changes to the RTP stack broke RFC2833 support,
perhaps you have one of the bad versions.

-Steve




On 19 January 2011 12:34, Travis Stutsman <tlstutsman at gmail.com> wrote:

> Yeah, I've been scouring the debug output and it's definitely not showing
> up.
>
>
>
>
> On Wed, Jan 19, 2011 at 7:27 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Looks ok then...
>>
>> At debug level in the logs DTMF digits showing up?
>>
>> It'll look something similar to:
>> 2011-01-19 12:02:54.855830 [DEBUG] switch_rtp.c:2925 RTP RECV DTMF 1:400
>>
>> -Steve
>>
>>
>>
>> On 19 January 2011 11:45, Travis Stutsman <tlstutsman at gmail.com> wrote:
>>
>>> First I want to thank you for taking the time, Steve.  I really
>>> appreciate it.
>>>
>>> Secondly, here is the SDP from the client end.  It appears to be 101 as
>>> expected.
>>>
>>> v=0
>>> o=- 12939910547141841 1 IN IP4 x.x.x.x
>>> s=CounterPath X-Lite 4.0
>>>
>>> c=IN IP4 x.x.x.x
>>> t=0 0
>>> a=ice-ufrag:8ef9c3
>>> a=ice-pwd:768def2c0edb34ec3cd4c49ed6b1279b
>>> m=audio 55982 RTP/AVP 8 101
>>>
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=sendrecv
>>> a=candidate:1 1 UDP 659136 x.x.x.x 55982 typ host
>>> a=candidate:1 2 UDP 659134 x.x.x.x 55983 typ host
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Jan 19, 2011 at 3:30 AM, Steven Ayre <steveayre at gmail.com>wrote:
>>>
>>>> What about the remote sdp? Just checking fs is expecting 101.
>>>>
>>>> Steve on iPhone
>>>>
>>>> On 18 Jan 2011, at 19:17, Travis Stutsman <tlstutsman at gmail.com> wrote:
>>>>
>>>> Call has been answered - I can hear the IVR.  Payload type is 101 -
>>>> packets on both ends verify this.  SDP example follows:
>>>>
>>>>
>>>> 2011-01-18 19:10:13.579728 [DEBUG] mod_sofia.c:681 Local SDP
>>>> sofia/internal/1000 at x.x.x.x:
>>>> v=0
>>>> o=FreeSWITCH 1295351291 1295351292 IN IP4 x.x.x.x
>>>> s=FreeSWITCH
>>>> c=IN IP4 x.x.x.x
>>>> t=0 0
>>>> m=audio 26522 RTP/AVP 8 101
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=silenceSupp:off - - - -
>>>> a=ptime:20
>>>> a=sendrecv
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Tue, Jan 18, 2011 at 2:02 PM, Steven Ayre < <steveayre at gmail.com>
>>>> steveayre at gmail.com> wrote:
>>>>
>>>>> Has the call been answered? (DTMF is ignored before the call is
>>>>> answered)
>>>>>
>>>>> What payload type are the RFC2833 packets using?
>>>>> What does the SDP look like?
>>>>> Does the payload type in the packets match the telephone-event in the
>>>>> SDP?
>>>>>
>>>>> -Steve
>>>>>
>>>>>
>>>>> On 18 January 2011 16:46, Travis Stutsman < <tlstutsman at gmail.com>
>>>>> tlstutsman at gmail.com> wrote:
>>>>>
>>>>>> I've looked through a lot of threads regarding DTMF troubles, but none
>>>>>> of them seem to relate to me.  I've just pulled a fresh copy of the
>>>>>> freeswitch source and compiled it and I cannot get my digits recognized -
>>>>>> neither in-band nor out-of-band.  I haven't messed with INFO yet as my
>>>>>> phones don't support it.  I've done packet captures on my local interface
>>>>>> and on the server's interface and I can see the RTP rfc2833 digits showing
>>>>>> up, but freeswitch completely ignores them.  I feel like I must be
>>>>>> overlooking something horribly obvious.  How can this be?  Any hints guys?
>>>>>> Thanks in advance!
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>
>>>>>
>>>> _______________________________________________
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>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
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>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
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>>>
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>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>>
>>
>> _______________________________________________
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>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>>
>
> _______________________________________________
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From steveayre at gmail.com  Wed Jan 19 20:33:44 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 19 Jan 2011 17:33:44 +0000
Subject: [Freeswitch-users] RFC 2833 DTMF detection - has something
	changed?
In-Reply-To: <9CCA8E46-03D7-4F13-B6BD-8AE8D2564006@freeswitch.org>
References: <549CFEF87AEDE841A38E9D15EAB4C04C57ED00ADDA@cooper>
	<9CCA8E46-03D7-4F13-B6BD-8AE8D2564006@freeswitch.org>
Message-ID: <AANLkTi=yhXDV7yE-BRpGuYU8rQwwMZU85VYBd87Cbo-+@mail.gmail.com>

Brian, he says he's on Git from today...


On 19 January 2011 17:10, Brian West <brian at freeswitch.org> wrote:

> Update we did have one day where it was messed up.
>
> /b
>
> On Jan 19, 2011, at 11:02 AM, Peter Olsson wrote:
>
> Hi,
>
> I have some problems with DTMF detection, which I?ve never seen earlier
> versions. I know that there was a bug in the rtp code a few days (or
> week(s)?) back, but I?m running on latest git (as of today). I?m also
> running on the same machine as before, and more or less no config changes
> (except for a few dialplan changes, but nothing that changes DTMF
> detection).
>
> Anyway, it seems to me that the 2833-detection is not as accurate as it was
> before. When using a couple of months old FS version I rarely missed any
> DTMF?s (I can?t remember I ever did..:)), but now it seems to happen once in
> a while. Also, I decided today to get wireshark up and running, and after 5
> DTMF?s, FS missed the last one. I looked inside my wireshark dump, and I
> could clearly see all DTMF packets in there, but FS somehow missed this.
>
> Is there some kind of debugging I could enable, for instance DEBUG_2833
> directive?
>
> What?s the best way to move this forward? I can send my wireshark dump if
> that helps, but I guess you will need some more debugging info from FS as
> well?
>
> Thanks,
>
> Peter Olsson
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Jan 19 20:41:01 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Jan 2011 09:41:01 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <AANLkTin1bO3QcTfoWn7MQnVgA_MCeZxQbuuDMxMT_9BP@mail.gmail.com>

Hello all! The agenda for today's conference call is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2011_01_19

We have Mitch Capper scheduled to speak to us on his experiences with
embedding FreeSWITCH in other applications. In his case he created a
FS-based softphone for Windows. We look forward to hearing what he's got!

-MC
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From michal.bielicki at seventhsignal.de  Wed Jan 19 20:48:32 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Wed, 19 Jan 2011 18:48:32 +0100
Subject: [Freeswitch-users] Ideas for FreeSWITCH Marketing Material ?
In-Reply-To: <AANLkTikzPH35X-2-VUGtaa2tGT2OiPm9GDD2xNV93V7E@mail.gmail.com>
References: <59B81A5D-5555-4BF0-82C1-ED44375046EA@seventhsignal.de>
	<AANLkTimj-=36JMFpGGY2LLfY1nmVnjd8XqzHQkHL9vQ8@mail.gmail.com>
	<B097DE9E-3D39-4A76-BDB6-C15747C4D386@ken-ton.com>
	<AANLkTikzPH35X-2-VUGtaa2tGT2OiPm9GDD2xNV93V7E@mail.gmail.com>
Message-ID: <666CF1E2-442E-49EB-BE5A-64C95916BE1A@seventhsignal.de>

All very nice but looks tupid on paper :)

Am 19.01.2011 um 14:26 schrieb EdPimentl:

> Michal,
> 
> If you are also looking for a Cinematic Intro, do you have the Images to include in it?
> See here http://wammawards.org/wammintro/index.html
> 
> -E
> vCardCloud.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de

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From peter.olsson at visionutveckling.se  Wed Jan 19 21:12:26 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 19 Jan 2011 19:12:26 +0100
Subject: [Freeswitch-users] RFC 2833 DTMF detection - has
	something	changed?
In-Reply-To: <AANLkTi=yhXDV7yE-BRpGuYU8rQwwMZU85VYBd87Cbo-+@mail.gmail.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C57ED00ADDA@cooper>
	<9CCA8E46-03D7-4F13-B6BD-8AE8D2564006@freeswitch.org>,
	<AANLkTi=yhXDV7yE-BRpGuYU8rQwwMZU85VYBd87Cbo-+@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57EC81C52A@cooper>

Yep, I'm on today's git, so I do have the last change/fix that was made for this in switch.rtp.c.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Steven Ayre [steveayre at gmail.com]
Skickat: den 19 januari 2011 18:33
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] RFC 2833 DTMF detection - has something    changed?

Brian, he says he's on Git from today...


On 19 January 2011 17:10, Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
Update we did have one day where it was messed up.

/b

On Jan 19, 2011, at 11:02 AM, Peter Olsson wrote:

Hi,

I have some problems with DTMF detection, which I?ve never seen earlier versions. I know that there was a bug in the rtp code a few days (or week(s)?) back, but I?m running on latest git (as of today). I?m also running on the same machine as before, and more or less no config changes (except for a few dialplan changes, but nothing that changes DTMF detection).

Anyway, it seems to me that the 2833-detection is not as accurate as it was before. When using a couple of months old FS version I rarely missed any DTMF?s (I can?t remember I ever did..:)), but now it seems to happen once in a while. Also, I decided today to get wireshark up and running, and after 5 DTMF?s, FS missed the last one. I looked inside my wireshark dump, and I could clearly see all DTMF packets in there, but FS somehow missed this.

Is there some kind of debugging I could enable, for instance DEBUG_2833 directive?

What?s the best way to move this forward? I can send my wireshark dump if that helps, but I guess you will need some more debugging info from FS as well?

Thanks,

Peter Olsson


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


!DSPAM:4d3721aa32761117311318!


From anthony.minessale at gmail.com  Wed Jan 19 21:22:09 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 19 Jan 2011 12:22:09 -0600
Subject: [Freeswitch-users] RFC 2833 DTMF detection - has something
	changed?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57EC81C52A@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C57ED00ADDA@cooper>
	<9CCA8E46-03D7-4F13-B6BD-8AE8D2564006@freeswitch.org>
	<AANLkTi=yhXDV7yE-BRpGuYU8rQwwMZU85VYBd87Cbo-+@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57EC81C52A@cooper>
Message-ID: <AANLkTimXXzVCFweC6x18yrNCf8_st9a_eRD8mxAmyega@mail.gmail.com>

comment line 2993 and see if its better.
If it works better, what is on the other end of the call?, I hate it already.


On Wed, Jan 19, 2011 at 12:12 PM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> Yep, I'm on today's git, so I do have the last change/fix that was made for this in switch.rtp.c.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Steven Ayre [steveayre at gmail.com]
> Skickat: den 19 januari 2011 18:33
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] RFC 2833 DTMF detection - has something ? ?changed?
>
> Brian, he says he's on Git from today...
>
>
> On 19 January 2011 17:10, Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
> Update we did have one day where it was messed up.
>
> /b
>
> On Jan 19, 2011, at 11:02 AM, Peter Olsson wrote:
>
> Hi,
>
> I have some problems with DTMF detection, which I?ve never seen earlier versions. I know that there was a bug in the rtp code a few days (or week(s)?) back, but I?m running on latest git (as of today). I?m also running on the same machine as before, and more or less no config changes (except for a few dialplan changes, but nothing that changes DTMF detection).
>
> Anyway, it seems to me that the 2833-detection is not as accurate as it was before. When using a couple of months old FS version I rarely missed any DTMF?s (I can?t remember I ever did..:)), but now it seems to happen once in a while. Also, I decided today to get wireshark up and running, and after 5 DTMF?s, FS missed the last one. I looked inside my wireshark dump, and I could clearly see all DTMF packets in there, but FS somehow missed this.
>
> Is there some kind of debugging I could enable, for instance DEBUG_2833 directive?
>
> What?s the best way to move this forward? I can send my wireshark dump if that helps, but I guess you will need some more debugging info from FS as well?
>
> Thanks,
>
> Peter Olsson
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> !DSPAM:4d3721aa32761117311318!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From vetali100 at gmail.com  Wed Jan 19 21:31:18 2011
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Wed, 19 Jan 2011 20:31:18 +0200
Subject: [Freeswitch-users] [freeswitch-users] mod_cdr_csv - rotate on
 hup is not working reliable
In-Reply-To: <AANLkTinNcbq-vuLXhvKxrzdYEEEOjyXYUfP1ty5mrT4c@mail.gmail.com>
References: <AANLkTimHO15CvUNBrbMETFgSxqvRSKNDpii-FzV7JzV0@mail.gmail.com>
	<AANLkTinL2QsbQ=PsxfyPFXHp1kLKQFLzCOeAn61hf5yi@mail.gmail.com>
	<AANLkTinNcbq-vuLXhvKxrzdYEEEOjyXYUfP1ty5mrT4c@mail.gmail.com>
Message-ID: <AANLkTi=nW24QV1CdSuT8PpBDy_ZEOtTB4YAi6C32CSOE@mail.gmail.com>

Thank you Anthony.

You are right twice.
I don't have timeout in xml_curl configuration.
And I am using core odbc as well.

So probably whole FS was blocked because of short network outage.

Just FYI, this is what I have in log between 16:57 and 17:01 - but I don't
think it will add more light to this problem:

2011-01-17 16:57:44.338974 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
/usr/local/freeswitch/log/cdr-csv/Master.csv
2011-01-17 16:59:04.382035 [ERR] switch_xml.c:1621 Error[[error near line
1]: root tag missing]
2011-01-17 16:59:10.739780 [WARNING] sofia_reg.c:1241 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [1000 at my.domain.com] from ip
*some_ip*
2011-01-17 17:00:24.443196 [ERR] switch_xml.c:1621 Error[[error near line
1]: root tag missing]
2011-01-17 17:00:30.997069 [ERR] switch_xml.c:1621 Error[[error near line
1]: root tag missing]
2011-01-17 17:00:30.997069 [WARNING] sofia_reg.c:2171 Can't find user [
1000 at my.domain.com]
You must define a domain called 'my.domain.com' in your directory and add a
user with the id="1000" attribute
and you must configure your device to use the proper domain in it's
authentication credentials.
2011-01-17 17:00:31.158235 [WARNING] sofia_reg.c:1199 SIP auth failure
(REGISTER) on sofia profile 'internal' for [1000 at my.domain.com] from ip
*some_ip*
2011-01-17 17:00:51.394967 [WARNING] sofia_reg.c:1241 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [1000 at my.domain.com] from ip
*some_ip*
2011-01-17 17:01:44.539778 [ERR] switch_xml.c:1621 Error[[error near line
1]: root tag missing]
2011-01-17 17:01:44.540795 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
/usr/local/freeswitch/log/cdr-csv/Master.csv
2011-01-17 17:01:44.540795 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
/usr/local/freeswitch/log/cdr-csv/Master.csv
2011-01-17 17:01:44.542071 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile
/usr/local/freeswitch/log/cdr-csv/Master.csv


I will optimize my configuration.

Thank you,
Vitalie



2011/1/18 Anthony Minessale <anthony.minessale at gmail.com>

> you probably did not have your timeout set on the xml_curl
> or are you using odbc as well? when networks time out during odbc you
> were probably blocking the whole FS waiting for it to return.
> It's best not to rely on public networks for critical data.
>
>
> On Mon, Jan 17, 2011 at 6:10 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> > Do you have any information about what was happening during that
> four-minute
> > period? I would have to wonder if there was something goofy happening
> with
> > disk i/o or some other weirdness. The hard part is going to be
> reproducing
> > it. If you are able to reproduce the symptoms then getting a gcore while
> the
> > system is in the weird state might be useful for debugging purposes.
> > What kind of load is on this system? Also, what are the system specs?
> > -MC
> >
> > On Mon, Jan 17, 2011 at 1:48 PM, Vitalii Colosov <vetali100 at gmail.com>
> > wrote:
> >>
> >> Hi,
> >> I am using cron job which executes: killall -HUP freeswitch
> >> Usually new cdr file is created every one minute:
> >> 2011-01-17 16:55:01.583043 [NOTICE] mod_cdr_csv.c:123 Rotated CDR
> logfile
> >> /usr/local/freeswitch/log/cdr-csv/Master.csv
> >> 2011-01-17 16:56:01.861371 [NOTICE] mod_cdr_csv.c:123 Rotated CDR
> logfile
> >> /usr/local/freeswitch/log/cdr-csv/Master.csv
> >> 2011-01-17 16:57:44.338974 [NOTICE] mod_cdr_csv.c:123 Rotated CDR
> logfile
> >> /usr/local/freeswitch/log/cdr-csv/Master.csv
> >>
> >> Today I got some problems with network DB connection, as this also can
> be
> >> seen by few lines (I am using xml_curl to serve registrations):
> >>
> >> 2011-01-17 16:59:04.382035 [ERR] switch_xml.c:1621 Error[[error near
> line
> >> 1]: root tag missing]
> >> 2011-01-17 17:00:24.443196 [ERR] switch_xml.c:1621 Error[[error near
> line
> >> 1]: root tag missing]
> >> 2011-01-17 17:00:30.997069 [ERR] switch_xml.c:1621 Error[[error near
> line
> >> 1]: root tag missing]
> >> 2011-01-17 17:00:30.997069 [WARNING] sofia_reg.c:2171 Can't find user
> >> [1000 at xxx]
> >> You must define a domain called 'xxx' in your directory and add a user
> >> with the id="1000" attribute
> >> and you must configure your device to use the proper domain in it's
> >> authentication credentials.
> >> ...
> >> And I did not get any CDR file between 16:57 and 17:01.
> >>
> >> CDR rotation was restored after 4 minutes - 17:01 - BUT 3 times in 1
> >> second!
> >> 2011-01-17 17:01:44.540795 [NOTICE] mod_cdr_csv.c:123 Rotated CDR
> logfile
> >> /usr/local/freeswitch/log/cdr-csv/Master.csv
> >> 2011-01-17 17:01:44.540795 [NOTICE] mod_cdr_csv.c:123 Rotated CDR
> logfile
> >> /usr/local/freeswitch/log/cdr-csv/Master.csv
> >> 2011-01-17 17:01:44.542071 [NOTICE] mod_cdr_csv.c:123 Rotated CDR
> logfile
> >> /usr/local/freeswitch/log/cdr-csv/Master.csv
> >>
> >> The same repeated again in few minutes:
> >> 2011-01-17 17:05:09.701507 [NOTICE] mod_cdr_csv.c:123 Rotated CDR
> logfile
> >> /usr/local/freeswitch/log/cdr-csv/Master.csv
> >> 2011-01-17 17:05:09.701507 [NOTICE] mod_cdr_csv.c:123 Rotated CDR
> logfile
> >> /usr/local/freeswitch/log/cdr-csv/Master.csv
> >> 2011-01-17 17:05:09.853446 [NOTICE] mod_cdr_csv.c:123 Rotated CDR
> logfile
> >> /usr/local/freeswitch/log/cdr-csv/Master.csv
> >> 2011-01-17 17:05:14.949815 [NOTICE] mod_cdr_csv.c:123 Rotated CDR
> logfile
> >> /usr/local/freeswitch/log/cdr-csv/Master.csv
> >> 2011-01-17 17:05:14.952841 [NOTICE] mod_cdr_csv.c:123 Rotated CDR
> logfile
> >> /usr/local/freeswitch/log/cdr-csv/Master.csv
> >>
> >> Because of this, I lost few call records (it is confirmed).
> >> File at 17:05:14.949815 (which contained 1 call record - as per logs)
> was
> >> overwritten by empty file at 17:05:14.952841.
> >>
> >>
> >> Basically my question is - is it expected behavior that such [rarely
> >> expected] errors make it behave this way?
> >>
> >> It is Freeswitch 1.0.7 on Ubuntu Server 10.4.
> >> Thank you,
> >> Vitalie
> >>
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rmartinez at redvoiss.net  Wed Jan 19 21:16:10 2011
From: rmartinez at redvoiss.net (Ricardo Martinez)
Date: Wed, 19 Jan 2011 15:16:10 -0300
Subject: [Freeswitch-users] Question aboutCPU usage.
Message-ID: <0c8ce8044287af7bd97d78ffa9a69857@mail.gmail.com>

Hello.

I?m new at Freeswitch.  I just installed the versi?n : FreeSWITCH version:
1.0.head (git-c64f475 2011-01-18 14-36-30 -0500).

I also installed the codec g729 for transcoding from g711 to g729  (for
testing purposes only).

I have near 120 simmultaneous calls and I?m having this use of CPU.



top - 15:14:29 up 21:49,  3 users,  load average: 15.15, 15.66, 16.11

Tasks: 118 total,   2 running, 116 sleeping,   0 stopped,   0 zombie

Cpu0  : 49.7%us,  1.7%sy,  0.0%ni, 48.1%id,  0.2%wa,  0.0%hi,  0.2%si,
0.0%st

Cpu1  : 49.5%us,  1.7%sy,  0.0%ni, 48.3%id,  0.3%wa,  0.0%hi,  0.2%si,
0.0%st

Cpu2  : 48.8%us,  1.8%sy,  0.0%ni, 48.3%id,  0.6%wa,  0.1%hi,  0.5%si,
0.0%st

Cpu3  : 47.2%us,  2.2%sy,  0.0%ni, 47.4%id,  0.4%wa,  0.5%hi,  2.4%si,
0.0%st

Mem:   2073316k total,  2024396k used,    48920k free,   159660k buffers

Swap:  4192944k total,       72k used,  4192872k free,  1678300k cached



  PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
COMMAND

21944 root      20   0  130m  70m 4784 S 385.5  3.5 420:08.69
freeswitch

    1 root      20   0  2012  776  572 S  0.0  0.0   0:00.95
init

    2 root      15  -5     0    0    0 S  0.0  0.0   0:00.00
kthreadd

    3 root      RT  -5     0    0    0 S  0.0  0.0   0:00.09
migration/0

    4 root      15  -5     0    0    0 S  0.0  0.0   0:02.70
ksoftirqd/0

    5 root      RT  -5     0    0    0 S  0.0  0.0   0:00.00
watchdog/0

    6 root      RT  -5     0    0    0 S  0.0  0.0   0:00.09
migration/1

    7 root      15  -5     0    0    0 S  0.0  0.0   0:02.66
ksoftirqd/1

    8 root      RT  -5     0    0    0 S  0.0  0.0   0:00.00
watchdog/1

    9 root      RT  -5     0    0    0 S  0.0  0.0   0:00.04
migration/2

   10 root      15  -5     0    0    0 S  0.0  0.0   0:02.25
ksoftirqd/2

   11 root      RT  -5     0    0    0 S  0.0  0.0   0:00.00
watchdog/2



I have a server with Intel(R) Xeon(R) CPU Quad Core @ 2.33GHz and 2 Gig of
RAM.



Is this use of CPU normal with this number of active calls?.  Remember that
I?m doing transcoding.



I also made this setup :



ulimit -c unlimited

ulimit -d unlimited

ulimit -f unlimited

ulimit -i unlimited

ulimit -n 999999

ulimit -q unlimited

ulimit -u unlimited

ulimit -v unlimited

ulimit -x unlimited

ulimit -s 240

ulimit -l unlimited



Can someone help me here?



Thanks

Ricardo.-
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From brian at freeswitch.org  Wed Jan 19 21:40:47 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 19 Jan 2011 12:40:47 -0600
Subject: [Freeswitch-users] Question aboutCPU usage.
In-Reply-To: <0c8ce8044287af7bd97d78ffa9a69857@mail.gmail.com>
References: <0c8ce8044287af7bd97d78ffa9a69857@mail.gmail.com>
Message-ID: <B81E0B19-9F11-4B07-8CA6-57F61972A806@freeswitch.org>

Depend what does uname -a say?

/b

On Jan 19, 2011, at 12:16 PM, Ricardo Martinez wrote:

> Hello.
> I?m new at Freeswitch.  I just installed the versi?n : FreeSWITCH version: 1.0.head (git-c64f475 2011-01-18 14-36-30 -0500).
> I also installed the codec g729 for transcoding from g711 to g729  (for testing purposes only).
> I have near 120 simmultaneous calls and I?m having this use of CPU.

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From peter.olsson at visionutveckling.se  Wed Jan 19 21:41:46 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 19 Jan 2011 19:41:46 +0100
Subject: [Freeswitch-users] RFC 2833 DTMF detection - has
	something	changed?
In-Reply-To: <AANLkTimXXzVCFweC6x18yrNCf8_st9a_eRD8mxAmyega@mail.gmail.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C57ED00ADDA@cooper>
	<9CCA8E46-03D7-4F13-B6BD-8AE8D2564006@freeswitch.org>
	<AANLkTi=yhXDV7yE-BRpGuYU8rQwwMZU85VYBd87Cbo-+@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57EC81C52A@cooper>,
	<AANLkTimXXzVCFweC6x18yrNCf8_st9a_eRD8mxAmyega@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57EC81C52B@cooper>

Thanks, I will try this out tomorrow.

I've used different endpoints while trying, calls via an Avaya PBX, Polycom's registered to FS (and a few other), but the symptoms seem to be the same for all of those. It's not happening very frequently though, but enough to notice something has changed...

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Anthony Minessale [anthony.minessale at gmail.com]
Skickat: den 19 januari 2011 19:22
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] RFC 2833 DTMF detection - has something    changed?

comment line 2993 and see if its better.
If it works better, what is on the other end of the call?, I hate it already.


On Wed, Jan 19, 2011 at 12:12 PM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> Yep, I'm on today's git, so I do have the last change/fix that was made for this in switch.rtp.c.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Steven Ayre [steveayre at gmail.com]
> Skickat: den 19 januari 2011 18:33
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] RFC 2833 DTMF detection - has something    changed?
>
> Brian, he says he's on Git from today...
>
>
> On 19 January 2011 17:10, Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
> Update we did have one day where it was messed up.
>
> /b
>
> On Jan 19, 2011, at 11:02 AM, Peter Olsson wrote:
>
> Hi,
>
> I have some problems with DTMF detection, which I?ve never seen earlier versions. I know that there was a bug in the rtp code a few days (or week(s)?) back, but I?m running on latest git (as of today). I?m also running on the same machine as before, and more or less no config changes (except for a few dialplan changes, but nothing that changes DTMF detection).
>
> Anyway, it seems to me that the 2833-detection is not as accurate as it was before. When using a couple of months old FS version I rarely missed any DTMF?s (I can?t remember I ever did..:)), but now it seems to happen once in a while. Also, I decided today to get wireshark up and running, and after 5 DTMF?s, FS missed the last one. I looked inside my wireshark dump, and I could clearly see all DTMF packets in there, but FS somehow missed this.
>
> Is there some kind of debugging I could enable, for instance DEBUG_2833 directive?
>
> What?s the best way to move this forward? I can send my wireshark dump if that helps, but I guess you will need some more debugging info from FS as well?
>
> Thanks,
>
> Peter Olsson
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4d372d0d32761021210236!



From infos at madovsky.org  Wed Jan 19 21:58:06 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 19 Jan 2011 13:58:06 -0500
Subject: [Freeswitch-users] missing start audio at conference bridge
Message-ID: <EF25E348B5C8445F81D76BB2735A9F45@e1705>

once the bridge is done,
it missing audio until this log line happens

2011-01-19 13:53:32.364254 [DEBUG] sofia.c:4646 Channel sofia/external/user124 at 11.11.11.11 entering state [ready][200]

after that the audio is coming.

is there a way to send a ready 200 before answer or at the pre_answer ?

Thanks

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From edpimentl at gmail.com  Wed Jan 19 22:07:25 2011
From: edpimentl at gmail.com (EdPimentl)
Date: Wed, 19 Jan 2011 14:07:25 -0500
Subject: [Freeswitch-users] Ideas for FreeSWITCH Marketing Material ?
In-Reply-To: <666CF1E2-442E-49EB-BE5A-64C95916BE1A@seventhsignal.de>
References: <59B81A5D-5555-4BF0-82C1-ED44375046EA@seventhsignal.de>
	<AANLkTimj-=36JMFpGGY2LLfY1nmVnjd8XqzHQkHL9vQ8@mail.gmail.com>
	<B097DE9E-3D39-4A76-BDB6-C15747C4D386@ken-ton.com>
	<AANLkTikzPH35X-2-VUGtaa2tGT2OiPm9GDD2xNV93V7E@mail.gmail.com>
	<666CF1E2-442E-49EB-BE5A-64C95916BE1A@seventhsignal.de>
Message-ID: <AANLkTinH4M+z4xWn4Ck_1q6WFpqTNc_nJHXshHXWVB8Y@mail.gmail.com>

? ?
I sent a video intro... not the marketing flyer.

Thanks for your comments, I think.

-E
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From rmartinez at redvoiss.net  Wed Jan 19 21:43:20 2011
From: rmartinez at redvoiss.net (Ricardo Martinez)
Date: Wed, 19 Jan 2011 15:43:20 -0300
Subject: [Freeswitch-users] Question aboutCPU usage.
In-Reply-To: <B81E0B19-9F11-4B07-8CA6-57F61972A806@freeswitch.org>
References: <0c8ce8044287af7bd97d78ffa9a69857@mail.gmail.com>
	<B81E0B19-9F11-4B07-8CA6-57F61972A806@freeswitch.org>
Message-ID: <ed7697940c474d5638d4981746e6caeb@mail.gmail.com>

Hello.



[root at ser-ng bin]# uname -a

Linux 2.6.33.5-124.fc13.i686.PAE #1 SMP Fri Jun 11 09:42:24 UTC 2010 i686
i686 i386 GNU/Linux



Ricardo.-



*De:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *En nombre de *Brian West
*Enviado el:* mi?rcoles, 19 de enero de 2011 15:41
*Para:* FreeSWITCH Users Help
*Asunto:* Re: [Freeswitch-users] Question aboutCPU usage.



Depend what does uname -a say?



/b



On Jan 19, 2011, at 12:16 PM, Ricardo Martinez wrote:



Hello.

I?m new at Freeswitch.  I just installed the versi?n : FreeSWITCH version:
1.0.head (git-c64f475 2011-01-18 14-36-30 -0500).

I also installed the codec g729 for transcoding from g711 to g729  (for
testing purposes only).

I have near 120 simmultaneous calls and I?m having this use of CPU.
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From peter.olsson at visionutveckling.se  Wed Jan 19 22:13:21 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 19 Jan 2011 20:13:21 +0100
Subject: [Freeswitch-users] missing start audio at conference bridge
In-Reply-To: <EF25E348B5C8445F81D76BB2735A9F45@e1705>
References: <EF25E348B5C8445F81D76BB2735A9F45@e1705>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57EC81C52C@cooper>

Answer is a ready/200. Seems like it takes some time to complete setup of the call.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Madovsky [infos at madovsky.org]
Skickat: den 19 januari 2011 19:58
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] missing start audio at conference bridge

once the bridge is done,
it missing audio until this log line happens

2011-01-19 13:53:32.364254 [DEBUG] sofia.c:4646 Channel sofia/external/user124 at 11.11.11.11<mailto:sofia/external/user124 at 11.11.11.11> entering state [ready][200]

after that the audio is coming.

is there a way to send a ready 200 before answer or at the pre_answer ?

Thanks

!DSPAM:4d3735e032761961616417!


From msc at freeswitch.org  Wed Jan 19 22:15:25 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Jan 2011 11:15:25 -0800
Subject: [Freeswitch-users] Possible mod_smpp
Message-ID: <AANLkTiksHkyBnVnbRamuNDK=kxfnE6jCoQdV1eWhXgbr@mail.gmail.com>

Hello all,

If you are interested in giving monetary support to have a professional
software firm create mod_smpp then please contact me off list and I will
give you more details.

Thanks,
Michael
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From brian at freeswitch.org  Wed Jan 19 22:21:44 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 19 Jan 2011 13:21:44 -0600
Subject: [Freeswitch-users] Question aboutCPU usage.
In-Reply-To: <ed7697940c474d5638d4981746e6caeb@mail.gmail.com>
References: <0c8ce8044287af7bd97d78ffa9a69857@mail.gmail.com>
	<B81E0B19-9F11-4B07-8CA6-57F61972A806@freeswitch.org>
	<ed7697940c474d5638d4981746e6caeb@mail.gmail.com>
Message-ID: <12686BDC-187A-478B-8B5E-36FFB0AD5E73@freeswitch.org>

Chances are if you would install a 64bit OS on that there NICE 64bit CPU it would work much better.

/b

On Jan 19, 2011, at 12:43 PM, Ricardo Martinez wrote:

> Hello.
>  
> [root at ser-ng bin]# uname -a
> Linux 2.6.33.5-124.fc13.i686.PAE #1 SMP Fri Jun 11 09:42:24 UTC 2010 i686 i686 i386 GNU/Linux
>  
> Ricardo.-
>  
> De: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de Brian West
> Enviado el: mi?rcoles, 19 de enero de 2011 15:41
> Para: FreeSWITCH Users Help
> Asunto: Re: [Freeswitch-users] Question aboutCPU usage.
>  
> Depend what does uname -a say?
>  
> /b

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From infos at madovsky.org  Wed Jan 19 22:27:44 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 19 Jan 2011 14:27:44 -0500
Subject: [Freeswitch-users] missing start audio at conference bridge
Message-ID: <B3D0290B57D14153BA4DFCD8D58B8421@e1705>

I resolved it by adding a sleep of about 15000 msec before
TTS and after answer
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, January 19, 2011 1:58 PM
  Subject: missing start audio at conference bridge


  once the bridge is done,
  it missing audio until this log line happens

  2011-01-19 13:53:32.364254 [DEBUG] sofia.c:4646 Channel sofia/external/user124 at 11.11.11.11 entering state [ready][200]

  after that the audio is coming.

  is there a way to send a ready 200 before answer or at the pre_answer ?

  Thanks

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From tlstutsman at gmail.com  Wed Jan 19 22:37:55 2011
From: tlstutsman at gmail.com (Travis Stutsman)
Date: Wed, 19 Jan 2011 14:37:55 -0500
Subject: [Freeswitch-users] Yet another DTMF question
In-Reply-To: <AANLkTi=Trve2-zDb60FEahz2TLD_=BAvhzisf7gSOoFo@mail.gmail.com>
References: <AANLkTik030ecFw2bvKe74pOs+NzfqHknX8keDO3mQ=cH@mail.gmail.com>
	<AANLkTikqQodjsMT_VB1xeQdtY-2DEgL68ndEnHTBTvkb@mail.gmail.com>
	<AANLkTiksK3T24TUDO-7aq_tbkFCkj_4BRNpLRt4=GtBj@mail.gmail.com>
	<62C5671A-8657-4F99-8184-272EEBEC8E77@gmail.com>
	<AANLkTimFMwJ1EkP_+1uCyiKUCKocgh-mSYVEd39navS7@mail.gmail.com>
	<AANLkTikaerx0b+A7EhFLXxuBKBBY+RBog8Lpgyd1ZZmR@mail.gmail.com>
	<AANLkTinL1NxOjMJjShOwXEo5CJERLbY4N7Z7re=QRp8z@mail.gmail.com>
	<AANLkTi=Trve2-zDb60FEahz2TLD_=BAvhzisf7gSOoFo@mail.gmail.com>
Message-ID: <AANLkTikL3VkLtM=EJVaGwfpxjXDo5AVyhM_gD_C5evGN@mail.gmail.com>

I've updated and it works now.  I definitely got the broken version.  Thank
you very much.



On Wed, Jan 19, 2011 at 12:32 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Try uploading to the latest Git, a post from Brian just reminded me that
> there was a day where some changes to the RTP stack broke RFC2833 support,
> perhaps you have one of the bad versions.
>
> -Steve
>
>
>
>
>
> On 19 January 2011 12:34, Travis Stutsman <tlstutsman at gmail.com> wrote:
>
>> Yeah, I've been scouring the debug output and it's definitely not showing
>> up.
>>
>>
>>
>>
>> On Wed, Jan 19, 2011 at 7:27 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> Looks ok then...
>>>
>>> At debug level in the logs DTMF digits showing up?
>>>
>>> It'll look something similar to:
>>> 2011-01-19 12:02:54.855830 [DEBUG] switch_rtp.c:2925 RTP RECV DTMF 1:400
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 19 January 2011 11:45, Travis Stutsman <tlstutsman at gmail.com> wrote:
>>>
>>>> First I want to thank you for taking the time, Steve.  I really
>>>> appreciate it.
>>>>
>>>> Secondly, here is the SDP from the client end.  It appears to be 101 as
>>>> expected.
>>>>
>>>> v=0
>>>> o=- 12939910547141841 1 IN IP4 x.x.x.x
>>>> s=CounterPath X-Lite 4.0
>>>>
>>>> c=IN IP4 x.x.x.x
>>>> t=0 0
>>>> a=ice-ufrag:8ef9c3
>>>> a=ice-pwd:768def2c0edb34ec3cd4c49ed6b1279b
>>>> m=audio 55982 RTP/AVP 8 101
>>>>
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-15
>>>> a=sendrecv
>>>> a=candidate:1 1 UDP 659136 x.x.x.x 55982 typ host
>>>> a=candidate:1 2 UDP 659134 x.x.x.x 55983 typ host
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Wed, Jan 19, 2011 at 3:30 AM, Steven Ayre <steveayre at gmail.com>wrote:
>>>>
>>>>> What about the remote sdp? Just checking fs is expecting 101.
>>>>>
>>>>> Steve on iPhone
>>>>>
>>>>> On 18 Jan 2011, at 19:17, Travis Stutsman <tlstutsman at gmail.com>
>>>>> wrote:
>>>>>
>>>>> Call has been answered - I can hear the IVR.  Payload type is 101 -
>>>>> packets on both ends verify this.  SDP example follows:
>>>>>
>>>>>
>>>>> 2011-01-18 19:10:13.579728 [DEBUG] mod_sofia.c:681 Local SDP
>>>>> sofia/internal/1000 at x.x.x.x:
>>>>> v=0
>>>>> o=FreeSWITCH 1295351291 1295351292 IN IP4 x.x.x.x
>>>>> s=FreeSWITCH
>>>>> c=IN IP4 x.x.x.x
>>>>> t=0 0
>>>>> m=audio 26522 RTP/AVP 8 101
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-16
>>>>> a=silenceSupp:off - - - -
>>>>> a=ptime:20
>>>>> a=sendrecv
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Jan 18, 2011 at 2:02 PM, Steven Ayre < <steveayre at gmail.com>
>>>>> steveayre at gmail.com> wrote:
>>>>>
>>>>>> Has the call been answered? (DTMF is ignored before the call is
>>>>>> answered)
>>>>>>
>>>>>> What payload type are the RFC2833 packets using?
>>>>>> What does the SDP look like?
>>>>>> Does the payload type in the packets match the telephone-event in the
>>>>>> SDP?
>>>>>>
>>>>>> -Steve
>>>>>>
>>>>>>
>>>>>> On 18 January 2011 16:46, Travis Stutsman < <tlstutsman at gmail.com>
>>>>>> tlstutsman at gmail.com> wrote:
>>>>>>
>>>>>>> I've looked through a lot of threads regarding DTMF troubles, but
>>>>>>> none of them seem to relate to me.  I've just pulled a fresh copy of the
>>>>>>> freeswitch source and compiled it and I cannot get my digits recognized -
>>>>>>> neither in-band nor out-of-band.  I haven't messed with INFO yet as my
>>>>>>> phones don't support it.  I've done packet captures on my local interface
>>>>>>> and on the server's interface and I can see the RTP rfc2833 digits showing
>>>>>>> up, but freeswitch completely ignores them.  I feel like I must be
>>>>>>> overlooking something horribly obvious.  How can this be?  Any hints guys?
>>>>>>> Thanks in advance!
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From johnrose at comtex.net  Wed Jan 19 22:43:13 2011
From: johnrose at comtex.net (John Rose)
Date: Wed, 19 Jan 2011 12:43:13 -0700
Subject: [Freeswitch-users] dp+ prefixed on From URI
Message-ID: <003701cbb811$1c8febe0$55afc3a0$@comtex.net>

> -----Original Message-----
> From: John Rose [mailto:johnrose at comtex.net]
> 
> Why does the chat API command prefix a "dp+" onto the From URI when I
> call the chat API?  Here is an argument that I am using:
> 
> "sip|+15186819448 at 65.41.13.124|external/sip:+12062990047 at 65.41.13.125|
> Test message."
> 
> Then the From header:
> 
> From: \"+15186819448\"
> <sip:dp++15186819448 at 65.41.13.124>;tag=FUetK564c4egm\\r\\n
> 


Argh, after looking through the code I think this is a FS bug where it was
intended for this "dp+" to be prefixed in the Jingle protocol under certain
circumstances not SIP.... looks very suspicious...

John



From steveu at coppice.org  Wed Jan 19 23:11:38 2011
From: steveu at coppice.org (Steve Underwood)
Date: Thu, 20 Jan 2011 04:11:38 +0800
Subject: [Freeswitch-users] Possible mod_smpp
In-Reply-To: <AANLkTiksHkyBnVnbRamuNDK=kxfnE6jCoQdV1eWhXgbr@mail.gmail.com>
References: <AANLkTiksHkyBnVnbRamuNDK=kxfnE6jCoQdV1eWhXgbr@mail.gmail.com>
Message-ID: <4D37457A.6040607@coppice.org>

On 01/20/2011 03:15 AM, Michael Collins wrote:
> Hello all,
>
> If you are interested in giving monetary support to have a 
> professional software firm create mod_smpp then please contact me off 
> list and I will give you more details.
>
> Thanks,
> Michael
SMPP as in Short Message Peer to Peer? I have floated the idea of open 
sourcing an implementation of that a couple of times, but interest 
seemed weak and I never bothered.

Steve


From potxoka at gmail.com  Wed Jan 19 23:52:04 2011
From: potxoka at gmail.com (Antonio)
Date: Wed, 19 Jan 2011 21:52:04 +0100
Subject: [Freeswitch-users] Fax gateway
Message-ID: <4D374EF4.2090104@gmail.com>

Hi,

I have some doubts about implementing a system FreeSwitch fax, maybe 
it's because I have not understood exactly how this. I have a sip proxy 
architecture and FreeSwitch (several servers) as gateway.

+------+                                 +-------------+
  | PROXY |------------------ | GW FreeSwitch |
+------+                                 +-------------+

                       +----------+
                        |Mail Server  |
                        |       +         |
                        | FreeSwitch  |
                       +----------+

I have read from top to bottom FreeSwitch wiki and I found a script to 
do it, but still I have doubts as to which there are several DID for 
sending and receiving, my biggest problem is the acounting, not very 
well if GW lodge in the application, as it can go to the proxy 
(acounting) to leave by the GW and what DID go between for those the 
proxy (no acounting, but to have control) and return to GW will be in 
the application that send the mail.

Has anyone implemented something like this or can advise me?. My biggest 
question is how to make it happen always for the proxy without making a 
loop and fail. Thanks.

Greetings


From potxoka at gmail.com  Wed Jan 19 23:57:36 2011
From: potxoka at gmail.com (Antonio)
Date: Wed, 19 Jan 2011 21:57:36 +0100
Subject: [Freeswitch-users] Freeswitch and mysql
In-Reply-To: <4CA11C69.70302@gmail.com>
References: <4CA11C69.70302@gmail.com>
Message-ID: <4D375040.1000209@gmail.com>

El 28/09/10 0:36, Antonio escribi?:
> Hello,
>
> I asked and more I searched, I found nothing, how to integrate 
> FreeSwitch with mysql. Some time ago I found Asterisk-realtime and 
> wanted to know if there is something similar in FreeSwitch. Is to 
> avoid double configurations, for example if there is a conference room 
> that can be accessed from multiple servers, etc. I'm new in FreeSwitch 
> and I have much knowledge ;-)
>
> Does anyone have any url or book on how to do? Lua?. Thanks.
>
> Greetings 
Hello,

Thanks, is that having to generate hundreds of xml files with the 
extensions can be very stressful, so we had planned to use php + mysql 
to configureextensions. Buy the book FreeSwitch and I have yet to start 
reading and see if I can find a solution to this problem, also tend to 
read the wiki, but being new to FreeSwitch not quite understand certain 
concepts :-(. Thanks for the aid; -).

Greetings
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From avi at avimarcus.net  Thu Jan 20 00:16:53 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 19 Jan 2011 23:16:53 +0200
Subject: [Freeswitch-users] Freeswitch and mysql
In-Reply-To: <4D375040.1000209@gmail.com>
References: <4CA11C69.70302@gmail.com> <4D375040.1000209@gmail.com>
Message-ID: <AANLkTinYebKhBXQGdwV02bi_kBPB4Vp7Q3iGCv9YRDD7@mail.gmail.com>

Integrate freeswitch with mysql is vague. That itself could mean something
like having freeswitch store it's database in mysql, but that doesn't seem
to be what you want.
You are asking how to use a database (any, not just mysql) to configure the
dialplan - users, extensions, conferences, whatever.
1) You can use lua to process the calls and have it query your database
2) If you just need simple sql queries, check the mod_odbc_query from the
git contrib. If you already understand the dialplan basics, then this can
easily let you query the database as part of that.
3) However, if you need more complicated things, then mod_xml_curl is your
friend - it lets you grab dynamicly generated XML files for each call.
I myself use php to query a mysql database for how much to charge for the
call, a custom LCR implementation, etc.
I posted the basic classes to github a while ago:
https://github.com/avimar/FreeSWITCH-mod_xml-with-PHP
 <https://github.com/avimar/FreeSWITCH-mod_xml-with-PHP>Also, intralanman
wrote a very modular, all inclusive xml_curl implementation in php - which
if you understand it (I didn't know it existed) should be really helpful.
You can find that in the git contrib also in: intralanman/PHP/fs_curl

-Avi Marcus

On Wed, Jan 19, 2011 at 10:57 PM, Antonio <potxoka at gmail.com> wrote:

>  El 28/09/10 0:36, Antonio escribi?:
>
> Hello,
>
> I asked and more I searched, I found nothing, how to integrate FreeSwitch
> with mysql. Some time ago I found Asterisk-realtime and wanted to know if
> there is something similar in FreeSwitch. Is to avoid double
> configurations, for example if there is a conference room that can be
> accessed from multiple servers, etc. I'm new in FreeSwitch and I have much
> knowledge ;-)
>
> Does anyone have any url or book on how to do? Lua?. Thanks.
>
> Greetings
>
> Hello,
>
> Thanks, is that having to generate hundreds of xml files with the
> extensions can be very stressful, so we had planned to use php + mysql to
> configure extensions. Buy the book FreeSwitch and I have yet to start
> reading and see if I can find a solution to this problem, also tend to
> read the wiki, but being new to FreeSwitch not quite understand certain concepts
> :-(. Thanks for the aid; -).
>
> Greetings
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Thu Jan 20 01:53:07 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 19 Jan 2011 23:53:07 +0100
Subject: [Freeswitch-users] Fax gateway
In-Reply-To: <4D374EF4.2090104@gmail.com>
References: <4D374EF4.2090104@gmail.com>
Message-ID: <DF86F0D8-4B36-431C-AD5E-BF1485630BED@ipeva.fr>

Antonio,

I dont know about others, but I was not able to understand what you're trying to accomplish.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 19/01/2011 ? 21:52, Antonio a ?crit :

> Hi,
> 
> I have some doubts about implementing a system FreeSwitch fax, maybe 
> it's because I have not understood exactly how this. I have a sip proxy 
> architecture and FreeSwitch (several servers) as gateway.
> 
> +------+                                 +-------------+
>  | PROXY |------------------ | GW FreeSwitch |
> +------+                                 +-------------+
> 
>                       +----------+
>                        |Mail Server  |
>                        |       +         |
>                        | FreeSwitch  |
>                       +----------+
> 
> I have read from top to bottom FreeSwitch wiki and I found a script to 
> do it, but still I have doubts as to which there are several DID for 
> sending and receiving, my biggest problem is the acounting, not very 
> well if GW lodge in the application, as it can go to the proxy 
> (acounting) to leave by the GW and what DID go between for those the 
> proxy (no acounting, but to have control) and return to GW will be in 
> the application that send the mail.
> 
> Has anyone implemented something like this or can advise me?. My biggest 
> question is how to make it happen always for the proxy without making a 
> loop and fail. Thanks.
> 
> Greetings
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From chat2jesse at gmail.com  Thu Jan 20 02:54:18 2011
From: chat2jesse at gmail.com (jesse)
Date: Wed, 19 Jan 2011 15:54:18 -0800
Subject: [Freeswitch-users] what is wrong with FS1.0.7
Message-ID: <AANLkTim1JZY+rmjZEhW5NFDqC_uAwK5-=jRH-2DM7s6N@mail.gmail.com>

originate sofia/gateway/xyz.com/7132432432

-USAGE: <call url> <exten>|&<application_name>(<app_args>)
[<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>]


originate user/1000

-USAGE: <call url> <exten>|&<application_   >(<app_args>) [<dialplan>]
[<context>] [<cid_name>] [<cid_num>] [<timeout_sec>]


what the heck is going wrong with latest 1.0.7?


From infos at madovsky.org  Thu Jan 20 03:03:33 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 19 Jan 2011 19:03:33 -0500
Subject: [Freeswitch-users] what is wrong with FS1.0.7
References: <AANLkTim1JZY+rmjZEhW5NFDqC_uAwK5-=jRH-2DM7s6N@mail.gmail.com>
Message-ID: <8FFF9C58DC204F5AB2E1A4F715B82B8D@e1705>

I think you have to provide the <exten> in your dialstring

----- Original Message ----- 
From: "jesse" <chat2jesse at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, January 19, 2011 6:54 PM
Subject: [Freeswitch-users] what is wrong with FS1.0.7


> originate sofia/gateway/xyz.com/7132432432
> 
> -USAGE: <call url> <exten>|&<application_name>(<app_args>)
> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>]
> 
> 
> originate user/1000
> 
> -USAGE: <call url> <exten>|&<application_   >(<app_args>) [<dialplan>]
> [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>]
> 
> 
> what the heck is going wrong with latest 1.0.7?
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From chat2jesse at gmail.com  Thu Jan 20 03:04:59 2011
From: chat2jesse at gmail.com (jesse)
Date: Wed, 19 Jan 2011 16:04:59 -0800
Subject: [Freeswitch-users] what is wrong with FS1.0.7
In-Reply-To: <8FFF9C58DC204F5AB2E1A4F715B82B8D@e1705>
References: <AANLkTim1JZY+rmjZEhW5NFDqC_uAwK5-=jRH-2DM7s6N@mail.gmail.com>
	<8FFF9C58DC204F5AB2E1A4F715B82B8D@e1705>
Message-ID: <AANLkTikgK_9JwjZ5kfB0fFzo06JAMvvBnyEL0UMOf9UQ@mail.gmail.com>

the above new command lines used to work in FS 1.0.6.



On Wed, Jan 19, 2011 at 4:03 PM, Madovsky <infos at madovsky.org> wrote:
> I think you have to provide the <exten> in your dialstring
>
> ----- Original Message -----
> From: "jesse" <chat2jesse at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, January 19, 2011 6:54 PM
> Subject: [Freeswitch-users] what is wrong with FS1.0.7
>
>
>> originate sofia/gateway/xyz.com/7132432432
>>
>> -USAGE: <call url> <exten>|&<application_name>(<app_args>)
>> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>]
>>
>>
>> originate user/1000
>>
>> -USAGE: <call url> <exten>|&<application_ ? >(<app_args>) [<dialplan>]
>> [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>]
>>
>>
>> what the heck is going wrong with latest 1.0.7?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Thu Jan 20 03:12:00 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 19 Jan 2011 18:12:00 -0600
Subject: [Freeswitch-users] what is wrong with FS1.0.7
In-Reply-To: <AANLkTikgK_9JwjZ5kfB0fFzo06JAMvvBnyEL0UMOf9UQ@mail.gmail.com>
References: <AANLkTim1JZY+rmjZEhW5NFDqC_uAwK5-=jRH-2DM7s6N@mail.gmail.com>
	<8FFF9C58DC204F5AB2E1A4F715B82B8D@e1705>
	<AANLkTikgK_9JwjZ5kfB0fFzo06JAMvvBnyEL0UMOf9UQ@mail.gmail.com>
Message-ID: <ABBE9766-8819-48C9-A4C1-4E37C3DAC03B@freeswitch.org>

No it didn't...

/b

On Jan 19, 2011, at 6:04 PM, jesse wrote:

> the above new command lines used to work in FS 1.0.6.
> 
> 



From david.ponzone at ipeva.fr  Thu Jan 20 03:16:36 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 20 Jan 2011 01:16:36 +0100
Subject: [Freeswitch-users] what is wrong with FS1.0.7
In-Reply-To: <AANLkTim1JZY+rmjZEhW5NFDqC_uAwK5-=jRH-2DM7s6N@mail.gmail.com>
References: <AANLkTim1JZY+rmjZEhW5NFDqC_uAwK5-=jRH-2DM7s6N@mail.gmail.com>
Message-ID: <55150B8C-CDA9-469E-AD53-D72F37ACE1CE@ipeva.fr>

Jesse,

that's not new.
It has always been like that, at least for the last year.

You can't just originate a one-leg call, without telling FreeSWITCH what to do with it.
Generally, if you want to test, you would play an audio file to it, or send it to echo:
originate user/1000 &echo

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 20/01/2011 ? 00:54, jesse a ?crit :

> originate sofia/gateway/xyz.com/7132432432
> 
> -USAGE: <call url> <exten>|&<application_name>(<app_args>)
> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>]
> 
> 
> originate user/1000
> 
> -USAGE: <call url> <exten>|&<application_   >(<app_args>) [<dialplan>]
> [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>]
> 
> 
> what the heck is going wrong with latest 1.0.7?
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From chat2jesse at gmail.com  Thu Jan 20 03:17:37 2011
From: chat2jesse at gmail.com (jesse)
Date: Wed, 19 Jan 2011 16:17:37 -0800
Subject: [Freeswitch-users] what is wrong with FS1.0.7
In-Reply-To: <ABBE9766-8819-48C9-A4C1-4E37C3DAC03B@freeswitch.org>
References: <AANLkTim1JZY+rmjZEhW5NFDqC_uAwK5-=jRH-2DM7s6N@mail.gmail.com>
	<8FFF9C58DC204F5AB2E1A4F715B82B8D@e1705>
	<AANLkTikgK_9JwjZ5kfB0fFzo06JAMvvBnyEL0UMOf9UQ@mail.gmail.com>
	<ABBE9766-8819-48C9-A4C1-4E37C3DAC03B@freeswitch.org>
Message-ID: <AANLkTimASw0-C96e=_yd0siwXRYhEcEStV3rYFjVDAfc@mail.gmail.com>

my bad .....

On Wed, Jan 19, 2011 at 4:12 PM, Brian West <brian at freeswitch.org> wrote:
> No it didn't...
>
> /b
>
> On Jan 19, 2011, at 6:04 PM, jesse wrote:
>
>> the above new command lines used to work in FS 1.0.6.
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From anthony.minessale at gmail.com  Thu Jan 20 04:02:22 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 19 Jan 2011 19:02:22 -0600
Subject: [Freeswitch-users] dp+ prefixed on From URI
In-Reply-To: <003701cbb811$1c8febe0$55afc3a0$@comtex.net>
References: <003701cbb811$1c8febe0$55afc3a0$@comtex.net>
Message-ID: <AANLkTi=1XUoC5a_Kw8pgmVP3DCxDAKgteSScr4zdQYm5@mail.gmail.com>

It's not a bug, you are being presumptuous.

Every chat interface has a proto prefix that helps FS to route the
messages to the right module.
you sent it from mod_commands which chose the dp realm to advertise dp+

if you came in from SIP it is sip+ and conference are conf+ etc.....


On Wed, Jan 19, 2011 at 1:43 PM, John Rose <johnrose at comtex.net> wrote:
>> -----Original Message-----
>> From: John Rose [mailto:johnrose at comtex.net]
>>
>> Why does the chat API command prefix a "dp+" onto the From URI when I
>> call the chat API? ?Here is an argument that I am using:
>>
>> "sip|+15186819448 at 65.41.13.124|external/sip:+12062990047 at 65.41.13.125|
>> Test message."
>>
>> Then the From header:
>>
>> From: \"+15186819448\"
>> <sip:dp++15186819448 at 65.41.13.124>;tag=FUetK564c4egm\\r\\n
>>
>
>
> Argh, after looking through the code I think this is a FS bug where it was
> intended for this "dp+" to be prefixed in the Jingle protocol under certain
> circumstances not SIP.... looks very suspicious...
>
> John
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From Nabble at slickdeals.endjunk.com  Thu Jan 20 04:54:45 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 19 Jan 2011 17:54:45 -0800 (PST)
Subject: [Freeswitch-users] Fax gateway
In-Reply-To: <DF86F0D8-4B36-431C-AD5E-BF1485630BED@ipeva.fr>
References: <4D374EF4.2090104@gmail.com>
	<DF86F0D8-4B36-431C-AD5E-BF1485630BED@ipeva.fr>
Message-ID: <1295488485340-5942137.post@n2.nabble.com>



David Ponzone wrote:
> I dont know about others, but I was not able to understand what you're
> trying to accomplish.
LOL... You are not alone in this case. 

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Fax-gateway-tp5941346p5942137.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From johnrose at comtex.net  Thu Jan 20 06:10:18 2011
From: johnrose at comtex.net (John Rose)
Date: Wed, 19 Jan 2011 20:10:18 -0700
Subject: [Freeswitch-users] dp+ prefixed on From URI
In-Reply-To: <AANLkTi=1XUoC5a_Kw8pgmVP3DCxDAKgteSScr4zdQYm5@mail.gmail.com>
References: <003701cbb811$1c8febe0$55afc3a0$@comtex.net>
	<AANLkTi=1XUoC5a_Kw8pgmVP3DCxDAKgteSScr4zdQYm5@mail.gmail.com>
Message-ID: <005f01cbb84f$91f857d0$b5e90770$@comtex.net>

 

Well from switch_core_chat_send it's hardcoded "dp" as the protocol when
argv[0] is the actual protocol being passed from the chat API argument.

 

//--------------------------------------------------------------------------
-----------------

SWITCH_STANDARD_API(chat_api_function)

{

      char *lbuf = NULL, *argv[5];

      int argc = 0;

 

      if (!zstr(cmd) && (lbuf = strdup(cmd))

            && (argc = switch_separate_string(lbuf, '|', argv, (sizeof(argv)
/ sizeof(argv[0])))) >= 4) {

 

            if (switch_core_chat_send(argv[0], "dp", argv[1], argv[2], "",
argv[3], !zstr(argv[4]) ? argv[4] : NULL, "") == SWITCH_STATUS_SUCCESS) {

                  stream->write_function(stream, "Sent");

            } else {

                  stream->write_function(stream, "Error! Message Not Sent");

            }

      } else {

            stream->write_function(stream, "Invalid");

      }

 

      switch_safe_free(lbuf);

      return SWITCH_STATUS_SUCCESS;

}

//--------------------------------------------------------------------------
-----------------

 

Then here in sophia_prescence.c  it prepends "dp+" to the sip From URI.  The
"dp+" can cause issues downstream when the sip MESSAGE is gets routed
outbound from the FS box... should be an option to turn it off. 

 

From: \"+15186819448\"
<sip:dp++15186819448 at 65.41.13.124>;tag=FUetK564c4egm\\r\\n

 

 

John

 

//--------------------------------------------------------------------------
-----------------

if (!strcasecmp(proto, SOFIA_CHAT_PROTO)) {

            from = hint;

      } else {

            char *fp, *p = NULL;

            

            fp = strdup(from);

 

            if (!fp) {

                  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT,
"Memory Error!\n");

                  goto end;

            }

 

            if ((p = strchr(fp, '@'))) {

                  *p++ = '\0';

            }

 

            if (zstr(p)) {

                  p = profile->domain_name;

                  if (zstr(p)) {

                        p = host;

                  }

            }

 

            ffrom = switch_mprintf("\"%s\" <sip:%s+%s@%s>", fp, proto, fp,
p);

 

            from = ffrom;

            switch_safe_free(fp);

      }

//--------------------------------------------------------------------------
-----------------

 

 

 

> -----Original Message-----

> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-

> 

> It's not a bug, you are being presumptuous.

> 

> Every chat interface has a proto prefix that helps FS to route the
messages to

> the right module.

> you sent it from mod_commands which chose the dp realm to advertise dp+

> 

> if you came in from SIP it is sip+ and conference are conf+ etc.....

> 

> 

> On Wed, Jan 19, 2011 at 1:43 PM, John Rose < <mailto:johnrose at comtex.net>
johnrose at comtex.net> wrote:

> >> -----Original Message-----

> >> From: John Rose [mailto:johnrose at comtex.net]

> >>

> >> Why does the chat API command prefix a "dp+" onto the From URI when I

> >> call the chat API?  Here is an argument that I am using:

> >>

> >>

> "sip|+15186819448 at 65.41.13.124|external/sip:+12062990047 at 65.41.13.125

> >> |

> >> Test message."

> >>

> >> Then the From header:

> >>

> >> From: \"+15186819448\"

> >> <sip:dp++15186819448 at 65.41.13.124>;tag=FUetK564c4egm\\r\\n

> >>

> >

> >

> > Argh, after looking through the code I think this is a FS bug where it

> > was intended for this "dp+" to be prefixed in the Jingle protocol

> > under certain circumstances not SIP.... looks very suspicious...

> >

> > John

> >

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From jmesquita at freeswitch.org  Thu Jan 20 06:15:12 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 20 Jan 2011 00:15:12 -0300
Subject: [Freeswitch-users] dp+ prefixed on From URI
In-Reply-To: <005f01cbb84f$91f857d0$b5e90770$@comtex.net>
References: <003701cbb811$1c8febe0$55afc3a0$@comtex.net>
	<AANLkTi=1XUoC5a_Kw8pgmVP3DCxDAKgteSScr4zdQYm5@mail.gmail.com>
	<005f01cbb84f$91f857d0$b5e90770$@comtex.net>
Message-ID: <AANLkTikCE9M-zCPRsoPerEw58ooNea2q2bOqHj37AuxR@mail.gmail.com>

John, you should read what he wrote one more time. The code shows precisely
what he said. I am going to quote here:

"you sent it from mod_commands which chose the dp realm to advertise dp+"

If you _received_ a message from SIP, you would've seen the sip+ prefix and
if you sent from the conference module, you would've seen the conf+ prefix
on the event. Makes more sense now?

Regards,
Jo?o Mesquita


On Thu, Jan 20, 2011 at 12:10 AM, John Rose <johnrose at comtex.net> wrote:

>
>
> Well from *switch_core_chat_send* it's hardcoded "dp" as the protocol when
> *argv[0]* is the actual protocol being passed from the *chat* API
> argument.
>
>
>
>
> //-------------------------------------------------------------------------------------------
>
> SWITCH_STANDARD_API(chat_api_function)
>
> {
>
>       char *lbuf = NULL, *argv[5];
>
>       int argc = 0;
>
>
>
>       if (!zstr(cmd) && (lbuf = strdup(cmd))
>
>             && (argc = switch_separate_string(lbuf, '|', argv, (sizeof(argv)
> / sizeof(argv[0])))) >= 4) {
>
>
>
>             if (switch_core_chat_send(argv[0], "dp", argv[1], argv[2], "",
> argv[3], !zstr(argv[4]) ? argv[4] : NULL, "") == SWITCH_STATUS_SUCCESS) {
>
>                   stream->write_function(stream, "Sent");
>
>             } else {
>
>                   stream->write_function(stream, "Error! Message Not Sent"
> );
>
>             }
>
>       } else {
>
>             stream->write_function(stream, "Invalid");
>
>       }
>
>
>
>       switch_safe_free(lbuf);
>
>       return SWITCH_STATUS_SUCCESS;
>
> }
>
>
> //-------------------------------------------------------------------------------------------
>
>
>
> Then here in sophia_prescence.c  it prepends ?dp+? to the sip *From* URI.
> The ?dp+? can cause issues downstream when the sip *MESSAGE* is gets
> routed outbound from the FS box... should be an option to turn it off?
>
>
>
> From: \"+15186819448\" <sip:dp++15186819448 at 65.41.13.124<sip%3Adp%2B%2B15186819448 at 65.41.13.124>
> >;tag=FUetK564c4egm\\r\\n
>
>
>
>
>
> John
>
>
>
>
> //-------------------------------------------------------------------------------------------
>
> if (!strcasecmp(proto, SOFIA_CHAT_PROTO)) {
>
>             from = hint;
>
>       } else {
>
>             char *fp, *p = NULL;
>
>
>
>             fp = strdup(from);
>
>
>
>             if (!fp) {
>
>                   switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Memory
> Error!\n");
>
>                   goto end;
>
>             }
>
>
>
>             if ((p = strchr(fp, '@'))) {
>
>                   *p++ = '\0';
>
>             }
>
>
>
>             if (zstr(p)) {
>
>                   p = profile->domain_name;
>
>                   if (zstr(p)) {
>
>                         p = host;
>
>                   }
>
>             }
>
>
>
>             ffrom = switch_mprintf("\"%s\" <sip:%s+%s@%s>", fp, proto, fp,
> p);
>
>
>
>             from = ffrom;
>
>             switch_safe_free(fp);
>
>       }
>
>
> //-------------------------------------------------------------------------------------------
>
>
>
>
>
>
>
> > -----Original Message-----
>
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
>
> >
>
> > It's not a bug, you are being presumptuous.
>
> >
>
> > Every chat interface has a proto prefix that helps FS to route the
> messages to
>
> > the right module.
>
> > you sent it from mod_commands which chose the dp realm to advertise dp+
>
> >
>
> > if you came in from SIP it is sip+ and conference are conf+ etc.....
>
> >
>
> >
>
> > On Wed, Jan 19, 2011 at 1:43 PM, John Rose <johnrose at comtex.net> wrote:
>
> > >> -----Original Message-----
>
> > >> From: John Rose [mailto:johnrose at comtex.net]
>
> > >>
>
> > >> Why does the chat API command prefix a "dp+" onto the From URI when I
>
> > >> call the chat API?  Here is an argument that I am using:
>
> > >>
>
> > >>
>
> > "sip|+15186819448 at 65.41.13.124|external/sip:+12062990047 at 65.41.13.125<sip%3A%2B12062990047 at 65.41.13.125>
>
> > >> |
>
> > >> Test message."
>
> > >>
>
> > >> Then the From header:
>
> > >>
>
> > >> From: \"+15186819448\"
>
> > >> <sip:dp++15186819448 at 65.41.13.124<sip%3Adp%2B%2B15186819448 at 65.41.13.124>
> >;tag=FUetK564c4egm\\r\\n
>
> > >>
>
> > >
>
> > >
>
> > > Argh, after looking through the code I think this is a FS bug where it
>
> > > was intended for this "dp+" to be prefixed in the Jingle protocol
>
> > > under certain circumstances not SIP.... looks very suspicious...
>
> > >
>
> > > John
>
> > >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From johnrose at comtex.net  Thu Jan 20 06:31:37 2011
From: johnrose at comtex.net (John Rose)
Date: Wed, 19 Jan 2011 20:31:37 -0700
Subject: [Freeswitch-users] dp+ prefixed on From URI
In-Reply-To: <AANLkTikCE9M-zCPRsoPerEw58ooNea2q2bOqHj37AuxR@mail.gmail.com>
References: <003701cbb811$1c8febe0$55afc3a0$@comtex.net>	<AANLkTi=1XUoC5a_Kw8pgmVP3DCxDAKgteSScr4zdQYm5@mail.gmail.com>	<005f01cbb84f$91f857d0$b5e90770$@comtex.net>
	<AANLkTikCE9M-zCPRsoPerEw58ooNea2q2bOqHj37AuxR@mail.gmail.com>
Message-ID: <006601cbb852$8c99cd70$a5cd6850$@comtex.net>

Yes but there should be an option to turn it off.  When a UA receives the
MESSAGE Request the ?dp+? gets displayed on some clients which is confusing
and the reply needs to route back potentially though many sip hops with
lookups etc.. Having a ?dp+? prepended on the From URI may not be the best
sip based solution for that, maybe adding a custom header perhaps?

 

John

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o
Mesquita
Sent: Wednesday, January 19, 2011 8:15 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] dp+ prefixed on From URI

 

John, you should read what he wrote one more time. The code shows precisely
what he said. I am going to quote here:

 

"you sent it from mod_commands which chose the dp realm to advertise dp+"

 

If you _received_ a message from SIP, you would've seen the sip+ prefix and
if you sent from the conference module, you would've seen the conf+ prefix
on the event. Makes more sense now?

 

Regards,

Jo?o Mesquita



On Thu, Jan 20, 2011 at 12:10 AM, John Rose <johnrose at comtex.net> wrote:

 

Well from switch_core_chat_send it's hardcoded "dp" as the protocol when
argv[0] is the actual protocol being passed from the chat API argument.

 

//--------------------------------------------------------------------------
-----------------

SWITCH_STANDARD_API(chat_api_function)

{

      char *lbuf = NULL, *argv[5];

      int argc = 0;

 

      if (!zstr(cmd) && (lbuf = strdup(cmd))

            && (argc = switch_separate_string(lbuf, '|', argv, (sizeof(argv)
/ sizeof(argv[0])))) >= 4) {

 

            if (switch_core_chat_send(argv[0], "dp", argv[1], argv[2], "",
argv[3], !zstr(argv[4]) ? argv[4] : NULL, "") == SWITCH_STATUS_SUCCESS) {

                  stream->write_function(stream, "Sent");

            } else {

                  stream->write_function(stream, "Error! Message Not Sent");

            }

      } else {

            stream->write_function(stream, "Invalid");

      }

 

      switch_safe_free(lbuf);

      return SWITCH_STATUS_SUCCESS;

}

//--------------------------------------------------------------------------
-----------------

 

Then here in sophia_prescence.c  it prepends ?dp+? to the sip From URI.  The
?dp+? can cause issues downstream when the sip MESSAGE is gets routed
outbound from the FS box... should be an option to turn it off
 

 

From: \"+15186819448\" <sip:dp++15186819448 at 65.41.13.124
<mailto:sip%3Adp%2B%2B15186819448 at 65.41.13.124> >;tag=FUetK564c4egm\\r\\n

 

 

John

 

//--------------------------------------------------------------------------
-----------------

if (!strcasecmp(proto, SOFIA_CHAT_PROTO)) {

            from = hint;

      } else {

            char *fp, *p = NULL;

            

            fp = strdup(from);

 

            if (!fp) {

                  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT,
"Memory Error!\n");

                  goto end;

            }

 

            if ((p = strchr(fp, '@'))) {

                  *p++ = '\0';

            }

 

            if (zstr(p)) {

                  p = profile->domain_name;

                  if (zstr(p)) {

                        p = host;

                  }

            }

 

            ffrom = switch_mprintf("\"%s\" <sip:%s+%s@%s
<sip:%25s+%25s@%25s> >", fp, proto, fp, p);

 

            from = ffrom;

            switch_safe_free(fp);

      }

//--------------------------------------------------------------------------
-----------------

 

 

 

> -----Original Message-----

> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-

> 

> It's not a bug, you are being presumptuous.

> 

> Every chat interface has a proto prefix that helps FS to route the
messages to

> the right module.

> you sent it from mod_commands which chose the dp realm to advertise dp+

> 

> if you came in from SIP it is sip+ and conference are conf+ etc.....

> 

> 

> On Wed, Jan 19, 2011 at 1:43 PM, John Rose < <mailto:johnrose at comtex.net>
johnrose at comtex.net> wrote:

> >> -----Original Message-----

> >> From: John Rose [mailto:johnrose at comtex.net]

> >>

> >> Why does the chat API command prefix a "dp+" onto the From URI when I

> >> call the chat API?  Here is an argument that I am using:

> >>

> >>

> "sip|+15186819448 at 65.41.13.124|external/sip:+12062990047 at 65.41.13.125
<mailto:sip%3A%2B12062990047 at 65.41.13.125> 

> >> |

> >> Test message."

> >>

> >> Then the From header:

> >>

> >> From: \"+15186819448\"

> >> <sip:dp++15186819448 at 65.41.13.124
<mailto:sip%3Adp%2B%2B15186819448 at 65.41.13.124> >;tag=FUetK564c4egm\\r\\n

> >>

> >

> >

> > Argh, after looking through the code I think this is a FS bug where it

> > was intended for this "dp+" to be prefixed in the Jingle protocol

> > under certain circumstances not SIP.... looks very suspicious...

> >

> > John

> >


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From jmesquita at freeswitch.org  Thu Jan 20 06:52:29 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 20 Jan 2011 00:52:29 -0300
Subject: [Freeswitch-users] dp+ prefixed on From URI
In-Reply-To: <006601cbb852$8c99cd70$a5cd6850$@comtex.net>
References: <003701cbb811$1c8febe0$55afc3a0$@comtex.net>
	<AANLkTi=1XUoC5a_Kw8pgmVP3DCxDAKgteSScr4zdQYm5@mail.gmail.com>
	<005f01cbb84f$91f857d0$b5e90770$@comtex.net>
	<AANLkTikCE9M-zCPRsoPerEw58ooNea2q2bOqHj37AuxR@mail.gmail.com>
	<006601cbb852$8c99cd70$a5cd6850$@comtex.net>
Message-ID: <AANLkTin86KHAEAzQ=1c4tQwALd4BQU9m-p18dXi3tik0@mail.gmail.com>

Someone else will have to kick in now, I haven't played with SIMPLE or the
chat api yet.

Regards,
Jo?o Mesquita


On Thu, Jan 20, 2011 at 12:31 AM, John Rose <johnrose at comtex.net> wrote:

> Yes but there should be an option to turn it off.  When a UA receives the
> MESSAGE Request the ?dp+? gets displayed on some clients which is confusing
> and the reply needs to route back potentially though many sip hops with
> lookups etc.. Having a ?dp+? prepended on the From URI may not be the best
> sip based solution for that, maybe adding a custom header perhaps?
>
>
>
> John
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jo?o
> Mesquita
> *Sent:* Wednesday, January 19, 2011 8:15 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] dp+ prefixed on From URI
>
>
>
> John, you should read what he wrote one more time. The code shows precisely
> what he said. I am going to quote here:
>
>
>
> "you sent it from mod_commands which chose the dp realm to advertise dp+"
>
>
>
> If you _received_ a message from SIP, you would've seen the sip+ prefix and
> if you sent from the conference module, you would've seen the conf+ prefix
> on the event. Makes more sense now?
>
>
>
> Regards,
>
> Jo?o Mesquita
>
> On Thu, Jan 20, 2011 at 12:10 AM, John Rose <johnrose at comtex.net> wrote:
>
>
>
> Well from *switch_core_chat_send* it's hardcoded "dp" as the protocol when
> *argv[0]* is the actual protocol being passed from the *chat* API
> argument.
>
>
>
>
> //-------------------------------------------------------------------------------------------
>
> SWITCH_STANDARD_API(chat_api_function)
>
> {
>
>       char *lbuf = NULL, *argv[5];
>
>       int argc = 0;
>
>
>
>       if (!zstr(cmd) && (lbuf = strdup(cmd))
>
>             && (argc = switch_separate_string(lbuf, '|', argv, (sizeof(argv)
> / sizeof(argv[0])))) >= 4) {
>
>
>
>             if (switch_core_chat_send(argv[0], "dp", argv[1], argv[2], "",
> argv[3], !zstr(argv[4]) ? argv[4] : NULL, "") == SWITCH_STATUS_SUCCESS) {
>
>                   stream->write_function(stream, "Sent");
>
>             } else {
>
>                   stream->write_function(stream, "Error! Message Not Sent"
> );
>
>             }
>
>       } else {
>
>             stream->write_function(stream, "Invalid");
>
>       }
>
>
>
>       switch_safe_free(lbuf);
>
>       return SWITCH_STATUS_SUCCESS;
>
> }
>
>
> //-------------------------------------------------------------------------------------------
>
>
>
> Then here in sophia_prescence.c  it prepends ?dp+? to the sip *From* URI.
> The ?dp+? can cause issues downstream when the sip *MESSAGE* is gets
> routed outbound from the FS box... should be an option to turn it off?
>
>
>
> From: \"+15186819448\" <sip:dp++15186819448 at 65.41.13.124<sip%3Adp%2B%2B15186819448 at 65.41.13.124>
> >;tag=FUetK564c4egm\\r\\n
>
>
>
>
>
> John
>
>
>
>
> //-------------------------------------------------------------------------------------------
>
> if (!strcasecmp(proto, SOFIA_CHAT_PROTO)) {
>
>             from = hint;
>
>       } else {
>
>             char *fp, *p = NULL;
>
>
>
>             fp = strdup(from);
>
>
>
>             if (!fp) {
>
>                   switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Memory
> Error!\n");
>
>                   goto end;
>
>             }
>
>
>
>             if ((p = strchr(fp, '@'))) {
>
>                   *p++ = '\0';
>
>             }
>
>
>
>             if (zstr(p)) {
>
>                   p = profile->domain_name;
>
>                   if (zstr(p)) {
>
>                         p = host;
>
>                   }
>
>             }
>
>
>
>             ffrom = switch_mprintf("\"%s\" <sip:%s+%s@%s>", fp, proto, fp,
> p);
>
>
>
>             from = ffrom;
>
>             switch_safe_free(fp);
>
>       }
>
>
> //-------------------------------------------------------------------------------------------
>
>
>
>
>
>
>
> > -----Original Message-----
>
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
>
> >
>
> > It's not a bug, you are being presumptuous.
>
> >
>
> > Every chat interface has a proto prefix that helps FS to route the
> messages to
>
> > the right module.
>
> > you sent it from mod_commands which chose the dp realm to advertise dp+
>
> >
>
> > if you came in from SIP it is sip+ and conference are conf+ etc.....
>
> >
>
> >
>
> > On Wed, Jan 19, 2011 at 1:43 PM, John Rose <johnrose at comtex.net> wrote:
>
> > >> -----Original Message-----
>
> > >> From: John Rose [mailto:johnrose at comtex.net]
>
> > >>
>
> > >> Why does the chat API command prefix a "dp+" onto the From URI when I
>
> > >> call the chat API?  Here is an argument that I am using:
>
> > >>
>
> > >>
>
> > "sip|+15186819448 at 65.41.13.124|external/sip:+12062990047 at 65.41.13.125<sip%3A%2B12062990047 at 65.41.13.125>
>
> > >> |
>
> > >> Test message."
>
> > >>
>
> > >> Then the From header:
>
> > >>
>
> > >> From: \"+15186819448\"
>
> > >> <sip:dp++15186819448 at 65.41.13.124<sip%3Adp%2B%2B15186819448 at 65.41.13.124>
> >;tag=FUetK564c4egm\\r\\n
>
> > >>
>
> > >
>
> > >
>
> > > Argh, after looking through the code I think this is a FS bug where it
>
> > > was intended for this "dp+" to be prefixed in the Jingle protocol
>
> > > under certain circumstances not SIP.... looks very suspicious...
>
> > >
>
> > > John
>
> > >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Thu Jan 20 09:29:00 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 20 Jan 2011 01:29:00 -0500
Subject: [Freeswitch-users] non blocking dialplan playback
Message-ID: <9FA12FE2EB6B412FAF98D195785E8AC6@e1705>

I'm trying to playback music on leg A while the dialplan
continues action. playback, uuid_broadcast (from dialplan)
appear to block the dialplan (maybe I'm wrong)
which is the best way to achieve that ?

Thanks
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From infos at madovsky.org  Thu Jan 20 10:11:24 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 20 Jan 2011 02:11:24 -0500
Subject: [Freeswitch-users] non blocking dialplan playback
Message-ID: <271155B89E3E4E57878A6E75A1111DF5@e1705>

don't know if it's crappy or not,
but this works

                      <action application="set" data="api_result=${sched_api(+16 none uuid_break ${uuid})}"/>
                      <action application="answer"/>
                      <action application="playback" data="$${hold_music}"/>

other suggestion welcome


  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, January 20, 2011 1:29 AM
  Subject: non blocking dialplan playback


  I'm trying to playback music on leg A while the dialplan
  continues action. playback, uuid_broadcast (from dialplan)
  appear to block the dialplan (maybe I'm wrong)
  which is the best way to achieve that ?

  Thanks
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From u2nsam at gmail.com  Thu Jan 20 12:43:14 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 20 Jan 2011 15:13:14 +0530
Subject: [Freeswitch-users] call barging
Message-ID: <AANLkTinMqT+Qs1guPphYWqN5okMAD9c1K9=Jey6uepRw@mail.gmail.com>

hello ,

I am using,

<extension name="global" continue="true">
     <condition>
       <action application="info"/>
       <action application="db"
data="insert/spymap/${caller_id_number}/${uuid}"/>
       <action application="db"
data="insert/last_dial/${caller_id_number}/${destination_number}"/>
       <action application="db" data="insert/last_dial/global/${uuid}"/>
     </condition>
   </extension>

   <extension name="eavesdrop">
     <condition field="destination_number" expression="^88(.*)$|^\*0(.*)$">
       <action application="answer"/>
       <action application="eavesdrop" data="${db(select/spymap/$1)}"/>
     </condition>
   </extension>



i am getting the below error:-

2011-01-20 14:54:01.581151 [NOTICE] mod_dptools.c:920 Channel
[sofia/internal/7006 at 192.168.2.190] has been answered
2011-01-20 14:54:01.581151 [DEBUG] sofia.c:4646 Channel sofia/internal/
7006 at 192.168.2.190 entering state [completed][200]
EXECUTE sofia/internal/7006 at 192.168.2.190 eavesdrop()
2011-01-20 14:54:01.582119 [ERR] mod_dptools.c:529 Usage: [all | <uuid>]
2011-01-20 14:54:01.582119 [NOTICE] switch_core_state_machine.c:189
sofia/internal/7006 at 192.168.2.190 has executed the last dialplan
instruction, hanging up.
2011-01-20 14:54:01.582119 [DEBUG] switch_channel.c:2535 (sofia/internal/
7006 at 192.168.2.190) Callstate Change ACTIVE -> HANGUP
2011-01-20 14:54:01.582119 [NOTICE] switch_core_state_machine.c:191 Hangup
sofia/internal/7006 at 192.168.2.190 [CS_EXECUTE] [NORMAL_CLEARING]


http://pastebin.freeswitch.org/15076


Regds
Sam
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From potxoka at gmail.com  Thu Jan 20 12:56:20 2011
From: potxoka at gmail.com (Antonio)
Date: Thu, 20 Jan 2011 10:56:20 +0100
Subject: [Freeswitch-users] Fax gateway
In-Reply-To: <DF86F0D8-4B36-431C-AD5E-BF1485630BED@ipeva.fr>
References: <4D374EF4.2090104@gmail.com>
	<DF86F0D8-4B36-431C-AD5E-BF1485630BED@ipeva.fr>
Message-ID: <4D3806C4.4050009@gmail.com>

El 19/01/11 23:53, David Ponzone escribi?:
>
> I dont know about others, but I was not able to understand what you're 
> trying to accomplish.
Hello

In the rush might not explain to me properly. I want my GW (FreeSwitch) 
process the fax (mailserver + python script), so that the sending and 
receiving signaling pass through the proxy, not directly send to be 
configured in the FreeSwitch provider. thank you very much ;-)

Greetings


From Prometheus001 at gmx.net  Thu Jan 20 13:00:09 2011
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 20 Jan 2011 11:00:09 +0100
Subject: [Freeswitch-users] SIP device for attaching a speaker
Message-ID: <4D3807A9.8040103@gmx.net>

Hello,

I need to attach a speaker to our Freeeswitch. I may use a phone where I
attach an amplifier via the headset plug and put the phone on auto
answer (just as a workaround).
But does anybody know of a SIP device which is just built for that, e.g.
without keypad and designed for humid environments?

Best regards
Peter






From steveayre at gmail.com  Thu Jan 20 13:10:30 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 20 Jan 2011 10:10:30 +0000
Subject: [Freeswitch-users] SIP device for attaching a speaker
In-Reply-To: <4D3807A9.8040103@gmx.net>
References: <4D3807A9.8040103@gmx.net>
Message-ID: <AANLkTimTasBFgS3Ob_L9BVQf-UOJiySGna3Xe9+kSmHv@mail.gmail.com>

Is this of any use?
http://wiki.freeswitch.org/wiki/Mod_portaudio

-Steve



On 20 January 2011 10:00, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Hello,
>
> I need to attach a speaker to our Freeeswitch. I may use a phone where I
> attach an amplifier via the headset plug and put the phone on auto
> answer (just as a workaround).
> But does anybody know of a SIP device which is just built for that, e.g.
> without keypad and designed for humid environments?
>
> Best regards
> Peter
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From u2nsam at gmail.com  Thu Jan 20 13:11:31 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 20 Jan 2011 15:41:31 +0530
Subject: [Freeswitch-users] call barging
In-Reply-To: <AANLkTinMqT+Qs1guPphYWqN5okMAD9c1K9=Jey6uepRw@mail.gmail.com>
References: <AANLkTinMqT+Qs1guPphYWqN5okMAD9c1K9=Jey6uepRw@mail.gmail.com>
Message-ID: <AANLkTimrbwUS7s1G+catXL-XVTr_pg9yOWorR04MNFY=@mail.gmail.com>

If i use

<variable name="eavesdrop_group" value="spy1"/>

at the user config, will the  dialplan barge the above extension where the
statement is stated ?

<extension name="eavesdrop">
     <condition field="destination_number" expression="^779$">
       <action application="answer"/>
       <action application="set"
data="eavesdrop_require_group=sales_call_eavesdrop"/>
       <action application="eavesdrop" data="all"/>
     </condition>
   </extension>


Regds
Sam











On Thu, Jan 20, 2011 at 3:13 PM, Sam <u2nsam at gmail.com> wrote:

> hello ,
>
> I am using,
>
> <extension name="global" continue="true">
>      <condition>
>        <action application="info"/>
>        <action application="db" data="insert/spymap/${caller_id_number}/${uuid}"/>
>
>        <action application="db" data="insert/last_dial/${caller_id_number}/${destination_number}"/>
>        <action application="db" data="insert/last_dial/global/${uuid}"/>
>
>      </condition>
>    </extension>
>
>    <extension name="eavesdrop">
>
>      <condition field="destination_number" expression="^88(.*)$|^\*0(.*)$">
>        <action application="answer"/>
>        <action application="eavesdrop" data="${db(select/spymap/$1)}"/>
>
>      </condition>
>    </extension>
>
>
>
> i am getting the below error:-
>
> 2011-01-20 14:54:01.581151 [NOTICE] mod_dptools.c:920 Channel
> [sofia/internal/7006 at 192.168.2.190] has been answered
> 2011-01-20 14:54:01.581151 [DEBUG] sofia.c:4646 Channel sofia/internal/
> 7006 at 192.168.2.190 entering state [completed][200]
> EXECUTE sofia/internal/7006 at 192.168.2.190 eavesdrop()
> 2011-01-20 14:54:01.582119 [ERR] mod_dptools.c:529 Usage: [all | <uuid>]
> 2011-01-20 14:54:01.582119 [NOTICE] switch_core_state_machine.c:189
> sofia/internal/7006 at 192.168.2.190 has executed the last dialplan
> instruction, hanging up.
> 2011-01-20 14:54:01.582119 [DEBUG] switch_channel.c:2535 (sofia/internal/
> 7006 at 192.168.2.190) Callstate Change ACTIVE -> HANGUP
> 2011-01-20 14:54:01.582119 [NOTICE] switch_core_state_machine.c:191 Hangup
> sofia/internal/7006 at 192.168.2.190 [CS_EXECUTE] [NORMAL_CLEARING]
>
>
> http://pastebin.freeswitch.org/15076
>
>
> Regds
> Sam
>
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From steveayre at gmail.com  Thu Jan 20 13:15:55 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 20 Jan 2011 10:15:55 +0000
Subject: [Freeswitch-users] call barging
In-Reply-To: <AANLkTinMqT+Qs1guPphYWqN5okMAD9c1K9=Jey6uepRw@mail.gmail.com>
References: <AANLkTinMqT+Qs1guPphYWqN5okMAD9c1K9=Jey6uepRw@mail.gmail.com>
Message-ID: <AANLkTi=uWC6f3j8-cCEAeS=DFA-yyLpj0J8Gxi49EcTf@mail.gmail.com>

>
> EXECUTE sofia/internal/7006 at 192.168.2.190 eavesdrop()
> 2011-01-20 14:54:01.582119 [ERR] mod_dptools.c:529 Usage: [all | <uuid>]
>

It's telling you you're using it wrong. You need to tell eavesdrop who you
want to eavesdrop on. You don't have a data attribute, which must either be
"all" (useful for eavesdropping on groups) or the UUID.

There's examples of both on the wiki:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop

-Steve



On 20 January 2011 09:43, Sam <u2nsam at gmail.com> wrote:

> hello ,
>
> I am using,
>
> <extension name="global" continue="true">
>      <condition>
>        <action application="info"/>
>        <action application="db" data="insert/spymap/${caller_id_number}/${uuid}"/>
>
>        <action application="db" data="insert/last_dial/${caller_id_number}/${destination_number}"/>
>        <action application="db" data="insert/last_dial/global/${uuid}"/>
>
>      </condition>
>    </extension>
>
>    <extension name="eavesdrop">
>
>      <condition field="destination_number" expression="^88(.*)$|^\*0(.*)$">
>        <action application="answer"/>
>        <action application="eavesdrop" data="${db(select/spymap/$1)}"/>
>
>      </condition>
>    </extension>
>
>
>
> i am getting the below error:-
>
> 2011-01-20 14:54:01.581151 [NOTICE] mod_dptools.c:920 Channel
> [sofia/internal/7006 at 192.168.2.190] has been answered
> 2011-01-20 14:54:01.581151 [DEBUG] sofia.c:4646 Channel sofia/internal/
> 7006 at 192.168.2.190 entering state [completed][200]
> EXECUTE sofia/internal/7006 at 192.168.2.190 eavesdrop()
> 2011-01-20 14:54:01.582119 [ERR] mod_dptools.c:529 Usage: [all | <uuid>]
> 2011-01-20 14:54:01.582119 [NOTICE] switch_core_state_machine.c:189
> sofia/internal/7006 at 192.168.2.190 has executed the last dialplan
> instruction, hanging up.
> 2011-01-20 14:54:01.582119 [DEBUG] switch_channel.c:2535 (sofia/internal/
> 7006 at 192.168.2.190) Callstate Change ACTIVE -> HANGUP
> 2011-01-20 14:54:01.582119 [NOTICE] switch_core_state_machine.c:191 Hangup
> sofia/internal/7006 at 192.168.2.190 [CS_EXECUTE] [NORMAL_CLEARING]
>
>
> http://pastebin.freeswitch.org/15076
>
>
> Regds
> Sam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brent at overthewire.com.au  Thu Jan 20 13:16:28 2011
From: brent at overthewire.com.au (Brent Paddon)
Date: Thu, 20 Jan 2011 20:16:28 +1000
Subject: [Freeswitch-users] SIP device for attaching a speaker
In-Reply-To: <4D3807A9.8040103@gmx.net>
References: <4D3807A9.8040103@gmx.net>
Message-ID: <AANLkTi=Pa=w9WzCNkzAJCbyuqJi7b7omMs6AAcn6C-9D@mail.gmail.com>

Something like this maybe :

http://www.snom.com/en/products/sip-paging/snom-pa1/

On Thu, Jan 20, 2011 at 8:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Hello,
>
> I need to attach a speaker to our Freeeswitch. I may use a phone where I
> attach an amplifier via the headset plug and put the phone on auto
> answer (just as a workaround).
> But does anybody know of a SIP device which is just built for that, e.g.
> without keypad and designed for humid environments?
>
> Best regards
> Peter
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From david.ponzone at ipeva.fr  Thu Jan 20 13:20:20 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 20 Jan 2011 11:20:20 +0100
Subject: [Freeswitch-users] Fax gateway
In-Reply-To: <4D3806C4.4050009@gmail.com>
References: <4D374EF4.2090104@gmail.com>
	<DF86F0D8-4B36-431C-AD5E-BF1485630BED@ipeva.fr>
	<4D3806C4.4050009@gmail.com>
Message-ID: <4B80C2D4-2BF4-40A0-96BF-AC231CDEDB3D@ipeva.fr>

Sorry, that's not better.
Perhaps you should write it in your native language, in the simplest form, and then google-translate it.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 20/01/2011 ? 10:56, Antonio a ?crit :

> El 19/01/11 23:53, David Ponzone escribi?:
>> 
>> I dont know about others, but I was not able to understand what you're 
>> trying to accomplish.
> Hello
> 
> In the rush might not explain to me properly. I want my GW (FreeSwitch) 
> process the fax (mailserver + python script), so that the sending and 
> receiving signaling pass through the proxy, not directly send to be 
> configured in the FreeSwitch provider. thank you very much ;-)
> 
> Greetings
> 
> _______________________________________________
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From steveayre at gmail.com  Thu Jan 20 13:23:48 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 20 Jan 2011 10:23:48 +0000
Subject: [Freeswitch-users] call barging
In-Reply-To: <AANLkTimrbwUS7s1G+catXL-XVTr_pg9yOWorR04MNFY=@mail.gmail.com>
References: <AANLkTinMqT+Qs1guPphYWqN5okMAD9c1K9=Jey6uepRw@mail.gmail.com>
	<AANLkTimrbwUS7s1G+catXL-XVTr_pg9yOWorR04MNFY=@mail.gmail.com>
Message-ID: <AANLkTinNrHFKY22Amsv-mMw=wMcxmKheo-WxT8kYTrqC@mail.gmail.com>

Yes, that will work. You can set the group either within the extension or in
the user config.

For example:

 <extension name="bridge">
     <condition field="destination_number" expression="^9(\d+)$">
       <action application="set" data="eavesdrop_group=spy1"/>
       <action application="bridge" data="sofia/gateway/gw1/$1"/>
     </condition>
   </extension>
  <extension name="eavesdrop">
     <condition field="destination_number" expression="^666$">
       <action application="answer"/>
       <action application="set" data="eavesdrop_require_group=spy1"/>
       <action application="eavesdrop" data="all"/>
     </condition>
   </extension>


-Steve


On 20 January 2011 10:11, Sam <u2nsam at gmail.com> wrote:

> If i use
>
> <variable name="eavesdrop_group" value="spy1"/>
>
> at the user config, will the  dialplan barge the above extension where the
> statement is stated ?
>
> <extension name="eavesdrop">
>      <condition field="destination_number" expression="^779$">
>        <action application="answer"/>
>        <action application="set" data="eavesdrop_require_group=sales_call_eavesdrop"/>
>
>        <action application="eavesdrop" data="all"/>
>      </condition>
>    </extension>
>
>
> Regds
> Sam
>
>
>
>
>
>
>
>
>
>
>
>
> On Thu, Jan 20, 2011 at 3:13 PM, Sam <u2nsam at gmail.com> wrote:
>
>> hello ,
>>
>> I am using,
>>
>> <extension name="global" continue="true">
>>      <condition>
>>        <action application="info"/>
>>        <action application="db" data="insert/spymap/${caller_id_number}/${uuid}"/>
>>
>>
>>        <action application="db" data="insert/last_dial/${caller_id_number}/${destination_number}"/>
>>        <action application="db" data="insert/last_dial/global/${uuid}"/>
>>
>>
>>      </condition>
>>    </extension>
>>
>>    <extension name="eavesdrop">
>>
>>
>>      <condition field="destination_number" expression="^88(.*)$|^\*0(.*)$">
>>        <action application="answer"/>
>>        <action application="eavesdrop" data="${db(select/spymap/$1)}"/>
>>
>>
>>      </condition>
>>    </extension>
>>
>>
>>
>> i am getting the below error:-
>>
>> 2011-01-20 14:54:01.581151 [NOTICE] mod_dptools.c:920 Channel
>> [sofia/internal/7006 at 192.168.2.190] has been answered
>> 2011-01-20 14:54:01.581151 [DEBUG] sofia.c:4646 Channel sofia/internal/
>> 7006 at 192.168.2.190 entering state [completed][200]
>> EXECUTE sofia/internal/7006 at 192.168.2.190 eavesdrop()
>> 2011-01-20 14:54:01.582119 [ERR] mod_dptools.c:529 Usage: [all | <uuid>]
>> 2011-01-20 14:54:01.582119 [NOTICE] switch_core_state_machine.c:189
>> sofia/internal/7006 at 192.168.2.190 has executed the last dialplan
>> instruction, hanging up.
>> 2011-01-20 14:54:01.582119 [DEBUG] switch_channel.c:2535 (sofia/internal/
>> 7006 at 192.168.2.190) Callstate Change ACTIVE -> HANGUP
>> 2011-01-20 14:54:01.582119 [NOTICE] switch_core_state_machine.c:191 Hangup
>> sofia/internal/7006 at 192.168.2.190 [CS_EXECUTE] [NORMAL_CLEARING]
>>
>>
>> http://pastebin.freeswitch.org/15076
>>
>>
>> Regds
>> Sam
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From steveayre at gmail.com  Thu Jan 20 13:26:14 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 20 Jan 2011 10:26:14 +0000
Subject: [Freeswitch-users] call barging
In-Reply-To: <AANLkTinMqT+Qs1guPphYWqN5okMAD9c1K9=Jey6uepRw@mail.gmail.com>
References: <AANLkTinMqT+Qs1guPphYWqN5okMAD9c1K9=Jey6uepRw@mail.gmail.com>
Message-ID: <AANLkTinEzv4H8_6kRtsDzWgWSd5yH3PLNP=VYzmxqZTL@mail.gmail.com>

Reading this again, the reason it would have been failing would have been:

       <action application="eavesdrop" data="${db(select/spymap/$1)}"/>


If the mod_db select query returned no UUIDs (no-one has inserted their UUID
there), data would be "" and the eavesdrop command would fail with the error
you saw (noone to eavesdrop on). UUIDs are automatically removed from the
table when they hang up, and during a transfer too I believe.

-Steve


On 20 January 2011 09:43, Sam <u2nsam at gmail.com> wrote:

> hello ,
>
> I am using,
>
> <extension name="global" continue="true">
>      <condition>
>        <action application="info"/>
>        <action application="db" data="insert/spymap/${caller_id_number}/${uuid}"/>
>
>        <action application="db" data="insert/last_dial/${caller_id_number}/${destination_number}"/>
>        <action application="db" data="insert/last_dial/global/${uuid}"/>
>
>      </condition>
>    </extension>
>
>    <extension name="eavesdrop">
>
>      <condition field="destination_number" expression="^88(.*)$|^\*0(.*)$">
>        <action application="answer"/>
>        <action application="eavesdrop" data="${db(select/spymap/$1)}"/>
>
>      </condition>
>    </extension>
>
>
>
> i am getting the below error:-
>
> 2011-01-20 14:54:01.581151 [NOTICE] mod_dptools.c:920 Channel
> [sofia/internal/7006 at 192.168.2.190] has been answered
> 2011-01-20 14:54:01.581151 [DEBUG] sofia.c:4646 Channel sofia/internal/
> 7006 at 192.168.2.190 entering state [completed][200]
> EXECUTE sofia/internal/7006 at 192.168.2.190 eavesdrop()
> 2011-01-20 14:54:01.582119 [ERR] mod_dptools.c:529 Usage: [all | <uuid>]
> 2011-01-20 14:54:01.582119 [NOTICE] switch_core_state_machine.c:189
> sofia/internal/7006 at 192.168.2.190 has executed the last dialplan
> instruction, hanging up.
> 2011-01-20 14:54:01.582119 [DEBUG] switch_channel.c:2535 (sofia/internal/
> 7006 at 192.168.2.190) Callstate Change ACTIVE -> HANGUP
> 2011-01-20 14:54:01.582119 [NOTICE] switch_core_state_machine.c:191 Hangup
> sofia/internal/7006 at 192.168.2.190 [CS_EXECUTE] [NORMAL_CLEARING]
>
>
> http://pastebin.freeswitch.org/15076
>
>
> Regds
> Sam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From boris at tagnet.ru  Thu Jan 20 13:35:36 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 20 Jan 2011 15:35:36 +0500
Subject: [Freeswitch-users] PCMA stranges
Message-ID: <4D380FF8.7080705@tagnet.ru>

Hello!

     My network configuration is:
Softphone (eyeBeam 1.5) with only G711alaw enabled -> FreeSWITCH Version 
1.0.head (git-0cf1d54 2011-01-19 16-36-04 -0500) -> Cisco AS5350

proxy_media=true in profile.
So, when I do a test call, there is SDP but no RTP between FS and CISCO, 
so I can't hear voice. With G711ulaw there are no problems. What is 
wrong? Siptrace below:

    ------------------------------------------------------------------------
    INVITE sip:73435327569 at default SIP/2.0
    Via: SIP/2.0/UDP 
X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:50001 at X.X.29.123:21556>
    To: "73435327569"<sip:73435327569 at default>
    From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
    CSeq: 1 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, 
SUBSCRIBE, INFO
    Content-Type: application/sdp
    User-Agent: eyeBeam release 1102u stamp 52345
    Content-Length: 337

    v=0
    o=- 1 2 IN IP4 192.168.3.253
    s=CounterPath eyeBeam 1.5
    c=IN IP4 192.168.3.253
    t=0 0
    m=audio 40032 RTP/AVP 8 101
    a=alt:1 3 : XdMyLUqb Fs2PLxKV 192.168.3.253 40032
    a=alt:2 2 : 2qrc/CQ7 Z/9XFQZl 172.16.3.253 40032
    a=alt:3 1 : 8BOwI53P +ENGEr/U X.X.29.123 40032
    a=fmtp:101 0-15
    a=rtpmap:101 telephone-event/8000
    a=sendrecv
    ------------------------------------------------------------------------
send 385 bytes to udp/[X.X.29.123]:21556 at 10:30:31.313366:
    ------------------------------------------------------------------------
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 
X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
    From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
    To: "73435327569"<sip:73435327569 at default>
    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
    CSeq: 1 INVITE
    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19 
16-36-04 -0500
    Content-Length: 0

    ------------------------------------------------------------------------
2011-01-20 15:30:31.313448 [NOTICE] switch_channel.c:808 New Channel 
sofia/epbx/50001 at default [95698968-8cf9-40a2-91ad-1322fcfc76af]
2011-01-20 15:30:31.322338 [INFO] mod_dialplan_xml.c:331 Processing 
TAGNet <50001>->73435327569 in context public
2011-01-20 15:30:31.323362 [INFO] mod_dialplan_xml.c:331 Processing 
TAGNet <50001>->ext_translate_extsrc in context features
2011-01-20 15:30:31.324357 [NOTICE] switch_ivr.c:1606 Transfer 
sofia/epbx/50001 at default to XML[73435327569 at top.ctx]
2011-01-20 15:30:31.325352 [INFO] mod_dialplan_xml.c:331 Processing 
TAGNet <50001>->73435327569 in context top.ctx
2011-01-20 15:30:31.326344 [NOTICE] mod_dptools.c:1174 [top.ctx] - ext_local
2011-01-20 15:30:31.329360 [NOTICE] switch_channel.c:808 New Channel 
sofia/epbx/73435327569 at X.X.16.83:5060 [b9654c75-039a-4780-bc00-0fae65f92a9a]
send 1276 bytes to udp/[X.X.16.83]:5060 at 10:30:31.330623:
    ------------------------------------------------------------------------
    INVITE sip:73435327569 at X.X.16.83:5060 SIP/2.0
    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
    Max-Forwards: 68
    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
    To: <sip:73435327569 at X.X.16.83:5060>
    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
    CSeq: 7423651 INVITE
    Contact: <sip:mod_sofia at Y.Y.138.187:5060>
    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19 
16-36-04 -0500
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
    Supported: timer, precondition, path, replaces
    Allow-Events: talk, hold, presence, dialog, line-seize, call-info, 
sla, include-session-description, presence.winfo, message-summary, refer
    Content-Type: application/sdp
    Content-Disposition: session
    Content-Length: 341
    X-FS-Support: update_display
    Remote-Party-ID: "TAGNet" 
<sip:50001 at Y.Y.138.187>;party=calling;screen=yes;privacy=off

    v=0
    o=FreeSWITCH 1627720692 1627720693 IN IP4 Y.Y.138.187
    s=FreeSWITCH
    c=IN IP4 Y.Y.138.187
    t=0 0
    m=audio 23442 RTP/AVP 8 101
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=alt:1 3 : XdMyLUqb Fs2PLxKV 192.168.3.253 40032
    a=alt:2 2 : 2qrc/CQ7 Z/9XFQZl 172.16.3.253 40032
    a=alt:3 1 : 8BOwI53P +ENGEr/U X.X.29.123 40032
    ------------------------------------------------------------------------
recv 396 bytes from udp/[X.X.16.83]:5060 at 10:30:31.341106:
    ------------------------------------------------------------------------
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
    To: <sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
    Date: Thu, 20 Jan 2011 10:30:31 GMT
    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
    Server: Cisco-SIPGateway/IOS-12.x
    CSeq: 7423651 INVITE
    Allow-Events: telephone-event
    Content-Length: 0

    ------------------------------------------------------------------------
recv 814 bytes from udp/[X.X.16.83]:5060 at 10:30:32.401494:
    ------------------------------------------------------------------------
    SIP/2.0 183 Session Progress
    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
    To: <sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
    Date: Thu, 20 Jan 2011 10:30:31 GMT
    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
    Server: Cisco-SIPGateway/IOS-12.x
    CSeq: 7423651 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, 
SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
    Allow-Events: telephone-event
    Contact: <sip:73435327569 at X.X.16.83:5060>
    Content-Disposition: session;handling=required
    Content-Type: application/sdp
    Content-Length: 176

    v=0
    o=CiscoSystemsSIP-GW-UserAgent 4564 1578 IN IP4 X.X.16.83
    s=SIP Call
    c=IN IP4 X.X.16.83
    t=0 0
    m=audio 17770 RTP/AVP 8
    c=IN IP4 X.X.16.83
    a=rtpmap:8 PCMA/8000
    ------------------------------------------------------------------------
2011-01-20 15:30:32.401362 [INFO] sofia.c:729 
sofia/epbx/73435327569 at X.X.16.83:5060 Update Callee ID to "Outbound 
Call" <73435327569>
2011-01-20 15:30:32.409363 [NOTICE] sofia.c:4739 Pre-Answer 
sofia/epbx/73435327569 at X.X.16.83:5060!
2011-01-20 15:30:32.415412 [INFO] switch_ivr_originate.c:3345 Sending 
early media
2011-01-20 15:30:32.416367 [NOTICE] mod_sofia.c:2252 Pre-Answer 
sofia/epbx/50001 at default!
send 1145 bytes to udp/[X.X.29.123]:21556 at 10:30:32.418195:
    ------------------------------------------------------------------------
    SIP/2.0 183 Session Progress
    Via: SIP/2.0/UDP 
X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
    From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
    To: "73435327569" <sip:73435327569 at default>;tag=BScUp0QXHFBgB
    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
    CSeq: 1 INVITE
    Contact: <sip:73435327569 at Y.Y.138.187:5060;transport=udp>
    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19 
16-36-04 -0500
    Accept: application/sdp
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
    Supported: timer, precondition, path, replaces
    Allow-Events: talk, hold, presence, dialog, line-seize, call-info, 
sla, include-session-description, presence.winfo, message-summary, refer
    Content-Type: application/sdp
    Content-Disposition: session
    Content-Length: 181
    Remote-Party-ID: "73435327569" 
<sip:73435327569 at Y.Y.138.187>;party=calling;privacy=off;screen=no

    v=0
    o=FreeSWITCH 1627752513 1627752514 IN IP4 Y.Y.138.187
    s=FreeSWITCH
    c=IN IP4 Y.Y.138.187
    t=0 0
    m=audio 25806 RTP/AVP 8
    c=IN IP4 Y.Y.138.187
    a=rtpmap:8 PCMA/8000

    ------------------------------------------------------------------------
recv 365 bytes from udp/[X.X.29.123]:21556 at 10:30:39.196791:
    ------------------------------------------------------------------------
    CANCEL sip:73435327569 at default SIP/2.0
    Via: SIP/2.0/UDP 
X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
    To: "73435327569"<sip:73435327569 at default>
    From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
    CSeq: 1 CANCEL
    User-Agent: eyeBeam release 1102u stamp 52345
    Content-Length: 0

    ------------------------------------------------------------------------
send 320 bytes to udp/[X.X.29.123]:21556 at 10:30:39.196905:
    ------------------------------------------------------------------------
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 
X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
    From: "TAGNet" <sip:50001 at default>;tag=6c41dc5a
    To: "73435327569" <sip:73435327569 at default>;tag=BScUp0QXHFBgB
    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
    CSeq: 1 CANCEL
    Content-Length: 0

    ------------------------------------------------------------------------
send 715 bytes to udp/[X.X.29.123]:21556 at 10:30:39.197002:
    ------------------------------------------------------------------------
    SIP/2.0 487 Request Terminated
    Via: SIP/2.0/UDP 
X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
    From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
    To: "73435327569" <sip:73435327569 at default>;tag=BScUp0QXHFBgB
    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
    CSeq: 1 INVITE
    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19 
16-36-04 -0500
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
    Supported: timer, precondition, path, replaces
    Allow-Events: talk, hold, presence, dialog, line-seize, call-info, 
sla, include-session-description, presence.winfo, message-summary, refer
    Content-Length: 0

    ------------------------------------------------------------------------
2011-01-20 15:30:39.196838 [NOTICE] sofia.c:5286 Hangup 
sofia/epbx/50001 at default [CS_EXECUTE] [ORIGINATOR_CANCEL]
2011-01-20 15:30:39.196838 [NOTICE] switch_ivr_bridge.c:653 Hangup 
sofia/epbx/73435327569 at X.X.16.83:5060 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
recv 331 bytes from udp/[X.X.29.123]:21556 at 10:30:39.198834:
    ------------------------------------------------------------------------
    ACK sip:73435327569 at default SIP/2.0
    Via: SIP/2.0/UDP 
X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
    To: "73435327569" <sip:73435327569 at default>;tag=BScUp0QXHFBgB
    From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
    CSeq: 1 ACK
    Content-Length: 0

    ------------------------------------------------------------------------
2011-01-20 15:30:39.204838 [NOTICE] switch_core_session.c:1306 Session 
99 (sofia/epbx/50001 at default) Ended
2011-01-20 15:30:39.204838 [NOTICE] switch_core_session.c:1308 Close 
Channel sofia/epbx/50001 at default [CS_DESTROY]
send 372 bytes to udp/[X.X.16.83]:5060 at 10:30:39.212574:
    ------------------------------------------------------------------------
    CANCEL sip:73435327569 at X.X.16.83:5060 SIP/2.0
    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
    Max-Forwards: 68
    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
    To: <sip:73435327569 at X.X.16.83:5060>
    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
    CSeq: 7423651 CANCEL
    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
    Content-Length: 0

    ------------------------------------------------------------------------
2011-01-20 15:30:39.212840 [NOTICE] switch_core_session.c:1306 Session 
100 (sofia/epbx/73435327569 at X.X.16.83:5060) Ended
2011-01-20 15:30:39.212840 [NOTICE] switch_core_session.c:1308 Close 
Channel sofia/epbx/73435327569 at X.X.16.83:5060 [CS_DESTROY]
recv 311 bytes from udp/[X.X.16.83]:5060 at 10:30:39.216838:
    ------------------------------------------------------------------------
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
    To: <sip:73435327569 at X.X.16.83:5060>
    Date: Thu, 20 Jan 2011 10:30:39 GMT
    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
    Content-Length: 0
    CSeq: 7423651 CANCEL

    ------------------------------------------------------------------------
recv 407 bytes from udp/[X.X.16.83]:5060 at 10:30:39.219180:
    ------------------------------------------------------------------------
    SIP/2.0 487 Request Cancelled
    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
    To: <sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
    Date: Thu, 20 Jan 2011 10:30:39 GMT
    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
    Server: Cisco-SIPGateway/IOS-12.x
    CSeq: 7423651 INVITE
    Allow-Events: telephone-event
    Content-Length: 0

    ------------------------------------------------------------------------
send 334 bytes to udp/[X.X.16.83]:5060 at 10:30:39.219276:
    ------------------------------------------------------------------------
    ACK sip:73435327569 at X.X.16.83:5060 SIP/2.0
    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
    Max-Forwards: 68
    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
    To: <sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
    CSeq: 7423651 ACK
    Content-Length: 0

    ------------------------------------------------------------------------


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From david.ponzone at ipeva.fr  Thu Jan 20 14:30:25 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 20 Jan 2011 12:30:25 +0100
Subject: [Freeswitch-users] PCMA stranges
In-Reply-To: <4D380FF8.7080705@tagnet.ru>
References: <4D380FF8.7080705@tagnet.ru>
Message-ID: <E676EB80-7927-4061-BC2C-497DBA81D75E@ipeva.fr>

The Cisco never sends the 200/OK after the 183, so the call is not established.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 20/01/2011 ? 11:35, Boris Kovalenko a ?crit :

> Hello!
> 
>     My network configuration is:
> Softphone (eyeBeam 1.5) with only G711alaw enabled -> FreeSWITCH Version 
> 1.0.head (git-0cf1d54 2011-01-19 16-36-04 -0500) -> Cisco AS5350
> 
> proxy_media=true in profile.
> So, when I do a test call, there is SDP but no RTP between FS and CISCO, 
> so I can't hear voice. With G711ulaw there are no problems. What is 
> wrong? Siptrace below:
> 
>    ------------------------------------------------------------------------
>    INVITE sip:73435327569 at default SIP/2.0
>    Via: SIP/2.0/UDP 
> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
>    Max-Forwards: 70
>    Contact: <sip:50001 at X.X.29.123:21556>
>    To: "73435327569"<sip:73435327569 at default>
>    From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>    CSeq: 1 INVITE
>    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, 
> SUBSCRIBE, INFO
>    Content-Type: application/sdp
>    User-Agent: eyeBeam release 1102u stamp 52345
>    Content-Length: 337
> 
>    v=0
>    o=- 1 2 IN IP4 192.168.3.253
>    s=CounterPath eyeBeam 1.5
>    c=IN IP4 192.168.3.253
>    t=0 0
>    m=audio 40032 RTP/AVP 8 101
>    a=alt:1 3 : XdMyLUqb Fs2PLxKV 192.168.3.253 40032
>    a=alt:2 2 : 2qrc/CQ7 Z/9XFQZl 172.16.3.253 40032
>    a=alt:3 1 : 8BOwI53P +ENGEr/U X.X.29.123 40032
>    a=fmtp:101 0-15
>    a=rtpmap:101 telephone-event/8000
>    a=sendrecv
>    ------------------------------------------------------------------------
> send 385 bytes to udp/[X.X.29.123]:21556 at 10:30:31.313366:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP 
> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>    From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>    To: "73435327569"<sip:73435327569 at default>
>    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>    CSeq: 1 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19 
> 16-36-04 -0500
>    Content-Length: 0
> 
>    ------------------------------------------------------------------------
> 2011-01-20 15:30:31.313448 [NOTICE] switch_channel.c:808 New Channel 
> sofia/epbx/50001 at default [95698968-8cf9-40a2-91ad-1322fcfc76af]
> 2011-01-20 15:30:31.322338 [INFO] mod_dialplan_xml.c:331 Processing 
> TAGNet <50001>->73435327569 in context public
> 2011-01-20 15:30:31.323362 [INFO] mod_dialplan_xml.c:331 Processing 
> TAGNet <50001>->ext_translate_extsrc in context features
> 2011-01-20 15:30:31.324357 [NOTICE] switch_ivr.c:1606 Transfer 
> sofia/epbx/50001 at default to XML[73435327569 at top.ctx]
> 2011-01-20 15:30:31.325352 [INFO] mod_dialplan_xml.c:331 Processing 
> TAGNet <50001>->73435327569 in context top.ctx
> 2011-01-20 15:30:31.326344 [NOTICE] mod_dptools.c:1174 [top.ctx] - ext_local
> 2011-01-20 15:30:31.329360 [NOTICE] switch_channel.c:808 New Channel 
> sofia/epbx/73435327569 at X.X.16.83:5060 [b9654c75-039a-4780-bc00-0fae65f92a9a]
> send 1276 bytes to udp/[X.X.16.83]:5060 at 10:30:31.330623:
>    ------------------------------------------------------------------------
>    INVITE sip:73435327569 at X.X.16.83:5060 SIP/2.0
>    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>    Max-Forwards: 68
>    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>    To: <sip:73435327569 at X.X.16.83:5060>
>    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>    CSeq: 7423651 INVITE
>    Contact: <sip:mod_sofia at Y.Y.138.187:5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19 
> 16-36-04 -0500
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info, 
> sla, include-session-description, presence.winfo, message-summary, refer
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 341
>    X-FS-Support: update_display
>    Remote-Party-ID: "TAGNet" 
> <sip:50001 at Y.Y.138.187>;party=calling;screen=yes;privacy=off
> 
>    v=0
>    o=FreeSWITCH 1627720692 1627720693 IN IP4 Y.Y.138.187
>    s=FreeSWITCH
>    c=IN IP4 Y.Y.138.187
>    t=0 0
>    m=audio 23442 RTP/AVP 8 101
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-15
>    a=alt:1 3 : XdMyLUqb Fs2PLxKV 192.168.3.253 40032
>    a=alt:2 2 : 2qrc/CQ7 Z/9XFQZl 172.16.3.253 40032
>    a=alt:3 1 : 8BOwI53P +ENGEr/U X.X.29.123 40032
>    ------------------------------------------------------------------------
> recv 396 bytes from udp/[X.X.16.83]:5060 at 10:30:31.341106:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>    To: <sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>    Date: Thu, 20 Jan 2011 10:30:31 GMT
>    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>    Server: Cisco-SIPGateway/IOS-12.x
>    CSeq: 7423651 INVITE
>    Allow-Events: telephone-event
>    Content-Length: 0
> 
>    ------------------------------------------------------------------------
> recv 814 bytes from udp/[X.X.16.83]:5060 at 10:30:32.401494:
>    ------------------------------------------------------------------------
>    SIP/2.0 183 Session Progress
>    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>    To: <sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>    Date: Thu, 20 Jan 2011 10:30:31 GMT
>    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>    Server: Cisco-SIPGateway/IOS-12.x
>    CSeq: 7423651 INVITE
>    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, 
> SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
>    Allow-Events: telephone-event
>    Contact: <sip:73435327569 at X.X.16.83:5060>
>    Content-Disposition: session;handling=required
>    Content-Type: application/sdp
>    Content-Length: 176
> 
>    v=0
>    o=CiscoSystemsSIP-GW-UserAgent 4564 1578 IN IP4 X.X.16.83
>    s=SIP Call
>    c=IN IP4 X.X.16.83
>    t=0 0
>    m=audio 17770 RTP/AVP 8
>    c=IN IP4 X.X.16.83
>    a=rtpmap:8 PCMA/8000
>    ------------------------------------------------------------------------
> 2011-01-20 15:30:32.401362 [INFO] sofia.c:729 
> sofia/epbx/73435327569 at X.X.16.83:5060 Update Callee ID to "Outbound 
> Call" <73435327569>
> 2011-01-20 15:30:32.409363 [NOTICE] sofia.c:4739 Pre-Answer 
> sofia/epbx/73435327569 at X.X.16.83:5060!
> 2011-01-20 15:30:32.415412 [INFO] switch_ivr_originate.c:3345 Sending 
> early media
> 2011-01-20 15:30:32.416367 [NOTICE] mod_sofia.c:2252 Pre-Answer 
> sofia/epbx/50001 at default!
> send 1145 bytes to udp/[X.X.29.123]:21556 at 10:30:32.418195:
>    ------------------------------------------------------------------------
>    SIP/2.0 183 Session Progress
>    Via: SIP/2.0/UDP 
> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>    From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>    To: "73435327569" <sip:73435327569 at default>;tag=BScUp0QXHFBgB
>    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>    CSeq: 1 INVITE
>    Contact: <sip:73435327569 at Y.Y.138.187:5060;transport=udp>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19 
> 16-36-04 -0500
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info, 
> sla, include-session-description, presence.winfo, message-summary, refer
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 181
>    Remote-Party-ID: "73435327569" 
> <sip:73435327569 at Y.Y.138.187>;party=calling;privacy=off;screen=no
> 
>    v=0
>    o=FreeSWITCH 1627752513 1627752514 IN IP4 Y.Y.138.187
>    s=FreeSWITCH
>    c=IN IP4 Y.Y.138.187
>    t=0 0
>    m=audio 25806 RTP/AVP 8
>    c=IN IP4 Y.Y.138.187
>    a=rtpmap:8 PCMA/8000
> 
>    ------------------------------------------------------------------------
> recv 365 bytes from udp/[X.X.29.123]:21556 at 10:30:39.196791:
>    ------------------------------------------------------------------------
>    CANCEL sip:73435327569 at default SIP/2.0
>    Via: SIP/2.0/UDP 
> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
>    To: "73435327569"<sip:73435327569 at default>
>    From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>    CSeq: 1 CANCEL
>    User-Agent: eyeBeam release 1102u stamp 52345
>    Content-Length: 0
> 
>    ------------------------------------------------------------------------
> send 320 bytes to udp/[X.X.29.123]:21556 at 10:30:39.196905:
>    ------------------------------------------------------------------------
>    SIP/2.0 200 OK
>    Via: SIP/2.0/UDP 
> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>    From: "TAGNet" <sip:50001 at default>;tag=6c41dc5a
>    To: "73435327569" <sip:73435327569 at default>;tag=BScUp0QXHFBgB
>    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>    CSeq: 1 CANCEL
>    Content-Length: 0
> 
>    ------------------------------------------------------------------------
> send 715 bytes to udp/[X.X.29.123]:21556 at 10:30:39.197002:
>    ------------------------------------------------------------------------
>    SIP/2.0 487 Request Terminated
>    Via: SIP/2.0/UDP 
> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>    From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>    To: "73435327569" <sip:73435327569 at default>;tag=BScUp0QXHFBgB
>    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>    CSeq: 1 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19 
> 16-36-04 -0500
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info, 
> sla, include-session-description, presence.winfo, message-summary, refer
>    Content-Length: 0
> 
>    ------------------------------------------------------------------------
> 2011-01-20 15:30:39.196838 [NOTICE] sofia.c:5286 Hangup 
> sofia/epbx/50001 at default [CS_EXECUTE] [ORIGINATOR_CANCEL]
> 2011-01-20 15:30:39.196838 [NOTICE] switch_ivr_bridge.c:653 Hangup 
> sofia/epbx/73435327569 at X.X.16.83:5060 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> recv 331 bytes from udp/[X.X.29.123]:21556 at 10:30:39.198834:
>    ------------------------------------------------------------------------
>    ACK sip:73435327569 at default SIP/2.0
>    Via: SIP/2.0/UDP 
> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
>    To: "73435327569" <sip:73435327569 at default>;tag=BScUp0QXHFBgB
>    From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>    CSeq: 1 ACK
>    Content-Length: 0
> 
>    ------------------------------------------------------------------------
> 2011-01-20 15:30:39.204838 [NOTICE] switch_core_session.c:1306 Session 
> 99 (sofia/epbx/50001 at default) Ended
> 2011-01-20 15:30:39.204838 [NOTICE] switch_core_session.c:1308 Close 
> Channel sofia/epbx/50001 at default [CS_DESTROY]
> send 372 bytes to udp/[X.X.16.83]:5060 at 10:30:39.212574:
>    ------------------------------------------------------------------------
>    CANCEL sip:73435327569 at X.X.16.83:5060 SIP/2.0
>    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>    Max-Forwards: 68
>    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>    To: <sip:73435327569 at X.X.16.83:5060>
>    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>    CSeq: 7423651 CANCEL
>    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>    Content-Length: 0
> 
>    ------------------------------------------------------------------------
> 2011-01-20 15:30:39.212840 [NOTICE] switch_core_session.c:1306 Session 
> 100 (sofia/epbx/73435327569 at X.X.16.83:5060) Ended
> 2011-01-20 15:30:39.212840 [NOTICE] switch_core_session.c:1308 Close 
> Channel sofia/epbx/73435327569 at X.X.16.83:5060 [CS_DESTROY]
> recv 311 bytes from udp/[X.X.16.83]:5060 at 10:30:39.216838:
>    ------------------------------------------------------------------------
>    SIP/2.0 200 OK
>    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>    To: <sip:73435327569 at X.X.16.83:5060>
>    Date: Thu, 20 Jan 2011 10:30:39 GMT
>    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>    Content-Length: 0
>    CSeq: 7423651 CANCEL
> 
>    ------------------------------------------------------------------------
> recv 407 bytes from udp/[X.X.16.83]:5060 at 10:30:39.219180:
>    ------------------------------------------------------------------------
>    SIP/2.0 487 Request Cancelled
>    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>    To: <sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>    Date: Thu, 20 Jan 2011 10:30:39 GMT
>    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>    Server: Cisco-SIPGateway/IOS-12.x
>    CSeq: 7423651 INVITE
>    Allow-Events: telephone-event
>    Content-Length: 0
> 
>    ------------------------------------------------------------------------
> send 334 bytes to udp/[X.X.16.83]:5060 at 10:30:39.219276:
>    ------------------------------------------------------------------------
>    ACK sip:73435327569 at X.X.16.83:5060 SIP/2.0
>    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>    Max-Forwards: 68
>    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>    To: <sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>    CSeq: 7423651 ACK
>    Content-Length: 0
> 
>    ------------------------------------------------------------------------
> 
> 
> -- 
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   (3435) 494991
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
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From boris at tagnet.ru  Thu Jan 20 14:35:59 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 20 Jan 2011 16:35:59 +0500
Subject: [Freeswitch-users] PCMA stranges
In-Reply-To: <E676EB80-7927-4061-BC2C-497DBA81D75E@ipeva.fr>
References: <4D380FF8.7080705@tagnet.ru>
	<E676EB80-7927-4061-BC2C-497DBA81D75E@ipeva.fr>
Message-ID: <4D381E1F.7040202@tagnet.ru>

Hello!

     David, may You explain why this problem is with G711A only? When I 
use G711U there are no problems and call is established and I may talk.
> The Cisco never sends the 200/OK after the 183, so the call is not 
> established.
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service ClientIPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 20/01/2011 ? 11:35, Boris Kovalenko a ?crit :
>
>> Hello!
>>
>>     My network configuration is:
>> Softphone (eyeBeam 1.5) with only G711alaw enabled -> FreeSWITCH Version
>> 1.0.head (git-0cf1d54 2011-01-19 16-36-04 -0500) -> Cisco AS5350
>>
>> proxy_media=true in profile.
>> So, when I do a test call, there is SDP but no RTP between FS and CISCO,
>> so I can't hear voice. With G711ulaw there are no problems. What is
>> wrong? Siptrace below:
>>
>>    ------------------------------------------------------------------------
>>    INVITE sip:73435327569 at default SIP/2.0
>>    Via: SIP/2.0/UDP
>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
>>    Max-Forwards: 70
>>    Contact: <sip:50001 at X.X.29.123:21556>
>>    To: "73435327569"<sip:73435327569 at default>
>>    From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>    CSeq: 1 INVITE
>>    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
>> SUBSCRIBE, INFO
>>    Content-Type: application/sdp
>>    User-Agent: eyeBeam release 1102u stamp 52345
>>    Content-Length: 337
>>
>>    v=0
>>    o=- 1 2 IN IP4 192.168.3.253
>>    s=CounterPath eyeBeam 1.5
>>    c=IN IP4 192.168.3.253
>>    t=0 0
>>    m=audio 40032 RTP/AVP 8 101
>>    a=alt:1 3 : XdMyLUqb Fs2PLxKV 192.168.3.253 40032
>>    a=alt:2 2 : 2qrc/CQ7 Z/9XFQZl 172.16.3.253 40032
>>    a=alt:3 1 : 8BOwI53P +ENGEr/U X.X.29.123 40032
>>    a=fmtp:101 0-15
>>    a=rtpmap:101 telephone-event/8000
>>    a=sendrecv
>>    ------------------------------------------------------------------------
>> send 385 bytes to udp/[X.X.29.123]:21556 at 10:30:31.313366:
>>    ------------------------------------------------------------------------
>>    SIP/2.0 100 Trying
>>    Via: SIP/2.0/UDP
>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>>    From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>    To: "73435327569"<sip:73435327569 at default>
>>    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>    CSeq: 1 INVITE
>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
>> 16-36-04 -0500
>>    Content-Length: 0
>>
>>    ------------------------------------------------------------------------
>> 2011-01-20 15:30:31.313448 [NOTICE] switch_channel.c:808 New Channel
>> sofia/epbx/50001 at default [95698968-8cf9-40a2-91ad-1322fcfc76af]
>> 2011-01-20 15:30:31.322338 [INFO] mod_dialplan_xml.c:331 Processing
>> TAGNet <50001>->73435327569 in context public
>> 2011-01-20 15:30:31.323362 [INFO] mod_dialplan_xml.c:331 Processing
>> TAGNet <50001>->ext_translate_extsrc in context features
>> 2011-01-20 15:30:31.324357 [NOTICE] switch_ivr.c:1606 Transfer
>> sofia/epbx/50001 at default to XML[73435327569 at top.ctx]
>> 2011-01-20 15:30:31.325352 [INFO] mod_dialplan_xml.c:331 Processing
>> TAGNet <50001>->73435327569 in context top.ctx
>> 2011-01-20 15:30:31.326344 [NOTICE] mod_dptools.c:1174 [top.ctx] - 
>> ext_local
>> 2011-01-20 15:30:31.329360 [NOTICE] switch_channel.c:808 New Channel
>> sofia/epbx/73435327569 at X.X.16.83:5060 
>> [b9654c75-039a-4780-bc00-0fae65f92a9a]
>> send 1276 bytes to udp/[X.X.16.83]:5060 at 10:30:31.330623:
>>    ------------------------------------------------------------------------
>>    INVITE sip:73435327569 at X.X.16.83:5060 SIP/2.0
>>    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>    Max-Forwards: 68
>>    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>    To: <sip:73435327569 at X.X.16.83:5060>
>>    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>    CSeq: 7423651 INVITE
>>    Contact: <sip:mod_sofia at Y.Y.138.187:5060>
>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
>> 16-36-04 -0500
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>    Supported: timer, precondition, path, replaces
>>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>> sla, include-session-description, presence.winfo, message-summary, refer
>>    Content-Type: application/sdp
>>    Content-Disposition: session
>>    Content-Length: 341
>>    X-FS-Support: update_display
>>    Remote-Party-ID: "TAGNet"
>> <sip:50001 at Y.Y.138.187>;party=calling;screen=yes;privacy=off
>>
>>    v=0
>>    o=FreeSWITCH 1627720692 1627720693 IN IP4 Y.Y.138.187
>>    s=FreeSWITCH
>>    c=IN IP4 Y.Y.138.187
>>    t=0 0
>>    m=audio 23442 RTP/AVP 8 101
>>    a=rtpmap:101 telephone-event/8000
>>    a=fmtp:101 0-15
>>    a=alt:1 3 : XdMyLUqb Fs2PLxKV 192.168.3.253 40032
>>    a=alt:2 2 : 2qrc/CQ7 Z/9XFQZl 172.16.3.253 40032
>>    a=alt:3 1 : 8BOwI53P +ENGEr/U X.X.29.123 40032
>>    ------------------------------------------------------------------------
>> recv 396 bytes from udp/[X.X.16.83]:5060 at 10:30:31.341106:
>>    ------------------------------------------------------------------------
>>    SIP/2.0 100 Trying
>>    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>    To: <sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>>    Date: Thu, 20 Jan 2011 10:30:31 GMT
>>    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>    Server: Cisco-SIPGateway/IOS-12.x
>>    CSeq: 7423651 INVITE
>>    Allow-Events: telephone-event
>>    Content-Length: 0
>>
>>    ------------------------------------------------------------------------
>> recv 814 bytes from udp/[X.X.16.83]:5060 at 10:30:32.401494:
>>    ------------------------------------------------------------------------
>>    SIP/2.0 183 Session Progress
>>    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>    To: <sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>>    Date: Thu, 20 Jan 2011 10:30:31 GMT
>>    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>    Server: Cisco-SIPGateway/IOS-12.x
>>    CSeq: 7423651 INVITE
>>    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
>> SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
>>    Allow-Events: telephone-event
>>    Contact: <sip:73435327569 at X.X.16.83:5060>
>>    Content-Disposition: session;handling=required
>>    Content-Type: application/sdp
>>    Content-Length: 176
>>
>>    v=0
>>    o=CiscoSystemsSIP-GW-UserAgent 4564 1578 IN IP4 X.X.16.83
>>    s=SIP Call
>>    c=IN IP4 X.X.16.83
>>    t=0 0
>>    m=audio 17770 RTP/AVP 8
>>    c=IN IP4 X.X.16.83
>>    a=rtpmap:8 PCMA/8000
>>    ------------------------------------------------------------------------
>> 2011-01-20 15:30:32.401362 [INFO] sofia.c:729
>> sofia/epbx/73435327569 at X.X.16.83:5060 Update Callee ID to "Outbound
>> Call" <73435327569>
>> 2011-01-20 15:30:32.409363 [NOTICE] sofia.c:4739 Pre-Answer
>> sofia/epbx/73435327569 at X.X.16.83:5060!
>> 2011-01-20 15:30:32.415412 [INFO] switch_ivr_originate.c:3345 Sending
>> early media
>> 2011-01-20 15:30:32.416367 [NOTICE] mod_sofia.c:2252 Pre-Answer
>> sofia/epbx/50001 at default!
>> send 1145 bytes to udp/[X.X.29.123]:21556 at 10:30:32.418195:
>>    ------------------------------------------------------------------------
>>    SIP/2.0 183 Session Progress
>>    Via: SIP/2.0/UDP
>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>>    From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>    To: "73435327569" <sip:73435327569 at default>;tag=BScUp0QXHFBgB
>>    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>    CSeq: 1 INVITE
>>    Contact: <sip:73435327569 at Y.Y.138.187:5060;transport=udp>
>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
>> 16-36-04 -0500
>>    Accept: application/sdp
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>    Supported: timer, precondition, path, replaces
>>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>> sla, include-session-description, presence.winfo, message-summary, refer
>>    Content-Type: application/sdp
>>    Content-Disposition: session
>>    Content-Length: 181
>>    Remote-Party-ID: "73435327569"
>> <sip:73435327569 at Y.Y.138.187>;party=calling;privacy=off;screen=no
>>
>>    v=0
>>    o=FreeSWITCH 1627752513 1627752514 IN IP4 Y.Y.138.187
>>    s=FreeSWITCH
>>    c=IN IP4 Y.Y.138.187
>>    t=0 0
>>    m=audio 25806 RTP/AVP 8
>>    c=IN IP4 Y.Y.138.187
>>    a=rtpmap:8 PCMA/8000
>>
>>    ------------------------------------------------------------------------
>> recv 365 bytes from udp/[X.X.29.123]:21556 at 10:30:39.196791:
>>    ------------------------------------------------------------------------
>>    CANCEL sip:73435327569 at default SIP/2.0
>>    Via: SIP/2.0/UDP
>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
>>    To: "73435327569"<sip:73435327569 at default>
>>    From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>    CSeq: 1 CANCEL
>>    User-Agent: eyeBeam release 1102u stamp 52345
>>    Content-Length: 0
>>
>>    ------------------------------------------------------------------------
>> send 320 bytes to udp/[X.X.29.123]:21556 at 10:30:39.196905:
>>    ------------------------------------------------------------------------
>>    SIP/2.0 200 OK
>>    Via: SIP/2.0/UDP
>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>>    From: "TAGNet" <sip:50001 at default>;tag=6c41dc5a
>>    To: "73435327569" <sip:73435327569 at default>;tag=BScUp0QXHFBgB
>>    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>    CSeq: 1 CANCEL
>>    Content-Length: 0
>>
>>    ------------------------------------------------------------------------
>> send 715 bytes to udp/[X.X.29.123]:21556 at 10:30:39.197002:
>>    ------------------------------------------------------------------------
>>    SIP/2.0 487 Request Terminated
>>    Via: SIP/2.0/UDP
>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>>    From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>    To: "73435327569" <sip:73435327569 at default>;tag=BScUp0QXHFBgB
>>    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>    CSeq: 1 INVITE
>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
>> 16-36-04 -0500
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>    Supported: timer, precondition, path, replaces
>>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>> sla, include-session-description, presence.winfo, message-summary, refer
>>    Content-Length: 0
>>
>>    ------------------------------------------------------------------------
>> 2011-01-20 15:30:39.196838 [NOTICE] sofia.c:5286 Hangup
>> sofia/epbx/50001 at default [CS_EXECUTE] [ORIGINATOR_CANCEL]
>> 2011-01-20 15:30:39.196838 [NOTICE] switch_ivr_bridge.c:653 Hangup
>> sofia/epbx/73435327569 at X.X.16.83:5060 [CS_EXCHANGE_MEDIA] 
>> [NORMAL_CLEARING]
>> recv 331 bytes from udp/[X.X.29.123]:21556 at 10:30:39.198834:
>>    ------------------------------------------------------------------------
>>    ACK sip:73435327569 at default SIP/2.0
>>    Via: SIP/2.0/UDP
>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
>>    To: "73435327569" <sip:73435327569 at default>;tag=BScUp0QXHFBgB
>>    From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>    CSeq: 1 ACK
>>    Content-Length: 0
>>
>>    ------------------------------------------------------------------------
>> 2011-01-20 15:30:39.204838 [NOTICE] switch_core_session.c:1306 Session
>> 99 (sofia/epbx/50001 at default) Ended
>> 2011-01-20 15:30:39.204838 [NOTICE] switch_core_session.c:1308 Close
>> Channel sofia/epbx/50001 at default [CS_DESTROY]
>> send 372 bytes to udp/[X.X.16.83]:5060 at 10:30:39.212574:
>>    ------------------------------------------------------------------------
>>    CANCEL sip:73435327569 at X.X.16.83:5060 SIP/2.0
>>    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>    Max-Forwards: 68
>>    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>    To: <sip:73435327569 at X.X.16.83:5060>
>>    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>    CSeq: 7423651 CANCEL
>>    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>>    Content-Length: 0
>>
>>    ------------------------------------------------------------------------
>> 2011-01-20 15:30:39.212840 [NOTICE] switch_core_session.c:1306 Session
>> 100 (sofia/epbx/73435327569 at X.X.16.83:5060) Ended
>> 2011-01-20 15:30:39.212840 [NOTICE] switch_core_session.c:1308 Close
>> Channel sofia/epbx/73435327569 at X.X.16.83:5060 [CS_DESTROY]
>> recv 311 bytes from udp/[X.X.16.83]:5060 at 10:30:39.216838:
>>    ------------------------------------------------------------------------
>>    SIP/2.0 200 OK
>>    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>    To: <sip:73435327569 at X.X.16.83:5060>
>>    Date: Thu, 20 Jan 2011 10:30:39 GMT
>>    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>    Content-Length: 0
>>    CSeq: 7423651 CANCEL
>>
>>    ------------------------------------------------------------------------
>> recv 407 bytes from udp/[X.X.16.83]:5060 at 10:30:39.219180:
>>    ------------------------------------------------------------------------
>>    SIP/2.0 487 Request Cancelled
>>    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>    To: <sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>>    Date: Thu, 20 Jan 2011 10:30:39 GMT
>>    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>    Server: Cisco-SIPGateway/IOS-12.x
>>    CSeq: 7423651 INVITE
>>    Allow-Events: telephone-event
>>    Content-Length: 0
>>
>>    ------------------------------------------------------------------------
>> send 334 bytes to udp/[X.X.16.83]:5060 at 10:30:39.219276:
>>    ------------------------------------------------------------------------
>>    ACK sip:73435327569 at X.X.16.83:5060 SIP/2.0
>>    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>    Max-Forwards: 68
>>    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>    To: <sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>>    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>    CSeq: 7423651 ACK
>>    Content-Length: 0
>>
>>    ------------------------------------------------------------------------
>>
>>
>> -- 
>> ? ?????????,
>>   ????? ?????????
>>   ??? "??????"
>>   (3435) 494991
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991


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From chris at cloudtel.com  Thu Jan 20 15:03:48 2011
From: chris at cloudtel.com (Chris Burns)
Date: Thu, 20 Jan 2011 07:03:48 -0500
Subject: [Freeswitch-users] Sharing a VM Mailbox?
In-Reply-To: <AANLkTikvgVSjNnbzzxYbNfc+iBt8ROQL52E3JYyZW-nj@mail.gmail.com>
References: <AANLkTi=1OMJ93+vrF0bQvytJAD1TMTRs0qBm-obOdDGV@mail.gmail.com>
	<AANLkTi=0Q=5ky+FXpNRSk-cHTqUj8D=k6uu4s1Rmf8Ff@mail.gmail.com>
	<AANLkTikvgVSjNnbzzxYbNfc+iBt8ROQL52E3JYyZW-nj@mail.gmail.com>
Message-ID: <AANLkTinFGS1t93Y2AtVNAPxZuvnpke7duViHKSxvO0g=@mail.gmail.com>

Your SIP clients need options for that. It's the client who needs to
SUBSCRIBE to a different contact than they REGISTER for MWI to work like
that.

Otherwise you need to code some C or post a bounty for some MWI features to
be built into mod_sofia. Currently the switch sends MWI only to clients
subbed (or regged) to that user and domain or presence_host.

On Wed, Jan 19, 2011 at 4:40 AM, Avi Marcus <avi at avimarcus.net> wrote:

> I suppose that's possible, but what about the MWI subscribe?
> On Jan 19, 2011 9:00 AM, "Chris Burns" <chris at cloudtel.com> wrote:
> > http://wiki.freeswitch.org/wiki/Mod_voicemail#Check_Voice_Mail
> >
> > Following that example, you could add ${mailbox} to data when you call
> > voicemail app in the dialplan (eg. "check default ${domain_name}
> > ${mailbox}"). You are setting the mailbox variable in the directory but
> you
> > need a bit of "funny stuff in the dialplan" to use it :)
> >
> > On Tue, Jan 18, 2011 at 6:17 PM, Avi Marcus <avi at avimarcus.net> wrote:
> >
> >> How do I set users to share a VM box?
> >> I'm referring mostly to 1) receiving MWI for the other account, and 2)
> >> dialing *98 to get the shared messages, without funny stuff in the
> dialplan.
> >> (Or a not too convoluted workaround...)
> >>
> >> I see <user id="1010" *mailbox*="1010"> on
> >> http://wiki.freeswitch.org/wiki/Variable_voicemail_domain
> >> but that didn't seem to work.
> >>
> >> Suggestions?
> >>
> >> Thanks,
> >> Avi Marcus
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lists at telefaks.de  Thu Jan 20 15:38:59 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Thu, 20 Jan 2011 13:38:59 +0100
Subject: [Freeswitch-users] SIP device for attaching a speaker
In-Reply-To: <AANLkTi=Pa=w9WzCNkzAJCbyuqJi7b7omMs6AAcn6C-9D@mail.gmail.com>
References: <4D3807A9.8040103@gmx.net>
	<AANLkTi=Pa=w9WzCNkzAJCbyuqJi7b7omMs6AAcn6C-9D@mail.gmail.com>
Message-ID: <4D382CE3.3040905@telefaks.de>

Thanks,

that's exactly what I looked for. And it's even affordable.

Best regards
Peter

Brent Paddon schrieb:
> Something like this maybe :
>
> http://www.snom.com/en/products/sip-paging/snom-pa1/
>
> On Thu, Jan 20, 2011 at 8:00 PM, Peter P GMX <Prometheus001 at gmx.net
> <mailto:Prometheus001 at gmx.net>> wrote:
>
>     Hello,
>
>     I need to attach a speaker to our Freeeswitch. I may use a phone
>     where I
>     attach an amplifier via the headset plug and put the phone on auto
>     answer (just as a workaround).
>     But does anybody know of a SIP device which is just built for
>     that, e.g.
>     without keypad and designed for humid environments?
>
>     Best regards
>     Peter
>
>
>
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> --
> Brent Paddon
>
> Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au
> <mailto:brent.paddon at overthewire.com.au> | www.overthewire.com.au
> <http://www.overthewire.com.au>
> Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de




From tculjaga at gmail.com  Thu Jan 20 16:10:21 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 20 Jan 2011 14:10:21 +0100
Subject: [Freeswitch-users] Fax gateway
In-Reply-To: <4B80C2D4-2BF4-40A0-96BF-AC231CDEDB3D@ipeva.fr>
References: <4D374EF4.2090104@gmail.com>
	<DF86F0D8-4B36-431C-AD5E-BF1485630BED@ipeva.fr>
	<4D3806C4.4050009@gmail.com>
	<4B80C2D4-2BF4-40A0-96BF-AC231CDEDB3D@ipeva.fr>
Message-ID: <AANLkTikgcEbUojNS9_joPUAc=vGyMk42oxQ=10=TPUrT@mail.gmail.com>

hmmm, I will try too guess :P

you have your customers and you are assigning DID to them. You want to
charge the customers for originated fax calls (mail2fax) but you don't know
how?
All accounting is done on proxy level ... and the outgoing fax calls need to
use the appropriate DID so your billing can charge the customers accordingly
?


Did i get it right?



On Thu, Jan 20, 2011 at 11:20 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Sorry, that's not better.
> Perhaps you should write it in your native language, in the simplest form,
> and then google-translate it.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 20/01/2011 ? 10:56, Antonio a ?crit :
>
> El 19/01/11 23:53, David Ponzone escribi?:
>
>
> I dont know about others, but I was not able to understand what you're
>
> trying to accomplish.
>
> Hello
>
> In the rush might not explain to me properly. I want my GW (FreeSwitch)
> process the fax (mailserver + python script), so that the sending and
> receiving signaling pass through the proxy, not directly send to be
> configured in the FreeSwitch provider. thank you very much ;-)
>
> Greetings
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From thisjoy0528 at gmail.com  Thu Jan 20 16:22:19 2011
From: thisjoy0528 at gmail.com (joy this)
Date: Thu, 20 Jan 2011 21:22:19 +0800
Subject: [Freeswitch-users]  Question about bind_digit_action
Message-ID: <AANLkTinRz4sOoxOOf_D1a+XiFSFe33o_0obTjHjgxwS-@mail.gmail.com>

Dear all:



        I want to use ?bind_digit_action?, but it did not work. I can only
use ?bind_meta_app?, but the function is limited a lot. This is my diaplan:



<extension name="test">

        <condition field="destination_number" expression="^#123$">

                        <action application="bind_digit_action"
data="myrealm,500,exec:playback,ivr/ivr-welcome_to_freeswitch.wav"/>

                        <action application="bind_digit_action"
data="test1,456,exec:playback,ivr/ivr-welcome_to_freeswitch.wav"/>

                        <action application="bind_digit_action"
data="test1,##,exec:execute_extension,mix_welcome_to_freeswitch"/>

                        <action application="digit_action_set_realm"
data="test1"/>

        </condition>

</extension>



        But it shows:

2011-01-20 21:03:16.359375 [ERR] switch_core_session.c:1791 Invalid
Application bind_digit_action



        Can anyone please help me? My FS version is ?FreeSWITCH Version
1.0.head (git-)?.



Sincerely yours,

Thisjoy.
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From vermeulen.deon at gmail.com  Thu Jan 20 16:36:10 2011
From: vermeulen.deon at gmail.com (Deon Vermeulen)
Date: Thu, 20 Jan 2011 15:36:10 +0200
Subject: [Freeswitch-users] GSM/PSTN Gateways
Message-ID: <0F8036C9-D129-419D-AAA0-C8EA3856D7DD@gmail.com>

Hi List

I've been searching the Net a lot the past couple of days looking for relatively good priced Internal and External GSM and PSTN Gateways.

I did come up with couple devices, but they still seem a bit expensive.

One device that did draw my attention though is the DuMV at PCI GSM Card. http://www.portech.com.tw/p3-product1_1.asp?Pid=15
Do no if this will work in FreeSwitch?

Would really appreciate suggestions on products used.


Thank you very much

Regards

Deon

From Nabble at slickdeals.endjunk.com  Thu Jan 20 16:49:23 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 20 Jan 2011 05:49:23 -0800 (PST)
Subject: [Freeswitch-users] call barging
In-Reply-To: <AANLkTinNrHFKY22Amsv-mMw=wMcxmKheo-WxT8kYTrqC@mail.gmail.com>
References: <AANLkTinMqT+Qs1guPphYWqN5okMAD9c1K9=Jey6uepRw@mail.gmail.com>
	<AANLkTimrbwUS7s1G+catXL-XVTr_pg9yOWorR04MNFY=@mail.gmail.com>
	<AANLkTinNrHFKY22Amsv-mMw=wMcxmKheo-WxT8kYTrqC@mail.gmail.com>
Message-ID: <1295531363114-5943853.post@n2.nabble.com>



Steven Ayre wrote:
> 
> Yes, that will work. You can set the group either within the extension or
> in
> the user config.
> 
> For example:
> 
>  <extension name="bridge">
>      <condition field="destination_number" expression="^9(\d+)$">
>        <action application="set" data="eavesdrop_group=spy1"/>
>        <action application="bridge" data="sofia/gateway/gw1/$1"/>
>      </condition>
>    </extension>
>   <extension name="eavesdrop">
>      <condition field="destination_number" expression="^666$">
>        <action application="answer"/>
>        <action application="set" data="eavesdrop_require_group=spy1"/>
>        <action application="eavesdrop" data="all"/>
>      </condition>
>    </extension>
> 
> 
> -Steve
Steve, I tried your above suggestion on my FreeSWITCH Version 1.0.head
(git-cf253c3 2011-01-11 15-31-40 -0600) with mod_spy loaded and the barge in
works, but only void (no conversation captured).

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/call-barging-tp5943099p5943853.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From boris at tagnet.ru  Thu Jan 20 17:07:18 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 20 Jan 2011 19:07:18 +0500
Subject: [Freeswitch-users] PCMA stranges
In-Reply-To: <4D381E1F.7040202@tagnet.ru>
References: <4D380FF8.7080705@tagnet.ru>	<E676EB80-7927-4061-BC2C-497DBA81D75E@ipeva.fr>
	<4D381E1F.7040202@tagnet.ru>
Message-ID: <4D384196.105@tagnet.ru>

Hello!

     I found that problem is somewhere inside freeswitch. Direct call 
between SoftPhone and Cisco is working fine.
> Hello!
>
>     David, may You explain why this problem is with G711A only? When I 
> use G711U there are no problems and call is established and I may talk.
>> The Cisco never sends the 200/OK after the 183, so the call is not 
>> established.
>>
>> David Ponzone Direction Technique
>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service ClientIPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
>> <BLOCKED::http://www.ipeva-studio.com/>
>>
>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>> utilisation ou diffusion non autoris?e est interdite. Tout message 
>> ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline toute 
>> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou 
>> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le 
>> d?truire imm?diatement et d'avertir l'exp?diteur./
>> /
>> /
>>
>>
>>
>> Le 20/01/2011 ? 11:35, Boris Kovalenko a ?crit :
>>
>>> Hello!
>>>
>>>     My network configuration is:
>>> Softphone (eyeBeam 1.5) with only G711alaw enabled -> FreeSWITCH 
>>> Version
>>> 1.0.head (git-0cf1d54 2011-01-19 16-36-04 -0500) -> Cisco AS5350
>>>
>>> proxy_media=true in profile.
>>> So, when I do a test call, there is SDP but no RTP between FS and 
>>> CISCO,
>>> so I can't hear voice. With G711ulaw there are no problems. What is
>>> wrong? Siptrace below:
>>>
>>>    ------------------------------------------------------------------------
>>>    INVITE sip:73435327569 at default SIP/2.0
>>>    Via: SIP/2.0/UDP
>>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
>>>    Max-Forwards: 70
>>>    Contact: <sip:50001 at X.X.29.123:21556>
>>>    To: "73435327569"<sip:73435327569 at default>
>>>    From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>>    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>>    CSeq: 1 INVITE
>>>    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
>>> SUBSCRIBE, INFO
>>>    Content-Type: application/sdp
>>>    User-Agent: eyeBeam release 1102u stamp 52345
>>>    Content-Length: 337
>>>
>>>    v=0
>>>    o=- 1 2 IN IP4 192.168.3.253
>>>    s=CounterPath eyeBeam 1.5
>>>    c=IN IP4 192.168.3.253
>>>    t=0 0
>>>    m=audio 40032 RTP/AVP 8 101
>>>    a=alt:1 3 : XdMyLUqb Fs2PLxKV 192.168.3.253 40032
>>>    a=alt:2 2 : 2qrc/CQ7 Z/9XFQZl 172.16.3.253 40032
>>>    a=alt:3 1 : 8BOwI53P +ENGEr/U X.X.29.123 40032
>>>    a=fmtp:101 0-15
>>>    a=rtpmap:101 telephone-event/8000
>>>    a=sendrecv
>>>    ------------------------------------------------------------------------
>>> send 385 bytes to udp/[X.X.29.123]:21556 at 10:30:31.313366:
>>>    ------------------------------------------------------------------------
>>>    SIP/2.0 100 Trying
>>>    Via: SIP/2.0/UDP
>>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>>>    From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>>    To: "73435327569"<sip:73435327569 at default>
>>>    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>>    CSeq: 1 INVITE
>>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
>>> 16-36-04 -0500
>>>    Content-Length: 0
>>>
>>>    ------------------------------------------------------------------------
>>> 2011-01-20 15:30:31.313448 [NOTICE] switch_channel.c:808 New Channel
>>> sofia/epbx/50001 at default [95698968-8cf9-40a2-91ad-1322fcfc76af]
>>> 2011-01-20 15:30:31.322338 [INFO] mod_dialplan_xml.c:331 Processing
>>> TAGNet <50001>->73435327569 in context public
>>> 2011-01-20 15:30:31.323362 [INFO] mod_dialplan_xml.c:331 Processing
>>> TAGNet <50001>->ext_translate_extsrc in context features
>>> 2011-01-20 15:30:31.324357 [NOTICE] switch_ivr.c:1606 Transfer
>>> sofia/epbx/50001 at default to XML[73435327569 at top.ctx]
>>> 2011-01-20 15:30:31.325352 [INFO] mod_dialplan_xml.c:331 Processing
>>> TAGNet <50001>->73435327569 in context top.ctx
>>> 2011-01-20 15:30:31.326344 [NOTICE] mod_dptools.c:1174 [top.ctx] - 
>>> ext_local
>>> 2011-01-20 15:30:31.329360 [NOTICE] switch_channel.c:808 New Channel
>>> sofia/epbx/73435327569 at X.X.16.83:5060 
>>> [b9654c75-039a-4780-bc00-0fae65f92a9a]
>>> send 1276 bytes to udp/[X.X.16.83]:5060 at 10:30:31.330623:
>>>    ------------------------------------------------------------------------
>>>    INVITE sip:73435327569 at X.X.16.83:5060 SIP/2.0
>>>    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>>    Max-Forwards: 68
>>>    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>>    To: <sip:73435327569 at X.X.16.83:5060>
>>>    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>>    CSeq: 7423651 INVITE
>>>    Contact: <sip:mod_sofia at Y.Y.138.187:5060>
>>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
>>> 16-36-04 -0500
>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>    Supported: timer, precondition, path, replaces
>>>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>>> sla, include-session-description, presence.winfo, message-summary, refer
>>>    Content-Type: application/sdp
>>>    Content-Disposition: session
>>>    Content-Length: 341
>>>    X-FS-Support: update_display
>>>    Remote-Party-ID: "TAGNet"
>>> <sip:50001 at Y.Y.138.187>;party=calling;screen=yes;privacy=off
>>>
>>>    v=0
>>>    o=FreeSWITCH 1627720692 1627720693 IN IP4 Y.Y.138.187
>>>    s=FreeSWITCH
>>>    c=IN IP4 Y.Y.138.187
>>>    t=0 0
>>>    m=audio 23442 RTP/AVP 8 101
>>>    a=rtpmap:101 telephone-event/8000
>>>    a=fmtp:101 0-15
>>>    a=alt:1 3 : XdMyLUqb Fs2PLxKV 192.168.3.253 40032
>>>    a=alt:2 2 : 2qrc/CQ7 Z/9XFQZl 172.16.3.253 40032
>>>    a=alt:3 1 : 8BOwI53P +ENGEr/U X.X.29.123 40032
>>>    ------------------------------------------------------------------------
>>> recv 396 bytes from udp/[X.X.16.83]:5060 at 10:30:31.341106:
>>>    ------------------------------------------------------------------------
>>>    SIP/2.0 100 Trying
>>>    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>>    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>>    To: <sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>>>    Date: Thu, 20 Jan 2011 10:30:31 GMT
>>>    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>>    Server: Cisco-SIPGateway/IOS-12.x
>>>    CSeq: 7423651 INVITE
>>>    Allow-Events: telephone-event
>>>    Content-Length: 0
>>>
>>>    ------------------------------------------------------------------------
>>> recv 814 bytes from udp/[X.X.16.83]:5060 at 10:30:32.401494:
>>>    ------------------------------------------------------------------------
>>>    SIP/2.0 183 Session Progress
>>>    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>>    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>>    To: <sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>>>    Date: Thu, 20 Jan 2011 10:30:31 GMT
>>>    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>>    Server: Cisco-SIPGateway/IOS-12.x
>>>    CSeq: 7423651 INVITE
>>>    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
>>> SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
>>>    Allow-Events: telephone-event
>>>    Contact: <sip:73435327569 at X.X.16.83:5060>
>>>    Content-Disposition: session;handling=required
>>>    Content-Type: application/sdp
>>>    Content-Length: 176
>>>
>>>    v=0
>>>    o=CiscoSystemsSIP-GW-UserAgent 4564 1578 IN IP4 X.X.16.83
>>>    s=SIP Call
>>>    c=IN IP4 X.X.16.83
>>>    t=0 0
>>>    m=audio 17770 RTP/AVP 8
>>>    c=IN IP4 X.X.16.83
>>>    a=rtpmap:8 PCMA/8000
>>>    ------------------------------------------------------------------------
>>> 2011-01-20 15:30:32.401362 [INFO] sofia.c:729
>>> sofia/epbx/73435327569 at X.X.16.83:5060 Update Callee ID to "Outbound
>>> Call" <73435327569>
>>> 2011-01-20 15:30:32.409363 [NOTICE] sofia.c:4739 Pre-Answer
>>> sofia/epbx/73435327569 at X.X.16.83:5060!
>>> 2011-01-20 15:30:32.415412 [INFO] switch_ivr_originate.c:3345 Sending
>>> early media
>>> 2011-01-20 15:30:32.416367 [NOTICE] mod_sofia.c:2252 Pre-Answer
>>> sofia/epbx/50001 at default!
>>> send 1145 bytes to udp/[X.X.29.123]:21556 at 10:30:32.418195:
>>>    ------------------------------------------------------------------------
>>>    SIP/2.0 183 Session Progress
>>>    Via: SIP/2.0/UDP
>>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>>>    From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>>    To: "73435327569" <sip:73435327569 at default>;tag=BScUp0QXHFBgB
>>>    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>>    CSeq: 1 INVITE
>>>    Contact: <sip:73435327569 at Y.Y.138.187:5060;transport=udp>
>>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
>>> 16-36-04 -0500
>>>    Accept: application/sdp
>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>    Supported: timer, precondition, path, replaces
>>>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>>> sla, include-session-description, presence.winfo, message-summary, refer
>>>    Content-Type: application/sdp
>>>    Content-Disposition: session
>>>    Content-Length: 181
>>>    Remote-Party-ID: "73435327569"
>>> <sip:73435327569 at Y.Y.138.187>;party=calling;privacy=off;screen=no
>>>
>>>    v=0
>>>    o=FreeSWITCH 1627752513 1627752514 IN IP4 Y.Y.138.187
>>>    s=FreeSWITCH
>>>    c=IN IP4 Y.Y.138.187
>>>    t=0 0
>>>    m=audio 25806 RTP/AVP 8
>>>    c=IN IP4 Y.Y.138.187
>>>    a=rtpmap:8 PCMA/8000
>>>
>>>    ------------------------------------------------------------------------
>>> recv 365 bytes from udp/[X.X.29.123]:21556 at 10:30:39.196791:
>>>    ------------------------------------------------------------------------
>>>    CANCEL sip:73435327569 at default SIP/2.0
>>>    Via: SIP/2.0/UDP
>>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
>>>    To: "73435327569"<sip:73435327569 at default>
>>>    From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>>    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>>    CSeq: 1 CANCEL
>>>    User-Agent: eyeBeam release 1102u stamp 52345
>>>    Content-Length: 0
>>>
>>>    ------------------------------------------------------------------------
>>> send 320 bytes to udp/[X.X.29.123]:21556 at 10:30:39.196905:
>>>    ------------------------------------------------------------------------
>>>    SIP/2.0 200 OK
>>>    Via: SIP/2.0/UDP
>>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>>>    From: "TAGNet" <sip:50001 at default>;tag=6c41dc5a
>>>    To: "73435327569" <sip:73435327569 at default>;tag=BScUp0QXHFBgB
>>>    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>>    CSeq: 1 CANCEL
>>>    Content-Length: 0
>>>
>>>    ------------------------------------------------------------------------
>>> send 715 bytes to udp/[X.X.29.123]:21556 at 10:30:39.197002:
>>>    ------------------------------------------------------------------------
>>>    SIP/2.0 487 Request Terminated
>>>    Via: SIP/2.0/UDP
>>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>>>    From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>>    To: "73435327569" <sip:73435327569 at default>;tag=BScUp0QXHFBgB
>>>    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>>    CSeq: 1 INVITE
>>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
>>> 16-36-04 -0500
>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>    Supported: timer, precondition, path, replaces
>>>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>>> sla, include-session-description, presence.winfo, message-summary, refer
>>>    Content-Length: 0
>>>
>>>    ------------------------------------------------------------------------
>>> 2011-01-20 15:30:39.196838 [NOTICE] sofia.c:5286 Hangup
>>> sofia/epbx/50001 at default [CS_EXECUTE] [ORIGINATOR_CANCEL]
>>> 2011-01-20 15:30:39.196838 [NOTICE] switch_ivr_bridge.c:653 Hangup
>>> sofia/epbx/73435327569 at X.X.16.83:5060 [CS_EXCHANGE_MEDIA] 
>>> [NORMAL_CLEARING]
>>> recv 331 bytes from udp/[X.X.29.123]:21556 at 10:30:39.198834:
>>>    ------------------------------------------------------------------------
>>>    ACK sip:73435327569 at default SIP/2.0
>>>    Via: SIP/2.0/UDP
>>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
>>>    To: "73435327569" <sip:73435327569 at default>;tag=BScUp0QXHFBgB
>>>    From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>>    Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>>    CSeq: 1 ACK
>>>    Content-Length: 0
>>>
>>>    ------------------------------------------------------------------------
>>> 2011-01-20 15:30:39.204838 [NOTICE] switch_core_session.c:1306 Session
>>> 99 (sofia/epbx/50001 at default) Ended
>>> 2011-01-20 15:30:39.204838 [NOTICE] switch_core_session.c:1308 Close
>>> Channel sofia/epbx/50001 at default [CS_DESTROY]
>>> send 372 bytes to udp/[X.X.16.83]:5060 at 10:30:39.212574:
>>>    ------------------------------------------------------------------------
>>>    CANCEL sip:73435327569 at X.X.16.83:5060 SIP/2.0
>>>    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>>    Max-Forwards: 68
>>>    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>>    To: <sip:73435327569 at X.X.16.83:5060>
>>>    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>>    CSeq: 7423651 CANCEL
>>>    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>>>    Content-Length: 0
>>>
>>>    ------------------------------------------------------------------------
>>> 2011-01-20 15:30:39.212840 [NOTICE] switch_core_session.c:1306 Session
>>> 100 (sofia/epbx/73435327569 at X.X.16.83:5060) Ended
>>> 2011-01-20 15:30:39.212840 [NOTICE] switch_core_session.c:1308 Close
>>> Channel sofia/epbx/73435327569 at X.X.16.83:5060 [CS_DESTROY]
>>> recv 311 bytes from udp/[X.X.16.83]:5060 at 10:30:39.216838:
>>>    ------------------------------------------------------------------------
>>>    SIP/2.0 200 OK
>>>    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>>    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>>    To: <sip:73435327569 at X.X.16.83:5060>
>>>    Date: Thu, 20 Jan 2011 10:30:39 GMT
>>>    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>>    Content-Length: 0
>>>    CSeq: 7423651 CANCEL
>>>
>>>    ------------------------------------------------------------------------
>>> recv 407 bytes from udp/[X.X.16.83]:5060 at 10:30:39.219180:
>>>    ------------------------------------------------------------------------
>>>    SIP/2.0 487 Request Cancelled
>>>    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>>    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>>    To: <sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>>>    Date: Thu, 20 Jan 2011 10:30:39 GMT
>>>    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>>    Server: Cisco-SIPGateway/IOS-12.x
>>>    CSeq: 7423651 INVITE
>>>    Allow-Events: telephone-event
>>>    Content-Length: 0
>>>
>>>    ------------------------------------------------------------------------
>>> send 334 bytes to udp/[X.X.16.83]:5060 at 10:30:39.219276:
>>>    ------------------------------------------------------------------------
>>>    ACK sip:73435327569 at X.X.16.83:5060 SIP/2.0
>>>    Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>>    Max-Forwards: 68
>>>    From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>>    To: <sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>>>    Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>>    CSeq: 7423651 ACK
>>>    Content-Length: 0
>>>
>>>    ------------------------------------------------------------------------
>>>
>>>
>>> -- 
>>> ? ?????????,
>>>   ????? ?????????
>>>   ??? "??????"
>>>   (3435) 494991
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org 
>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> -- 
> ? ?????????,
>    ????? ?????????
>    ??? "??????"
>    (3435) 494991
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991


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From jmesquita at freeswitch.org  Thu Jan 20 17:12:20 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 20 Jan 2011 11:12:20 -0300
Subject: [Freeswitch-users] GSM/PSTN Gateways
In-Reply-To: <0F8036C9-D129-419D-AAA0-C8EA3856D7DD@gmail.com>
References: <0F8036C9-D129-419D-AAA0-C8EA3856D7DD@gmail.com>
Message-ID: <AANLkTi=yvaG2mNd_je1MhQu_+=-TWFN8dJWWtBVxTQP=@mail.gmail.com>

Normally, cheap and quality are inversely proportional concepts so be
careful. :-)

Have you seen this?
http://www.khomp.com.br/?menu=produto&content=produtos&type=SPX&base=20

Regards,
Jo?o Mesquita


On Thu, Jan 20, 2011 at 10:36 AM, Deon Vermeulen
<vermeulen.deon at gmail.com>wrote:

> Hi List
>
> I've been searching the Net a lot the past couple of days looking for
> relatively good priced Internal and External GSM and PSTN Gateways.
>
> I did come up with couple devices, but they still seem a bit expensive.
>
> One device that did draw my attention though is the DuMV at PCI GSM Card.
> http://www.portech.com.tw/p3-product1_1.asp?Pid=15
> Do no if this will work in FreeSwitch?
>
> Would really appreciate suggestions on products used.
>
>
> Thank you very much
>
> Regards
>
> Deon
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vermeulen.deon at gmail.com  Thu Jan 20 17:26:32 2011
From: vermeulen.deon at gmail.com (Deon Vermeulen)
Date: Thu, 20 Jan 2011 16:26:32 +0200
Subject: [Freeswitch-users] GSM/PSTN Gateways
In-Reply-To: <AANLkTi=yvaG2mNd_je1MhQu_+=-TWFN8dJWWtBVxTQP=@mail.gmail.com>
References: <0F8036C9-D129-419D-AAA0-C8EA3856D7DD@gmail.com>
	<AANLkTi=yvaG2mNd_je1MhQu_+=-TWFN8dJWWtBVxTQP=@mail.gmail.com>
Message-ID: <5A45E6A5-252B-4AB9-8EAC-9B3FF25F07AE@gmail.com>

Hi Joao


Yeah. Cheap and quality don't normally go together, but Quality at a better price does... ;-)

Thanks for the link. Really Appreciate.
Do these devices Interface well with Freeswitch, especially the USB Devices?


Regards
Deon


On Jan 20, 2011, at 4:12 PM, Jo?o Mesquita wrote:

> Normally, cheap and quality are inversely proportional concepts so be careful. :-)
> 
> Have you seen this? http://www.khomp.com.br/?menu=produto&content=produtos&type=SPX&base=20
> 
> Regards,
> Jo?o Mesquita
> 
> 
> On Thu, Jan 20, 2011 at 10:36 AM, Deon Vermeulen <vermeulen.deon at gmail.com> wrote:
> Hi List
> 
> I've been searching the Net a lot the past couple of days looking for relatively good priced Internal and External GSM and PSTN Gateways.
> 
> I did come up with couple devices, but they still seem a bit expensive.
> 
> One device that did draw my attention though is the DuMV at PCI GSM Card. http://www.portech.com.tw/p3-product1_1.asp?Pid=15
> Do no if this will work in FreeSwitch?
> 
> Would really appreciate suggestions on products used.
> 
> 
> Thank you very much
> 
> Regards
> 
> Deon
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jmesquita at freeswitch.org  Thu Jan 20 17:34:16 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 20 Jan 2011 11:34:16 -0300
Subject: [Freeswitch-users] GSM/PSTN Gateways
In-Reply-To: <5A45E6A5-252B-4AB9-8EAC-9B3FF25F07AE@gmail.com>
References: <0F8036C9-D129-419D-AAA0-C8EA3856D7DD@gmail.com>
	<AANLkTi=yvaG2mNd_je1MhQu_+=-TWFN8dJWWtBVxTQP=@mail.gmail.com>
	<5A45E6A5-252B-4AB9-8EAC-9B3FF25F07AE@gmail.com>
Message-ID: <AANLkTikvyQ=ZBMAH-rUmuMoAuDnejaZttT=RjwCLoR8z@mail.gmail.com>

Yes they do, they use the mod_khomp, which is maintained by Khomp
themselves. It is in tree, you can take a look at it! :-)

Regards,
Jo?o Mesquita


On Thu, Jan 20, 2011 at 11:26 AM, Deon Vermeulen
<vermeulen.deon at gmail.com>wrote:

> Hi Joao
>
>
> Yeah. Cheap and quality don't normally go together, but Quality at a better
> price does... ;-)
>
> Thanks for the link. Really Appreciate.
> Do these devices Interface well with Freeswitch, especially the USB
> Devices?
>
>
> Regards
> Deon
>
>
> On Jan 20, 2011, at 4:12 PM, Jo?o Mesquita wrote:
>
> Normally, cheap and quality are inversely proportional concepts so be
> careful. :-)
>
> Have you seen this?
> http://www.khomp.com.br/?menu=produto&content=produtos&type=SPX&base=20
>
> Regards,
> Jo?o Mesquita
>
>
> On Thu, Jan 20, 2011 at 10:36 AM, Deon Vermeulen <vermeulen.deon at gmail.com
> > wrote:
>
>> Hi List
>>
>> I've been searching the Net a lot the past couple of days looking for
>> relatively good priced Internal and External GSM and PSTN Gateways.
>>
>> I did come up with couple devices, but they still seem a bit expensive.
>>
>> One device that did draw my attention though is the DuMV at PCI GSM Card.
>> http://www.portech.com.tw/p3-product1_1.asp?Pid=15
>> Do no if this will work in FreeSwitch?
>>
>> Would really appreciate suggestions on products used.
>>
>>
>> Thank you very much
>>
>> Regards
>>
>> Deon
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Thu Jan 20 17:34:55 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 20 Jan 2011 11:34:55 -0300
Subject: [Freeswitch-users] GSM/PSTN Gateways
In-Reply-To: <5A45E6A5-252B-4AB9-8EAC-9B3FF25F07AE@gmail.com>
References: <0F8036C9-D129-419D-AAA0-C8EA3856D7DD@gmail.com>
	<AANLkTi=yvaG2mNd_je1MhQu_+=-TWFN8dJWWtBVxTQP=@mail.gmail.com>
	<5A45E6A5-252B-4AB9-8EAC-9B3FF25F07AE@gmail.com>
Message-ID: <AANLkTiksVF-OnK-bM7L9Vujm=_u6T6ay0Mzr0TxdXC-Q@mail.gmail.com>

Yes they do, they use the mod_khomp, which is maintained by Khomp
themselves. It is in tree, you can take a look at it! :-)

Regards,
Jo?o Mesquita


On Thu, Jan 20, 2011 at 11:26 AM, Deon Vermeulen
<vermeulen.deon at gmail.com>wrote:

> Hi Joao
>
>
> Yeah. Cheap and quality don't normally go together, but Quality at a better
> price does... ;-)
>
> Thanks for the link. Really Appreciate.
> Do these devices Interface well with Freeswitch, especially the USB
> Devices?
>
>
> Regards
> Deon
>
>
> On Jan 20, 2011, at 4:12 PM, Jo?o Mesquita wrote:
>
> Normally, cheap and quality are inversely proportional concepts so be
> careful. :-)
>
> Have you seen this?
> http://www.khomp.com.br/?menu=produto&content=produtos&type=SPX&base=20
>
> Regards,
> Jo?o Mesquita
>
>
> On Thu, Jan 20, 2011 at 10:36 AM, Deon Vermeulen <vermeulen.deon at gmail.com
> > wrote:
>
>> Hi List
>>
>> I've been searching the Net a lot the past couple of days looking for
>> relatively good priced Internal and External GSM and PSTN Gateways.
>>
>> I did come up with couple devices, but they still seem a bit expensive.
>>
>> One device that did draw my attention though is the DuMV at PCI GSM Card.
>> http://www.portech.com.tw/p3-product1_1.asp?Pid=15
>> Do no if this will work in FreeSwitch?
>>
>> Would really appreciate suggestions on products used.
>>
>>
>> Thank you very much
>>
>> Regards
>>
>> Deon
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From moises.silva at gmail.com  Thu Jan 20 17:37:26 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 20 Jan 2011 09:37:26 -0500
Subject: [Freeswitch-users] GSM/PSTN Gateways
In-Reply-To: <5A45E6A5-252B-4AB9-8EAC-9B3FF25F07AE@gmail.com>
References: <0F8036C9-D129-419D-AAA0-C8EA3856D7DD@gmail.com>
	<AANLkTi=yvaG2mNd_je1MhQu_+=-TWFN8dJWWtBVxTQP=@mail.gmail.com>
	<5A45E6A5-252B-4AB9-8EAC-9B3FF25F07AE@gmail.com>
Message-ID: <AANLkTimjf1D26e06GYbZ8_xUC9cgh8R8Lsrabg_c4q8a@mail.gmail.com>

9:26 AM, Deon Vermeulen <vermeulen.deon at gmail.com> wrote:
> Hi Joao
>
> Yeah. Cheap and quality don't normally go together, but Quality at a better
> price does... ;-)

I do not know about price, but I know Giovanni (one developer
contributing frequently to FreeSWITCH) is involved in the development
of this GSM gateways: http://www.mobigater.bg/

They make emphasis in "low-cost" so I guess the price shouldn't be that bad.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From freeswitch-list at puzzled.xs4all.nl  Thu Jan 20 17:53:00 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Thu, 20 Jan 2011 15:53:00 +0100
Subject: [Freeswitch-users] GSM/PSTN Gateways
In-Reply-To: <0F8036C9-D129-419D-AAA0-C8EA3856D7DD@gmail.com>
References: <0F8036C9-D129-419D-AAA0-C8EA3856D7DD@gmail.com>
Message-ID: <4D384C4C.60309@puzzled.xs4all.nl>

On 01/20/2011 02:36 PM, Deon Vermeulen wrote:
> Hi List
>
> I've been searching the Net a lot the past couple of days looking for relatively good priced Internal and External GSM and PSTN Gateways.
>
> I did come up with couple devices, but they still seem a bit expensive.
>
> One device that did draw my attention though is the DuMV at PCI GSM Card. http://www.portech.com.tw/p3-product1_1.asp?Pid=15
> Do no if this will work in FreeSwitch?
>
> Would really appreciate suggestions on products used.

A more premium solution would be 2N at http://www.2n.cz/en/
These work very well and in my experience are reliable.

Portech also have external boxes:
http://www.portech.com.tw/p3-product1.asp?Cid=6

Haven't used them myself but someone in #freenode once mentioned 
Mobigator, a GSM gateway compatible with FreeSWITCH. Uses a simple USB 
link to hook up to your freeswitch server. On the software side I think 
you need to install gsmopen on your FreeSWITCH box (howto on wiki).

http://www.mobigater.com/index.php?p=2&s=4
http://www.mobigater.com/index.php?p=2&s=6

At about $120 or ?90 they are quite affordable but I don't know how well 
they work.

Regards,
Patrick


From steveayre at gmail.com  Thu Jan 20 17:59:22 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 20 Jan 2011 14:59:22 +0000
Subject: [Freeswitch-users] GSM/PSTN Gateways
In-Reply-To: <4D384C4C.60309@puzzled.xs4all.nl>
References: <0F8036C9-D129-419D-AAA0-C8EA3856D7DD@gmail.com>
	<4D384C4C.60309@puzzled.xs4all.nl>
Message-ID: <AANLkTi=t8_ddWNcabFsJRiZh1M9_TUPDjARG3CwVXJ=b@mail.gmail.com>

Another manufacturer is http://www.quescom.com/

Some more information on your requirements might be useful - how many
channels do you want? Do you want to have SIMs at a different location to
the gateway? (can be easier to manage if you're managing gateways at
multiple offices)

The cheapest simplest option is probably using mod_gsmopen with an old
mobile.

-Steve



On 20 January 2011 14:53, Patrick Lists
<freeswitch-list at puzzled.xs4all.nl>wrote:

> On 01/20/2011 02:36 PM, Deon Vermeulen wrote:
> > Hi List
> >
> > I've been searching the Net a lot the past couple of days looking for
> relatively good priced Internal and External GSM and PSTN Gateways.
> >
> > I did come up with couple devices, but they still seem a bit expensive.
> >
> > One device that did draw my attention though is the DuMV at PCI GSM Card.
> http://www.portech.com.tw/p3-product1_1.asp?Pid=15
> > Do no if this will work in FreeSwitch?
> >
> > Would really appreciate suggestions on products used.
>
> A more premium solution would be 2N at http://www.2n.cz/en/
> These work very well and in my experience are reliable.
>
> Portech also have external boxes:
> http://www.portech.com.tw/p3-product1.asp?Cid=6
>
> Haven't used them myself but someone in #freenode once mentioned
> Mobigator, a GSM gateway compatible with FreeSWITCH. Uses a simple USB
> link to hook up to your freeswitch server. On the software side I think
> you need to install gsmopen on your FreeSWITCH box (howto on wiki).
>
> http://www.mobigater.com/index.php?p=2&s=4
> http://www.mobigater.com/index.php?p=2&s=6
>
> At about $120 or ?90 they are quite affordable but I don't know how well
> they work.
>
> Regards,
> Patrick
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Thu Jan 20 18:03:57 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 20 Jan 2011 15:03:57 +0000
Subject: [Freeswitch-users] call barging
In-Reply-To: <1295531363114-5943853.post@n2.nabble.com>
References: <AANLkTinMqT+Qs1guPphYWqN5okMAD9c1K9=Jey6uepRw@mail.gmail.com>
	<AANLkTimrbwUS7s1G+catXL-XVTr_pg9yOWorR04MNFY=@mail.gmail.com>
	<AANLkTinNrHFKY22Amsv-mMw=wMcxmKheo-WxT8kYTrqC@mail.gmail.com>
	<1295531363114-5943853.post@n2.nabble.com>
Message-ID: <AANLkTi=sZ9G-QNHJXUFiphzG04RYSmd1r_y6DiEF7J3h@mail.gmail.com>

Is someone speaking on the other call? What codecs are being used?

It might help to see your debug log, for both calls.

-Steve


On 20 January 2011 13:49, mazilo <Nabble at slickdeals.endjunk.com> wrote:

>
>
> Steven Ayre wrote:
> >
> > Yes, that will work. You can set the group either within the extension or
> > in
> > the user config.
> >
> > For example:
> >
> >  <extension name="bridge">
> >      <condition field="destination_number" expression="^9(\d+)$">
> >        <action application="set" data="eavesdrop_group=spy1"/>
> >        <action application="bridge" data="sofia/gateway/gw1/$1"/>
> >      </condition>
> >    </extension>
> >   <extension name="eavesdrop">
> >      <condition field="destination_number" expression="^666$">
> >        <action application="answer"/>
> >        <action application="set" data="eavesdrop_require_group=spy1"/>
> >        <action application="eavesdrop" data="all"/>
> >      </condition>
> >    </extension>
> >
> >
> > -Steve
> Steve, I tried your above suggestion on my FreeSWITCH Version 1.0.head
> (git-cf253c3 2011-01-11 15-31-40 -0600) with mod_spy loaded and the barge
> in
> works, but only void (no conversation captured).
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/call-barging-tp5943099p5943853.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Thu Jan 20 18:06:26 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 20 Jan 2011 15:06:26 +0000
Subject: [Freeswitch-users] Question about bind_digit_action
In-Reply-To: <AANLkTinRz4sOoxOOf_D1a+XiFSFe33o_0obTjHjgxwS-@mail.gmail.com>
References: <AANLkTinRz4sOoxOOf_D1a+XiFSFe33o_0obTjHjgxwS-@mail.gmail.com>
Message-ID: <AANLkTin_4ZxsXTAujCkfV7Vit7JJgzjfwxOviYBBEbkB@mail.gmail.com>

It's in mod_dptools, you can check that's loaded but I would be surprised if
it wasn't since so many useful things are in there that it's pretty much
standard.

How old is your git version? Your version command isn't showing it and this
was only added recently. Try updating.

Steve



On 20 January 2011 13:22, joy this <thisjoy0528 at gmail.com> wrote:

> Dear all:
>
>
>
>         I want to use ?bind_digit_action?, but it did not work. I can only
> use ?bind_meta_app?, but the function is limited a lot. This is my diaplan:
>
>
>
> <extension name="test">
>
>         <condition field="destination_number" expression="^#123$">
>
>                         <action application="bind_digit_action"
> data="myrealm,500,exec:playback,ivr/ivr-welcome_to_freeswitch.wav"/>
>
>                         <action application="bind_digit_action"
> data="test1,456,exec:playback,ivr/ivr-welcome_to_freeswitch.wav"/>
>
>                         <action application="bind_digit_action"
> data="test1,##,exec:execute_extension,mix_welcome_to_freeswitch"/>
>
>                         <action application="digit_action_set_realm"
> data="test1"/>
>
>         </condition>
>
> </extension>
>
>
>
>         But it shows:
>
> 2011-01-20 21:03:16.359375 [ERR] switch_core_session.c:1791 Invalid
> Application bind_digit_action
>
>
>
>         Can anyone please help me? My FS version is ?FreeSWITCH Version
> 1.0.head (git-)?.
>
>
>
> Sincerely yours,
>
> Thisjoy.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ross at ossiantelecom.co.uk  Thu Jan 20 19:02:51 2011
From: ross at ossiantelecom.co.uk (Ross McKillop)
Date: Thu, 20 Jan 2011 16:02:51 +0000
Subject: [Freeswitch-users] SIP device for attaching a speaker
In-Reply-To: <4D382CE3.3040905@telefaks.de>
References: <4D3807A9.8040103@gmx.net>
	<AANLkTi=Pa=w9WzCNkzAJCbyuqJi7b7omMs6AAcn6C-9D@mail.gmail.com>
	<4D382CE3.3040905@telefaks.de>
Message-ID: <468AAF0E-2A92-4BD7-8623-2E7555DFBED9@ossiantelecom.co.uk>


On 20 Jan 2011, at 12:38, Peter Steinbach wrote:

> that's exactly what I looked for. And it's even affordable

I've also used these before;

http://www.cyberdata.net/products/voip/digitalanalog/pagingamp/index.html

Similar device but just to give you an alternative.

Ross



From Nabble at slickdeals.endjunk.com  Thu Jan 20 19:03:00 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 20 Jan 2011 08:03:00 -0800 (PST)
Subject: [Freeswitch-users] call barging
In-Reply-To: <AANLkTi=sZ9G-QNHJXUFiphzG04RYSmd1r_y6DiEF7J3h@mail.gmail.com>
References: <AANLkTinMqT+Qs1guPphYWqN5okMAD9c1K9=Jey6uepRw@mail.gmail.com>
	<AANLkTimrbwUS7s1G+catXL-XVTr_pg9yOWorR04MNFY=@mail.gmail.com>
	<AANLkTinNrHFKY22Amsv-mMw=wMcxmKheo-WxT8kYTrqC@mail.gmail.com>
	<1295531363114-5943853.post@n2.nabble.com>
	<AANLkTi=sZ9G-QNHJXUFiphzG04RYSmd1r_y6DiEF7J3h@mail.gmail.com>
Message-ID: <1295539380742-5944311.post@n2.nabble.com>



Steven Ayre wrote:
> 
> Is someone speaking on the other call? What codecs are being used?
> 
> It might help to see your debug log, for both calls.
> 
> -Steve
My configuration uses a default conf/vars.xml settings and the calls were in
PCMU codec. I had an Ekiga softphone on my OpenSuSE v11.3 Linux computer
calling a Gizmo5 ECHO line (1-747-474-3246) while using my Uniden UIP1869V
to barge in. Now, when I tried it, all I got from the barge in is static
noise.  Here is the cli dump on  http://pastebin.com/B4fuNdX0 ekiga call to
1-747-474-3246  and  http://pastebin.com/Qbidsi3s call to 666 to barge in .

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/call-barging-tp5943099p5944311.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From imthiyaz at peopletech.co.in  Thu Jan 20 19:11:59 2011
From: imthiyaz at peopletech.co.in (Imthiyaz Ahmed)
Date: Thu, 20 Jan 2011 21:41:59 +0530
Subject: [Freeswitch-users] GSM/PSTN Gateways
In-Reply-To: <AANLkTi=t8_ddWNcabFsJRiZh1M9_TUPDjARG3CwVXJ=b@mail.gmail.com>
References: <0F8036C9-D129-419D-AAA0-C8EA3856D7DD@gmail.com>
	<4D384C4C.60309@puzzled.xs4all.nl>
	<AANLkTi=t8_ddWNcabFsJRiZh1M9_TUPDjARG3CwVXJ=b@mail.gmail.com>
Message-ID: <AANLkTin0afY88bBMtug7Fy9edP_HmrsLQ0vhd5yOkH5T@mail.gmail.com>

Pls try allywll Gateways , it is good

On Thu, Jan 20, 2011 at 8:29 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Another manufacturer is http://www.quescom.com/
>
> Some more information on your requirements might be useful - how many
> channels do you want? Do you want to have SIMs at a different location to
> the gateway? (can be easier to manage if you're managing gateways at
> multiple offices)
>
> The cheapest simplest option is probably using mod_gsmopen with an old
> mobile.
>
> -Steve
>
>
>
>
> On 20 January 2011 14:53, Patrick Lists <freeswitch-list at puzzled.xs4all.nl
> > wrote:
>
>> On 01/20/2011 02:36 PM, Deon Vermeulen wrote:
>> > Hi List
>> >
>> > I've been searching the Net a lot the past couple of days looking for
>> relatively good priced Internal and External GSM and PSTN Gateways.
>> >
>> > I did come up with couple devices, but they still seem a bit expensive.
>> >
>> > One device that did draw my attention though is the DuMV at PCI GSM Card.
>> http://www.portech.com.tw/p3-product1_1.asp?Pid=15
>> > Do no if this will work in FreeSwitch?
>> >
>> > Would really appreciate suggestions on products used.
>>
>> A more premium solution would be 2N at http://www.2n.cz/en/
>> These work very well and in my experience are reliable.
>>
>> Portech also have external boxes:
>> http://www.portech.com.tw/p3-product1.asp?Cid=6
>>
>> Haven't used them myself but someone in #freenode once mentioned
>> Mobigator, a GSM gateway compatible with FreeSWITCH. Uses a simple USB
>> link to hook up to your freeswitch server. On the software side I think
>> you need to install gsmopen on your FreeSWITCH box (howto on wiki).
>>
>> http://www.mobigater.com/index.php?p=2&s=4
>> http://www.mobigater.com/index.php?p=2&s=6
>>
>> At about $120 or ?90 they are quite affordable but I don't know how well
>> they work.
>>
>> Regards,
>> Patrick
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Best Regards
G.Imthiyaz Ahmed
PeopleTech systems (P) ltd
http://peopletech.co.in
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From anthony.minessale at gmail.com  Thu Jan 20 19:14:51 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 20 Jan 2011 10:14:51 -0600
Subject: [Freeswitch-users] PCMA stranges
In-Reply-To: <4D384196.105@tagnet.ru>
References: <4D380FF8.7080705@tagnet.ru>
	<E676EB80-7927-4061-BC2C-497DBA81D75E@ipeva.fr>
	<4D381E1F.7040202@tagnet.ru> <4D384196.105@tagnet.ru>
Message-ID: <AANLkTikVAt6t8K4fxg0GXisw=0_+s5gBYe9Hyvep5KN2@mail.gmail.com>

proxy media means media is ignored, its not ever touched.  Its
designed to transparently pass media even if FS is unaware of the
codec.
So if you say you are using proxy media and you have codec specific
problems it seems peculiar.


On Thu, Jan 20, 2011 at 8:07 AM, Boris Kovalenko <boris at tagnet.ru> wrote:
> Hello!
>
> ??? I found that problem is somewhere inside freeswitch. Direct call between
> SoftPhone and Cisco is working fine.
>
> Hello!
>
> ??? David, may You explain why this problem is with G711A only? When I use
> G711U there are no problems and call is established and I may talk.
>
> The Cisco never sends the 200/OK after the 183, so the call is not
> established.
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 20/01/2011 ? 11:35, Boris Kovalenko a ?crit :
>
> Hello!
>
> ????My network configuration is:
> Softphone (eyeBeam 1.5) with only G711alaw enabled -> FreeSWITCH Version
> 1.0.head (git-0cf1d54 2011-01-19 16-36-04 -0500) -> Cisco AS5350
>
> proxy_media=true in profile.
> So, when I do a test call, there is SDP but no RTP between FS and CISCO,
> so I can't hear voice. With G711ulaw there are no problems. What is
> wrong? Siptrace below:
>
> ???------------------------------------------------------------------------
> ???INVITE sip:73435327569 at default SIP/2.0
> ???Via: SIP/2.0/UDP
> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
> ???Max-Forwards: 70
> ???Contact: <sip:50001 at X.X.29.123:21556>
> ???To: "73435327569"<sip:73435327569 at default>
> ???From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
> ???Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
> ???CSeq: 1 INVITE
> ???Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE, INFO
> ???Content-Type: application/sdp
> ???User-Agent: eyeBeam release 1102u stamp 52345
> ???Content-Length: 337
>
> ???v=0
> ???o=- 1 2 IN IP4 192.168.3.253
> ???s=CounterPath eyeBeam 1.5
> ???c=IN IP4 192.168.3.253
> ???t=0 0
> ???m=audio 40032 RTP/AVP 8 101
> ???a=alt:1 3 : XdMyLUqb Fs2PLxKV 192.168.3.253 40032
> ???a=alt:2 2 : 2qrc/CQ7 Z/9XFQZl 172.16.3.253 40032
> ???a=alt:3 1 : 8BOwI53P +ENGEr/U X.X.29.123 40032
> ???a=fmtp:101 0-15
> ???a=rtpmap:101 telephone-event/8000
> ???a=sendrecv
> ???------------------------------------------------------------------------
> send 385 bytes to udp/[X.X.29.123]:21556 at 10:30:31.313366:
> ???------------------------------------------------------------------------
> ???SIP/2.0 100 Trying
> ???Via: SIP/2.0/UDP
> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
> ???From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
> ???To: "73435327569"<sip:73435327569 at default>
> ???Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
> ???CSeq: 1 INVITE
> ???User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
> 16-36-04 -0500
> ???Content-Length: 0
>
> ???------------------------------------------------------------------------
> 2011-01-20 15:30:31.313448 [NOTICE] switch_channel.c:808 New Channel
> sofia/epbx/50001 at default [95698968-8cf9-40a2-91ad-1322fcfc76af]
> 2011-01-20 15:30:31.322338 [INFO] mod_dialplan_xml.c:331 Processing
> TAGNet <50001>->73435327569 in context public
> 2011-01-20 15:30:31.323362 [INFO] mod_dialplan_xml.c:331 Processing
> TAGNet <50001>->ext_translate_extsrc in context features
> 2011-01-20 15:30:31.324357 [NOTICE] switch_ivr.c:1606 Transfer
> sofia/epbx/50001 at default to XML[73435327569 at top.ctx]
> 2011-01-20 15:30:31.325352 [INFO] mod_dialplan_xml.c:331 Processing
> TAGNet <50001>->73435327569 in context top.ctx
> 2011-01-20 15:30:31.326344 [NOTICE] mod_dptools.c:1174 [top.ctx] - ext_local
> 2011-01-20 15:30:31.329360 [NOTICE] switch_channel.c:808 New Channel
> sofia/epbx/73435327569 at X.X.16.83:5060 [b9654c75-039a-4780-bc00-0fae65f92a9a]
> send 1276 bytes to udp/[X.X.16.83]:5060 at 10:30:31.330623:
> ???------------------------------------------------------------------------
> ???INVITE sip:73435327569 at X.X.16.83:5060 SIP/2.0
> ???Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
> ???Max-Forwards: 68
> ???From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
> ???To: <sip:73435327569 at X.X.16.83:5060>
> ???Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
> ???CSeq: 7423651 INVITE
> ???Contact: <sip:mod_sofia at Y.Y.138.187:5060>
> ???User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
> 16-36-04 -0500
> ???Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> ???Supported: timer, precondition, path, replaces
> ???Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
> sla, include-session-description, presence.winfo, message-summary, refer
> ???Content-Type: application/sdp
> ???Content-Disposition: session
> ???Content-Length: 341
> ???X-FS-Support: update_display
> ???Remote-Party-ID: "TAGNet"
> <sip:50001 at Y.Y.138.187>;party=calling;screen=yes;privacy=off
>
> ???v=0
> ???o=FreeSWITCH 1627720692 1627720693 IN IP4 Y.Y.138.187
> ???s=FreeSWITCH
> ???c=IN IP4 Y.Y.138.187
> ???t=0 0
> ???m=audio 23442 RTP/AVP 8 101
> ???a=rtpmap:101 telephone-event/8000
> ???a=fmtp:101 0-15
> ???a=alt:1 3 : XdMyLUqb Fs2PLxKV 192.168.3.253 40032
> ???a=alt:2 2 : 2qrc/CQ7 Z/9XFQZl 172.16.3.253 40032
> ???a=alt:3 1 : 8BOwI53P +ENGEr/U X.X.29.123 40032
> ???------------------------------------------------------------------------
> recv 396 bytes from udp/[X.X.16.83]:5060 at 10:30:31.341106:
> ???------------------------------------------------------------------------
> ???SIP/2.0 100 Trying
> ???Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
> ???From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
> ???To: <sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
> ???Date: Thu, 20 Jan 2011 10:30:31 GMT
> ???Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
> ???Server: Cisco-SIPGateway/IOS-12.x
> ???CSeq: 7423651 INVITE
> ???Allow-Events: telephone-event
> ???Content-Length: 0
>
> ???------------------------------------------------------------------------
> recv 814 bytes from udp/[X.X.16.83]:5060 at 10:30:32.401494:
> ???------------------------------------------------------------------------
> ???SIP/2.0 183 Session Progress
> ???Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
> ???From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
> ???To: <sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
> ???Date: Thu, 20 Jan 2011 10:30:31 GMT
> ???Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
> ???Server: Cisco-SIPGateway/IOS-12.x
> ???CSeq: 7423651 INVITE
> ???Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
> SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
> ???Allow-Events: telephone-event
> ???Contact: <sip:73435327569 at X.X.16.83:5060>
> ???Content-Disposition: session;handling=required
> ???Content-Type: application/sdp
> ???Content-Length: 176
>
> ???v=0
> ???o=CiscoSystemsSIP-GW-UserAgent 4564 1578 IN IP4 X.X.16.83
> ???s=SIP Call
> ???c=IN IP4 X.X.16.83
> ???t=0 0
> ???m=audio 17770 RTP/AVP 8
> ???c=IN IP4 X.X.16.83
> ???a=rtpmap:8 PCMA/8000
> ???------------------------------------------------------------------------
> 2011-01-20 15:30:32.401362 [INFO] sofia.c:729
> sofia/epbx/73435327569 at X.X.16.83:5060 Update Callee ID to "Outbound
> Call" <73435327569>
> 2011-01-20 15:30:32.409363 [NOTICE] sofia.c:4739 Pre-Answer
> sofia/epbx/73435327569 at X.X.16.83:5060!
> 2011-01-20 15:30:32.415412 [INFO] switch_ivr_originate.c:3345 Sending
> early media
> 2011-01-20 15:30:32.416367 [NOTICE] mod_sofia.c:2252 Pre-Answer
> sofia/epbx/50001 at default!
> send 1145 bytes to udp/[X.X.29.123]:21556 at 10:30:32.418195:
> ???------------------------------------------------------------------------
> ???SIP/2.0 183 Session Progress
> ???Via: SIP/2.0/UDP
> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
> ???From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
> ???To: "73435327569" <sip:73435327569 at default>;tag=BScUp0QXHFBgB
> ???Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
> ???CSeq: 1 INVITE
> ???Contact: <sip:73435327569 at Y.Y.138.187:5060;transport=udp>
> ???User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
> 16-36-04 -0500
> ???Accept: application/sdp
> ???Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> ???Supported: timer, precondition, path, replaces
> ???Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
> sla, include-session-description, presence.winfo, message-summary, refer
> ???Content-Type: application/sdp
> ???Content-Disposition: session
> ???Content-Length: 181
> ???Remote-Party-ID: "73435327569"
> <sip:73435327569 at Y.Y.138.187>;party=calling;privacy=off;screen=no
>
> ???v=0
> ???o=FreeSWITCH 1627752513 1627752514 IN IP4 Y.Y.138.187
> ???s=FreeSWITCH
> ???c=IN IP4 Y.Y.138.187
> ???t=0 0
> ???m=audio 25806 RTP/AVP 8
> ???c=IN IP4 Y.Y.138.187
> ???a=rtpmap:8 PCMA/8000
>
> ???------------------------------------------------------------------------
> recv 365 bytes from udp/[X.X.29.123]:21556 at 10:30:39.196791:
> ???------------------------------------------------------------------------
> ???CANCEL sip:73435327569 at default SIP/2.0
> ???Via: SIP/2.0/UDP
> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
> ???To: "73435327569"<sip:73435327569 at default>
> ???From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
> ???Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
> ???CSeq: 1 CANCEL
> ???User-Agent: eyeBeam release 1102u stamp 52345
> ???Content-Length: 0
>
> ???------------------------------------------------------------------------
> send 320 bytes to udp/[X.X.29.123]:21556 at 10:30:39.196905:
> ???------------------------------------------------------------------------
> ???SIP/2.0 200 OK
> ???Via: SIP/2.0/UDP
> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
> ???From: "TAGNet" <sip:50001 at default>;tag=6c41dc5a
> ???To: "73435327569" <sip:73435327569 at default>;tag=BScUp0QXHFBgB
> ???Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
> ???CSeq: 1 CANCEL
> ???Content-Length: 0
>
> ???------------------------------------------------------------------------
> send 715 bytes to udp/[X.X.29.123]:21556 at 10:30:39.197002:
> ???------------------------------------------------------------------------
> ???SIP/2.0 487 Request Terminated
> ???Via: SIP/2.0/UDP
> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
> ???From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
> ???To: "73435327569" <sip:73435327569 at default>;tag=BScUp0QXHFBgB
> ???Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
> ???CSeq: 1 INVITE
> ???User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
> 16-36-04 -0500
> ???Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> ???Supported: timer, precondition, path, replaces
> ???Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
> sla, include-session-description, presence.winfo, message-summary, refer
> ???Content-Length: 0
>
> ???------------------------------------------------------------------------
> 2011-01-20 15:30:39.196838 [NOTICE] sofia.c:5286 Hangup
> sofia/epbx/50001 at default [CS_EXECUTE] [ORIGINATOR_CANCEL]
> 2011-01-20 15:30:39.196838 [NOTICE] switch_ivr_bridge.c:653 Hangup
> sofia/epbx/73435327569 at X.X.16.83:5060 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> recv 331 bytes from udp/[X.X.29.123]:21556 at 10:30:39.198834:
> ???------------------------------------------------------------------------
> ???ACK sip:73435327569 at default SIP/2.0
> ???Via: SIP/2.0/UDP
> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
> ???To: "73435327569" <sip:73435327569 at default>;tag=BScUp0QXHFBgB
> ???From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
> ???Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
> ???CSeq: 1 ACK
> ???Content-Length: 0
>
> ???------------------------------------------------------------------------
> 2011-01-20 15:30:39.204838 [NOTICE] switch_core_session.c:1306 Session
> 99 (sofia/epbx/50001 at default) Ended
> 2011-01-20 15:30:39.204838 [NOTICE] switch_core_session.c:1308 Close
> Channel sofia/epbx/50001 at default [CS_DESTROY]
> send 372 bytes to udp/[X.X.16.83]:5060 at 10:30:39.212574:
> ???------------------------------------------------------------------------
> ???CANCEL sip:73435327569 at X.X.16.83:5060 SIP/2.0
> ???Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
> ???Max-Forwards: 68
> ???From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
> ???To: <sip:73435327569 at X.X.16.83:5060>
> ???Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
> ???CSeq: 7423651 CANCEL
> ???Reason: Q.850;cause=16;text="NORMAL_CLEARING"
> ???Content-Length: 0
>
> ???------------------------------------------------------------------------
> 2011-01-20 15:30:39.212840 [NOTICE] switch_core_session.c:1306 Session
> 100 (sofia/epbx/73435327569 at X.X.16.83:5060) Ended
> 2011-01-20 15:30:39.212840 [NOTICE] switch_core_session.c:1308 Close
> Channel sofia/epbx/73435327569 at X.X.16.83:5060 [CS_DESTROY]
> recv 311 bytes from udp/[X.X.16.83]:5060 at 10:30:39.216838:
> ???------------------------------------------------------------------------
> ???SIP/2.0 200 OK
> ???Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
> ???From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
> ???To: <sip:73435327569 at X.X.16.83:5060>
> ???Date: Thu, 20 Jan 2011 10:30:39 GMT
> ???Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
> ???Content-Length: 0
> ???CSeq: 7423651 CANCEL
>
> ???------------------------------------------------------------------------
> recv 407 bytes from udp/[X.X.16.83]:5060 at 10:30:39.219180:
> ???------------------------------------------------------------------------
> ???SIP/2.0 487 Request Cancelled
> ???Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
> ???From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
> ???To: <sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
> ???Date: Thu, 20 Jan 2011 10:30:39 GMT
> ???Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
> ???Server: Cisco-SIPGateway/IOS-12.x
> ???CSeq: 7423651 INVITE
> ???Allow-Events: telephone-event
> ???Content-Length: 0
>
> ???------------------------------------------------------------------------
> send 334 bytes to udp/[X.X.16.83]:5060 at 10:30:39.219276:
> ???------------------------------------------------------------------------
> ???ACK sip:73435327569 at X.X.16.83:5060 SIP/2.0
> ???Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
> ???Max-Forwards: 68
> ???From: "TAGNet" <sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
> ???To: <sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
> ???Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
> ???CSeq: 7423651 ACK
> ???Content-Length: 0
>
> ???------------------------------------------------------------------------
>
>
> --
> ? ?????????,
> ??????? ?????????
> ????? "??????"
> ??(3435) 494991
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
>
> _______________________________________________
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> http://www.freeswitch.org
>
>
> --
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   (3435) 494991
>
>
> _______________________________________________
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>
>
> --
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   (3435) 494991
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
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>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
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FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From steveayre at gmail.com  Thu Jan 20 19:27:52 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 20 Jan 2011 16:27:52 +0000
Subject: [Freeswitch-users] call barging
In-Reply-To: <1295539380742-5944311.post@n2.nabble.com>
References: <AANLkTinMqT+Qs1guPphYWqN5okMAD9c1K9=Jey6uepRw@mail.gmail.com>
	<AANLkTimrbwUS7s1G+catXL-XVTr_pg9yOWorR04MNFY=@mail.gmail.com>
	<AANLkTinNrHFKY22Amsv-mMw=wMcxmKheo-WxT8kYTrqC@mail.gmail.com>
	<1295531363114-5943853.post@n2.nabble.com>
	<AANLkTi=sZ9G-QNHJXUFiphzG04RYSmd1r_y6DiEF7J3h@mail.gmail.com>
	<1295539380742-5944311.post@n2.nabble.com>
Message-ID: <AANLkTinDdXEkE7_-1vcnsfgOTbtM7i8=NUgb82d8C-iB@mail.gmail.com>

   1. 2011-01-20 10:42:47.723158 [DEBUG] switch_core_media_bug.c:360
   Attaching BUG to sofia/internal/1000 at 192.168.1.123

Looks like it is eavesdropping on the other channel, I'm not sure why you're
not getting any audio but the eavesdrop app itself does appear to be
working.

I take it both phones can dial in and both speak and hear audio?

-Steve



On 20 January 2011 16:03, mazilo <Nabble at slickdeals.endjunk.com> wrote:

>
>
> Steven Ayre wrote:
> >
> > Is someone speaking on the other call? What codecs are being used?
> >
> > It might help to see your debug log, for both calls.
> >
> > -Steve
> My configuration uses a default conf/vars.xml settings and the calls were
> in
> PCMU codec. I had an Ekiga softphone on my OpenSuSE v11.3 Linux computer
> calling a Gizmo5 ECHO line (1-747-474-3246) while using my Uniden UIP1869V
> to barge in. Now, when I tried it, all I got from the barge in is static
> noise.  Here is the cli dump on  http://pastebin.com/B4fuNdX0 ekiga call
> to
> 1-747-474-3246  and  http://pastebin.com/Qbidsi3s call to 666 to barge in
> .
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/call-barging-tp5943099p5944311.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch-list at puzzled.xs4all.nl  Thu Jan 20 19:30:25 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Thu, 20 Jan 2011 17:30:25 +0100
Subject: [Freeswitch-users] GSM/PSTN Gateways
In-Reply-To: <AANLkTin0afY88bBMtug7Fy9edP_HmrsLQ0vhd5yOkH5T@mail.gmail.com>
References: <0F8036C9-D129-419D-AAA0-C8EA3856D7DD@gmail.com>	<4D384C4C.60309@puzzled.xs4all.nl>	<AANLkTi=t8_ddWNcabFsJRiZh1M9_TUPDjARG3CwVXJ=b@mail.gmail.com>
	<AANLkTin0afY88bBMtug7Fy9edP_HmrsLQ0vhd5yOkH5T@mail.gmail.com>
Message-ID: <4D386321.5090402@puzzled.xs4all.nl>

On 01/20/2011 05:11 PM, Imthiyaz Ahmed wrote:
> allywll Gateways

I thought that name was a spelling error. It was not. That's quite an 
unpronounceable company name. Image saying that with at the end of the 
FOSDEM beer party. Or maybe it will go beter then. Anyway their GSM 
gateways are here: http://www.allywll.com/Product.asp?lbID=31

Regards,
Patrick


From boris at tagnet.ru  Thu Jan 20 19:39:04 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 20 Jan 2011 21:39:04 +0500
Subject: [Freeswitch-users] PCMA stranges
In-Reply-To: <AANLkTikVAt6t8K4fxg0GXisw=0_+s5gBYe9Hyvep5KN2@mail.gmail.com>
References: <4D380FF8.7080705@tagnet.ru>	<E676EB80-7927-4061-BC2C-497DBA81D75E@ipeva.fr>	<4D381E1F.7040202@tagnet.ru>
	<4D384196.105@tagnet.ru>
	<AANLkTikVAt6t8K4fxg0GXisw=0_+s5gBYe9Hyvep5KN2@mail.gmail.com>
Message-ID: <4D386528.2010506@tagnet.ru>

Hello!

     Yes, I know the media is ignored, so I wondering about this codec 
strangenes. What test I have done:

SoftPhone (g711ulaw) -> FS -> Cisco 5350 : working
SoftPhone (g711alaw, proxy_media=true) -> FS -> Cisco 5350 : not working
SoftPhone (g711alaw, proxy_media=false) -> FS -> Cisco 5350 : working
SoftPhone (g711alaw) -> Cisco 5350 : working


> proxy media means media is ignored, its not ever touched.  Its
> designed to transparently pass media even if FS is unaware of the
> codec.
> So if you say you are using proxy media and you have codec specific
> problems it seems peculiar.
>
>
> On Thu, Jan 20, 2011 at 8:07 AM, Boris Kovalenko<boris at tagnet.ru>  wrote:
>> Hello!
>>
>>      I found that problem is somewhere inside freeswitch. Direct call between
>> SoftPhone and Cisco is working fine.
>>
>> Hello!
>>
>>      David, may You explain why this problem is with G711A only? When I use
>> G711U there are no problems and call is established and I may talk.
>>
>> The Cisco never sends the 200/OK after the 183, so the call is not
>> established.
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>> Le 20/01/2011 ? 11:35, Boris Kovalenko a ?crit :
>>
>> Hello!
>>
>>      My network configuration is:
>> Softphone (eyeBeam 1.5) with only G711alaw enabled ->  FreeSWITCH Version
>> 1.0.head (git-0cf1d54 2011-01-19 16-36-04 -0500) ->  Cisco AS5350
>>
>> proxy_media=true in profile.
>> So, when I do a test call, there is SDP but no RTP between FS and CISCO,
>> so I can't hear voice. With G711ulaw there are no problems. What is
>> wrong? Siptrace below:
>>
>>     ------------------------------------------------------------------------
>>     INVITE sip:73435327569 at default SIP/2.0
>>     Via: SIP/2.0/UDP
>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
>>     Max-Forwards: 70
>>     Contact:<sip:50001 at X.X.29.123:21556>
>>     To: "73435327569"<sip:73435327569 at default>
>>     From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>     Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>     CSeq: 1 INVITE
>>     Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
>> SUBSCRIBE, INFO
>>     Content-Type: application/sdp
>>     User-Agent: eyeBeam release 1102u stamp 52345
>>     Content-Length: 337
>>
>>     v=0
>>     o=- 1 2 IN IP4 192.168.3.253
>>     s=CounterPath eyeBeam 1.5
>>     c=IN IP4 192.168.3.253
>>     t=0 0
>>     m=audio 40032 RTP/AVP 8 101
>>     a=alt:1 3 : XdMyLUqb Fs2PLxKV 192.168.3.253 40032
>>     a=alt:2 2 : 2qrc/CQ7 Z/9XFQZl 172.16.3.253 40032
>>     a=alt:3 1 : 8BOwI53P +ENGEr/U X.X.29.123 40032
>>     a=fmtp:101 0-15
>>     a=rtpmap:101 telephone-event/8000
>>     a=sendrecv
>>     ------------------------------------------------------------------------
>> send 385 bytes to udp/[X.X.29.123]:21556 at 10:30:31.313366:
>>     ------------------------------------------------------------------------
>>     SIP/2.0 100 Trying
>>     Via: SIP/2.0/UDP
>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>>     From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>     To: "73435327569"<sip:73435327569 at default>
>>     Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>     CSeq: 1 INVITE
>>     User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
>> 16-36-04 -0500
>>     Content-Length: 0
>>
>>     ------------------------------------------------------------------------
>> 2011-01-20 15:30:31.313448 [NOTICE] switch_channel.c:808 New Channel
>> sofia/epbx/50001 at default [95698968-8cf9-40a2-91ad-1322fcfc76af]
>> 2011-01-20 15:30:31.322338 [INFO] mod_dialplan_xml.c:331 Processing
>> TAGNet<50001>->73435327569 in context public
>> 2011-01-20 15:30:31.323362 [INFO] mod_dialplan_xml.c:331 Processing
>> TAGNet<50001>->ext_translate_extsrc in context features
>> 2011-01-20 15:30:31.324357 [NOTICE] switch_ivr.c:1606 Transfer
>> sofia/epbx/50001 at default to XML[73435327569 at top.ctx]
>> 2011-01-20 15:30:31.325352 [INFO] mod_dialplan_xml.c:331 Processing
>> TAGNet<50001>->73435327569 in context top.ctx
>> 2011-01-20 15:30:31.326344 [NOTICE] mod_dptools.c:1174 [top.ctx] - ext_local
>> 2011-01-20 15:30:31.329360 [NOTICE] switch_channel.c:808 New Channel
>> sofia/epbx/73435327569 at X.X.16.83:5060 [b9654c75-039a-4780-bc00-0fae65f92a9a]
>> send 1276 bytes to udp/[X.X.16.83]:5060 at 10:30:31.330623:
>>     ------------------------------------------------------------------------
>>     INVITE sip:73435327569 at X.X.16.83:5060 SIP/2.0
>>     Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>     Max-Forwards: 68
>>     From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>     To:<sip:73435327569 at X.X.16.83:5060>
>>     Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>     CSeq: 7423651 INVITE
>>     Contact:<sip:mod_sofia at Y.Y.138.187:5060>
>>     User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
>> 16-36-04 -0500
>>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>     Supported: timer, precondition, path, replaces
>>     Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>> sla, include-session-description, presence.winfo, message-summary, refer
>>     Content-Type: application/sdp
>>     Content-Disposition: session
>>     Content-Length: 341
>>     X-FS-Support: update_display
>>     Remote-Party-ID: "TAGNet"
>> <sip:50001 at Y.Y.138.187>;party=calling;screen=yes;privacy=off
>>
>>     v=0
>>     o=FreeSWITCH 1627720692 1627720693 IN IP4 Y.Y.138.187
>>     s=FreeSWITCH
>>     c=IN IP4 Y.Y.138.187
>>     t=0 0
>>     m=audio 23442 RTP/AVP 8 101
>>     a=rtpmap:101 telephone-event/8000
>>     a=fmtp:101 0-15
>>     a=alt:1 3 : XdMyLUqb Fs2PLxKV 192.168.3.253 40032
>>     a=alt:2 2 : 2qrc/CQ7 Z/9XFQZl 172.16.3.253 40032
>>     a=alt:3 1 : 8BOwI53P +ENGEr/U X.X.29.123 40032
>>     ------------------------------------------------------------------------
>> recv 396 bytes from udp/[X.X.16.83]:5060 at 10:30:31.341106:
>>     ------------------------------------------------------------------------
>>     SIP/2.0 100 Trying
>>     Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>     From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>     To:<sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>>     Date: Thu, 20 Jan 2011 10:30:31 GMT
>>     Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>     Server: Cisco-SIPGateway/IOS-12.x
>>     CSeq: 7423651 INVITE
>>     Allow-Events: telephone-event
>>     Content-Length: 0
>>
>>     ------------------------------------------------------------------------
>> recv 814 bytes from udp/[X.X.16.83]:5060 at 10:30:32.401494:
>>     ------------------------------------------------------------------------
>>     SIP/2.0 183 Session Progress
>>     Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>     From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>     To:<sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>>     Date: Thu, 20 Jan 2011 10:30:31 GMT
>>     Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>     Server: Cisco-SIPGateway/IOS-12.x
>>     CSeq: 7423651 INVITE
>>     Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
>> SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
>>     Allow-Events: telephone-event
>>     Contact:<sip:73435327569 at X.X.16.83:5060>
>>     Content-Disposition: session;handling=required
>>     Content-Type: application/sdp
>>     Content-Length: 176
>>
>>     v=0
>>     o=CiscoSystemsSIP-GW-UserAgent 4564 1578 IN IP4 X.X.16.83
>>     s=SIP Call
>>     c=IN IP4 X.X.16.83
>>     t=0 0
>>     m=audio 17770 RTP/AVP 8
>>     c=IN IP4 X.X.16.83
>>     a=rtpmap:8 PCMA/8000
>>     ------------------------------------------------------------------------
>> 2011-01-20 15:30:32.401362 [INFO] sofia.c:729
>> sofia/epbx/73435327569 at X.X.16.83:5060 Update Callee ID to "Outbound
>> Call"<73435327569>
>> 2011-01-20 15:30:32.409363 [NOTICE] sofia.c:4739 Pre-Answer
>> sofia/epbx/73435327569 at X.X.16.83:5060!
>> 2011-01-20 15:30:32.415412 [INFO] switch_ivr_originate.c:3345 Sending
>> early media
>> 2011-01-20 15:30:32.416367 [NOTICE] mod_sofia.c:2252 Pre-Answer
>> sofia/epbx/50001 at default!
>> send 1145 bytes to udp/[X.X.29.123]:21556 at 10:30:32.418195:
>>     ------------------------------------------------------------------------
>>     SIP/2.0 183 Session Progress
>>     Via: SIP/2.0/UDP
>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>>     From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>     To: "73435327569"<sip:73435327569 at default>;tag=BScUp0QXHFBgB
>>     Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>     CSeq: 1 INVITE
>>     Contact:<sip:73435327569 at Y.Y.138.187:5060;transport=udp>
>>     User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
>> 16-36-04 -0500
>>     Accept: application/sdp
>>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>     Supported: timer, precondition, path, replaces
>>     Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>> sla, include-session-description, presence.winfo, message-summary, refer
>>     Content-Type: application/sdp
>>     Content-Disposition: session
>>     Content-Length: 181
>>     Remote-Party-ID: "73435327569"
>> <sip:73435327569 at Y.Y.138.187>;party=calling;privacy=off;screen=no
>>
>>     v=0
>>     o=FreeSWITCH 1627752513 1627752514 IN IP4 Y.Y.138.187
>>     s=FreeSWITCH
>>     c=IN IP4 Y.Y.138.187
>>     t=0 0
>>     m=audio 25806 RTP/AVP 8
>>     c=IN IP4 Y.Y.138.187
>>     a=rtpmap:8 PCMA/8000
>>
>>     ------------------------------------------------------------------------
>> recv 365 bytes from udp/[X.X.29.123]:21556 at 10:30:39.196791:
>>     ------------------------------------------------------------------------
>>     CANCEL sip:73435327569 at default SIP/2.0
>>     Via: SIP/2.0/UDP
>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
>>     To: "73435327569"<sip:73435327569 at default>
>>     From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>     Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>     CSeq: 1 CANCEL
>>     User-Agent: eyeBeam release 1102u stamp 52345
>>     Content-Length: 0
>>
>>     ------------------------------------------------------------------------
>> send 320 bytes to udp/[X.X.29.123]:21556 at 10:30:39.196905:
>>     ------------------------------------------------------------------------
>>     SIP/2.0 200 OK
>>     Via: SIP/2.0/UDP
>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>>     From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>     To: "73435327569"<sip:73435327569 at default>;tag=BScUp0QXHFBgB
>>     Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>     CSeq: 1 CANCEL
>>     Content-Length: 0
>>
>>     ------------------------------------------------------------------------
>> send 715 bytes to udp/[X.X.29.123]:21556 at 10:30:39.197002:
>>     ------------------------------------------------------------------------
>>     SIP/2.0 487 Request Terminated
>>     Via: SIP/2.0/UDP
>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>>     From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>     To: "73435327569"<sip:73435327569 at default>;tag=BScUp0QXHFBgB
>>     Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>     CSeq: 1 INVITE
>>     User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
>> 16-36-04 -0500
>>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>     Supported: timer, precondition, path, replaces
>>     Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>> sla, include-session-description, presence.winfo, message-summary, refer
>>     Content-Length: 0
>>
>>     ------------------------------------------------------------------------
>> 2011-01-20 15:30:39.196838 [NOTICE] sofia.c:5286 Hangup
>> sofia/epbx/50001 at default [CS_EXECUTE] [ORIGINATOR_CANCEL]
>> 2011-01-20 15:30:39.196838 [NOTICE] switch_ivr_bridge.c:653 Hangup
>> sofia/epbx/73435327569 at X.X.16.83:5060 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>> recv 331 bytes from udp/[X.X.29.123]:21556 at 10:30:39.198834:
>>     ------------------------------------------------------------------------
>>     ACK sip:73435327569 at default SIP/2.0
>>     Via: SIP/2.0/UDP
>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
>>     To: "73435327569"<sip:73435327569 at default>;tag=BScUp0QXHFBgB
>>     From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>     Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>     CSeq: 1 ACK
>>     Content-Length: 0
>>
>>     ------------------------------------------------------------------------
>> 2011-01-20 15:30:39.204838 [NOTICE] switch_core_session.c:1306 Session
>> 99 (sofia/epbx/50001 at default) Ended
>> 2011-01-20 15:30:39.204838 [NOTICE] switch_core_session.c:1308 Close
>> Channel sofia/epbx/50001 at default [CS_DESTROY]
>> send 372 bytes to udp/[X.X.16.83]:5060 at 10:30:39.212574:
>>     ------------------------------------------------------------------------
>>     CANCEL sip:73435327569 at X.X.16.83:5060 SIP/2.0
>>     Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>     Max-Forwards: 68
>>     From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>     To:<sip:73435327569 at X.X.16.83:5060>
>>     Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>     CSeq: 7423651 CANCEL
>>     Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>>     Content-Length: 0
>>
>>     ------------------------------------------------------------------------
>> 2011-01-20 15:30:39.212840 [NOTICE] switch_core_session.c:1306 Session
>> 100 (sofia/epbx/73435327569 at X.X.16.83:5060) Ended
>> 2011-01-20 15:30:39.212840 [NOTICE] switch_core_session.c:1308 Close
>> Channel sofia/epbx/73435327569 at X.X.16.83:5060 [CS_DESTROY]
>> recv 311 bytes from udp/[X.X.16.83]:5060 at 10:30:39.216838:
>>     ------------------------------------------------------------------------
>>     SIP/2.0 200 OK
>>     Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>     From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>     To:<sip:73435327569 at X.X.16.83:5060>
>>     Date: Thu, 20 Jan 2011 10:30:39 GMT
>>     Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>     Content-Length: 0
>>     CSeq: 7423651 CANCEL
>>
>>     ------------------------------------------------------------------------
>> recv 407 bytes from udp/[X.X.16.83]:5060 at 10:30:39.219180:
>>     ------------------------------------------------------------------------
>>     SIP/2.0 487 Request Cancelled
>>     Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>     From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>     To:<sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>>     Date: Thu, 20 Jan 2011 10:30:39 GMT
>>     Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>     Server: Cisco-SIPGateway/IOS-12.x
>>     CSeq: 7423651 INVITE
>>     Allow-Events: telephone-event
>>     Content-Length: 0
>>
>>     ------------------------------------------------------------------------
>> send 334 bytes to udp/[X.X.16.83]:5060 at 10:30:39.219276:
>>     ------------------------------------------------------------------------
>>     ACK sip:73435327569 at X.X.16.83:5060 SIP/2.0
>>     Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>     Max-Forwards: 68
>>     From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>     To:<sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>>     Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>     CSeq: 7423651 ACK
>>     Content-Length: 0
>>
>>     ------------------------------------------------------------------------
>>
>>
>> --
>> ? ?????????,
>>    ????? ?????????
>>    ??? "??????"
>>    (3435) 494991
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> --
>> ? ?????????,
>>    ????? ?????????
>>    ??? "??????"
>>    (3435) 494991
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> --
>> ? ?????????,
>>    ????? ?????????
>>    ??? "??????"
>>    (3435) 494991
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From Nabble at slickdeals.endjunk.com  Thu Jan 20 21:44:17 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 20 Jan 2011 10:44:17 -0800 (PST)
Subject: [Freeswitch-users] call barging
In-Reply-To: <AANLkTinDdXEkE7_-1vcnsfgOTbtM7i8=NUgb82d8C-iB@mail.gmail.com>
References: <AANLkTinMqT+Qs1guPphYWqN5okMAD9c1K9=Jey6uepRw@mail.gmail.com>
	<AANLkTimrbwUS7s1G+catXL-XVTr_pg9yOWorR04MNFY=@mail.gmail.com>
	<AANLkTinNrHFKY22Amsv-mMw=wMcxmKheo-WxT8kYTrqC@mail.gmail.com>
	<1295531363114-5943853.post@n2.nabble.com>
	<AANLkTi=sZ9G-QNHJXUFiphzG04RYSmd1r_y6DiEF7J3h@mail.gmail.com>
	<1295539380742-5944311.post@n2.nabble.com>
	<AANLkTinDdXEkE7_-1vcnsfgOTbtM7i8=NUgb82d8C-iB@mail.gmail.com>
Message-ID: <1295549057834-5944895.post@n2.nabble.com>



Steven Ayre wrote:
> 
>    1. 2011-01-20 10:42:47.723158 [DEBUG] switch_core_media_bug.c:360
>    Attaching BUG to sofia/internal/1000 at 192.168.1.123
> 
> Looks like it is eavesdropping on the other channel, I'm not sure why
> you're
> not getting any audio but the eavesdrop app itself does appear to be
> working.
> 
> I take it both phones can dial in and both speak and hear audio?
> 
> -Steve
Yes. Each extension can call each other without any audio problems.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/call-barging-tp5943099p5944895.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kris at kriskinc.com  Thu Jan 20 21:47:18 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 20 Jan 2011 13:47:18 -0500
Subject: [Freeswitch-users] uuid_media on?
Message-ID: <AANLkTi=Yhb=wnG1cRtS2CMXj5t-bhSZCM7-azGARq70p@mail.gmail.com>

Why doesn't uuid_media support an "on" argument to re-INVITE
FreeSWITCH back into the media path on demand?

-- 
Kristian Kielhofner


From brian at freeswitch.org  Thu Jan 20 21:49:35 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 20 Jan 2011 12:49:35 -0600
Subject: [Freeswitch-users] uuid_media on?
In-Reply-To: <AANLkTi=Yhb=wnG1cRtS2CMXj5t-bhSZCM7-azGARq70p@mail.gmail.com>
References: <AANLkTi=Yhb=wnG1cRtS2CMXj5t-bhSZCM7-azGARq70p@mail.gmail.com>
Message-ID: <248B52E1-EE5D-469E-BAA7-020809D8454B@freeswitch.org>

uuid_meida <uuid> will do exactly that.

/b

On Jan 20, 2011, at 12:47 PM, Kristian Kielhofner wrote:

> Why doesn't uuid_media support an "on" argument to re-INVITE
> FreeSWITCH back into the media path on demand?



From msc at freeswitch.org  Thu Jan 20 21:52:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 20 Jan 2011 10:52:02 -0800
Subject: [Freeswitch-users] Possible mod_smpp
In-Reply-To: <4D37457A.6040607@coppice.org>
References: <AANLkTiksHkyBnVnbRamuNDK=kxfnE6jCoQdV1eWhXgbr@mail.gmail.com>
	<4D37457A.6040607@coppice.org>
Message-ID: <AANLkTimjx4Ggj5Aw2zM3eU0PQTOcy2ShY_2u1JDZFLhN@mail.gmail.com>

Our IRC buddy Delphiworld wants it for something. Math quoted him a price to
do it and he decided that he doesn't want to pay it all himself so he's
fishing around for other interested parties. Judging by the lack of response
to my message I'd have to say that interest is still very weak...

-MC

On Wed, Jan 19, 2011 at 12:11 PM, Steve Underwood <steveu at coppice.org>wrote:

> On 01/20/2011 03:15 AM, Michael Collins wrote:
> > Hello all,
> >
> > If you are interested in giving monetary support to have a
> > professional software firm create mod_smpp then please contact me off
> > list and I will give you more details.
> >
> > Thanks,
> > Michael
> SMPP as in Short Message Peer to Peer? I have floated the idea of open
> sourcing an implementation of that a couple of times, but interest
> seemed weak and I never bothered.
>
> Steve
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Thu Jan 20 21:55:29 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 20 Jan 2011 18:55:29 +0000
Subject: [Freeswitch-users] call barging
In-Reply-To: <1295549057834-5944895.post@n2.nabble.com>
References: <AANLkTinMqT+Qs1guPphYWqN5okMAD9c1K9=Jey6uepRw@mail.gmail.com>
	<AANLkTimrbwUS7s1G+catXL-XVTr_pg9yOWorR04MNFY=@mail.gmail.com>
	<AANLkTinNrHFKY22Amsv-mMw=wMcxmKheo-WxT8kYTrqC@mail.gmail.com>
	<1295531363114-5943853.post@n2.nabble.com>
	<AANLkTi=sZ9G-QNHJXUFiphzG04RYSmd1r_y6DiEF7J3h@mail.gmail.com>
	<1295539380742-5944311.post@n2.nabble.com>
	<AANLkTinDdXEkE7_-1vcnsfgOTbtM7i8=NUgb82d8C-iB@mail.gmail.com>
	<1295549057834-5944895.post@n2.nabble.com>
Message-ID: <1C581132-1B25-4787-90C4-E730BDBE276C@gmail.com>

Via freeswitch? Not direct.

Steve on iPhone

On 20 Jan 2011, at 18:44, mazilo <Nabble at slickdeals.endjunk.com> wrote:

> 
> 
> Steven Ayre wrote:
>> 
>>   1. 2011-01-20 10:42:47.723158 [DEBUG] switch_core_media_bug.c:360
>>   Attaching BUG to sofia/internal/1000 at 192.168.1.123
>> 
>> Looks like it is eavesdropping on the other channel, I'm not sure why
>> you're
>> not getting any audio but the eavesdrop app itself does appear to be
>> working.
>> 
>> I take it both phones can dial in and both speak and hear audio?
>> 
>> -Steve
> Yes. Each extension can call each other without any audio problems.
> 
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> -- 
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/call-barging-tp5943099p5944895.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From kris at kriskinc.com  Thu Jan 20 22:06:01 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 20 Jan 2011 14:06:01 -0500
Subject: [Freeswitch-users] uuid_media on?
In-Reply-To: <248B52E1-EE5D-469E-BAA7-020809D8454B@freeswitch.org>
References: <AANLkTi=Yhb=wnG1cRtS2CMXj5t-bhSZCM7-azGARq70p@mail.gmail.com>
	<248B52E1-EE5D-469E-BAA7-020809D8454B@freeswitch.org>
Message-ID: <AANLkTin1dsM=ApgoJY0Xc4PYXoMCvf2A3JBY6D_dRnLm@mail.gmail.com>

Brian,

  Awesome!  I'll update the wiki with this info.

On Thu, Jan 20, 2011 at 1:49 PM, Brian West <brian at freeswitch.org> wrote:
> uuid_meida <uuid> will do exactly that.
>
> /b

-- 
Kristian Kielhofner


From msc at freeswitch.org  Thu Jan 20 23:08:42 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 20 Jan 2011 12:08:42 -0800
Subject: [Freeswitch-users] Caller ID Number not going out on PRI
In-Reply-To: <4D35F532.2080802@gmail.com>
References: <4D35F532.2080802@gmail.com>
Message-ID: <AANLkTimsZ04PiNVupqwSymVmW6j=khrLin7skAxd4icT@mail.gmail.com>

I'd recommend pastebin of your relevant dialplan and freetdm configs. It
would also help to see a debug console log trace of a call that is
displaying the behavior.
-MC

On Tue, Jan 18, 2011 at 12:16 PM, Phone <phone.bytes at gmail.com> wrote:

> A few upgrades ago, the Caller ID Number on outbound calls stopped
> working.  Now it always displays the extension number, where before it
> was using the value specified in the directory entry.  Seems like maybe
> the variables are not passing from the directory entry to the dial plan.
>
> Any ideas?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Nabble at slickdeals.endjunk.com  Thu Jan 20 23:16:02 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 20 Jan 2011 12:16:02 -0800 (PST)
Subject: [Freeswitch-users] call barging
In-Reply-To: <1C581132-1B25-4787-90C4-E730BDBE276C@gmail.com>
References: <AANLkTinMqT+Qs1guPphYWqN5okMAD9c1K9=Jey6uepRw@mail.gmail.com>
	<AANLkTimrbwUS7s1G+catXL-XVTr_pg9yOWorR04MNFY=@mail.gmail.com>
	<AANLkTinNrHFKY22Amsv-mMw=wMcxmKheo-WxT8kYTrqC@mail.gmail.com>
	<1295531363114-5943853.post@n2.nabble.com>
	<AANLkTi=sZ9G-QNHJXUFiphzG04RYSmd1r_y6DiEF7J3h@mail.gmail.com>
	<1295539380742-5944311.post@n2.nabble.com>
	<AANLkTinDdXEkE7_-1vcnsfgOTbtM7i8=NUgb82d8C-iB@mail.gmail.com>
	<1295549057834-5944895.post@n2.nabble.com>
	<1C581132-1B25-4787-90C4-E730BDBE276C@gmail.com>
Message-ID: <1295554562750-5945210.post@n2.nabble.com>



Steven Ayre wrote:
> 
> Via freeswitch? Not direct.
Both the Ekiga softphone and Uniden UIP1869V device are configured as
extensions 1000 and 1007, respectively. I can call from extension 1000 to
1007 by dialing 1000 and 1007 to 1000 by dialing 1007 without an audio
problems (if that's what you meant dialing via FS).

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/call-barging-tp5943099p5945210.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From lfurrea at gmail.com  Thu Jan 20 22:50:05 2011
From: lfurrea at gmail.com (Luis F Urrea)
Date: Fri, 21 Jan 2011 03:50:05 +0800
Subject: [Freeswitch-users] FW:RE
Message-ID: <AANLkTim8O+nvUUEP7Fhb8nHQqVp26ssH3TdD_oScVOW4@mail.gmail.com>

dear, I found a good company several days ago, and try to buy some goods,
and I received satisfactory item 5 days later.
Introducing to you: fallinele.com , maybe also useful for you.


From steveayre at gmail.com  Thu Jan 20 23:54:38 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 20 Jan 2011 20:54:38 +0000
Subject: [Freeswitch-users] call barging
In-Reply-To: <1295554562750-5945210.post@n2.nabble.com>
References: <AANLkTinMqT+Qs1guPphYWqN5okMAD9c1K9=Jey6uepRw@mail.gmail.com>
	<AANLkTimrbwUS7s1G+catXL-XVTr_pg9yOWorR04MNFY=@mail.gmail.com>
	<AANLkTinNrHFKY22Amsv-mMw=wMcxmKheo-WxT8kYTrqC@mail.gmail.com>
	<1295531363114-5943853.post@n2.nabble.com>
	<AANLkTi=sZ9G-QNHJXUFiphzG04RYSmd1r_y6DiEF7J3h@mail.gmail.com>
	<1295539380742-5944311.post@n2.nabble.com>
	<AANLkTinDdXEkE7_-1vcnsfgOTbtM7i8=NUgb82d8C-iB@mail.gmail.com>
	<1295549057834-5944895.post@n2.nabble.com>
	<1C581132-1B25-4787-90C4-E730BDBE276C@gmail.com>
	<1295554562750-5945210.post@n2.nabble.com>
Message-ID: <AANLkTik7b7E6NSUQ97HQz1KLVFLBcmXNdC4TqSjt5Kcx@mail.gmail.com>

Yes it was. Hmmm....

On 20 January 2011 20:16, mazilo <Nabble at slickdeals.endjunk.com> wrote:

>
>
> Steven Ayre wrote:
> >
> > Via freeswitch? Not direct.
> Both the Ekiga softphone and Uniden UIP1869V device are configured as
> extensions 1000 and 1007, respectively. I can call from extension 1000 to
> 1007 by dialing 1000 and 1007 to 1000 by dialing 1007 without an audio
> problems (if that's what you meant dialing via FS).
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/call-barging-tp5943099p5945210.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From peter at phpwerks.com  Fri Jan 21 00:10:24 2011
From: peter at phpwerks.com (Peter Brenner)
Date: Thu, 20 Jan 2011 16:10:24 -0500
Subject: [Freeswitch-users] Making a Call from PHP
Message-ID: <4D38A4C0.20800@phpwerks.com>

Very new to Freeswitch and I am trying to work through some examples of 
connecting/interacting with Freeswitch from PHP.  The example that I am 
trying to work through is one where a user enters 2 extension numbers on 
a page, submits the page which then originates a call from one extension 
to the other.

I used the follow in example as a starting point:

http://wiki.freeswitch.org/wiki/PHP_Event_Socket
I changed the command:
$cmd = "api help";

To:

$cmd = "api originate sofia/internal/1000&bridge(sofia/internal/1001)";

I want to call from extension 1000 to 1001

I receive the following error

2011-01-20 16:01:16.482373 [WARNING] mod_sofia.c:4022 Cannot locate registered user 1000 at internal
2011-01-20 16:01:16.482373 [NOTICE] mod_sofia.c:4221 Close Channel N/A [CS_NEW]
2011-01-20 16:01:16.482373 [ERR] switch_ivr_originate.c:2628 Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED]

If I change the command to
$cmd = "api originate sofia/internal/1000 at 192.168.1.211&bridge(sofia/internal/1001)";


2011-01-20 16:04:00.602758 [NOTICE] switch_channel.c:808 New Channel sofia/internal/1000 at 192.168.1.211 [c721e1e2-7eb7-46d5-9507-dcf02ac33828]
2011-01-20 16:04:00.602758 [NOTICE] switch_channel.c:808 New Channel sofia/internal/0000000000 at 192.168.1.211 [b7b163eb-3c90-4b86-9fde-070abdfbb45b]
2011-01-20 16:04:00.612832 [INFO] mod_dialplan_xml.c:331 Processing<0000000000>->1000 in context public
2011-01-20 16:04:00.625071 [ERR] sofia.c:5869 Cannot Blind Transfer 1 Legged calls
2011-01-20 16:04:00.625071 [NOTICE] sofia.c:5286 Hangup sofia/internal/1000 at 192.168.1.211 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE]
2011-01-20 16:04:00.654406 [NOTICE] switch_core_session.c:1306 Session 2 (sofia/internal/1000 at 192.168.1.211) Ended
2011-01-20 16:04:00.654406 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/1000 at 192.168.1.211 [CS_DESTROY]
2011-01-20 16:04:00.725081 [NOTICE] switch_core_state_machine.c:189 sofia/internal/0000000000 at 192.168.1.211 has executed the last dialplan instruction, hanging up.
2011-01-20 16:04:00.725081 [NOTICE] switch_core_state_machine.c:191 Hangup sofia/internal/0000000000 at 192.168.1.211 [CS_EXECUTE] [NORMAL_CLEARING]
2011-01-20 16:04:00.730022 [NOTICE] switch_core_session.c:1306 Session 3 (sofia/internal/0000000000 at 192.168.1.211) Ended
2011-01-20 16:04:00.730022 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/0000000000 at 192.168.1.211 [CS_DESTROY]

Here is the output from sofia status

                      Name          Type                                       Data      State
=================================================================================================
                  external       profile           sip:mod_sofia at 192.168.1.211:5080      RUNNING (0)
     external::example.com       gateway                    sip:joeuser at example.com      NOREG
             internal-ipv6       profile                   sip:mod_sofia@[::1]:5060      RUNNING (0)
                  internal       profile           sip:mod_sofia at 192.168.1.211:5060      RUNNING (0)
             192.168.1.211         alias                                   internal      ALIASED
=================================================================================================
3 profiles 1 alias

Any advice or help would be greatly appreciated.

thanks!
Peter




From msc at freeswitch.org  Fri Jan 21 00:29:51 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 20 Jan 2011 13:29:51 -0800
Subject: [Freeswitch-users] PCMA stranges
In-Reply-To: <4D386528.2010506@tagnet.ru>
References: <4D380FF8.7080705@tagnet.ru>
	<E676EB80-7927-4061-BC2C-497DBA81D75E@ipeva.fr>
	<4D381E1F.7040202@tagnet.ru> <4D384196.105@tagnet.ru>
	<AANLkTikVAt6t8K4fxg0GXisw=0_+s5gBYe9Hyvep5KN2@mail.gmail.com>
	<4D386528.2010506@tagnet.ru>
Message-ID: <AANLkTi=wcUEe89em_-c2DaFbBPunWum9Oh10yjj1KTXw@mail.gmail.com>

Have you compared the SIP traces between the working g711u and the
non-working g711a? What seems to be different between the two?
-MC

On Thu, Jan 20, 2011 at 8:39 AM, Boris Kovalenko <boris at tagnet.ru> wrote:

> Hello!
>
>     Yes, I know the media is ignored, so I wondering about this codec
> strangenes. What test I have done:
>
> SoftPhone (g711ulaw) -> FS -> Cisco 5350 : working
> SoftPhone (g711alaw, proxy_media=true) -> FS -> Cisco 5350 : not working
> SoftPhone (g711alaw, proxy_media=false) -> FS -> Cisco 5350 : working
> SoftPhone (g711alaw) -> Cisco 5350 : working
>
>
> > proxy media means media is ignored, its not ever touched.  Its
> > designed to transparently pass media even if FS is unaware of the
> > codec.
> > So if you say you are using proxy media and you have codec specific
> > problems it seems peculiar.
> >
> >
> > On Thu, Jan 20, 2011 at 8:07 AM, Boris Kovalenko<boris at tagnet.ru>
>  wrote:
> >> Hello!
> >>
> >>      I found that problem is somewhere inside freeswitch. Direct call
> between
> >> SoftPhone and Cisco is working fine.
> >>
> >> Hello!
> >>
> >>      David, may You explain why this problem is with G711A only? When I
> use
> >> G711U there are no problems and call is established and I may talk.
> >>
> >> The Cisco never sends the 200/OK after the 183, so the call is not
> >> established.
> >> David Ponzone  Direction Technique
> >> email: david.ponzone at ipeva.fr
> >> tel:      01 74 03 18 97
> >> gsm:   06 66 98 76 34
> >> Service Client IPeva
> >> tel:      0811 46 26 26
> >> www.ipeva.fr  -   www.ipeva-studio.com
> >> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> >> l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion
> >> non autoris?e est interdite. Tout message ?lectronique est susceptible
> >> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il
> >> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> >> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> >>
> >>
> >>
> >> Le 20/01/2011 ? 11:35, Boris Kovalenko a ?crit :
> >>
> >> Hello!
> >>
> >>      My network configuration is:
> >> Softphone (eyeBeam 1.5) with only G711alaw enabled ->  FreeSWITCH
> Version
> >> 1.0.head (git-0cf1d54 2011-01-19 16-36-04 -0500) ->  Cisco AS5350
> >>
> >> proxy_media=true in profile.
> >> So, when I do a test call, there is SDP but no RTP between FS and CISCO,
> >> so I can't hear voice. With G711ulaw there are no problems. What is
> >> wrong? Siptrace below:
> >>
> >>
> ------------------------------------------------------------------------
> >>     INVITE sip:73435327569 at default SIP/2.0
> >>     Via: SIP/2.0/UDP
> >>
> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
> >>     Max-Forwards: 70
> >>     Contact:<sip:50001 at X.X.29.123:21556>
> >>     To: "73435327569"<sip:73435327569 at default>
> >>     From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
> >>     Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
> >>     CSeq: 1 INVITE
> >>     Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> >> SUBSCRIBE, INFO
> >>     Content-Type: application/sdp
> >>     User-Agent: eyeBeam release 1102u stamp 52345
> >>     Content-Length: 337
> >>
> >>     v=0
> >>     o=- 1 2 IN IP4 192.168.3.253
> >>     s=CounterPath eyeBeam 1.5
> >>     c=IN IP4 192.168.3.253
> >>     t=0 0
> >>     m=audio 40032 RTP/AVP 8 101
> >>     a=alt:1 3 : XdMyLUqb Fs2PLxKV 192.168.3.253 40032
> >>     a=alt:2 2 : 2qrc/CQ7 Z/9XFQZl 172.16.3.253 40032
> >>     a=alt:3 1 : 8BOwI53P +ENGEr/U X.X.29.123 40032
> >>     a=fmtp:101 0-15
> >>     a=rtpmap:101 telephone-event/8000
> >>     a=sendrecv
> >>
> ------------------------------------------------------------------------
> >> send 385 bytes to udp/[X.X.29.123]:21556 at 10:30:31.313366:
> >>
> ------------------------------------------------------------------------
> >>     SIP/2.0 100 Trying
> >>     Via: SIP/2.0/UDP
> >>
> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
> >>     From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
> >>     To: "73435327569"<sip:73435327569 at default>
> >>     Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
> >>     CSeq: 1 INVITE
> >>     User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
> >> 16-36-04 -0500
> >>     Content-Length: 0
> >>
> >>
> ------------------------------------------------------------------------
> >> 2011-01-20 15:30:31.313448 [NOTICE] switch_channel.c:808 New Channel
> >> sofia/epbx/50001 at default [95698968-8cf9-40a2-91ad-1322fcfc76af]
> >> 2011-01-20 15:30:31.322338 [INFO] mod_dialplan_xml.c:331 Processing
> >> TAGNet<50001>->73435327569 in context public
> >> 2011-01-20 15:30:31.323362 [INFO] mod_dialplan_xml.c:331 Processing
> >> TAGNet<50001>->ext_translate_extsrc in context features
> >> 2011-01-20 15:30:31.324357 [NOTICE] switch_ivr.c:1606 Transfer
> >> sofia/epbx/50001 at default to XML[73435327569 at top.ctx]
> >> 2011-01-20 15:30:31.325352 [INFO] mod_dialplan_xml.c:331 Processing
> >> TAGNet<50001>->73435327569 in context top.ctx
> >> 2011-01-20 15:30:31.326344 [NOTICE] mod_dptools.c:1174 [top.ctx] -
> ext_local
> >> 2011-01-20 15:30:31.329360 [NOTICE] switch_channel.c:808 New Channel
> >> sofia/epbx/73435327569 at X.X.16.83:5060
> [b9654c75-039a-4780-bc00-0fae65f92a9a]
> >> send 1276 bytes to udp/[X.X.16.83]:5060 at 10:30:31.330623:
> >>
> ------------------------------------------------------------------------
> >>     INVITE sip:73435327569 at X.X.16.83:5060 SIP/2.0
> >>     Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
> >>     Max-Forwards: 68
> >>     From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
> >>     To:<sip:73435327569 at X.X.16.83:5060>
> >>     Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
> >>     CSeq: 7423651 INVITE
> >>     Contact:<sip:mod_sofia at Y.Y.138.187:5060>
> >>     User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
> >> 16-36-04 -0500
> >>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >>     Supported: timer, precondition, path, replaces
> >>     Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
> >> sla, include-session-description, presence.winfo, message-summary, refer
> >>     Content-Type: application/sdp
> >>     Content-Disposition: session
> >>     Content-Length: 341
> >>     X-FS-Support: update_display
> >>     Remote-Party-ID: "TAGNet"
> >> <sip:50001 at Y.Y.138.187>;party=calling;screen=yes;privacy=off
> >>
> >>     v=0
> >>     o=FreeSWITCH 1627720692 1627720693 IN IP4 Y.Y.138.187
> >>     s=FreeSWITCH
> >>     c=IN IP4 Y.Y.138.187
> >>     t=0 0
> >>     m=audio 23442 RTP/AVP 8 101
> >>     a=rtpmap:101 telephone-event/8000
> >>     a=fmtp:101 0-15
> >>     a=alt:1 3 : XdMyLUqb Fs2PLxKV 192.168.3.253 40032
> >>     a=alt:2 2 : 2qrc/CQ7 Z/9XFQZl 172.16.3.253 40032
> >>     a=alt:3 1 : 8BOwI53P +ENGEr/U X.X.29.123 40032
> >>
> ------------------------------------------------------------------------
> >> recv 396 bytes from udp/[X.X.16.83]:5060 at 10:30:31.341106:
> >>
> ------------------------------------------------------------------------
> >>     SIP/2.0 100 Trying
> >>     Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
> >>     From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
> >>     To:<sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
> >>     Date: Thu, 20 Jan 2011 10:30:31 GMT
> >>     Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
> >>     Server: Cisco-SIPGateway/IOS-12.x
> >>     CSeq: 7423651 INVITE
> >>     Allow-Events: telephone-event
> >>     Content-Length: 0
> >>
> >>
> ------------------------------------------------------------------------
> >> recv 814 bytes from udp/[X.X.16.83]:5060 at 10:30:32.401494:
> >>
> ------------------------------------------------------------------------
> >>     SIP/2.0 183 Session Progress
> >>     Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
> >>     From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
> >>     To:<sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
> >>     Date: Thu, 20 Jan 2011 10:30:31 GMT
> >>     Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
> >>     Server: Cisco-SIPGateway/IOS-12.x
> >>     CSeq: 7423651 INVITE
> >>     Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
> >> SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
> >>     Allow-Events: telephone-event
> >>     Contact:<sip:73435327569 at X.X.16.83:5060>
> >>     Content-Disposition: session;handling=required
> >>     Content-Type: application/sdp
> >>     Content-Length: 176
> >>
> >>     v=0
> >>     o=CiscoSystemsSIP-GW-UserAgent 4564 1578 IN IP4 X.X.16.83
> >>     s=SIP Call
> >>     c=IN IP4 X.X.16.83
> >>     t=0 0
> >>     m=audio 17770 RTP/AVP 8
> >>     c=IN IP4 X.X.16.83
> >>     a=rtpmap:8 PCMA/8000
> >>
> ------------------------------------------------------------------------
> >> 2011-01-20 15:30:32.401362 [INFO] sofia.c:729
> >> sofia/epbx/73435327569 at X.X.16.83:5060 Update Callee ID to "Outbound
> >> Call"<73435327569>
> >> 2011-01-20 15:30:32.409363 [NOTICE] sofia.c:4739 Pre-Answer
> >> sofia/epbx/73435327569 at X.X.16.83:5060!
> >> 2011-01-20 15:30:32.415412 [INFO] switch_ivr_originate.c:3345 Sending
> >> early media
> >> 2011-01-20 15:30:32.416367 [NOTICE] mod_sofia.c:2252 Pre-Answer
> >> sofia/epbx/50001 at default!
> >> send 1145 bytes to udp/[X.X.29.123]:21556 at 10:30:32.418195:
> >>
> ------------------------------------------------------------------------
> >>     SIP/2.0 183 Session Progress
> >>     Via: SIP/2.0/UDP
> >>
> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
> >>     From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
> >>     To: "73435327569"<sip:73435327569 at default>;tag=BScUp0QXHFBgB
> >>     Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
> >>     CSeq: 1 INVITE
> >>     Contact:<sip:73435327569 at Y.Y.138.187:5060;transport=udp>
> >>     User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
> >> 16-36-04 -0500
> >>     Accept: application/sdp
> >>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >>     Supported: timer, precondition, path, replaces
> >>     Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
> >> sla, include-session-description, presence.winfo, message-summary, refer
> >>     Content-Type: application/sdp
> >>     Content-Disposition: session
> >>     Content-Length: 181
> >>     Remote-Party-ID: "73435327569"
> >> <sip:73435327569 at Y.Y.138.187>;party=calling;privacy=off;screen=no
> >>
> >>     v=0
> >>     o=FreeSWITCH 1627752513 1627752514 IN IP4 Y.Y.138.187
> >>     s=FreeSWITCH
> >>     c=IN IP4 Y.Y.138.187
> >>     t=0 0
> >>     m=audio 25806 RTP/AVP 8
> >>     c=IN IP4 Y.Y.138.187
> >>     a=rtpmap:8 PCMA/8000
> >>
> >>
> ------------------------------------------------------------------------
> >> recv 365 bytes from udp/[X.X.29.123]:21556 at 10:30:39.196791:
> >>
> ------------------------------------------------------------------------
> >>     CANCEL sip:73435327569 at default SIP/2.0
> >>     Via: SIP/2.0/UDP
> >>
> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
> >>     To: "73435327569"<sip:73435327569 at default>
> >>     From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
> >>     Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
> >>     CSeq: 1 CANCEL
> >>     User-Agent: eyeBeam release 1102u stamp 52345
> >>     Content-Length: 0
> >>
> >>
> ------------------------------------------------------------------------
> >> send 320 bytes to udp/[X.X.29.123]:21556 at 10:30:39.196905:
> >>
> ------------------------------------------------------------------------
> >>     SIP/2.0 200 OK
> >>     Via: SIP/2.0/UDP
> >>
> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
> >>     From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
> >>     To: "73435327569"<sip:73435327569 at default>;tag=BScUp0QXHFBgB
> >>     Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
> >>     CSeq: 1 CANCEL
> >>     Content-Length: 0
> >>
> >>
> ------------------------------------------------------------------------
> >> send 715 bytes to udp/[X.X.29.123]:21556 at 10:30:39.197002:
> >>
> ------------------------------------------------------------------------
> >>     SIP/2.0 487 Request Terminated
> >>     Via: SIP/2.0/UDP
> >>
> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
> >>     From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
> >>     To: "73435327569"<sip:73435327569 at default>;tag=BScUp0QXHFBgB
> >>     Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
> >>     CSeq: 1 INVITE
> >>     User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
> >> 16-36-04 -0500
> >>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >>     Supported: timer, precondition, path, replaces
> >>     Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
> >> sla, include-session-description, presence.winfo, message-summary, refer
> >>     Content-Length: 0
> >>
> >>
> ------------------------------------------------------------------------
> >> 2011-01-20 15:30:39.196838 [NOTICE] sofia.c:5286 Hangup
> >> sofia/epbx/50001 at default [CS_EXECUTE] [ORIGINATOR_CANCEL]
> >> 2011-01-20 15:30:39.196838 [NOTICE] switch_ivr_bridge.c:653 Hangup
> >> sofia/epbx/73435327569 at X.X.16.83:5060 [CS_EXCHANGE_MEDIA]
> [NORMAL_CLEARING]
> >> recv 331 bytes from udp/[X.X.29.123]:21556 at 10:30:39.198834:
> >>
> ------------------------------------------------------------------------
> >>     ACK sip:73435327569 at default SIP/2.0
> >>     Via: SIP/2.0/UDP
> >>
> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
> >>     To: "73435327569"<sip:73435327569 at default>;tag=BScUp0QXHFBgB
> >>     From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
> >>     Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
> >>     CSeq: 1 ACK
> >>     Content-Length: 0
> >>
> >>
> ------------------------------------------------------------------------
> >> 2011-01-20 15:30:39.204838 [NOTICE] switch_core_session.c:1306 Session
> >> 99 (sofia/epbx/50001 at default) Ended
> >> 2011-01-20 15:30:39.204838 [NOTICE] switch_core_session.c:1308 Close
> >> Channel sofia/epbx/50001 at default [CS_DESTROY]
> >> send 372 bytes to udp/[X.X.16.83]:5060 at 10:30:39.212574:
> >>
> ------------------------------------------------------------------------
> >>     CANCEL sip:73435327569 at X.X.16.83:5060 SIP/2.0
> >>     Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
> >>     Max-Forwards: 68
> >>     From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
> >>     To:<sip:73435327569 at X.X.16.83:5060>
> >>     Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
> >>     CSeq: 7423651 CANCEL
> >>     Reason: Q.850;cause=16;text="NORMAL_CLEARING"
> >>     Content-Length: 0
> >>
> >>
> ------------------------------------------------------------------------
> >> 2011-01-20 15:30:39.212840 [NOTICE] switch_core_session.c:1306 Session
> >> 100 (sofia/epbx/73435327569 at X.X.16.83:5060) Ended
> >> 2011-01-20 15:30:39.212840 [NOTICE] switch_core_session.c:1308 Close
> >> Channel sofia/epbx/73435327569 at X.X.16.83:5060 [CS_DESTROY]
> >> recv 311 bytes from udp/[X.X.16.83]:5060 at 10:30:39.216838:
> >>
> ------------------------------------------------------------------------
> >>     SIP/2.0 200 OK
> >>     Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
> >>     From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
> >>     To:<sip:73435327569 at X.X.16.83:5060>
> >>     Date: Thu, 20 Jan 2011 10:30:39 GMT
> >>     Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
> >>     Content-Length: 0
> >>     CSeq: 7423651 CANCEL
> >>
> >>
> ------------------------------------------------------------------------
> >> recv 407 bytes from udp/[X.X.16.83]:5060 at 10:30:39.219180:
> >>
> ------------------------------------------------------------------------
> >>     SIP/2.0 487 Request Cancelled
> >>     Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
> >>     From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
> >>     To:<sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
> >>     Date: Thu, 20 Jan 2011 10:30:39 GMT
> >>     Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
> >>     Server: Cisco-SIPGateway/IOS-12.x
> >>     CSeq: 7423651 INVITE
> >>     Allow-Events: telephone-event
> >>     Content-Length: 0
> >>
> >>
> ------------------------------------------------------------------------
> >> send 334 bytes to udp/[X.X.16.83]:5060 at 10:30:39.219276:
> >>
> ------------------------------------------------------------------------
> >>     ACK sip:73435327569 at X.X.16.83:5060 SIP/2.0
> >>     Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
> >>     Max-Forwards: 68
> >>     From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
> >>     To:<sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
> >>     Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
> >>     CSeq: 7423651 ACK
> >>     Content-Length: 0
> >>
> >>
> ------------------------------------------------------------------------
> >>
> >>
> >> --
> >> ? ?????????,
> >>    ????? ?????????
> >>    ??? "??????"
> >>    (3435) 494991
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >> --
> >> ? ?????????,
> >>    ????? ?????????
> >>    ??? "??????"
> >>    (3435) 494991
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >> --
> >> ? ?????????,
> >>    ????? ?????????
> >>    ??? "??????"
> >>    (3435) 494991
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
>
>
> --
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   (3435) 494991
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Fri Jan 21 00:35:47 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 20 Jan 2011 13:35:47 -0800
Subject: [Freeswitch-users] Making a Call from PHP
In-Reply-To: <4D38A4C0.20800@phpwerks.com>
References: <4D38A4C0.20800@phpwerks.com>
Message-ID: <AANLkTinCLKSKJQHBCZWPcnyuj+BZdYpfsZuE3VyL+N=K@mail.gmail.com>

I believe we talked about this on IRC but for posterity's sake I'll repeat
here: your originate syntax is not correct. It can be confusing because
there are several ways to accomplish the same thing. In your case I
recommend using this syntax:

api originate user/1000 1001

The first argument to originate is a dialstring. If you know that you are
going to be dialing a locally registered user then the "user" channel is
easier than saying "sofia/internal/1000%${domain}". The second argument to
originate is a dialplan extension or and application to execute. In the
above example the first leg calls user/1000 and when he/she answers it then
sends the other leg through the dialplan as if user 1000 had dialed "1001"
and pressed send. I hope that makes sense.

Let us know if you continue to have issues with this. You are VERY close to
having this working - just a few characters in the dialstring.

-MC

On Thu, Jan 20, 2011 at 1:10 PM, Peter Brenner <peter at phpwerks.com> wrote:

> Very new to Freeswitch and I am trying to work through some examples of
> connecting/interacting with Freeswitch from PHP.  The example that I am
> trying to work through is one where a user enters 2 extension numbers on
> a page, submits the page which then originates a call from one extension
> to the other.
>
> I used the follow in example as a starting point:
>
> http://wiki.freeswitch.org/wiki/PHP_Event_Socket
> I changed the command:
> $cmd = "api help";
>
> To:
>
> $cmd = "api originate sofia/internal/1000&bridge(sofia/internal/1001)";
>
> I want to call from extension 1000 to 1001
>
> I receive the following error
>
> 2011-01-20 16:01:16.482373 [WARNING] mod_sofia.c:4022 Cannot locate
> registered user 1000 at internal
> 2011-01-20 16:01:16.482373 [NOTICE] mod_sofia.c:4221 Close Channel N/A
> [CS_NEW]
> 2011-01-20 16:01:16.482373 [ERR] switch_ivr_originate.c:2628 Cannot create
> outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED]
>
> If I change the command to
> $cmd = "api originate sofia/internal/1000 at 192.168.1.211
> &bridge(sofia/internal/1001)";
>
>
> 2011-01-20 16:04:00.602758 [NOTICE] switch_channel.c:808 New Channel
> sofia/internal/1000 at 192.168.1.211 [c721e1e2-7eb7-46d5-9507-dcf02ac33828]
> 2011-01-20 16:04:00.602758 [NOTICE] switch_channel.c:808 New Channel
> sofia/internal/0000000000 at 192.168.1.211[b7b163eb-3c90-4b86-9fde-070abdfbb45b]
> 2011-01-20 16:04:00.612832 [INFO] mod_dialplan_xml.c:331
> Processing<0000000000>->1000 in context public
> 2011-01-20 16:04:00.625071 [ERR] sofia.c:5869 Cannot Blind Transfer 1
> Legged calls
> 2011-01-20 16:04:00.625071 [NOTICE] sofia.c:5286 Hangup sofia/internal/
> 1000 at 192.168.1.211 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE]
> 2011-01-20 16:04:00.654406 [NOTICE] switch_core_session.c:1306 Session 2
> (sofia/internal/1000 at 192.168.1.211) Ended
> 2011-01-20 16:04:00.654406 [NOTICE] switch_core_session.c:1308 Close
> Channel sofia/internal/1000 at 192.168.1.211 [CS_DESTROY]
> 2011-01-20 16:04:00.725081 [NOTICE] switch_core_state_machine.c:189
> sofia/internal/0000000000 at 192.168.1.211 has executed the last dialplan
> instruction, hanging up.
> 2011-01-20 16:04:00.725081 [NOTICE] switch_core_state_machine.c:191 Hangup
> sofia/internal/0000000000 at 192.168.1.211 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-01-20 16:04:00.730022 [NOTICE] switch_core_session.c:1306 Session 3
> (sofia/internal/0000000000 at 192.168.1.211) Ended
> 2011-01-20 16:04:00.730022 [NOTICE] switch_core_session.c:1308 Close
> Channel sofia/internal/0000000000 at 192.168.1.211 [CS_DESTROY]
>
> Here is the output from sofia status
>
>                      Name          Type
>   Data      State
>
> =================================================================================================
>                  external       profile
> sip:mod_sofia at 192.168.1.211:5080      RUNNING (0)
>     external::example.com       gateway
> sip:joeuser at example.com <sip%3Ajoeuser at example.com>      NOREG
>             internal-ipv6       profile                   sip:mod_sofia@[::1]:5060
>      RUNNING (0)
>                  internal       profile
> sip:mod_sofia at 192.168.1.211:5060      RUNNING (0)
>             192.168.1.211         alias
> internal      ALIASED
>
> =================================================================================================
> 3 profiles 1 alias
>
> Any advice or help would be greatly appreciated.
>
> thanks!
> Peter
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Fri Jan 21 00:37:34 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 20 Jan 2011 13:37:34 -0800
Subject: [Freeswitch-users] call disconnects on call forwarding from
	sangoma
In-Reply-To: <AANLkTinQWTD=NuGE77qAjJHt4o-WLBHc=Dz+bEdROj5T@mail.gmail.com>
References: <AANLkTinQWTD=NuGE77qAjJHt4o-WLBHc=Dz+bEdROj5T@mail.gmail.com>
Message-ID: <AANLkTikUK4O58vuJxa5s2vVDPcrh6=s9zqvt36p5My72@mail.gmail.com>

Wow that's definitely a question for Moy. Moy, can you let us know about
this error?
-MC

On Tue, Jan 18, 2011 at 11:55 PM, Sam <u2nsam at gmail.com> wrote:

> Hi,
>
> When i make a call from extension to extension, after timeout it gets
> forwarded properly on failure(i.e no answer) to my mobile number,
>
> but when i call from outside phone to my extension it do not get forwarded
> properly giving me the sangoma error  ,
>
> which i have pasted it in pastebin link ( check line 244 ) : :
> http://pastebin.freeswitch.org/15064
>
> What could be the reason ?
>
> Regards
> Sam
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri Jan 21 00:39:44 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 20 Jan 2011 13:39:44 -0800
Subject: [Freeswitch-users] sip_ph_X-var in conference
In-Reply-To: <09F27B7B578941D688EB8717D621F330@e1705>
References: <09F27B7B578941D688EB8717D621F330@e1705>
Message-ID: <AANLkTinph-RoM8DsRBVgxFAARM=8VF7Ub9=UK=JF63Qv@mail.gmail.com>

I believe in chapter 9 of the book Darren covers how to send NOTFIY messages
to a given phone/IP/endpoint.

-MC

On Wed, Jan 19, 2011 at 9:32 AM, Madovsky <infos at madovsky.org> wrote:

>  is it possilbe to send
> sip_ph_X-var to the caller in a conference ?
> if yes, how to do it ?
>
> thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From phone.bytes at gmail.com  Fri Jan 21 00:47:12 2011
From: phone.bytes at gmail.com (Phone Bytes)
Date: Thu, 20 Jan 2011 14:47:12 -0700
Subject: [Freeswitch-users] Caller ID Number not going out on PRI
In-Reply-To: <AANLkTimsZ04PiNVupqwSymVmW6j=khrLin7skAxd4icT@mail.gmail.com>
References: <4D35F532.2080802@gmail.com>
	<AANLkTimsZ04PiNVupqwSymVmW6j=khrLin7skAxd4icT@mail.gmail.com>
Message-ID: <1295560032.7918.16.camel@douga-desktop>

In looking further at this, we are wondering if this is an issue of
variable values not being passed from the directory xml's to the
dialplan.

It seems that neither the Caller ID nor the Toll variables are passing
to the dialplan?

The xml and dialplan are here http://freeswitch.pastebin.com/hrj0uauU

The Debug logs are here http://freeswitch.pastebin.com/vgJccwYY

Thanks








On Thu, 2011-01-20 at 12:08 -0800, Michael Collins wrote:
> I'd recommend pastebin of your relevant dialplan and freetdm configs.
> It would also help to see a debug console log trace of a call that is
> displaying the behavior.
> -MC
> 
> On Tue, Jan 18, 2011 at 12:16 PM, Phone <phone.bytes at gmail.com> wrote:
>         A few upgrades ago, the Caller ID Number on outbound calls
>         stopped
>         working.  Now it always displays the extension number, where
>         before it
>         was using the value specified in the directory entry.  Seems
>         like maybe
>         the variables are not passing from the directory entry to the
>         dial plan.
>         
>         Any ideas?
>         





From brian at freeswitch.org  Fri Jan 21 01:33:11 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 20 Jan 2011 16:33:11 -0600
Subject: [Freeswitch-users] sip_ph_X-var in conference
In-Reply-To: <09F27B7B578941D688EB8717D621F330@e1705>
References: <09F27B7B578941D688EB8717D621F330@e1705>
Message-ID: <27A30140-1246-4D4F-A451-FD6E4448626E@freeswitch.org>

What exactly are you building?  You appear to be building some sort of commercial conferencing application.  Care to share?

Thanks,
Brian

On Jan 19, 2011, at 11:32 AM, Madovsky wrote:

> is it possilbe to send
> sip_ph_X-var to the caller in a conference ?
> if yes, how to do it ?
>  
> thanks

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From infos at madovsky.org  Fri Jan 21 01:52:17 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 20 Jan 2011 17:52:17 -0500
Subject: [Freeswitch-users] sip_ph_X-var in conference
References: <09F27B7B578941D688EB8717D621F330@e1705>
	<27A30140-1246-4D4F-A451-FD6E4448626E@freeswitch.org>
Message-ID: <E901ADDA09894E50B8B1F809598B1D24@e1705>

LOL !!
of course, once money will arrive (if it arrives one day..)
  ----- Original Message ----- 
  From: Brian West 
  To: FreeSWITCH Users Help 
  Sent: Thursday, January 20, 2011 5:33 PM
  Subject: Re: [Freeswitch-users] sip_ph_X-var in conference


  What exactly are you building?  You appear to be building some sort of commercial conferencing application.  Care to share?


  Thanks,
  Brian


  On Jan 19, 2011, at 11:32 AM, Madovsky wrote:


    is it possilbe to send
    sip_ph_X-var to the caller in a conference ?
    if yes, how to do it ?

    thanks




------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Fri Jan 21 02:07:50 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 20 Jan 2011 18:07:50 -0500
Subject: [Freeswitch-users] sip_ph_X-var in conference
References: <09F27B7B578941D688EB8717D621F330@e1705>
	<AANLkTinph-RoM8DsRBVgxFAARM=8VF7Ub9=UK=JF63Qv@mail.gmail.com>
Message-ID: <C11A309C6B264AD39637D3069044810E@e1705>

ok thanks will read it.
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Thursday, January 20, 2011 4:39 PM
  Subject: Re: [Freeswitch-users] sip_ph_X-var in conference


  I believe in chapter 9 of the book Darren covers how to send NOTFIY messages to a given phone/IP/endpoint.


  -MC


  On Wed, Jan 19, 2011 at 9:32 AM, Madovsky <infos at madovsky.org> wrote:

    is it possilbe to send
    sip_ph_X-var to the caller in a conference ?
    if yes, how to do it ?

    thanks

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org






------------------------------------------------------------------------------


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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From george.niculae79 at gmail.com  Fri Jan 21 03:05:34 2011
From: george.niculae79 at gmail.com (George Niculae)
Date: Fri, 21 Jan 2011 02:05:34 +0200
Subject: [Freeswitch-users] conference pin issue
Message-ID: <AANLkTik8HaAUMpf0_=7q6h_xcTAVqGuP4HNQ9Ave3Xkn@mail.gmail.com>

Hi All,

I noticed that in a pin protected conference scenario, if the user inputs
more digits than the length of the correct one the remaining digits are
accounted for next retry - e.g. if the conf pin is 1234 and user enter 12345
then 5 will be used as first digit in 2nd retry. Extension is configured as:

  <extension name="5678">
    <condition field="destination_number" expression="^uuu$">
      <action application="set" data="supplied_pin=${sip_h_X-ConfPin}"/>
      <action application="export" data="supplied_pin=${sip_h_X-ConfPin}"/>
      <action application="bridge" data="loopback/app=conference:uuu at 5678
+1234"/>
    </condition>
  </extension>

I attached a patch for mod_conference that checks pin buffer to be 0 before
starting to collect digits again and reset it if > 0. I also modified the
code to play "please enter the conference access number" prompt just first
time and only "the passcode you entered is not correct, please try again" on
subsequent retries.

Please review the patch and let me know your comments.

Thanks,
George
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From anthony.minessale at gmail.com  Fri Jan 21 03:06:18 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 20 Jan 2011 18:06:18 -0600
Subject: [Freeswitch-users] PCMA stranges
In-Reply-To: <AANLkTi=wcUEe89em_-c2DaFbBPunWum9Oh10yjj1KTXw@mail.gmail.com>
References: <4D380FF8.7080705@tagnet.ru>
	<E676EB80-7927-4061-BC2C-497DBA81D75E@ipeva.fr>
	<4D381E1F.7040202@tagnet.ru> <4D384196.105@tagnet.ru>
	<AANLkTikVAt6t8K4fxg0GXisw=0_+s5gBYe9Hyvep5KN2@mail.gmail.com>
	<4D386528.2010506@tagnet.ru>
	<AANLkTi=wcUEe89em_-c2DaFbBPunWum9Oh10yjj1KTXw@mail.gmail.com>
Message-ID: <AANLkTimbouEk+WRzarw7q6PrjcBfDupd+xxS=6_XdvO0@mail.gmail.com>

I just added a patch for another fix that I think might help you, try latest GIT

On Thu, Jan 20, 2011 at 3:29 PM, Michael Collins <msc at freeswitch.org> wrote:
> Have you compared the SIP traces between the working g711u and the
> non-working g711a? What seems to be different between the two?
> -MC
>
> On Thu, Jan 20, 2011 at 8:39 AM, Boris Kovalenko <boris at tagnet.ru> wrote:
>>
>> Hello!
>>
>> ? ? Yes, I know the media is ignored, so I wondering about this codec
>> strangenes. What test I have done:
>>
>> SoftPhone (g711ulaw) -> FS -> Cisco 5350 : working
>> SoftPhone (g711alaw, proxy_media=true) -> FS -> Cisco 5350 : not working
>> SoftPhone (g711alaw, proxy_media=false) -> FS -> Cisco 5350 : working
>> SoftPhone (g711alaw) -> Cisco 5350 : working
>>
>>
>> > proxy media means media is ignored, its not ever touched. ?Its
>> > designed to transparently pass media even if FS is unaware of the
>> > codec.
>> > So if you say you are using proxy media and you have codec specific
>> > problems it seems peculiar.
>> >
>> >
>> > On Thu, Jan 20, 2011 at 8:07 AM, Boris Kovalenko<boris at tagnet.ru>
>> > ?wrote:
>> >> Hello!
>> >>
>> >> ? ? ?I found that problem is somewhere inside freeswitch. Direct call
>> >> between
>> >> SoftPhone and Cisco is working fine.
>> >>
>> >> Hello!
>> >>
>> >> ? ? ?David, may You explain why this problem is with G711A only? When I
>> >> use
>> >> G711U there are no problems and call is established and I may talk.
>> >>
>> >> The Cisco never sends the 200/OK after the 183, so the call is not
>> >> established.
>> >> David Ponzone ?Direction Technique
>> >> email: david.ponzone at ipeva.fr
>> >> tel: ? ? ?01 74 03 18 97
>> >> gsm: ? 06 66 98 76 34
>> >> Service Client IPeva
>> >> tel: ? ? ?0811 46 26 26
>> >> www.ipeva.fr ?- ? www.ipeva-studio.com
>> >> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> >> l'intention exclusive de ses destinataires. Toute utilisation ou
>> >> diffusion
>> >> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> >> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
>> >> s'il
>> >> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de
>> >> ce
>> >> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> >>
>> >>
>> >>
>> >> Le 20/01/2011 ? 11:35, Boris Kovalenko a ?crit :
>> >>
>> >> Hello!
>> >>
>> >> ? ? ?My network configuration is:
>> >> Softphone (eyeBeam 1.5) with only G711alaw enabled -> ?FreeSWITCH
>> >> Version
>> >> 1.0.head (git-0cf1d54 2011-01-19 16-36-04 -0500) -> ?Cisco AS5350
>> >>
>> >> proxy_media=true in profile.
>> >> So, when I do a test call, there is SDP but no RTP between FS and
>> >> CISCO,
>> >> so I can't hear voice. With G711ulaw there are no problems. What is
>> >> wrong? Siptrace below:
>> >>
>> >>
>> >> ------------------------------------------------------------------------
>> >> ? ? INVITE sip:73435327569 at default SIP/2.0
>> >> ? ? Via: SIP/2.0/UDP
>> >>
>> >> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
>> >> ? ? Max-Forwards: 70
>> >> ? ? Contact:<sip:50001 at X.X.29.123:21556>
>> >> ? ? To: "73435327569"<sip:73435327569 at default>
>> >> ? ? From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>> >> ? ? Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>> >> ? ? CSeq: 1 INVITE
>> >> ? ? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
>> >> SUBSCRIBE, INFO
>> >> ? ? Content-Type: application/sdp
>> >> ? ? User-Agent: eyeBeam release 1102u stamp 52345
>> >> ? ? Content-Length: 337
>> >>
>> >> ? ? v=0
>> >> ? ? o=- 1 2 IN IP4 192.168.3.253
>> >> ? ? s=CounterPath eyeBeam 1.5
>> >> ? ? c=IN IP4 192.168.3.253
>> >> ? ? t=0 0
>> >> ? ? m=audio 40032 RTP/AVP 8 101
>> >> ? ? a=alt:1 3 : XdMyLUqb Fs2PLxKV 192.168.3.253 40032
>> >> ? ? a=alt:2 2 : 2qrc/CQ7 Z/9XFQZl 172.16.3.253 40032
>> >> ? ? a=alt:3 1 : 8BOwI53P +ENGEr/U X.X.29.123 40032
>> >> ? ? a=fmtp:101 0-15
>> >> ? ? a=rtpmap:101 telephone-event/8000
>> >> ? ? a=sendrecv
>> >>
>> >> ------------------------------------------------------------------------
>> >> send 385 bytes to udp/[X.X.29.123]:21556 at 10:30:31.313366:
>> >>
>> >> ------------------------------------------------------------------------
>> >> ? ? SIP/2.0 100 Trying
>> >> ? ? Via: SIP/2.0/UDP
>> >>
>> >> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>> >> ? ? From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>> >> ? ? To: "73435327569"<sip:73435327569 at default>
>> >> ? ? Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>> >> ? ? CSeq: 1 INVITE
>> >> ? ? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
>> >> 16-36-04 -0500
>> >> ? ? Content-Length: 0
>> >>
>> >>
>> >> ------------------------------------------------------------------------
>> >> 2011-01-20 15:30:31.313448 [NOTICE] switch_channel.c:808 New Channel
>> >> sofia/epbx/50001 at default [95698968-8cf9-40a2-91ad-1322fcfc76af]
>> >> 2011-01-20 15:30:31.322338 [INFO] mod_dialplan_xml.c:331 Processing
>> >> TAGNet<50001>->73435327569 in context public
>> >> 2011-01-20 15:30:31.323362 [INFO] mod_dialplan_xml.c:331 Processing
>> >> TAGNet<50001>->ext_translate_extsrc in context features
>> >> 2011-01-20 15:30:31.324357 [NOTICE] switch_ivr.c:1606 Transfer
>> >> sofia/epbx/50001 at default to XML[73435327569 at top.ctx]
>> >> 2011-01-20 15:30:31.325352 [INFO] mod_dialplan_xml.c:331 Processing
>> >> TAGNet<50001>->73435327569 in context top.ctx
>> >> 2011-01-20 15:30:31.326344 [NOTICE] mod_dptools.c:1174 [top.ctx] -
>> >> ext_local
>> >> 2011-01-20 15:30:31.329360 [NOTICE] switch_channel.c:808 New Channel
>> >> sofia/epbx/73435327569 at X.X.16.83:5060
>> >> [b9654c75-039a-4780-bc00-0fae65f92a9a]
>> >> send 1276 bytes to udp/[X.X.16.83]:5060 at 10:30:31.330623:
>> >>
>> >> ------------------------------------------------------------------------
>> >> ? ? INVITE sip:73435327569 at X.X.16.83:5060 SIP/2.0
>> >> ? ? Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>> >> ? ? Max-Forwards: 68
>> >> ? ? From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>> >> ? ? To:<sip:73435327569 at X.X.16.83:5060>
>> >> ? ? Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>> >> ? ? CSeq: 7423651 INVITE
>> >> ? ? Contact:<sip:mod_sofia at Y.Y.138.187:5060>
>> >> ? ? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
>> >> 16-36-04 -0500
>> >> ? ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> >> ? ? Supported: timer, precondition, path, replaces
>> >> ? ? Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>> >> sla, include-session-description, presence.winfo, message-summary,
>> >> refer
>> >> ? ? Content-Type: application/sdp
>> >> ? ? Content-Disposition: session
>> >> ? ? Content-Length: 341
>> >> ? ? X-FS-Support: update_display
>> >> ? ? Remote-Party-ID: "TAGNet"
>> >> <sip:50001 at Y.Y.138.187>;party=calling;screen=yes;privacy=off
>> >>
>> >> ? ? v=0
>> >> ? ? o=FreeSWITCH 1627720692 1627720693 IN IP4 Y.Y.138.187
>> >> ? ? s=FreeSWITCH
>> >> ? ? c=IN IP4 Y.Y.138.187
>> >> ? ? t=0 0
>> >> ? ? m=audio 23442 RTP/AVP 8 101
>> >> ? ? a=rtpmap:101 telephone-event/8000
>> >> ? ? a=fmtp:101 0-15
>> >> ? ? a=alt:1 3 : XdMyLUqb Fs2PLxKV 192.168.3.253 40032
>> >> ? ? a=alt:2 2 : 2qrc/CQ7 Z/9XFQZl 172.16.3.253 40032
>> >> ? ? a=alt:3 1 : 8BOwI53P +ENGEr/U X.X.29.123 40032
>> >>
>> >> ------------------------------------------------------------------------
>> >> recv 396 bytes from udp/[X.X.16.83]:5060 at 10:30:31.341106:
>> >>
>> >> ------------------------------------------------------------------------
>> >> ? ? SIP/2.0 100 Trying
>> >> ? ? Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>> >> ? ? From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>> >> ? ? To:<sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>> >> ? ? Date: Thu, 20 Jan 2011 10:30:31 GMT
>> >> ? ? Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>> >> ? ? Server: Cisco-SIPGateway/IOS-12.x
>> >> ? ? CSeq: 7423651 INVITE
>> >> ? ? Allow-Events: telephone-event
>> >> ? ? Content-Length: 0
>> >>
>> >>
>> >> ------------------------------------------------------------------------
>> >> recv 814 bytes from udp/[X.X.16.83]:5060 at 10:30:32.401494:
>> >>
>> >> ------------------------------------------------------------------------
>> >> ? ? SIP/2.0 183 Session Progress
>> >> ? ? Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>> >> ? ? From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>> >> ? ? To:<sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>> >> ? ? Date: Thu, 20 Jan 2011 10:30:31 GMT
>> >> ? ? Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>> >> ? ? Server: Cisco-SIPGateway/IOS-12.x
>> >> ? ? CSeq: 7423651 INVITE
>> >> ? ? Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
>> >> SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
>> >> ? ? Allow-Events: telephone-event
>> >> ? ? Contact:<sip:73435327569 at X.X.16.83:5060>
>> >> ? ? Content-Disposition: session;handling=required
>> >> ? ? Content-Type: application/sdp
>> >> ? ? Content-Length: 176
>> >>
>> >> ? ? v=0
>> >> ? ? o=CiscoSystemsSIP-GW-UserAgent 4564 1578 IN IP4 X.X.16.83
>> >> ? ? s=SIP Call
>> >> ? ? c=IN IP4 X.X.16.83
>> >> ? ? t=0 0
>> >> ? ? m=audio 17770 RTP/AVP 8
>> >> ? ? c=IN IP4 X.X.16.83
>> >> ? ? a=rtpmap:8 PCMA/8000
>> >>
>> >> ------------------------------------------------------------------------
>> >> 2011-01-20 15:30:32.401362 [INFO] sofia.c:729
>> >> sofia/epbx/73435327569 at X.X.16.83:5060 Update Callee ID to "Outbound
>> >> Call"<73435327569>
>> >> 2011-01-20 15:30:32.409363 [NOTICE] sofia.c:4739 Pre-Answer
>> >> sofia/epbx/73435327569 at X.X.16.83:5060!
>> >> 2011-01-20 15:30:32.415412 [INFO] switch_ivr_originate.c:3345 Sending
>> >> early media
>> >> 2011-01-20 15:30:32.416367 [NOTICE] mod_sofia.c:2252 Pre-Answer
>> >> sofia/epbx/50001 at default!
>> >> send 1145 bytes to udp/[X.X.29.123]:21556 at 10:30:32.418195:
>> >>
>> >> ------------------------------------------------------------------------
>> >> ? ? SIP/2.0 183 Session Progress
>> >> ? ? Via: SIP/2.0/UDP
>> >>
>> >> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>> >> ? ? From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>> >> ? ? To: "73435327569"<sip:73435327569 at default>;tag=BScUp0QXHFBgB
>> >> ? ? Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>> >> ? ? CSeq: 1 INVITE
>> >> ? ? Contact:<sip:73435327569 at Y.Y.138.187:5060;transport=udp>
>> >> ? ? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
>> >> 16-36-04 -0500
>> >> ? ? Accept: application/sdp
>> >> ? ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> >> ? ? Supported: timer, precondition, path, replaces
>> >> ? ? Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>> >> sla, include-session-description, presence.winfo, message-summary,
>> >> refer
>> >> ? ? Content-Type: application/sdp
>> >> ? ? Content-Disposition: session
>> >> ? ? Content-Length: 181
>> >> ? ? Remote-Party-ID: "73435327569"
>> >> <sip:73435327569 at Y.Y.138.187>;party=calling;privacy=off;screen=no
>> >>
>> >> ? ? v=0
>> >> ? ? o=FreeSWITCH 1627752513 1627752514 IN IP4 Y.Y.138.187
>> >> ? ? s=FreeSWITCH
>> >> ? ? c=IN IP4 Y.Y.138.187
>> >> ? ? t=0 0
>> >> ? ? m=audio 25806 RTP/AVP 8
>> >> ? ? c=IN IP4 Y.Y.138.187
>> >> ? ? a=rtpmap:8 PCMA/8000
>> >>
>> >>
>> >> ------------------------------------------------------------------------
>> >> recv 365 bytes from udp/[X.X.29.123]:21556 at 10:30:39.196791:
>> >>
>> >> ------------------------------------------------------------------------
>> >> ? ? CANCEL sip:73435327569 at default SIP/2.0
>> >> ? ? Via: SIP/2.0/UDP
>> >>
>> >> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
>> >> ? ? To: "73435327569"<sip:73435327569 at default>
>> >> ? ? From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>> >> ? ? Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>> >> ? ? CSeq: 1 CANCEL
>> >> ? ? User-Agent: eyeBeam release 1102u stamp 52345
>> >> ? ? Content-Length: 0
>> >>
>> >>
>> >> ------------------------------------------------------------------------
>> >> send 320 bytes to udp/[X.X.29.123]:21556 at 10:30:39.196905:
>> >>
>> >> ------------------------------------------------------------------------
>> >> ? ? SIP/2.0 200 OK
>> >> ? ? Via: SIP/2.0/UDP
>> >>
>> >> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>> >> ? ? From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>> >> ? ? To: "73435327569"<sip:73435327569 at default>;tag=BScUp0QXHFBgB
>> >> ? ? Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>> >> ? ? CSeq: 1 CANCEL
>> >> ? ? Content-Length: 0
>> >>
>> >>
>> >> ------------------------------------------------------------------------
>> >> send 715 bytes to udp/[X.X.29.123]:21556 at 10:30:39.197002:
>> >>
>> >> ------------------------------------------------------------------------
>> >> ? ? SIP/2.0 487 Request Terminated
>> >> ? ? Via: SIP/2.0/UDP
>> >>
>> >> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>> >> ? ? From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>> >> ? ? To: "73435327569"<sip:73435327569 at default>;tag=BScUp0QXHFBgB
>> >> ? ? Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>> >> ? ? CSeq: 1 INVITE
>> >> ? ? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
>> >> 16-36-04 -0500
>> >> ? ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> >> ? ? Supported: timer, precondition, path, replaces
>> >> ? ? Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>> >> sla, include-session-description, presence.winfo, message-summary,
>> >> refer
>> >> ? ? Content-Length: 0
>> >>
>> >>
>> >> ------------------------------------------------------------------------
>> >> 2011-01-20 15:30:39.196838 [NOTICE] sofia.c:5286 Hangup
>> >> sofia/epbx/50001 at default [CS_EXECUTE] [ORIGINATOR_CANCEL]
>> >> 2011-01-20 15:30:39.196838 [NOTICE] switch_ivr_bridge.c:653 Hangup
>> >> sofia/epbx/73435327569 at X.X.16.83:5060 [CS_EXCHANGE_MEDIA]
>> >> [NORMAL_CLEARING]
>> >> recv 331 bytes from udp/[X.X.29.123]:21556 at 10:30:39.198834:
>> >>
>> >> ------------------------------------------------------------------------
>> >> ? ? ACK sip:73435327569 at default SIP/2.0
>> >> ? ? Via: SIP/2.0/UDP
>> >>
>> >> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
>> >> ? ? To: "73435327569"<sip:73435327569 at default>;tag=BScUp0QXHFBgB
>> >> ? ? From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>> >> ? ? Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>> >> ? ? CSeq: 1 ACK
>> >> ? ? Content-Length: 0
>> >>
>> >>
>> >> ------------------------------------------------------------------------
>> >> 2011-01-20 15:30:39.204838 [NOTICE] switch_core_session.c:1306 Session
>> >> 99 (sofia/epbx/50001 at default) Ended
>> >> 2011-01-20 15:30:39.204838 [NOTICE] switch_core_session.c:1308 Close
>> >> Channel sofia/epbx/50001 at default [CS_DESTROY]
>> >> send 372 bytes to udp/[X.X.16.83]:5060 at 10:30:39.212574:
>> >>
>> >> ------------------------------------------------------------------------
>> >> ? ? CANCEL sip:73435327569 at X.X.16.83:5060 SIP/2.0
>> >> ? ? Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>> >> ? ? Max-Forwards: 68
>> >> ? ? From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>> >> ? ? To:<sip:73435327569 at X.X.16.83:5060>
>> >> ? ? Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>> >> ? ? CSeq: 7423651 CANCEL
>> >> ? ? Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>> >> ? ? Content-Length: 0
>> >>
>> >>
>> >> ------------------------------------------------------------------------
>> >> 2011-01-20 15:30:39.212840 [NOTICE] switch_core_session.c:1306 Session
>> >> 100 (sofia/epbx/73435327569 at X.X.16.83:5060) Ended
>> >> 2011-01-20 15:30:39.212840 [NOTICE] switch_core_session.c:1308 Close
>> >> Channel sofia/epbx/73435327569 at X.X.16.83:5060 [CS_DESTROY]
>> >> recv 311 bytes from udp/[X.X.16.83]:5060 at 10:30:39.216838:
>> >>
>> >> ------------------------------------------------------------------------
>> >> ? ? SIP/2.0 200 OK
>> >> ? ? Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>> >> ? ? From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>> >> ? ? To:<sip:73435327569 at X.X.16.83:5060>
>> >> ? ? Date: Thu, 20 Jan 2011 10:30:39 GMT
>> >> ? ? Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>> >> ? ? Content-Length: 0
>> >> ? ? CSeq: 7423651 CANCEL
>> >>
>> >>
>> >> ------------------------------------------------------------------------
>> >> recv 407 bytes from udp/[X.X.16.83]:5060 at 10:30:39.219180:
>> >>
>> >> ------------------------------------------------------------------------
>> >> ? ? SIP/2.0 487 Request Cancelled
>> >> ? ? Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>> >> ? ? From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>> >> ? ? To:<sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>> >> ? ? Date: Thu, 20 Jan 2011 10:30:39 GMT
>> >> ? ? Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>> >> ? ? Server: Cisco-SIPGateway/IOS-12.x
>> >> ? ? CSeq: 7423651 INVITE
>> >> ? ? Allow-Events: telephone-event
>> >> ? ? Content-Length: 0
>> >>
>> >>
>> >> ------------------------------------------------------------------------
>> >> send 334 bytes to udp/[X.X.16.83]:5060 at 10:30:39.219276:
>> >>
>> >> ------------------------------------------------------------------------
>> >> ? ? ACK sip:73435327569 at X.X.16.83:5060 SIP/2.0
>> >> ? ? Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>> >> ? ? Max-Forwards: 68
>> >> ? ? From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>> >> ? ? To:<sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>> >> ? ? Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>> >> ? ? CSeq: 7423651 ACK
>> >> ? ? Content-Length: 0
>> >>
>> >>
>> >> ------------------------------------------------------------------------
>> >>
>> >>
>> >> --
>> >> ? ?????????,
>> >> ? ?????? ?????????
>> >> ? ???? "??????"
>> >> ? ?(3435) 494991
>> >>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >> --
>> >> ? ?????????,
>> >> ? ?????? ?????????
>> >> ? ???? "??????"
>> >> ? ?(3435) 494991
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >> --
>> >> ? ?????????,
>> >> ? ?????? ?????????
>> >> ? ???? "??????"
>> >> ? ?(3435) 494991
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> >
>>
>>
>> --
>> ? ?????????,
>> ? ????? ?????????
>> ? ??? "??????"
>> ? (3435) 494991
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From brian at freeswitch.org  Fri Jan 21 03:11:34 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 20 Jan 2011 18:11:34 -0600
Subject: [Freeswitch-users] conference pin issue
In-Reply-To: <AANLkTik8HaAUMpf0_=7q6h_xcTAVqGuP4HNQ9Ave3Xkn@mail.gmail.com>
References: <AANLkTik8HaAUMpf0_=7q6h_xcTAVqGuP4HNQ9Ave3Xkn@mail.gmail.com>
Message-ID: <0C8CA92D-9EAA-4C51-8182-F1CB594CA868@freeswitch.org>

Please post to jira.freeswitch.org

/b

On Jan 20, 2011, at 6:05 PM, George Niculae wrote:

> Hi All,
> 
> I noticed that in a pin protected conference scenario, if the user inputs more digits than the length of the correct one the remaining digits are accounted for next retry - e.g. if the conf pin is 1234 and user enter 12345 then 5 will be used as first digit in 2nd retry. Extension is configured as:
> 
>   <extension name="5678">
>     <condition field="destination_number" expression="^uuu$">
>       <action application="set" data="supplied_pin=${sip_h_X-ConfPin}"/>
>       <action application="export" data="supplied_pin=${sip_h_X-ConfPin}"/>
>       <action application="bridge" data="loopback/app=conference:uuu at 5678+1234"/>
>     </condition>
>   </extension>
> 
> I attached a patch for mod_conference that checks pin buffer to be 0 before starting to collect digits again and reset it if > 0. I also modified the code to play "please enter the conference access number" prompt just first time and only "the passcode you entered is not correct, please try again" on subsequent retries.
> 
> Please review the patch and let me know your comments.
> 
> Thanks,
> George
> 
> <0001-Conference-pin.patch>_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Fri Jan 21 03:13:10 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 20 Jan 2011 18:13:10 -0600
Subject: [Freeswitch-users] conference pin issue
In-Reply-To: <0C8CA92D-9EAA-4C51-8182-F1CB594CA868@freeswitch.org>
References: <AANLkTik8HaAUMpf0_=7q6h_xcTAVqGuP4HNQ9Ave3Xkn@mail.gmail.com>
	<0C8CA92D-9EAA-4C51-8182-F1CB594CA868@freeswitch.org>
Message-ID: <AANLkTinvKg8BMnGRRr0Fvz8afJ_GrYZB_ZgGWwjve5PO@mail.gmail.com>

yes it seems ok but be sure to post it to jira

On Thu, Jan 20, 2011 at 6:11 PM, Brian West <brian at freeswitch.org> wrote:
> Please post to jira.freeswitch.org
>
> /b
>
> On Jan 20, 2011, at 6:05 PM, George Niculae wrote:
>
>> Hi All,
>>
>> I noticed that in a pin protected conference scenario, if the user inputs more digits than the length of the correct one the remaining digits are accounted for next retry - e.g. if the conf pin is 1234 and user enter 12345 then 5 will be used as first digit in 2nd retry. Extension is configured as:
>>
>> ? <extension name="5678">
>> ? ? <condition field="destination_number" expression="^uuu$">
>> ? ? ? <action application="set" data="supplied_pin=${sip_h_X-ConfPin}"/>
>> ? ? ? <action application="export" data="supplied_pin=${sip_h_X-ConfPin}"/>
>> ? ? ? <action application="bridge" data="loopback/app=conference:uuu at 5678+1234"/>
>> ? ? </condition>
>> ? </extension>
>>
>> I attached a patch for mod_conference that checks pin buffer to be 0 before starting to collect digits again and reset it if > 0. I also modified the code to play "please enter the conference access number" prompt just first time and only "the passcode you entered is not correct, please try again" on subsequent retries.
>>
>> Please review the patch and let me know your comments.
>>
>> Thanks,
>> George
>>
>> <0001-Conference-pin.patch>_______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From jmesquita at freeswitch.org  Fri Jan 21 03:13:55 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 20 Jan 2011 21:13:55 -0300
Subject: [Freeswitch-users] conference pin issue
In-Reply-To: <AANLkTik8HaAUMpf0_=7q6h_xcTAVqGuP4HNQ9Ave3Xkn@mail.gmail.com>
References: <AANLkTik8HaAUMpf0_=7q6h_xcTAVqGuP4HNQ9Ave3Xkn@mail.gmail.com>
Message-ID: <AANLkTi=Y=Hz_Z1yhV=9H9aQtQWCwec6=F=R+7mO-w8Xg@mail.gmail.com>

This belongs to Jira, doesn't it?

I am not skilled enough to review it, but the core devs really rather have
this on Jira. Thank you for stepping up to solve a problem you've found.

Regards,
Jo?o Mesquita


On Thu, Jan 20, 2011 at 9:05 PM, George Niculae
<george.niculae79 at gmail.com>wrote:

> Hi All,
>
> I noticed that in a pin protected conference scenario, if the user inputs
> more digits than the length of the correct one the remaining digits are
> accounted for next retry - e.g. if the conf pin is 1234 and user enter 12345
> then 5 will be used as first digit in 2nd retry. Extension is configured as:
>
>   <extension name="5678">
>     <condition field="destination_number" expression="^uuu$">
>       <action application="set" data="supplied_pin=${sip_h_X-ConfPin}"/>
>       <action application="export" data="supplied_pin=${sip_h_X-ConfPin}"/>
>       <action application="bridge" data="loopback/app=conference:uuu at 5678
> +1234"/>
>     </condition>
>   </extension>
>
> I attached a patch for mod_conference that checks pin buffer to be 0 before
> starting to collect digits again and reset it if > 0. I also modified the
> code to play "please enter the conference access number" prompt just first
> time and only "the passcode you entered is not correct, please try again" on
> subsequent retries.
>
> Please review the patch and let me know your comments.
>
> Thanks,
> George
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From michal.bielicki at seventhsignal.de  Fri Jan 21 03:19:39 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Fri, 21 Jan 2011 01:19:39 +0100
Subject: [Freeswitch-users] conference pin issue
In-Reply-To: <0C8CA92D-9EAA-4C51-8182-F1CB594CA868@freeswitch.org>
References: <AANLkTik8HaAUMpf0_=7q6h_xcTAVqGuP4HNQ9Ave3Xkn@mail.gmail.com>
	<0C8CA92D-9EAA-4C51-8182-F1CB594CA868@freeswitch.org>
Message-ID: <26221ECF-AA7D-4937-9F23-477FD4B0A681@seventhsignal.de>

Nevertheless very cool patch :)

Am 21.01.2011 um 01:11 schrieb Brian West:

> Please post to jira.freeswitch.org
> 
> /b
> 
> On Jan 20, 2011, at 6:05 PM, George Niculae wrote:
> 
>> Hi All,
>> 
>> I noticed that in a pin protected conference scenario, if the user inputs more digits than the length of the correct one the remaining digits are accounted for next retry - e.g. if the conf pin is 1234 and user enter 12345 then 5 will be used as first digit in 2nd retry. Extension is configured as:
>> 
>>  <extension name="5678">
>>    <condition field="destination_number" expression="^uuu$">
>>      <action application="set" data="supplied_pin=${sip_h_X-ConfPin}"/>
>>      <action application="export" data="supplied_pin=${sip_h_X-ConfPin}"/>
>>      <action application="bridge" data="loopback/app=conference:uuu at 5678+1234"/>
>>    </condition>
>>  </extension>
>> 
>> I attached a patch for mod_conference that checks pin buffer to be 0 before starting to collect digits again and reset it if > 0. I also modified the code to play "please enter the conference access number" prompt just first time and only "the passcode you entered is not correct, please try again" on subsequent retries.
>> 
>> Please review the patch and let me know your comments.
>> 
>> Thanks,
>> George
>> 
>> <0001-Conference-pin.patch>_______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de

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From brian at freeswitch.org  Fri Jan 21 03:23:38 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 20 Jan 2011 18:23:38 -0600
Subject: [Freeswitch-users] conference pin issue
In-Reply-To: <26221ECF-AA7D-4937-9F23-477FD4B0A681@seventhsignal.de>
References: <AANLkTik8HaAUMpf0_=7q6h_xcTAVqGuP4HNQ9Ave3Xkn@mail.gmail.com>
	<0C8CA92D-9EAA-4C51-8182-F1CB594CA868@freeswitch.org>
	<26221ECF-AA7D-4937-9F23-477FD4B0A681@seventhsignal.de>
Message-ID: <902C3395-87EE-4084-A771-CE911419D9BC@freeswitch.org>

Yep.

/b

On Jan 20, 2011, at 6:19 PM, Michal Bielicki wrote:

> Nevertheless very cool patch :)
> 
> Am 21.01.2011 um 01:11 schrieb Brian West:
> 
>> Please post to jira.freeswitch.org
>> 
>> /b

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From infos at madovsky.org  Fri Jan 21 03:45:16 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 20 Jan 2011 19:45:16 -0500
Subject: [Freeswitch-users] mod_conference member-flags
References: <69D8E2E35ADF4A9EB28E9EE4BF4970F7@e1705><AANLkTikYiAOGtRnaehtwk_Ag9o_Egh0Dm9nvNfLGcXjz@mail.gmail.com><69B27340DF63444AB2001FB610FE9E6C@e1705><AANLkTi=2PpkY-RxRi_iyzvBAKm8NPC-sgUq=6EJXS83z@mail.gmail.com><339DD15147E24681AC1A175628B1D2A4@e1705>
	<AANLkTikbFiQ3xGBS_wheD_2-3Tt9akUojdCoaZy+KoKA@mail.gmail.com>
Message-ID: <B60BA1F7000040268849A6ED2A67C094@e1705>

Just tried flags{moderator,endconf}
for the conference owner
and flags{waste} for other
and works fine.

thanks
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 17, 2011 7:22 PM
  Subject: Re: [Freeswitch-users] mod_conference member-flags


  >From the wiki:


  "Can be any combination of: deaf, waste, mute-detect, dist-dtmf, moderator, endconf, mintwo."


  Give it a try and see what happens. Let us know.
  -MC


  On Mon, Jan 17, 2011 at 4:19 PM, Madovsky <infos at madovsky.org> wrote:

    so can I use membe-flags=moderator,endconf ?
      ----- Original Message ----- 
      From: Michael Collins 
      To: FreeSWITCH Users Help 
      Sent: Monday, January 17, 2011 7:06 PM
      Subject: Re: [Freeswitch-users] mod_conference member-flags


      Where did you read that? I've not tried it but I didn't see any indication that the two items are mutually exclusive. 


      -MC


      On Mon, Jan 17, 2011 at 3:54 PM, Madovsky <infos at madovsky.org> wrote:

        yes, but if I understand together it's not possible ?
          ----- Original Message ----- 
          From: Michael Collins 
          To: FreeSWITCH Users Help 
          Sent: Monday, January 17, 2011 2:44 PM
          Subject: Re: [Freeswitch-users] mod_conference member-flags


          Please confirm: you want the conference moderator also to have the flag where if he/she leaves the conference that the conference ends? 


          -MC


          On Sat, Jan 15, 2011 at 3:36 PM, Madovsky <infos at madovsky.org> wrote:

            How a "moderator" can be
            also "endconf" in same time ?

            Thanks

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From infos at madovsky.org  Fri Jan 21 03:52:08 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 20 Jan 2011 19:52:08 -0500
Subject: [Freeswitch-users] cepstral and SSML
References: <B619ED95945044D5934B36A8D1AEBEF2@e1705>
	<AANLkTikUgmfGH-4cBpPjuc57Rr6NPan8tQFibSbwvz_j@mail.gmail.com>
Message-ID: <12751511EAE149B0A7778CFC22E77C44@e1705>

Michael,

after pump up the debug level to 6,

I havve nothing as an error on logs :

EXECUTE sofia/internal/11111 at default speak(cepstral|david|<prosody volume='14'>Conference transfer. Please wait</prosody>)
2011-01-20 19:48:25.036446 [DEBUG] switch_ivr_play_say.c:2290 OPEN TTS cepstral
2011-01-20 19:48:25.037448 [DEBUG] switch_ivr_play_say.c:2299 Raw Codec Activated
2011-01-20 19:48:25.038454 [DEBUG] switch_ivr_play_say.c:1988 Speaking text: <prosody volume=14>Conference transfer. Please wait</prosody>
2011-01-20 19:48:25.827624 [DEBUG] switch_rtp.c:2699 Correct ip/port confirmed.
2011-01-20 19:48:28.031178 [DEBUG] switch_ivr_play_say.c:2180 done speaking text

I tried to put single, double and triple backslashes on every special char without success.
the voice says "slash blabla(don't understand)


  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 17, 2011 2:32 PM
  Subject: Re: [Freeswitch-users] cepstral and SSML


  Can you pastebin the debug output? That might give us a clue as to where the escaping needs to occur.
  -MC


  On Tue, Jan 11, 2011 at 6:08 PM, Madovsky <infos at madovsky.org> wrote:

    if I use

    <action application="speak" data="cepstral|david|<prosody volume='65'>Your credit is too low to reach the landline user phone. Transfer to voicemail.</prosody>"/>

    it seems that SSML isn't supported like this,
    the voice says "/ prosody" only. the whole sentence is not said.

    do I escape the SSML code in the data ?

    Thanks

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From infos at madovsky.org  Fri Jan 21 04:03:37 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 20 Jan 2011 20:03:37 -0500
Subject: [Freeswitch-users] moderator conference flag and nibblebill
Message-ID: <BEC4FD4A5E3A474EA4E9A1B6DB7D33F1@e1705>

if a caller in waste mode go to a conference
where the moderator isn't into nibblebill starts
to debit his account. I can't guess hwo I can manage
that in a dialplan to start debit once the conference really starts

Any ?
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From msc at freeswitch.org  Fri Jan 21 04:33:05 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 20 Jan 2011 17:33:05 -0800
Subject: [Freeswitch-users] conference pin issue
In-Reply-To: <AANLkTik8HaAUMpf0_=7q6h_xcTAVqGuP4HNQ9Ave3Xkn@mail.gmail.com>
References: <AANLkTik8HaAUMpf0_=7q6h_xcTAVqGuP4HNQ9Ave3Xkn@mail.gmail.com>
Message-ID: <AANLkTikiCV0zSR0AZyBS5Lv_tm7iTghFBKVMZTsf8_S9@mail.gmail.com>

George,

Thanks for taking the time to look at the source code and creating a simple
but useful patch. Your effort is definitely appreciated. Thank you also for
putting this into Jira so that we can keep track of these sorts of things.

-MC

On Thu, Jan 20, 2011 at 4:05 PM, George Niculae
<george.niculae79 at gmail.com>wrote:

> Hi All,
>
> I noticed that in a pin protected conference scenario, if the user inputs
> more digits than the length of the correct one the remaining digits are
> accounted for next retry - e.g. if the conf pin is 1234 and user enter 12345
> then 5 will be used as first digit in 2nd retry. Extension is configured as:
>
>   <extension name="5678">
>     <condition field="destination_number" expression="^uuu$">
>       <action application="set" data="supplied_pin=${sip_h_X-ConfPin}"/>
>       <action application="export" data="supplied_pin=${sip_h_X-ConfPin}"/>
>       <action application="bridge" data="loopback/app=conference:uuu at 5678
> +1234"/>
>     </condition>
>   </extension>
>
> I attached a patch for mod_conference that checks pin buffer to be 0 before
> starting to collect digits again and reset it if > 0. I also modified the
> code to play "please enter the conference access number" prompt just first
> time and only "the passcode you entered is not correct, please try again" on
> subsequent retries.
>
> Please review the patch and let me know your comments.
>
> Thanks,
> George
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
>
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From infos at madovsky.org  Fri Jan 21 05:33:12 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 20 Jan 2011 21:33:12 -0500
Subject: [Freeswitch-users] sample rate of a leg
Message-ID: <37089F18951B49A8BAA089D3451F88C4@e1705>

Have IVR 16000khz and 8000khz
and like to know in dialplan which sample rate
the legA is, as this I redirect to the right IVR rate
and avoid transcoding.
Is there a var for that ?

Thanks
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From infos at madovsky.org  Fri Jan 21 05:50:38 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 20 Jan 2011 21:50:38 -0500
Subject: [Freeswitch-users] conference list
Message-ID: <DBCC446AC5A943F1AEB555BF1707ED6C@e1705>

Sorry Brian, it's my last question before next week ;)

it's more a suggestion that a question,
when use api conference myconf list command
maybe it would be useful to know in the csv list 
which channel is moderator etc...

Thanks
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From fraserredmond at gmail.com  Fri Jan 21 06:13:08 2011
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Thu, 20 Jan 2011 22:13:08 -0500
Subject: [Freeswitch-users] Run dialplan tools from event socket
Message-ID: <AANLkTikg5+1aMoWxYr3UYO13iO2qA6FsG9FFi-Zg3_bD@mail.gmail.com>

Is there any way to run a dialplan tool from the event socket?

I have a dialplan that uses a dtmf to set up and perform an attended
transfer, like so:
<action application="bind_meta_app" data="8 a s
execute_extension::TransferCall XML transfer_call"/>

But I can't see any way to run the same thing from the event socket. I
thought doing an  "api uuid_transfer"  might do it, but that hangs up one of
the legs (no good for attended transfer.)

api uuid_transfer Uuid -bleg TransferCall XML transfer_call

As far as I can see, the closest thing is "sendmsg execute", but it looks
like you have to park a call/channel first to use that, so I'm not sure that
that is much use for attended transfer either.

Or should I be lame and do "api uuid_send_dtmf" to send * 8.

Cheers,
Fraser
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From thisjoy0528 at gmail.com  Fri Jan 21 06:20:20 2011
From: thisjoy0528 at gmail.com (joy this)
Date: Fri, 21 Jan 2011 11:20:20 +0800
Subject: [Freeswitch-users] Question about bind_digit_action
In-Reply-To: <AANLkTin_4ZxsXTAujCkfV7Vit7JJgzjfwxOviYBBEbkB@mail.gmail.com>
References: <AANLkTinRz4sOoxOOf_D1a+XiFSFe33o_0obTjHjgxwS-@mail.gmail.com>
	<AANLkTin_4ZxsXTAujCkfV7Vit7JJgzjfwxOviYBBEbkB@mail.gmail.com>
Message-ID: <AANLkTinjDTrzKUufVHdw=hGB=245mq5ZWzdvC9Z5mF5s@mail.gmail.com>

I only know that my FS is built on October. I will try to update. Thank you
for helping.
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From jmesquita at freeswitch.org  Fri Jan 21 06:48:36 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 21 Jan 2011 00:48:36 -0300
Subject: [Freeswitch-users] Run dialplan tools from event socket
In-Reply-To: <AANLkTikg5+1aMoWxYr3UYO13iO2qA6FsG9FFi-Zg3_bD@mail.gmail.com>
References: <AANLkTikg5+1aMoWxYr3UYO13iO2qA6FsG9FFi-Zg3_bD@mail.gmail.com>
Message-ID: <AANLkTin5fn9qMre-8xFAZD5_SUzbLbp_YueTRPPyh-=K@mail.gmail.com>

Lucky for you I have been working on this lately and the bad news is ...
there's no easy way to do it....

You can execute an extension like you said, but you have to park the legs
first... It would help to know how's the transfer_call extension so that I
can try to help you out, but maybe it is easier if you think of it this way:

When you use an app like att_xfer, the core already knows what to do next
with a call and parks the legs for you. If you do it on ESL, you've done it
half way and you didn't really park anything before you transfered the call.
When the bridge is undone, the leg that was not transfered doesn't know what
to do, has no applications to be run at this moment and so all it's left for
it is to let go.

A bit clearer? Att_xfer is a bit of a pain in the butt and it kinda requires
you to know a bit more of the inner workings of the state machine.

You can always execute att_xfer using ESL's execute (
http://wiki.freeswitch.org/wiki/Event_Socket_Library#execute) if you don't
care what happens to the legs afterwards but if you want to have control
over all 3 legs, no luck for you...

Regards,
Jo?o Mesquita


On Fri, Jan 21, 2011 at 12:13 AM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> Is there any way to run a dialplan tool from the event socket?
>
> I have a dialplan that uses a dtmf to set up and perform an attended
> transfer, like so:
> <action application="bind_meta_app" data="8 a s
> execute_extension::TransferCall XML transfer_call"/>
>
> But I can't see any way to run the same thing from the event socket. I
> thought doing an  "api uuid_transfer"  might do it, but that hangs up one of
> the legs (no good for attended transfer.)
>
> api uuid_transfer Uuid -bleg TransferCall XML transfer_call
>
> As far as I can see, the closest thing is "sendmsg execute", but it looks
> like you have to park a call/channel first to use that, so I'm not sure that
> that is much use for attended transfer either.
>
> Or should I be lame and do "api uuid_send_dtmf" to send * 8.
>
> Cheers,
> Fraser
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Fri Jan 21 06:50:09 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 21 Jan 2011 00:50:09 -0300
Subject: [Freeswitch-users] Question about bind_digit_action
In-Reply-To: <AANLkTinjDTrzKUufVHdw=hGB=245mq5ZWzdvC9Z5mF5s@mail.gmail.com>
References: <AANLkTinRz4sOoxOOf_D1a+XiFSFe33o_0obTjHjgxwS-@mail.gmail.com>
	<AANLkTin_4ZxsXTAujCkfV7Vit7JJgzjfwxOviYBBEbkB@mail.gmail.com>
	<AANLkTinjDTrzKUufVHdw=hGB=245mq5ZWzdvC9Z5mF5s@mail.gmail.com>
Message-ID: <AANLkTinyfq2myUxftuvxgbxL-5wP59tzWxbPECJa-hsE@mail.gmail.com>

I am almost positive that bind_digit_action was NOT in the code back in
October.. That's like dinosaur version nowadays.

Regards,
Jo?o Mesquita


On Fri, Jan 21, 2011 at 12:20 AM, joy this <thisjoy0528 at gmail.com> wrote:

> I only know that my FS is built on October. I will try to update. Thank you
> for helping.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Fri Jan 21 07:07:42 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 20 Jan 2011 23:07:42 -0500
Subject: [Freeswitch-users] sample rate of a leg
References: <4d38fd8f.017b0e0a.2151.2701@mx.google.com>
Message-ID: <5737230E3AAE4C0182E3A74CD2B066C9@e1705>

oops, didn't see before,
it's read_rate var

thanks
  ----- Original Message ----- 
  From: msc at freeswitch.org 
  To: Madovsky 
  Sent: Thursday, January 20, 2011 10:29 PM
  Subject: Re: [Freeswitch-users] sample rate of a leg


  I believe there is. I am afk atm so I recommend sending a call to the info ext and looking for a variable like "rate"

  -MC

  Sent from my HTC on the Now Network from Sprint!


  ----- Reply message -----
  From: "Madovsky" <infos at madovsky.org>
  Date: Thu, Jan 20, 2011 6:33 pm
  Subject: [Freeswitch-users] sample rate of a leg
  To: <freeswitch-users at lists.freeswitch.org>


  Have IVR 16000khz and 8000khz
  and like to know in dialplan which sample rate
  the legA is, as this I redirect to the right IVR rate
  and avoid transcoding.
  Is there a var for that ?

  Thanks
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From cmrienzo at gmail.com  Fri Jan 21 07:18:13 2011
From: cmrienzo at gmail.com (Chris Rienzo)
Date: Thu, 20 Jan 2011 23:18:13 -0500
Subject: [Freeswitch-users] Run dialplan tools from event socket
In-Reply-To: <AANLkTikg5+1aMoWxYr3UYO13iO2qA6FsG9FFi-Zg3_bD@mail.gmail.com>
References: <AANLkTikg5+1aMoWxYr3UYO13iO2qA6FsG9FFi-Zg3_bD@mail.gmail.com>
Message-ID: <A4DD8D97-0D17-40B2-B41A-53C6622F48E7@gmail.com>

Why not go with the lame solution if it works?  It seems simple.  



On Jan 20, 2011, at 22:13, Fraser Redmond <fraserredmond at gmail.com> wrote:

> Is there any way to run a dialplan tool from the event socket?
> 
> I have a dialplan that uses a dtmf to set up and perform an attended transfer, like so:
> <action application="bind_meta_app" data="8 a s execute_extension::TransferCall XML transfer_call"/>
> 
> But I can't see any way to run the same thing from the event socket. I thought doing an  "api uuid_transfer"  might do it, but that hangs up one of the legs (no good for attended transfer.)
> 
> api uuid_transfer Uuid -bleg TransferCall XML transfer_call
> 
> As far as I can see, the closest thing is "sendmsg execute", but it looks like you have to park a call/channel first to use that, so I'm not sure that that is much use for attended transfer either.
> 
> Or should I be lame and do "api uuid_send_dtmf" to send * 8.
> 
> Cheers,
> Fraser
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From boris at tagnet.ru  Fri Jan 21 07:25:04 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Fri, 21 Jan 2011 09:25:04 +0500
Subject: [Freeswitch-users] PCMA stranges
In-Reply-To: <AANLkTi=wcUEe89em_-c2DaFbBPunWum9Oh10yjj1KTXw@mail.gmail.com>
References: <4D380FF8.7080705@tagnet.ru>	<E676EB80-7927-4061-BC2C-497DBA81D75E@ipeva.fr>	<4D381E1F.7040202@tagnet.ru>
	<4D384196.105@tagnet.ru>	<AANLkTikVAt6t8K4fxg0GXisw=0_+s5gBYe9Hyvep5KN2@mail.gmail.com>	<4D386528.2010506@tagnet.ru>
	<AANLkTi=wcUEe89em_-c2DaFbBPunWum9Oh10yjj1KTXw@mail.gmail.com>
Message-ID: <4D390AA0.8060908@tagnet.ru>

Hello!

     Yes, I tried and saw no difference but only audio attribute. I may 
post traces here if someone want to look at them.

> Have you compared the SIP traces between the working g711u and the 
> non-working g711a? What seems to be different between the two?
> -MC
>
> On Thu, Jan 20, 2011 at 8:39 AM, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     Hello!
>
>         Yes, I know the media is ignored, so I wondering about this codec
>     strangenes. What test I have done:
>
>     SoftPhone (g711ulaw) -> FS -> Cisco 5350 : working
>     SoftPhone (g711alaw, proxy_media=true) -> FS -> Cisco 5350 : not
>     working
>     SoftPhone (g711alaw, proxy_media=false) -> FS -> Cisco 5350 : working
>     SoftPhone (g711alaw) -> Cisco 5350 : working
>
>
>     > proxy media means media is ignored, its not ever touched.  Its
>     > designed to transparently pass media even if FS is unaware of the
>     > codec.
>     > So if you say you are using proxy media and you have codec specific
>     > problems it seems peculiar.
>     >
>     >
>     > On Thu, Jan 20, 2011 at 8:07 AM, Boris Kovalenko<boris at tagnet.ru
>     <mailto:boris at tagnet.ru>>  wrote:
>     >> Hello!
>     >>
>     >>      I found that problem is somewhere inside freeswitch.
>     Direct call between
>     >> SoftPhone and Cisco is working fine.
>     >>
>     >> Hello!
>     >>
>     >>      David, may You explain why this problem is with G711A
>     only? When I use
>     >> G711U there are no problems and call is established and I may talk.
>     >>
>     >> The Cisco never sends the 200/OK after the 183, so the call is not
>     >> established.
>     >> David Ponzone  Direction Technique
>     >> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>     >> tel:      01 74 03 18 97
>     >> gsm:   06 66 98 76 34
>     >> Service Client IPeva
>     >> tel:      0811 46 26 26
>     >> www.ipeva.fr <http://www.ipeva.fr>  - www.ipeva-studio.com
>     <http://www.ipeva-studio.com>
>     >> Ce message et toutes les pi?ces jointes sont confidentiels et
>     ?tablis ?
>     >> l'intention exclusive de ses destinataires. Toute utilisation
>     ou diffusion
>     >> non autoris?e est interdite. Tout message ?lectronique est
>     susceptible
>     >> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce
>     message s'il
>     >> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>     destinataire de ce
>     >> message, merci de le d?truire imm?diatement et d'avertir
>     l'exp?diteur.
>     >>
>     >>
>     >>
>     >> Le 20/01/2011 ? 11:35, Boris Kovalenko a ?crit :
>     >>
>     >> Hello!
>     >>
>     >>      My network configuration is:
>     >> Softphone (eyeBeam 1.5) with only G711alaw enabled ->
>      FreeSWITCH Version
>     >> 1.0.head (git-0cf1d54 2011-01-19 16-36-04 -0500) ->  Cisco AS5350
>     >>
>     >> proxy_media=true in profile.
>     >> So, when I do a test call, there is SDP but no RTP between FS
>     and CISCO,
>     >> so I can't hear voice. With G711ulaw there are no problems. What is
>     >> wrong? Siptrace below:
>     >>
>     >>    
>     ------------------------------------------------------------------------
>     >>     INVITE sip:73435327569 at default SIP/2.0
>     >>     Via: SIP/2.0/UDP
>     >>
>     X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
>     >>     Max-Forwards: 70
>     >>     Contact:<sip:50001 at X.X.29.123:21556>
>     >>     To: "73435327569"<sip:73435327569 at default>
>     >>     From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>     >>     Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>     >>     CSeq: 1 INVITE
>     >>     Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY,
>     MESSAGE,
>     >> SUBSCRIBE, INFO
>     >>     Content-Type: application/sdp
>     >>     User-Agent: eyeBeam release 1102u stamp 52345
>     >>     Content-Length: 337
>     >>
>     >>     v=0
>     >>     o=- 1 2 IN IP4 192.168.3.253
>     >>     s=CounterPath eyeBeam 1.5
>     >>     c=IN IP4 192.168.3.253
>     >>     t=0 0
>     >>     m=audio 40032 RTP/AVP 8 101
>     >>     a=alt:1 3 : XdMyLUqb Fs2PLxKV 192.168.3.253 40032
>     >>     a=alt:2 2 : 2qrc/CQ7 Z/9XFQZl 172.16.3.253 40032
>     >>     a=alt:3 1 : 8BOwI53P +ENGEr/U X.X.29.123 40032
>     >>     a=fmtp:101 0-15
>     >>     a=rtpmap:101 telephone-event/8000
>     >>     a=sendrecv
>     >>    
>     ------------------------------------------------------------------------
>     >> send 385 bytes to udp/[X.X.29.123]:21556 at 10:30:31.313366:
>     >>    
>     ------------------------------------------------------------------------
>     >>     SIP/2.0 100 Trying
>     >>     Via: SIP/2.0/UDP
>     >>
>     X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>     >>     From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>     >>     To: "73435327569"<sip:73435327569 at default>
>     >>     Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>     >>     CSeq: 1 INVITE
>     >>     User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54
>     2011-01-19
>     >> 16-36-04 -0500
>     >>     Content-Length: 0
>     >>
>     >>    
>     ------------------------------------------------------------------------
>     >> 2011-01-20 15:30:31.313448 [NOTICE] switch_channel.c:808 New
>     Channel
>     >> sofia/epbx/50001 at default [95698968-8cf9-40a2-91ad-1322fcfc76af]
>     >> 2011-01-20 15:30:31.322338 [INFO] mod_dialplan_xml.c:331 Processing
>     >> TAGNet<50001>->73435327569 in context public
>     >> 2011-01-20 15:30:31.323362 [INFO] mod_dialplan_xml.c:331 Processing
>     >> TAGNet<50001>->ext_translate_extsrc in context features
>     >> 2011-01-20 15:30:31.324357 [NOTICE] switch_ivr.c:1606 Transfer
>     >> sofia/epbx/50001 at default to XML[73435327569 at top.ctx]
>     >> 2011-01-20 15:30:31.325352 [INFO] mod_dialplan_xml.c:331 Processing
>     >> TAGNet<50001>->73435327569 in context top.ctx
>     >> 2011-01-20 15:30:31.326344 [NOTICE] mod_dptools.c:1174
>     [top.ctx] - ext_local
>     >> 2011-01-20 15:30:31.329360 [NOTICE] switch_channel.c:808 New
>     Channel
>     >> sofia/epbx/73435327569 at X.X.16.83:5060
>     [b9654c75-039a-4780-bc00-0fae65f92a9a]
>     >> send 1276 bytes to udp/[X.X.16.83]:5060 at 10:30:31.330623:
>     >>    
>     ------------------------------------------------------------------------
>     >>     INVITE sip:73435327569 at X.X.16.83:5060 SIP/2.0
>     >>     Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>     >>     Max-Forwards: 68
>     >>     From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>     >>     To:<sip:73435327569 at X.X.16.83:5060>
>     >>     Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>     >>     CSeq: 7423651 INVITE
>     >>     Contact:<sip:mod_sofia at Y.Y.138.187:5060>
>     >>     User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54
>     2011-01-19
>     >> 16-36-04 -0500
>     >>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
>     INFO,
>     >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>     >>     Supported: timer, precondition, path, replaces
>     >>     Allow-Events: talk, hold, presence, dialog, line-seize,
>     call-info,
>     >> sla, include-session-description, presence.winfo,
>     message-summary, refer
>     >>     Content-Type: application/sdp
>     >>     Content-Disposition: session
>     >>     Content-Length: 341
>     >>     X-FS-Support: update_display
>     >>     Remote-Party-ID: "TAGNet"
>     >> <sip:50001 at Y.Y.138.187>;party=calling;screen=yes;privacy=off
>     >>
>     >>     v=0
>     >>     o=FreeSWITCH 1627720692 1627720693 IN IP4 Y.Y.138.187
>     >>     s=FreeSWITCH
>     >>     c=IN IP4 Y.Y.138.187
>     >>     t=0 0
>     >>     m=audio 23442 RTP/AVP 8 101
>     >>     a=rtpmap:101 telephone-event/8000
>     >>     a=fmtp:101 0-15
>     >>     a=alt:1 3 : XdMyLUqb Fs2PLxKV 192.168.3.253 40032
>     >>     a=alt:2 2 : 2qrc/CQ7 Z/9XFQZl 172.16.3.253 40032
>     >>     a=alt:3 1 : 8BOwI53P +ENGEr/U X.X.29.123 40032
>     >>    
>     ------------------------------------------------------------------------
>     >> recv 396 bytes from udp/[X.X.16.83]:5060 at 10:30:31.341106:
>     >>    
>     ------------------------------------------------------------------------
>     >>     SIP/2.0 100 Trying
>     >>     Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>     >>     From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>     >>     To:<sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>     >>     Date: Thu, 20 Jan 2011 10:30:31 GMT
>     >>     Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>     >>     Server: Cisco-SIPGateway/IOS-12.x
>     >>     CSeq: 7423651 INVITE
>     >>     Allow-Events: telephone-event
>     >>     Content-Length: 0
>     >>
>     >>    
>     ------------------------------------------------------------------------
>     >> recv 814 bytes from udp/[X.X.16.83]:5060 at 10:30:32.401494:
>     >>    
>     ------------------------------------------------------------------------
>     >>     SIP/2.0 183 Session Progress
>     >>     Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>     >>     From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>     >>     To:<sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>     >>     Date: Thu, 20 Jan 2011 10:30:31 GMT
>     >>     Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>     >>     Server: Cisco-SIPGateway/IOS-12.x
>     >>     CSeq: 7423651 INVITE
>     >>     Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
>     >> SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
>     >>     Allow-Events: telephone-event
>     >>     Contact:<sip:73435327569 at X.X.16.83:5060>
>     >>     Content-Disposition: session;handling=required
>     >>     Content-Type: application/sdp
>     >>     Content-Length: 176
>     >>
>     >>     v=0
>     >>     o=CiscoSystemsSIP-GW-UserAgent 4564 1578 IN IP4 X.X.16.83
>     >>     s=SIP Call
>     >>     c=IN IP4 X.X.16.83
>     >>     t=0 0
>     >>     m=audio 17770 RTP/AVP 8
>     >>     c=IN IP4 X.X.16.83
>     >>     a=rtpmap:8 PCMA/8000
>     >>    
>     ------------------------------------------------------------------------
>     >> 2011-01-20 15:30:32.401362 [INFO] sofia.c:729
>     >> sofia/epbx/73435327569 at X.X.16.83:5060 Update Callee ID to "Outbound
>     >> Call"<73435327569>
>     >> 2011-01-20 15:30:32.409363 [NOTICE] sofia.c:4739 Pre-Answer
>     >> sofia/epbx/73435327569 at X.X.16.83:5060!
>     >> 2011-01-20 15:30:32.415412 [INFO] switch_ivr_originate.c:3345
>     Sending
>     >> early media
>     >> 2011-01-20 15:30:32.416367 [NOTICE] mod_sofia.c:2252 Pre-Answer
>     >> sofia/epbx/50001 at default!
>     >> send 1145 bytes to udp/[X.X.29.123]:21556 at 10:30:32.418195:
>     >>    
>     ------------------------------------------------------------------------
>     >>     SIP/2.0 183 Session Progress
>     >>     Via: SIP/2.0/UDP
>     >>
>     X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>     >>     From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>     >>     To: "73435327569"<sip:73435327569 at default>;tag=BScUp0QXHFBgB
>     >>     Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>     >>     CSeq: 1 INVITE
>     >>     Contact:<sip:73435327569 at Y.Y.138.187:5060;transport=udp>
>     >>     User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54
>     2011-01-19
>     >> 16-36-04 -0500
>     >>     Accept: application/sdp
>     >>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
>     INFO,
>     >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>     >>     Supported: timer, precondition, path, replaces
>     >>     Allow-Events: talk, hold, presence, dialog, line-seize,
>     call-info,
>     >> sla, include-session-description, presence.winfo,
>     message-summary, refer
>     >>     Content-Type: application/sdp
>     >>     Content-Disposition: session
>     >>     Content-Length: 181
>     >>     Remote-Party-ID: "73435327569"
>     >> <sip:73435327569 at Y.Y.138.187>;party=calling;privacy=off;screen=no
>     >>
>     >>     v=0
>     >>     o=FreeSWITCH 1627752513 1627752514 IN IP4 Y.Y.138.187
>     >>     s=FreeSWITCH
>     >>     c=IN IP4 Y.Y.138.187
>     >>     t=0 0
>     >>     m=audio 25806 RTP/AVP 8
>     >>     c=IN IP4 Y.Y.138.187
>     >>     a=rtpmap:8 PCMA/8000
>     >>
>     >>    
>     ------------------------------------------------------------------------
>     >> recv 365 bytes from udp/[X.X.29.123]:21556 at 10:30:39.196791:
>     >>    
>     ------------------------------------------------------------------------
>     >>     CANCEL sip:73435327569 at default SIP/2.0
>     >>     Via: SIP/2.0/UDP
>     >>
>     X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
>     >>     To: "73435327569"<sip:73435327569 at default>
>     >>     From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>     >>     Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>     >>     CSeq: 1 CANCEL
>     >>     User-Agent: eyeBeam release 1102u stamp 52345
>     >>     Content-Length: 0
>     >>
>     >>    
>     ------------------------------------------------------------------------
>     >> send 320 bytes to udp/[X.X.29.123]:21556 at 10:30:39.196905:
>     >>    
>     ------------------------------------------------------------------------
>     >>     SIP/2.0 200 OK
>     >>     Via: SIP/2.0/UDP
>     >>
>     X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>     >>     From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>     >>     To: "73435327569"<sip:73435327569 at default>;tag=BScUp0QXHFBgB
>     >>     Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>     >>     CSeq: 1 CANCEL
>     >>     Content-Length: 0
>     >>
>     >>    
>     ------------------------------------------------------------------------
>     >> send 715 bytes to udp/[X.X.29.123]:21556 at 10:30:39.197002:
>     >>    
>     ------------------------------------------------------------------------
>     >>     SIP/2.0 487 Request Terminated
>     >>     Via: SIP/2.0/UDP
>     >>
>     X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>     >>     From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>     >>     To: "73435327569"<sip:73435327569 at default>;tag=BScUp0QXHFBgB
>     >>     Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>     >>     CSeq: 1 INVITE
>     >>     User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54
>     2011-01-19
>     >> 16-36-04 -0500
>     >>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
>     INFO,
>     >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>     >>     Supported: timer, precondition, path, replaces
>     >>     Allow-Events: talk, hold, presence, dialog, line-seize,
>     call-info,
>     >> sla, include-session-description, presence.winfo,
>     message-summary, refer
>     >>     Content-Length: 0
>     >>
>     >>    
>     ------------------------------------------------------------------------
>     >> 2011-01-20 15:30:39.196838 [NOTICE] sofia.c:5286 Hangup
>     >> sofia/epbx/50001 at default [CS_EXECUTE] [ORIGINATOR_CANCEL]
>     >> 2011-01-20 15:30:39.196838 [NOTICE] switch_ivr_bridge.c:653 Hangup
>     >> sofia/epbx/73435327569 at X.X.16.83:5060 [CS_EXCHANGE_MEDIA]
>     [NORMAL_CLEARING]
>     >> recv 331 bytes from udp/[X.X.29.123]:21556 at 10:30:39.198834:
>     >>    
>     ------------------------------------------------------------------------
>     >>     ACK sip:73435327569 at default SIP/2.0
>     >>     Via: SIP/2.0/UDP
>     >>
>     X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
>     >>     To: "73435327569"<sip:73435327569 at default>;tag=BScUp0QXHFBgB
>     >>     From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>     >>     Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>     >>     CSeq: 1 ACK
>     >>     Content-Length: 0
>     >>
>     >>    
>     ------------------------------------------------------------------------
>     >> 2011-01-20 15:30:39.204838 [NOTICE] switch_core_session.c:1306
>     Session
>     >> 99 (sofia/epbx/50001 at default) Ended
>     >> 2011-01-20 15:30:39.204838 [NOTICE] switch_core_session.c:1308
>     Close
>     >> Channel sofia/epbx/50001 at default [CS_DESTROY]
>     >> send 372 bytes to udp/[X.X.16.83]:5060 at 10:30:39.212574:
>     >>    
>     ------------------------------------------------------------------------
>     >>     CANCEL sip:73435327569 at X.X.16.83:5060 SIP/2.0
>     >>     Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>     >>     Max-Forwards: 68
>     >>     From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>     >>     To:<sip:73435327569 at X.X.16.83:5060>
>     >>     Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>     >>     CSeq: 7423651 CANCEL
>     >>     Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>     >>     Content-Length: 0
>     >>
>     >>    
>     ------------------------------------------------------------------------
>     >> 2011-01-20 15:30:39.212840 [NOTICE] switch_core_session.c:1306
>     Session
>     >> 100 (sofia/epbx/73435327569 at X.X.16.83:5060) Ended
>     >> 2011-01-20 15:30:39.212840 [NOTICE] switch_core_session.c:1308
>     Close
>     >> Channel sofia/epbx/73435327569 at X.X.16.83:5060 [CS_DESTROY]
>     >> recv 311 bytes from udp/[X.X.16.83]:5060 at 10:30:39.216838:
>     >>    
>     ------------------------------------------------------------------------
>     >>     SIP/2.0 200 OK
>     >>     Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>     >>     From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>     >>     To:<sip:73435327569 at X.X.16.83:5060>
>     >>     Date: Thu, 20 Jan 2011 10:30:39 GMT
>     >>     Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>     >>     Content-Length: 0
>     >>     CSeq: 7423651 CANCEL
>     >>
>     >>    
>     ------------------------------------------------------------------------
>     >> recv 407 bytes from udp/[X.X.16.83]:5060 at 10:30:39.219180:
>     >>    
>     ------------------------------------------------------------------------
>     >>     SIP/2.0 487 Request Cancelled
>     >>     Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>     >>     From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>     >>     To:<sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>     >>     Date: Thu, 20 Jan 2011 10:30:39 GMT
>     >>     Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>     >>     Server: Cisco-SIPGateway/IOS-12.x
>     >>     CSeq: 7423651 INVITE
>     >>     Allow-Events: telephone-event
>     >>     Content-Length: 0
>     >>
>     >>    
>     ------------------------------------------------------------------------
>     >> send 334 bytes to udp/[X.X.16.83]:5060 at 10:30:39.219276:
>     >>    
>     ------------------------------------------------------------------------
>     >>     ACK sip:73435327569 at X.X.16.83:5060 SIP/2.0
>     >>     Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>     >>     Max-Forwards: 68
>     >>     From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>     >>     To:<sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>     >>     Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>     >>     CSeq: 7423651 ACK
>     >>     Content-Length: 0
>     >>
>     >>    
>     ------------------------------------------------------------------------
>     >>
>     >>
>     >> --
>     >> ? ?????????,
>     >>    ????? ?????????
>     >>    ??? "??????"
>     >>    (3435) 494991
>     >>
>     >>
>     >>
>     >> _______________________________________________
>     >> FreeSWITCH-users mailing list
>     >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >> http://www.freeswitch.org
>     >>
>     >>
>     >> _______________________________________________
>     >> FreeSWITCH-users mailing list
>     >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >> http://www.freeswitch.org
>     >>
>     >>
>     >> --
>     >> ? ?????????,
>     >>    ????? ?????????
>     >>    ??? "??????"
>     >>    (3435) 494991
>     >>
>     >>
>     >> _______________________________________________
>     >> FreeSWITCH-users mailing list
>     >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >> http://www.freeswitch.org
>     >>
>     >>
>     >> --
>     >> ? ?????????,
>     >>    ????? ?????????
>     >>    ??? "??????"
>     >>    (3435) 494991
>     >>
>     >>
>     >> _______________________________________________
>     >> FreeSWITCH-users mailing list
>     >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >> http://www.freeswitch.org
>     >>
>     >>
>     >
>     >
>
>
>     --
>     ? ?????????,
>       ????? ?????????
>       ??? "??????"
>       (3435) 494991
>
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991


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From george at ezuce.com  Fri Jan 21 03:31:54 2011
From: george at ezuce.com (George Niculae)
Date: Fri, 21 Jan 2011 02:31:54 +0200
Subject: [Freeswitch-users] conference pin issue
In-Reply-To: <902C3395-87EE-4084-A771-CE911419D9BC@freeswitch.org>
References: <AANLkTik8HaAUMpf0_=7q6h_xcTAVqGuP4HNQ9Ave3Xkn@mail.gmail.com>
	<0C8CA92D-9EAA-4C51-8182-F1CB594CA868@freeswitch.org>
	<26221ECF-AA7D-4937-9F23-477FD4B0A681@seventhsignal.de>
	<902C3395-87EE-4084-A771-CE911419D9BC@freeswitch.org>
Message-ID: <AANLkTikANX2OXsi+_5BzUFSn49APeU0Z55Rrdn2C2Hdg@mail.gmail.com>

Moved to JIRA, please see http://jira.freeswitch.org/browse/FS-3000

Thanks,
George

On Fri, Jan 21, 2011 at 2:23 AM, Brian West <brian at freeswitch.org> wrote:
> Yep.
> /b
> On Jan 20, 2011, at 6:19 PM, Michal Bielicki wrote:
>
> Nevertheless very cool patch :)
> Am 21.01.2011 um 01:11 schrieb Brian West:
>
> Please post to?jira.freeswitch.org
>
> /b
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From peter at phpwerks.com  Fri Jan 21 05:19:43 2011
From: peter at phpwerks.com (Peter Brenner)
Date: Thu, 20 Jan 2011 21:19:43 -0500
Subject: [Freeswitch-users] Making a Call from PHP
In-Reply-To: <AANLkTinCLKSKJQHBCZWPcnyuj+BZdYpfsZuE3VyL+N=K@mail.gmail.com>
References: <4D38A4C0.20800@phpwerks.com>
	<AANLkTinCLKSKJQHBCZWPcnyuj+BZdYpfsZuE3VyL+N=K@mail.gmail.com>
Message-ID: <4D38ED3F.9060507@phpwerks.com>

Michael,

Thank you for you response and suggestions.  I tried using the syntax 
you provided but received this error
[ERR] switch_ivr_originate.c:2628 Cannot create outgoing channel of type 
[error] cause: [USER_NOT_REGISTERED]

Which indicates that I do not have a user registered.  This is where I 
am lost, I have searched the documentation and can't determine how to 
register a user.  I have softphone (xlite phone) configured with an 
account (user 1001 @ 192.168.1.254) from which I can make called to 
extensions. From PHP it fails. The phone is available when I attempt to 
run my php script.

Is there a config setting I need to set?  My freeswitch installation is 
a generic install with no modifications - the /conf/directory/default 
contains definitions for users 1000 - 1019 - these appear to get run by 
default.xml

The ultimate goal for this test is to create a simple page that accepts 
a number, dials and connects to an internal number.

Sorry for what is probably an elementary question, but any guidance 
would be greatly appreciated.

Peter


On 1/20/11 4:35 PM, Michael Collins wrote:
> I believe we talked about this on IRC but for posterity's sake I'll 
> repeat here: your originate syntax is not correct. It can be confusing 
> because there are several ways to accomplish the same thing. In your 
> case I recommend using this syntax:
>
> api originate user/1000 1001
>
> The first argument to originate is a dialstring. If you know that you 
> are going to be dialing a locally registered user then the "user" 
> channel is easier than saying "sofia/internal/1000%${domain}". The 
> second argument to originate is a dialplan extension or and 
> application to execute. In the above example the first leg calls 
> user/1000 and when he/she answers it then sends the other leg through 
> the dialplan as if user 1000 had dialed "1001" and pressed send. I 
> hope that makes sense.
>
> Let us know if you continue to have issues with this. You are VERY 
> close to having this working - just a few characters in the dialstring.
>
> -MC
>
> On Thu, Jan 20, 2011 at 1:10 PM, Peter Brenner <peter at phpwerks.com 
> <mailto:peter at phpwerks.com>> wrote:
>
>     Very new to Freeswitch and I am trying to work through some
>     examples of
>     connecting/interacting with Freeswitch from PHP.  The example that
>     I am
>     trying to work through is one where a user enters 2 extension
>     numbers on
>     a page, submits the page which then originates a call from one
>     extension
>     to the other.
>
>     I used the follow in example as a starting point:
>
>     http://wiki.freeswitch.org/wiki/PHP_Event_Socket
>     I changed the command:
>     $cmd = "api help";
>
>     To:
>
>     $cmd = "api originate
>     sofia/internal/1000&bridge(sofia/internal/1001)";
>
>     I want to call from extension 1000 to 1001
>
>     I receive the following error
>
>     2011-01-20 16:01:16.482373 [WARNING] mod_sofia.c:4022 Cannot
>     locate registered user 1000 at internal
>     2011-01-20 16:01:16.482373 [NOTICE] mod_sofia.c:4221 Close Channel
>     N/A [CS_NEW]
>     2011-01-20 16:01:16.482373 [ERR] switch_ivr_originate.c:2628
>     Cannot create outgoing channel of type [sofia] cause:
>     [USER_NOT_REGISTERED]
>
>     If I change the command to
>     $cmd = "api originate sofia/internal/1000 at 192.168.1.211
>     <mailto:1000 at 192.168.1.211>&bridge(sofia/internal/1001)";
>
>
>     2011-01-20 16:04:00.602758 [NOTICE] switch_channel.c:808 New
>     Channel sofia/internal/1000 at 192.168.1.211
>     <mailto:1000 at 192.168.1.211> [c721e1e2-7eb7-46d5-9507-dcf02ac33828]
>     2011-01-20 16:04:00.602758 [NOTICE] switch_channel.c:808 New
>     Channel sofia/internal/0000000000 at 192.168.1.211
>     <mailto:0000000000 at 192.168.1.211>
>     [b7b163eb-3c90-4b86-9fde-070abdfbb45b]
>     2011-01-20 16:04:00.612832 [INFO] mod_dialplan_xml.c:331
>     Processing<0000000000>->1000 in context public
>     2011-01-20 16:04:00.625071 [ERR] sofia.c:5869 Cannot Blind
>     Transfer 1 Legged calls
>     2011-01-20 16:04:00.625071 [NOTICE] sofia.c:5286 Hangup
>     sofia/internal/1000 at 192.168.1.211 <mailto:1000 at 192.168.1.211>
>     [CS_CONSUME_MEDIA] [NO_USER_RESPONSE]
>     2011-01-20 16:04:00.654406 [NOTICE] switch_core_session.c:1306
>     Session 2 (sofia/internal/1000 at 192.168.1.211
>     <mailto:1000 at 192.168.1.211>) Ended
>     2011-01-20 16:04:00.654406 [NOTICE] switch_core_session.c:1308
>     Close Channel sofia/internal/1000 at 192.168.1.211
>     <mailto:1000 at 192.168.1.211> [CS_DESTROY]
>     2011-01-20 16:04:00.725081 [NOTICE]
>     switch_core_state_machine.c:189
>     sofia/internal/0000000000 at 192.168.1.211
>     <mailto:0000000000 at 192.168.1.211> has executed the last dialplan
>     instruction, hanging up.
>     2011-01-20 16:04:00.725081 [NOTICE]
>     switch_core_state_machine.c:191 Hangup
>     sofia/internal/0000000000 at 192.168.1.211
>     <mailto:0000000000 at 192.168.1.211> [CS_EXECUTE] [NORMAL_CLEARING]
>     2011-01-20 16:04:00.730022 [NOTICE] switch_core_session.c:1306
>     Session 3 (sofia/internal/0000000000 at 192.168.1.211
>     <mailto:0000000000 at 192.168.1.211>) Ended
>     2011-01-20 16:04:00.730022 [NOTICE] switch_core_session.c:1308
>     Close Channel sofia/internal/0000000000 at 192.168.1.211
>     <mailto:0000000000 at 192.168.1.211> [CS_DESTROY]
>
>     Here is the output from sofia status
>
>                          Name          Type                          
>                 Data      State
>     =================================================================================================
>                      external       profile
>     sip:mod_sofia at 192.168.1.211:5080
>     <http://sip:mod_sofia at 192.168.1.211:5080>      RUNNING (0)
>         external::example.com <http://example.com>       gateway
>     sip:joeuser at example.com <mailto:sip%3Ajoeuser at example.com>      NOREG
>                 internal-ipv6       profile                  
>     sip:mod_sofia@[::1]:5060      RUNNING (0)
>                      internal       profile
>     sip:mod_sofia at 192.168.1.211:5060
>     <http://sip:mod_sofia at 192.168.1.211:5060>      RUNNING (0)
>                 192.168.1.211         alias                          
>             internal      ALIASED
>     =================================================================================================
>     3 profiles 1 alias
>
>     Any advice or help would be greatly appreciated.
>
>     thanks!
>     Peter
>
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From boris at tagnet.ru  Fri Jan 21 07:37:33 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Fri, 21 Jan 2011 09:37:33 +0500
Subject: [Freeswitch-users] PCMA stranges
In-Reply-To: <AANLkTimbouEk+WRzarw7q6PrjcBfDupd+xxS=6_XdvO0@mail.gmail.com>
References: <4D380FF8.7080705@tagnet.ru>	<E676EB80-7927-4061-BC2C-497DBA81D75E@ipeva.fr>	<4D381E1F.7040202@tagnet.ru>
	<4D384196.105@tagnet.ru>	<AANLkTikVAt6t8K4fxg0GXisw=0_+s5gBYe9Hyvep5KN2@mail.gmail.com>	<4D386528.2010506@tagnet.ru>	<AANLkTi=wcUEe89em_-c2DaFbBPunWum9Oh10yjj1KTXw@mail.gmail.com>
	<AANLkTimbouEk+WRzarw7q6PrjcBfDupd+xxS=6_XdvO0@mail.gmail.com>
Message-ID: <4D390D8D.10000@tagnet.ru>

Hello!

     Yes! It's fixed! Thank You, Anthony!
> I just added a patch for another fix that I think might help you, try latest GIT
>
> On Thu, Jan 20, 2011 at 3:29 PM, Michael Collins<msc at freeswitch.org>  wrote:
>> Have you compared the SIP traces between the working g711u and the
>> non-working g711a? What seems to be different between the two?
>> -MC
>>
>> On Thu, Jan 20, 2011 at 8:39 AM, Boris Kovalenko<boris at tagnet.ru>  wrote:
>>> Hello!
>>>
>>>      Yes, I know the media is ignored, so I wondering about this codec
>>> strangenes. What test I have done:
>>>
>>> SoftPhone (g711ulaw) ->  FS ->  Cisco 5350 : working
>>> SoftPhone (g711alaw, proxy_media=true) ->  FS ->  Cisco 5350 : not working
>>> SoftPhone (g711alaw, proxy_media=false) ->  FS ->  Cisco 5350 : working
>>> SoftPhone (g711alaw) ->  Cisco 5350 : working
>>>
>>>
>>>> proxy media means media is ignored, its not ever touched.  Its
>>>> designed to transparently pass media even if FS is unaware of the
>>>> codec.
>>>> So if you say you are using proxy media and you have codec specific
>>>> problems it seems peculiar.
>>>>
>>>>
>>>> On Thu, Jan 20, 2011 at 8:07 AM, Boris Kovalenko<boris at tagnet.ru>
>>>>   wrote:
>>>>> Hello!
>>>>>
>>>>>       I found that problem is somewhere inside freeswitch. Direct call
>>>>> between
>>>>> SoftPhone and Cisco is working fine.
>>>>>
>>>>> Hello!
>>>>>
>>>>>       David, may You explain why this problem is with G711A only? When I
>>>>> use
>>>>> G711U there are no problems and call is established and I may talk.
>>>>>
>>>>> The Cisco never sends the 200/OK after the 183, so the call is not
>>>>> established.
>>>>> David Ponzone  Direction Technique
>>>>> email: david.ponzone at ipeva.fr
>>>>> tel:      01 74 03 18 97
>>>>> gsm:   06 66 98 76 34
>>>>> Service Client IPeva
>>>>> tel:      0811 46 26 26
>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou
>>>>> diffusion
>>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
>>>>> s'il
>>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de
>>>>> ce
>>>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>
>>>>>
>>>>>
>>>>> Le 20/01/2011 ? 11:35, Boris Kovalenko a ?crit :
>>>>>
>>>>> Hello!
>>>>>
>>>>>       My network configuration is:
>>>>> Softphone (eyeBeam 1.5) with only G711alaw enabled ->    FreeSWITCH
>>>>> Version
>>>>> 1.0.head (git-0cf1d54 2011-01-19 16-36-04 -0500) ->    Cisco AS5350
>>>>>
>>>>> proxy_media=true in profile.
>>>>> So, when I do a test call, there is SDP but no RTP between FS and
>>>>> CISCO,
>>>>> so I can't hear voice. With G711ulaw there are no problems. What is
>>>>> wrong? Siptrace below:
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>      INVITE sip:73435327569 at default SIP/2.0
>>>>>      Via: SIP/2.0/UDP
>>>>>
>>>>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
>>>>>      Max-Forwards: 70
>>>>>      Contact:<sip:50001 at X.X.29.123:21556>
>>>>>      To: "73435327569"<sip:73435327569 at default>
>>>>>      From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>>>>      Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>>>>      CSeq: 1 INVITE
>>>>>      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
>>>>> SUBSCRIBE, INFO
>>>>>      Content-Type: application/sdp
>>>>>      User-Agent: eyeBeam release 1102u stamp 52345
>>>>>      Content-Length: 337
>>>>>
>>>>>      v=0
>>>>>      o=- 1 2 IN IP4 192.168.3.253
>>>>>      s=CounterPath eyeBeam 1.5
>>>>>      c=IN IP4 192.168.3.253
>>>>>      t=0 0
>>>>>      m=audio 40032 RTP/AVP 8 101
>>>>>      a=alt:1 3 : XdMyLUqb Fs2PLxKV 192.168.3.253 40032
>>>>>      a=alt:2 2 : 2qrc/CQ7 Z/9XFQZl 172.16.3.253 40032
>>>>>      a=alt:3 1 : 8BOwI53P +ENGEr/U X.X.29.123 40032
>>>>>      a=fmtp:101 0-15
>>>>>      a=rtpmap:101 telephone-event/8000
>>>>>      a=sendrecv
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> send 385 bytes to udp/[X.X.29.123]:21556 at 10:30:31.313366:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>      SIP/2.0 100 Trying
>>>>>      Via: SIP/2.0/UDP
>>>>>
>>>>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>>>>>      From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>>>>      To: "73435327569"<sip:73435327569 at default>
>>>>>      Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>>>>      CSeq: 1 INVITE
>>>>>      User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
>>>>> 16-36-04 -0500
>>>>>      Content-Length: 0
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> 2011-01-20 15:30:31.313448 [NOTICE] switch_channel.c:808 New Channel
>>>>> sofia/epbx/50001 at default [95698968-8cf9-40a2-91ad-1322fcfc76af]
>>>>> 2011-01-20 15:30:31.322338 [INFO] mod_dialplan_xml.c:331 Processing
>>>>> TAGNet<50001>->73435327569 in context public
>>>>> 2011-01-20 15:30:31.323362 [INFO] mod_dialplan_xml.c:331 Processing
>>>>> TAGNet<50001>->ext_translate_extsrc in context features
>>>>> 2011-01-20 15:30:31.324357 [NOTICE] switch_ivr.c:1606 Transfer
>>>>> sofia/epbx/50001 at default to XML[73435327569 at top.ctx]
>>>>> 2011-01-20 15:30:31.325352 [INFO] mod_dialplan_xml.c:331 Processing
>>>>> TAGNet<50001>->73435327569 in context top.ctx
>>>>> 2011-01-20 15:30:31.326344 [NOTICE] mod_dptools.c:1174 [top.ctx] -
>>>>> ext_local
>>>>> 2011-01-20 15:30:31.329360 [NOTICE] switch_channel.c:808 New Channel
>>>>> sofia/epbx/73435327569 at X.X.16.83:5060
>>>>> [b9654c75-039a-4780-bc00-0fae65f92a9a]
>>>>> send 1276 bytes to udp/[X.X.16.83]:5060 at 10:30:31.330623:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>      INVITE sip:73435327569 at X.X.16.83:5060 SIP/2.0
>>>>>      Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>>>>      Max-Forwards: 68
>>>>>      From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>>>>      To:<sip:73435327569 at X.X.16.83:5060>
>>>>>      Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>>>>      CSeq: 7423651 INVITE
>>>>>      Contact:<sip:mod_sofia at Y.Y.138.187:5060>
>>>>>      User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
>>>>> 16-36-04 -0500
>>>>>      Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>>      Supported: timer, precondition, path, replaces
>>>>>      Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>>>>> sla, include-session-description, presence.winfo, message-summary,
>>>>> refer
>>>>>      Content-Type: application/sdp
>>>>>      Content-Disposition: session
>>>>>      Content-Length: 341
>>>>>      X-FS-Support: update_display
>>>>>      Remote-Party-ID: "TAGNet"
>>>>> <sip:50001 at Y.Y.138.187>;party=calling;screen=yes;privacy=off
>>>>>
>>>>>      v=0
>>>>>      o=FreeSWITCH 1627720692 1627720693 IN IP4 Y.Y.138.187
>>>>>      s=FreeSWITCH
>>>>>      c=IN IP4 Y.Y.138.187
>>>>>      t=0 0
>>>>>      m=audio 23442 RTP/AVP 8 101
>>>>>      a=rtpmap:101 telephone-event/8000
>>>>>      a=fmtp:101 0-15
>>>>>      a=alt:1 3 : XdMyLUqb Fs2PLxKV 192.168.3.253 40032
>>>>>      a=alt:2 2 : 2qrc/CQ7 Z/9XFQZl 172.16.3.253 40032
>>>>>      a=alt:3 1 : 8BOwI53P +ENGEr/U X.X.29.123 40032
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> recv 396 bytes from udp/[X.X.16.83]:5060 at 10:30:31.341106:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>      SIP/2.0 100 Trying
>>>>>      Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>>>>      From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>>>>      To:<sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>>>>>      Date: Thu, 20 Jan 2011 10:30:31 GMT
>>>>>      Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>>>>      Server: Cisco-SIPGateway/IOS-12.x
>>>>>      CSeq: 7423651 INVITE
>>>>>      Allow-Events: telephone-event
>>>>>      Content-Length: 0
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> recv 814 bytes from udp/[X.X.16.83]:5060 at 10:30:32.401494:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>      SIP/2.0 183 Session Progress
>>>>>      Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>>>>      From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>>>>      To:<sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>>>>>      Date: Thu, 20 Jan 2011 10:30:31 GMT
>>>>>      Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>>>>      Server: Cisco-SIPGateway/IOS-12.x
>>>>>      CSeq: 7423651 INVITE
>>>>>      Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
>>>>> SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
>>>>>      Allow-Events: telephone-event
>>>>>      Contact:<sip:73435327569 at X.X.16.83:5060>
>>>>>      Content-Disposition: session;handling=required
>>>>>      Content-Type: application/sdp
>>>>>      Content-Length: 176
>>>>>
>>>>>      v=0
>>>>>      o=CiscoSystemsSIP-GW-UserAgent 4564 1578 IN IP4 X.X.16.83
>>>>>      s=SIP Call
>>>>>      c=IN IP4 X.X.16.83
>>>>>      t=0 0
>>>>>      m=audio 17770 RTP/AVP 8
>>>>>      c=IN IP4 X.X.16.83
>>>>>      a=rtpmap:8 PCMA/8000
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> 2011-01-20 15:30:32.401362 [INFO] sofia.c:729
>>>>> sofia/epbx/73435327569 at X.X.16.83:5060 Update Callee ID to "Outbound
>>>>> Call"<73435327569>
>>>>> 2011-01-20 15:30:32.409363 [NOTICE] sofia.c:4739 Pre-Answer
>>>>> sofia/epbx/73435327569 at X.X.16.83:5060!
>>>>> 2011-01-20 15:30:32.415412 [INFO] switch_ivr_originate.c:3345 Sending
>>>>> early media
>>>>> 2011-01-20 15:30:32.416367 [NOTICE] mod_sofia.c:2252 Pre-Answer
>>>>> sofia/epbx/50001 at default!
>>>>> send 1145 bytes to udp/[X.X.29.123]:21556 at 10:30:32.418195:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>      SIP/2.0 183 Session Progress
>>>>>      Via: SIP/2.0/UDP
>>>>>
>>>>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>>>>>      From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>>>>      To: "73435327569"<sip:73435327569 at default>;tag=BScUp0QXHFBgB
>>>>>      Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>>>>      CSeq: 1 INVITE
>>>>>      Contact:<sip:73435327569 at Y.Y.138.187:5060;transport=udp>
>>>>>      User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
>>>>> 16-36-04 -0500
>>>>>      Accept: application/sdp
>>>>>      Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>>      Supported: timer, precondition, path, replaces
>>>>>      Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>>>>> sla, include-session-description, presence.winfo, message-summary,
>>>>> refer
>>>>>      Content-Type: application/sdp
>>>>>      Content-Disposition: session
>>>>>      Content-Length: 181
>>>>>      Remote-Party-ID: "73435327569"
>>>>> <sip:73435327569 at Y.Y.138.187>;party=calling;privacy=off;screen=no
>>>>>
>>>>>      v=0
>>>>>      o=FreeSWITCH 1627752513 1627752514 IN IP4 Y.Y.138.187
>>>>>      s=FreeSWITCH
>>>>>      c=IN IP4 Y.Y.138.187
>>>>>      t=0 0
>>>>>      m=audio 25806 RTP/AVP 8
>>>>>      c=IN IP4 Y.Y.138.187
>>>>>      a=rtpmap:8 PCMA/8000
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> recv 365 bytes from udp/[X.X.29.123]:21556 at 10:30:39.196791:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>      CANCEL sip:73435327569 at default SIP/2.0
>>>>>      Via: SIP/2.0/UDP
>>>>>
>>>>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
>>>>>      To: "73435327569"<sip:73435327569 at default>
>>>>>      From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>>>>      Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>>>>      CSeq: 1 CANCEL
>>>>>      User-Agent: eyeBeam release 1102u stamp 52345
>>>>>      Content-Length: 0
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> send 320 bytes to udp/[X.X.29.123]:21556 at 10:30:39.196905:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>      SIP/2.0 200 OK
>>>>>      Via: SIP/2.0/UDP
>>>>>
>>>>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>>>>>      From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>>>>      To: "73435327569"<sip:73435327569 at default>;tag=BScUp0QXHFBgB
>>>>>      Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>>>>      CSeq: 1 CANCEL
>>>>>      Content-Length: 0
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> send 715 bytes to udp/[X.X.29.123]:21556 at 10:30:39.197002:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>      SIP/2.0 487 Request Terminated
>>>>>      Via: SIP/2.0/UDP
>>>>>
>>>>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport=21556
>>>>>      From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>>>>      To: "73435327569"<sip:73435327569 at default>;tag=BScUp0QXHFBgB
>>>>>      Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>>>>      CSeq: 1 INVITE
>>>>>      User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0cf1d54 2011-01-19
>>>>> 16-36-04 -0500
>>>>>      Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>>      Supported: timer, precondition, path, replaces
>>>>>      Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>>>>> sla, include-session-description, presence.winfo, message-summary,
>>>>> refer
>>>>>      Content-Length: 0
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> 2011-01-20 15:30:39.196838 [NOTICE] sofia.c:5286 Hangup
>>>>> sofia/epbx/50001 at default [CS_EXECUTE] [ORIGINATOR_CANCEL]
>>>>> 2011-01-20 15:30:39.196838 [NOTICE] switch_ivr_bridge.c:653 Hangup
>>>>> sofia/epbx/73435327569 at X.X.16.83:5060 [CS_EXCHANGE_MEDIA]
>>>>> [NORMAL_CLEARING]
>>>>> recv 331 bytes from udp/[X.X.29.123]:21556 at 10:30:39.198834:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>      ACK sip:73435327569 at default SIP/2.0
>>>>>      Via: SIP/2.0/UDP
>>>>>
>>>>> X.X.29.123:21556;branch=z9hG4bK-d8754z-dd2b0e57f511a000-1---d8754z-;rport
>>>>>      To: "73435327569"<sip:73435327569 at default>;tag=BScUp0QXHFBgB
>>>>>      From: "TAGNet"<sip:50001 at default>;tag=6c41dc5a
>>>>>      Call-ID: YWEwYjEzNWE3ZTI1NWY0YTEyN2Y0M2Y4ZjZjMjU4YzE.
>>>>>      CSeq: 1 ACK
>>>>>      Content-Length: 0
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> 2011-01-20 15:30:39.204838 [NOTICE] switch_core_session.c:1306 Session
>>>>> 99 (sofia/epbx/50001 at default) Ended
>>>>> 2011-01-20 15:30:39.204838 [NOTICE] switch_core_session.c:1308 Close
>>>>> Channel sofia/epbx/50001 at default [CS_DESTROY]
>>>>> send 372 bytes to udp/[X.X.16.83]:5060 at 10:30:39.212574:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>      CANCEL sip:73435327569 at X.X.16.83:5060 SIP/2.0
>>>>>      Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>>>>      Max-Forwards: 68
>>>>>      From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>>>>      To:<sip:73435327569 at X.X.16.83:5060>
>>>>>      Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>>>>      CSeq: 7423651 CANCEL
>>>>>      Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>>>>>      Content-Length: 0
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> 2011-01-20 15:30:39.212840 [NOTICE] switch_core_session.c:1306 Session
>>>>> 100 (sofia/epbx/73435327569 at X.X.16.83:5060) Ended
>>>>> 2011-01-20 15:30:39.212840 [NOTICE] switch_core_session.c:1308 Close
>>>>> Channel sofia/epbx/73435327569 at X.X.16.83:5060 [CS_DESTROY]
>>>>> recv 311 bytes from udp/[X.X.16.83]:5060 at 10:30:39.216838:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>      SIP/2.0 200 OK
>>>>>      Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>>>>      From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>>>>      To:<sip:73435327569 at X.X.16.83:5060>
>>>>>      Date: Thu, 20 Jan 2011 10:30:39 GMT
>>>>>      Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>>>>      Content-Length: 0
>>>>>      CSeq: 7423651 CANCEL
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> recv 407 bytes from udp/[X.X.16.83]:5060 at 10:30:39.219180:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>      SIP/2.0 487 Request Cancelled
>>>>>      Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>>>>      From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>>>>      To:<sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>>>>>      Date: Thu, 20 Jan 2011 10:30:39 GMT
>>>>>      Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>>>>      Server: Cisco-SIPGateway/IOS-12.x
>>>>>      CSeq: 7423651 INVITE
>>>>>      Allow-Events: telephone-event
>>>>>      Content-Length: 0
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> send 334 bytes to udp/[X.X.16.83]:5060 at 10:30:39.219276:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>      ACK sip:73435327569 at X.X.16.83:5060 SIP/2.0
>>>>>      Via: SIP/2.0/UDP Y.Y.138.187;rport;branch=z9hG4bKar3ygD17yt4Fc
>>>>>      Max-Forwards: 68
>>>>>      From: "TAGNet"<sip:50001 at Y.Y.138.187>;tag=c25KrU80er12p
>>>>>      To:<sip:73435327569 at X.X.16.83:5060>;tag=D57E4-1D19
>>>>>      Call-ID: 25e9ebed-9f23-122e-b98f-002354cb08ac
>>>>>      CSeq: 7423651 ACK
>>>>>      Content-Length: 0
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>>
>>>>> --
>>>>> ? ?????????,
>>>>>     ????? ?????????
>>>>>     ??? "??????"
>>>>>     (3435) 494991
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> --
>>>>> ? ?????????,
>>>>>     ????? ?????????
>>>>>     ??? "??????"
>>>>>     (3435) 494991
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> --
>>>>> ? ?????????,
>>>>>     ????? ?????????
>>>>>     ??? "??????"
>>>>>     (3435) 494991
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>> --
>>> ? ?????????,
>>>    ????? ?????????
>>>    ??? "??????"
>>>    (3435) 494991
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From fraserredmond at gmail.com  Fri Jan 21 07:37:28 2011
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Thu, 20 Jan 2011 23:37:28 -0500
Subject: [Freeswitch-users] Run dialplan tools from event socket
In-Reply-To: <AANLkTin5fn9qMre-8xFAZD5_SUzbLbp_YueTRPPyh-=K@mail.gmail.com>
References: <AANLkTikg5+1aMoWxYr3UYO13iO2qA6FsG9FFi-Zg3_bD@mail.gmail.com>
	<AANLkTin5fn9qMre-8xFAZD5_SUzbLbp_YueTRPPyh-=K@mail.gmail.com>
Message-ID: <AANLkTin84SkDdDsBE0wfU2QnG025bNK8BhcW=JDvz9gv@mail.gmail.com>

Thanks Jo?o.

My transfer_call extension runs a couple of js scripts to get and validate
the number to transfer to, then does
                <action application="att_xfer" data="${dial_string}"/>
(and it has a couple of steps after that to handle failed transfers.)

So could I use ESL's execute command to run the execute_extension? Not sure
how I missed that option in the wiki. I"ll give it a try, see what happens.


I forgot to say in the original post, but execute_extension seems to be
particularly nice for this use-case, as it falls back through the dialplan
gracefully if there's a problem.

Cheers,
Fraser




2011/1/20 Jo?o Mesquita <jmesquita at freeswitch.org>

> Lucky for you I have been working on this lately and the bad news is ...
> there's no easy way to do it....
>
> You can execute an extension like you said, but you have to park the legs
> first... It would help to know how's the transfer_call extension so that I
> can try to help you out, but maybe it is easier if you think of it this way:
>
> When you use an app like att_xfer, the core already knows what to do next
> with a call and parks the legs for you. If you do it on ESL, you've done it
> half way and you didn't really park anything before you transfered the call.
> When the bridge is undone, the leg that was not transfered doesn't know what
> to do, has no applications to be run at this moment and so all it's left for
> it is to let go.
>
> A bit clearer? Att_xfer is a bit of a pain in the butt and it kinda
> requires you to know a bit more of the inner workings of the state machine.
>
> You can always execute att_xfer using ESL's execute (
> http://wiki.freeswitch.org/wiki/Event_Socket_Library#execute) if you don't
> care what happens to the legs afterwards but if you want to have control
> over all 3 legs, no luck for you...
>
> Regards,
> Jo?o Mesquita
>
>
> On Fri, Jan 21, 2011 at 12:13 AM, Fraser Redmond <fraserredmond at gmail.com>wrote:
>
>> Is there any way to run a dialplan tool from the event socket?
>>
>> I have a dialplan that uses a dtmf to set up and perform an attended
>> transfer, like so:
>> <action application="bind_meta_app" data="8 a s
>> execute_extension::TransferCall XML transfer_call"/>
>>
>> But I can't see any way to run the same thing from the event socket. I
>> thought doing an  "api uuid_transfer"  might do it, but that hangs up one of
>> the legs (no good for attended transfer.)
>>
>> api uuid_transfer Uuid -bleg TransferCall XML transfer_call
>>
>> As far as I can see, the closest thing is "sendmsg execute", but it looks
>> like you have to park a call/channel first to use that, so I'm not sure that
>> that is much use for attended transfer either.
>>
>> Or should I be lame and do "api uuid_send_dtmf" to send * 8.
>>
>> Cheers,
>> Fraser
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Fri Jan 21 08:07:54 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 21 Jan 2011 10:37:54 +0530
Subject: [Freeswitch-users] call barging
In-Reply-To: <AANLkTinNrHFKY22Amsv-mMw=wMcxmKheo-WxT8kYTrqC@mail.gmail.com>
References: <AANLkTinMqT+Qs1guPphYWqN5okMAD9c1K9=Jey6uepRw@mail.gmail.com>
	<AANLkTimrbwUS7s1G+catXL-XVTr_pg9yOWorR04MNFY=@mail.gmail.com>
	<AANLkTinNrHFKY22Amsv-mMw=wMcxmKheo-WxT8kYTrqC@mail.gmail.com>
Message-ID: <AANLkTimeyiq92cGvtoxYuUMEJGyu-Oo2sWAGB6L3Hu1n@mail.gmail.com>

kool, it works.

Regds
Sam

On 1/20/11, Steven Ayre <steveayre at gmail.com> wrote:
> Yes, that will work. You can set the group either within the extension or in
> the user config.
>
> For example:
>
>  <extension name="bridge">
>      <condition field="destination_number" expression="^9(\d+)$">
>        <action application="set" data="eavesdrop_group=spy1"/>
>        <action application="bridge" data="sofia/gateway/gw1/$1"/>
>      </condition>
>    </extension>
>   <extension name="eavesdrop">
>      <condition field="destination_number" expression="^666$">
>        <action application="answer"/>
>        <action application="set" data="eavesdrop_require_group=spy1"/>
>        <action application="eavesdrop" data="all"/>
>      </condition>
>    </extension>
>
>
> -Steve
>
>
> On 20 January 2011 10:11, Sam <u2nsam at gmail.com> wrote:
>
>> If i use
>>
>> <variable name="eavesdrop_group" value="spy1"/>
>>
>> at the user config, will the  dialplan barge the above extension where the
>> statement is stated ?
>>
>> <extension name="eavesdrop">
>>      <condition field="destination_number" expression="^779$">
>>        <action application="answer"/>
>>        <action application="set"
>> data="eavesdrop_require_group=sales_call_eavesdrop"/>
>>
>>        <action application="eavesdrop" data="all"/>
>>      </condition>
>>    </extension>
>>
>>
>> Regds
>> Sam
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> On Thu, Jan 20, 2011 at 3:13 PM, Sam <u2nsam at gmail.com> wrote:
>>
>>> hello ,
>>>
>>> I am using,
>>>
>>> <extension name="global" continue="true">
>>>      <condition>
>>>        <action application="info"/>
>>>        <action application="db"
>>> data="insert/spymap/${caller_id_number}/${uuid}"/>
>>>
>>>
>>>        <action application="db"
>>> data="insert/last_dial/${caller_id_number}/${destination_number}"/>
>>>        <action application="db" data="insert/last_dial/global/${uuid}"/>
>>>
>>>
>>>      </condition>
>>>    </extension>
>>>
>>>    <extension name="eavesdrop">
>>>
>>>
>>>      <condition field="destination_number"
>>> expression="^88(.*)$|^\*0(.*)$">
>>>        <action application="answer"/>
>>>        <action application="eavesdrop" data="${db(select/spymap/$1)}"/>
>>>
>>>
>>>      </condition>
>>>    </extension>
>>>
>>>
>>>
>>> i am getting the below error:-
>>>
>>> 2011-01-20 14:54:01.581151 [NOTICE] mod_dptools.c:920 Channel
>>> [sofia/internal/7006 at 192.168.2.190] has been answered
>>> 2011-01-20 14:54:01.581151 [DEBUG] sofia.c:4646 Channel sofia/internal/
>>> 7006 at 192.168.2.190 entering state [completed][200]
>>> EXECUTE sofia/internal/7006 at 192.168.2.190 eavesdrop()
>>> 2011-01-20 14:54:01.582119 [ERR] mod_dptools.c:529 Usage: [all | <uuid>]
>>> 2011-01-20 14:54:01.582119 [NOTICE] switch_core_state_machine.c:189
>>> sofia/internal/7006 at 192.168.2.190 has executed the last dialplan
>>> instruction, hanging up.
>>> 2011-01-20 14:54:01.582119 [DEBUG] switch_channel.c:2535 (sofia/internal/
>>> 7006 at 192.168.2.190) Callstate Change ACTIVE -> HANGUP
>>> 2011-01-20 14:54:01.582119 [NOTICE] switch_core_state_machine.c:191
>>> Hangup
>>> sofia/internal/7006 at 192.168.2.190 [CS_EXECUTE] [NORMAL_CLEARING]
>>>
>>>
>>> http://pastebin.freeswitch.org/15076
>>>
>>>
>>> Regds
>>> Sam
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>


From ewin.hogan at gmail.com  Fri Jan 21 08:06:53 2011
From: ewin.hogan at gmail.com (Edwin Moedano)
Date: Thu, 20 Jan 2011 23:06:53 -0600
Subject: [Freeswitch-users] Run dialplan tools from event socket
In-Reply-To: <AANLkTin84SkDdDsBE0wfU2QnG025bNK8BhcW=JDvz9gv@mail.gmail.com>
References: <AANLkTikg5+1aMoWxYr3UYO13iO2qA6FsG9FFi-Zg3_bD@mail.gmail.com>
	<AANLkTin5fn9qMre-8xFAZD5_SUzbLbp_YueTRPPyh-=K@mail.gmail.com>
	<AANLkTin84SkDdDsBE0wfU2QnG025bNK8BhcW=JDvz9gv@mail.gmail.com>
Message-ID: <36597691-BCB9-4D4F-AD24-28698DBF15F3@gmail.com>

Hi Well for attended transfers what I do is to answer the call, do something then I originate a call, check the status and then I bridge them, if I want to take out fs I use uuid_simplify ;)

Edwin Moedano
El 20/01/2011, a las 10:37 p.m., Fraser Redmond <fraserredmond at gmail.com> escribi?:

> Thanks Jo?o.
> 
> My transfer_call extension runs a couple of js scripts to get and validate the number to transfer to, then does
>                 <action application="att_xfer" data="${dial_string}"/>
> (and it has a couple of steps after that to handle failed transfers.)
> 
> So could I use ESL's execute command to run the execute_extension? Not sure how I missed that option in the wiki. I"ll give it a try, see what happens.
> 
> 
> I forgot to say in the original post, but execute_extension seems to be particularly nice for this use-case, as it falls back through the dialplan gracefully if there's a problem.
> 
> Cheers,
> Fraser
> 
> 
> 
> 
> 2011/1/20 Jo?o Mesquita <jmesquita at freeswitch.org>
> Lucky for you I have been working on this lately and the bad news is ... there's no easy way to do it....
> 
> You can execute an extension like you said, but you have to park the legs first... It would help to know how's the transfer_call extension so that I can try to help you out, but maybe it is easier if you think of it this way:
> 
> When you use an app like att_xfer, the core already knows what to do next with a call and parks the legs for you. If you do it on ESL, you've done it half way and you didn't really park anything before you transfered the call. When the bridge is undone, the leg that was not transfered doesn't know what to do, has no applications to be run at this moment and so all it's left for it is to let go.
> 
> A bit clearer? Att_xfer is a bit of a pain in the butt and it kinda requires you to know a bit more of the inner workings of the state machine.
> 
> You can always execute att_xfer using ESL's execute (http://wiki.freeswitch.org/wiki/Event_Socket_Library#execute) if you don't care what happens to the legs afterwards but if you want to have control over all 3 legs, no luck for you...
> 
> Regards,
> Jo?o Mesquita
> 
> 
> On Fri, Jan 21, 2011 at 12:13 AM, Fraser Redmond <fraserredmond at gmail.com> wrote:
> Is there any way to run a dialplan tool from the event socket?
> 
> I have a dialplan that uses a dtmf to set up and perform an attended transfer, like so:
> <action application="bind_meta_app" data="8 a s execute_extension::TransferCall XML transfer_call"/>
> 
> But I can't see any way to run the same thing from the event socket. I thought doing an  "api uuid_transfer"  might do it, but that hangs up one of the legs (no good for attended transfer.)
> 
> api uuid_transfer Uuid -bleg TransferCall XML transfer_call
> 
> As far as I can see, the closest thing is "sendmsg execute", but it looks like you have to park a call/channel first to use that, so I'm not sure that that is much use for attended transfer either.
> 
> Or should I be lame and do "api uuid_send_dtmf" to send * 8.
> 
> Cheers,
> Fraser
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From rupa at rupa.com  Fri Jan 21 09:03:40 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 21 Jan 2011 00:03:40 -0600
Subject: [Freeswitch-users] moderator conference flag and nibblebill
In-Reply-To: <BEC4FD4A5E3A474EA4E9A1B6DB7D33F1@e1705>
References: <BEC4FD4A5E3A474EA4E9A1B6DB7D33F1@e1705>
Message-ID: <AANLkTimz_Ro8Tk1Qsrjy-DoF-qrjedVEbc=HnG+h4XK9@mail.gmail.com>

Not sure from the dialplan.  I'd approach it like this:

have the callers connect to the conference with nibblebill paused.

Have a process connected to FS via ESL monitoring conference events.
I assume we get enough information via the ESL events to determine
when the moderator connects (if not, it would be  a good addition so
suggest it).  When the moderator joins, then iterate through the
currently joined members and unpause each one.  Ensure any new members
that join also get unpaused on join.

Should be pretty straightforward via ESL.  Strictly from dialplan?  No idea.

On Thu, Jan 20, 2011 at 7:03 PM, Madovsky <infos at madovsky.org> wrote:
> if a caller in waste mode go to a conference
> where the moderator isn't into nibblebill starts
> to debit his account. I can't guess hwo I can manage
> that in a dialplan to start debit once the conference really starts
>
> Any ?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From rupa at rupa.com  Fri Jan 21 09:08:14 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 21 Jan 2011 00:08:14 -0600
Subject: [Freeswitch-users] conference list
In-Reply-To: <DBCC446AC5A943F1AEB555BF1707ED6C@e1705>
References: <DBCC446AC5A943F1AEB555BF1707ED6C@e1705>
Message-ID: <AANLkTi=zp_5gMY51Puc-4bpmJ0duZtAZnYM9Vw0t_xF1@mail.gmail.com>

Dunno about csv, but xml_list has a flags node for flags for each member.

On Thu, Jan 20, 2011 at 8:50 PM, Madovsky <infos at madovsky.org> wrote:
> Sorry Brian, it's my last question before next week ;)
>
> it's more a suggestion that a question,
> when use api conference myconf list command
> maybe it would be useful to know in the csv list
> which channel is moderator etc...
>
> Thanks
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From infos at madovsky.org  Fri Jan 21 09:13:03 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 21 Jan 2011 01:13:03 -0500
Subject: [Freeswitch-users] conference list
References: <DBCC446AC5A943F1AEB555BF1707ED6C@e1705>
	<AANLkTi=zp_5gMY51Puc-4bpmJ0duZtAZnYM9Vw0t_xF1@mail.gmail.com>
Message-ID: <021F1447C64046159BEE423C61E947A0@e1705>

sorry when I say csv it's the csv like list when
you type api conference list
but there is only flags as speak|hear|floor,
the flags-member is not listed

----- Original Message ----- 
From: "Rupa Schomaker" <rupa at rupa.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, January 21, 2011 1:08 AM
Subject: Re: [Freeswitch-users] conference list


> Dunno about csv, but xml_list has a flags node for flags for each member.
> 
> On Thu, Jan 20, 2011 at 8:50 PM, Madovsky <infos at madovsky.org> wrote:
>> Sorry Brian, it's my last question before next week ;)
>>
>> it's more a suggestion that a question,
>> when use api conference myconf list command
>> maybe it would be useful to know in the csv list
>> which channel is moderator etc...
>>
>> Thanks
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> 
> 
> 
> -- 
> -Rupa
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From infos at madovsky.org  Fri Jan 21 09:15:26 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 21 Jan 2011 01:15:26 -0500
Subject: [Freeswitch-users] moderator conference flag and nibblebill
References: <BEC4FD4A5E3A474EA4E9A1B6DB7D33F1@e1705>
	<AANLkTimz_Ro8Tk1Qsrjy-DoF-qrjedVEbc=HnG+h4XK9@mail.gmail.com>
Message-ID: <3B561E18A1594FE180422EA2FC679EA6@e1705>

ok thanks.
interesting solution.
I really need to have time to learn esl I think....

----- Original Message ----- 
From: "Rupa Schomaker" <rupa at rupa.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, January 21, 2011 1:03 AM
Subject: Re: [Freeswitch-users] moderator conference flag and nibblebill


> Not sure from the dialplan.  I'd approach it like this:
>
> have the callers connect to the conference with nibblebill paused.
>
> Have a process connected to FS via ESL monitoring conference events.
> I assume we get enough information via the ESL events to determine
> when the moderator connects (if not, it would be  a good addition so
> suggest it).  When the moderator joins, then iterate through the
> currently joined members and unpause each one.  Ensure any new members
> that join also get unpaused on join.
>
> Should be pretty straightforward via ESL.  Strictly from dialplan?  No 
> idea.
>
> On Thu, Jan 20, 2011 at 7:03 PM, Madovsky <infos at madovsky.org> wrote:
>> if a caller in waste mode go to a conference
>> where the moderator isn't into nibblebill starts
>> to debit his account. I can't guess hwo I can manage
>> that in a dialplan to start debit once the conference really starts
>>
>> Any ?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> -- 
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From u2nsam at gmail.com  Fri Jan 21 09:46:52 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 21 Jan 2011 12:16:52 +0530
Subject: [Freeswitch-users] call barging
In-Reply-To: <AANLkTimeyiq92cGvtoxYuUMEJGyu-Oo2sWAGB6L3Hu1n@mail.gmail.com>
References: <AANLkTinMqT+Qs1guPphYWqN5okMAD9c1K9=Jey6uepRw@mail.gmail.com>
	<AANLkTimrbwUS7s1G+catXL-XVTr_pg9yOWorR04MNFY=@mail.gmail.com>
	<AANLkTinNrHFKY22Amsv-mMw=wMcxmKheo-WxT8kYTrqC@mail.gmail.com>
	<AANLkTimeyiq92cGvtoxYuUMEJGyu-Oo2sWAGB6L3Hu1n@mail.gmail.com>
Message-ID: <AANLkTimhjr_fm1ZyWRchXjL_JGBjBgknZoqQsXSp3M6f@mail.gmail.com>

But i gives error for g729 codec !

2011-01-21 12:04:44.817605 [ERR] mod_g729.c:102 This codec is only usable in
passthrough mode!
2011-01-21 12:04:44.817605 [ERR] switch_core_io.c:1042 Codec RAW Signed
Linear (16 bit) encoder error!
2011-01-21 12:04:44.838101 [ERR] mod_g729.c:102 This codec is only usable in
passthrough mode!
2011-01-21 12:04:44.838101 [ERR] switch_core_io.c:1042 Codec RAW Signed
Linear (16 bit) encoder error!
2011-01-21 12:04:44.858131 [ERR] mod_g729.c:145 This codec is only usable in
passthrough mode!
2011-01-21 12:04:44.858131 [ERR] switch_core_io.c:433 Codec RAW Signed
Linear (16 bit) decoder error!

Regds
Sam

On Fri, Jan 21, 2011 at 10:37 AM, Sam <u2nsam at gmail.com> wrote:

> kool, it works.
>
> Regds
> Sam
>
> On 1/20/11, Steven Ayre <steveayre at gmail.com> wrote:
> > Yes, that will work. You can set the group either within the extension or
> in
> > the user config.
> >
> > For example:
> >
> >  <extension name="bridge">
> >      <condition field="destination_number" expression="^9(\d+)$">
> >        <action application="set" data="eavesdrop_group=spy1"/>
> >        <action application="bridge" data="sofia/gateway/gw1/$1"/>
> >      </condition>
> >    </extension>
> >   <extension name="eavesdrop">
> >      <condition field="destination_number" expression="^666$">
> >        <action application="answer"/>
> >        <action application="set" data="eavesdrop_require_group=spy1"/>
> >        <action application="eavesdrop" data="all"/>
> >      </condition>
> >    </extension>
> >
> >
> > -Steve
> >
> >
> > On 20 January 2011 10:11, Sam <u2nsam at gmail.com> wrote:
> >
> >> If i use
> >>
> >> <variable name="eavesdrop_group" value="spy1"/>
> >>
> >> at the user config, will the  dialplan barge the above extension where
> the
> >> statement is stated ?
> >>
> >> <extension name="eavesdrop">
> >>      <condition field="destination_number" expression="^779$">
> >>        <action application="answer"/>
> >>        <action application="set"
> >> data="eavesdrop_require_group=sales_call_eavesdrop"/>
> >>
> >>        <action application="eavesdrop" data="all"/>
> >>      </condition>
> >>    </extension>
> >>
> >>
> >> Regds
> >> Sam
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >> On Thu, Jan 20, 2011 at 3:13 PM, Sam <u2nsam at gmail.com> wrote:
> >>
> >>> hello ,
> >>>
> >>> I am using,
> >>>
> >>> <extension name="global" continue="true">
> >>>      <condition>
> >>>        <action application="info"/>
> >>>        <action application="db"
> >>> data="insert/spymap/${caller_id_number}/${uuid}"/>
> >>>
> >>>
> >>>        <action application="db"
> >>> data="insert/last_dial/${caller_id_number}/${destination_number}"/>
> >>>        <action application="db"
> data="insert/last_dial/global/${uuid}"/>
> >>>
> >>>
> >>>      </condition>
> >>>    </extension>
> >>>
> >>>    <extension name="eavesdrop">
> >>>
> >>>
> >>>      <condition field="destination_number"
> >>> expression="^88(.*)$|^\*0(.*)$">
> >>>        <action application="answer"/>
> >>>        <action application="eavesdrop" data="${db(select/spymap/$1)}"/>
> >>>
> >>>
> >>>      </condition>
> >>>    </extension>
> >>>
> >>>
> >>>
> >>> i am getting the below error:-
> >>>
> >>> 2011-01-20 14:54:01.581151 [NOTICE] mod_dptools.c:920 Channel
> >>> [sofia/internal/7006 at 192.168.2.190] has been answered
> >>> 2011-01-20 14:54:01.581151 [DEBUG] sofia.c:4646 Channel sofia/internal/
> >>> 7006 at 192.168.2.190 entering state [completed][200]
> >>> EXECUTE sofia/internal/7006 at 192.168.2.190 eavesdrop()
> >>> 2011-01-20 14:54:01.582119 [ERR] mod_dptools.c:529 Usage: [all |
> <uuid>]
> >>> 2011-01-20 14:54:01.582119 [NOTICE] switch_core_state_machine.c:189
> >>> sofia/internal/7006 at 192.168.2.190 has executed the last dialplan
> >>> instruction, hanging up.
> >>> 2011-01-20 14:54:01.582119 [DEBUG] switch_channel.c:2535
> (sofia/internal/
> >>> 7006 at 192.168.2.190) Callstate Change ACTIVE -> HANGUP
> >>> 2011-01-20 14:54:01.582119 [NOTICE] switch_core_state_machine.c:191
> >>> Hangup
> >>> sofia/internal/7006 at 192.168.2.190 [CS_EXECUTE] [NORMAL_CLEARING]
> >>>
> >>>
> >>> http://pastebin.freeswitch.org/15076
> >>>
> >>>
> >>> Regds
> >>> Sam
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
>
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From vermeulen.deon at gmail.com  Fri Jan 21 10:42:35 2011
From: vermeulen.deon at gmail.com (Deon Vermeulen)
Date: Fri, 21 Jan 2011 09:42:35 +0200
Subject: [Freeswitch-users] GSM/PSTN Gateways
In-Reply-To: <4D386321.5090402@puzzled.xs4all.nl>
References: <0F8036C9-D129-419D-AAA0-C8EA3856D7DD@gmail.com>	<4D384C4C.60309@puzzled.xs4all.nl>	<AANLkTi=t8_ddWNcabFsJRiZh1M9_TUPDjARG3CwVXJ=b@mail.gmail.com>
	<AANLkTin0afY88bBMtug7Fy9edP_HmrsLQ0vhd5yOkH5T@mail.gmail.com>
	<4D386321.5090402@puzzled.xs4all.nl>
Message-ID: <B116FE8E-E7A0-46F5-AF5C-748126BA7B2D@gmail.com>

Everyone.

Thanks for the info.

Really appreciate it.


Kind Regards
Deon

On Jan 20, 2011, at 6:30 PM, Patrick Lists wrote:

> On 01/20/2011 05:11 PM, Imthiyaz Ahmed wrote:
>> allywll Gateways
> 
> I thought that name was a spelling error. It was not. That's quite an 
> unpronounceable company name. Image saying that with at the end of the 
> FOSDEM beer party. Or maybe it will go beter then. Anyway their GSM 
> gateways are here: http://www.allywll.com/Product.asp?lbID=31
> 
> Regards,
> Patrick
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From kond at nstel.ru  Fri Jan 21 10:44:28 2011
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Fri, 21 Jan 2011 10:44:28 +0300
Subject: [Freeswitch-users] sangoma a101: can't load freetdm module
Message-ID: <20110121074428.CB3FA12330@mail.nstel.ru>

Hi all, 
 
i have a problem with A101 sangoma pri card.
I spent couple of days trying to make it work but alas...
 
My problem is that i can't load mod_freetdm module.
I'm sure i'm doing something wrong, but i can't find out what is my mistake...
So.. please help to solve the problem...
 
I setup my test system according to http://wiki.freeswitch.org/wiki/FreeTDM
I use freeswitch 1.0.7, wanpipe 3.5.18 and libsng_isdn-7.0.0.
 
 
wanrouter can see the card:
 
[root at sipx4 conf]# wanrouter hwprobe
-------------------------------
| Wanpipe Hardware Probe Info |
-------------------------------
1 . AFT-A101-SH : SLOT=4 : BUS=8 : IRQ=177 : CPU=A : PORT=1 : HWEC=32 : V=37
Card Cnt: A101-2=1
 
[root at sipx4 conf]# wanrouter status
Devices currently active:
        wanpipe1
Wanpipe Config:
Device name | Protocol Map | Adapter  | IRQ | Slot/IO | If's | CLK | Baud rate |
wanpipe1    | N/A          | A101/1D/A102/2D/4/4D/8| 177 | 4       | 1    | N/A | 0         |
Wanrouter Status:
Device name | Protocol | Station | Status        |
wanpipe1    | AFT TE1  | N/A     | Connected     |
 
Here is my config files:
 
[root at sipx4 conf]# cat /etc/wanpipe/wanpipe1.conf
#================================================
# WANPIPE1 Configuration File
#================================================
#
# Date: Wed Dec  6 20:29:03 UTC 2006
#
# Note: This file was generated automatically
#       by /usr/local/sbin/setup-sangoma program.
#
#       If you want to edit this file, it is
#       recommended that you use wancfg program
#       to do so.
#================================================
# Sangoma Technologies Inc.
#================================================
 
[devices]
wanpipe1 = WAN_AFT_TE1, Comment
 
[interfaces]
w1g1 = wanpipe1, , TDM_VOICE_API, Comment
 
[wanpipe1]
CARD_TYPE       = AFT
S514CPU         = A
CommPort        = PRI
AUTO_PCISLOT    = NO
PCISLOT         = 4
PCIBUS          = 8
FE_MEDIA        = E1
FE_LCODE        = HDB3
FE_FRAME        = CRC4
FE_LINE         = 1
TE_CLOCK        = MASTER
TE_REF_CLOCK    = 0
TE_SIG_MODE     = CCS
TE_HIGHIMPEDANCE        = NO
TE_RX_SLEVEL    = 430
LBO             = 120OH
FE_TXTRISTATE   = NO
MTU             = 1500
UDPPORT         = 9000
TTL             = 255
IGNORE_FRONT_END        = NO
TDMV_SPAN               = 1
TDMV_DCHAN              = 16
TE_AIS_MAINTENANCE = NO         #NO: defualt  YES: Start port in AIS Blue Alarm and keep line down
                                #wanpipemon -i w1g1 -c Ttx_ais_off to disable AIS maintenance mode
                                                                #wanpipemon -i w1g1 -c Ttx_ais_on to enable AIS maintenance mode
TDMV_HW_DTMF            = YES           # YES: receive dtmf events from hardware
TDMV_HW_FAX_DETECT              = YES           # YES: receive fax 1100hz events from hardware
HWEC_OPERATION_MODE     = OCT_NORMAL    # OCT_NORMAL: echo cancelation enabled with nlp (default)
                                                                                # OCT_SPEECH: improves software tone detection by disabling NLP (echo possible)
                                                                                # OCT_NO_ECHO:disables echo cancelation but allows VQE/tone functions.
HWEC_DTMF_REMOVAL       = NO    # NO: default  YES: remove dtmf out of incoming media (must have hwdtmf enabled)
HWEC_NOISE_REDUCTION    = NO    # NO: default  YES: reduces noise on the line - could break fax
HWEC_ACUSTIC_ECHO       = NO    # NO: default  YES: enables acustic echo cancelation
HWEC_NLP_DISABLE        = NO    # NO: default  YES: guarantees software tone detection (possible echo)
HWEC_TX_AUTO_GAIN       = 0     # 0: disable   -40-0: default tx audio level to be maintained (-20 default)
HWEC_RX_AUTO_GAIN       = 0     # 0: disable   -40-0: default tx audio level to be maintained (-20 default)
HWEC_TX_GAIN            = 0             # 0: disable   -24-24: db values to be applied to tx signal
HWEC_RX_GAIN            = 0             # 0: disable   -24-24: db values to be applied to tx signal
 
[w1g1]
ACTIVE_CH       = ALL
TDMV_HWEC       = YES
MTU             = 80
[root at sipx4 conf]#

[root at sipx4 conf]# cat /usr/local/freeswitch-107/conf/wanpipe.conf  (i installed FS into /usr/local/freeswitch-107)
[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750

[root at sipx4 conf]# cat /usr/local/freeswitch-107/conf/freetdm.conf
[span wanpipe wp1]
trunk_type => e1
group=1
b-channel => 1:1-15
b-channel => 1:17-31
d-channel => 1:16

[root at sipx4 conf]# cat /usr/local/freeswitch-107/conf/autoload_configs/freetdm.conf.xml
<configuration name="freetdm.conf" description="Freetdm Configuration">
  <settings>
    <param name="debug" value="0"/>
    <!--<param name="hold-music" value="$${moh_uri}"/>-->
    <!--<param name="enable-analog-option" value="call-swap"/>-->
    <!--<param name="enable-analog-option" value="3-way"/>-->
  </settings>
                <config_profiles>
                <profile name="my_pri_nt_1">
                        <param name="switchtype" value="euroisdn" />
                        <param name="interface" value="net"/>
                </profile>
                </config_profiles>
        <sangoma_pri_spans>
                <span name="wp1" cfgprofile="my_pri_nt_1">
                        <param name="dialplan" value="XML"/>
                        <param name="context" value="default"/>
                </span>
        </sangoma_pri_spans>
        <analog_spans>
        </analog_spans>
</configuration>
[root at sipx4 conf]#

The other side of the E1  connection is mediant 2000 E1 port. Mediant shows d-channel alarm.
 
And here is what i got in the log, when tried to load module:
[freeswitch107 at sipx4 log]$ cat freeswitch.log.2011-01-21-10-24-01.1
2011-01-21 10:23:15.775934 [NOTICE] mod_logfile.c:158 New log started.
2011-01-21 10:23:57.495838 [INFO] mod_enum.c:808 ENUM Reloaded
2011-01-21 10:23:57.495838 [INFO] switch_time.c:954 Timezone reloaded 530 definitions
2011-01-21 10:23:57.495838 [DEBUG] ftdm_config.c:52 New mod directory: /usr/local/freeswitch-107/mod
2011-01-21 10:23:57.495838 [DEBUG] ftdm_config.c:58 New config directory: /usr/local/freeswitch-107/conf
2011-01-21 10:23:57.495838 [DEBUG] ftdm_sched.c:154 Initializing scheduling API
2011-01-21 10:23:57.495838 [DEBUG] ftdm_sched.c:251 Created schedule freetdm-master
2011-01-21 10:23:57.495838 [NOTICE] ftdm_sched.c:178 Launching main schedule thread
2011-01-21 10:23:57.495838 [DEBUG] ftdm_sched.c:187 Running schedule freetdm-master in the main schedule thread
2011-01-21 10:23:57.495838 [DEBUG] ftdm_config.c:80 Configuration file is /usr/local/freeswitch-107/conf/modules.conf.
2011-01-21 10:23:57.495838 [NOTICE] ftdm_io.c:5731 Modules configured: 1
2011-01-21 10:23:57.495838 [DEBUG] ftdm_config.c:80 Configuration file is /usr/local/freeswitch-107/conf/freetdm.conf.
2011-01-21 10:23:57.495838 [DEBUG] ftdm_io.c:4676 Reading FreeTDM configuration file
2011-01-21 10:23:57.495838 [DEBUG] ftdm_io.c:4692 found config for span
2011-01-21 10:23:57.495838 [INFO] ftdm_io.c:4976 Loading IO from /usr/local/freeswitch-107/mod/ftmod_wanpipe.so [wanpipe]
2011-01-21 10:23:57.495838 [DEBUG] ftdm_config.c:80 Configuration file is /usr/local/freeswitch-107/conf/wanpipe.conf.
2011-01-21 10:23:57.495838 [INFO] ftdm_io.c:800 Auto-loaded I/O module 'wanpipe'
2011-01-21 10:23:57.495838 [DEBUG] ftdm_io.c:4706 created span 1 (wp1) of type wanpipe
2011-01-21 10:23:57.495838 [DEBUG] ftdm_io.c:4722 span 1 [trunk_type]=[e1]
2011-01-21 10:23:57.495838 [DEBUG] ftdm_io.c:4727 setting trunk type to 'E1'
2011-01-21 10:23:57.495838 [DEBUG] ftdm_io.c:4722 span 1 [group]=[1]
2011-01-21 10:23:57.495838 [DEBUG] ftdm_io.c:4722 span 1 [b-channel]=[1:1-15]
2011-01-21 10:23:57.495838 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 1
2011-01-21 10:23:57.495838 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 2
2011-01-21 10:23:57.495838 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 3
2011-01-21 10:23:57.495838 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 4
2011-01-21 10:23:57.495838 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 5
2011-01-21 10:23:57.495838 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 6
2011-01-21 10:23:57.495838 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 7
2011-01-21 10:23:57.495838 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 8
2011-01-21 10:23:57.495838 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 9
2011-01-21 10:23:57.495838 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 10
2011-01-21 10:23:57.495838 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 11
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 12
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 13
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 14
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 15
2011-01-21 10:23:57.497841 [ERR] ftdm_io.c:4623 1:Failed to configure span2011-01-21 10:23:57.497841 [DEBUG] ftdm_io.c:4722 span 1 [b-channel]=[1:17-31]
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 17
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 18
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 19
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 20
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 21
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 22
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 23
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 24
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 25
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 26
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 27
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 28
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 29
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 30
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 31
2011-01-21 10:23:57.497841 [ERR] ftdm_io.c:4623 1:Failed to configure span2011-01-21 10:23:57.497841 [DEBUG] ftdm_io.c:4722 span 1 [d-channel]=[1:16]
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 16
2011-01-21 10:23:57.497841 [ERR] ftdm_io.c:4623 1:Failed to configure span2011-01-21 10:23:57.497841 [INFO] ftdm_io.c:4901 Configured 0 channel(s)
2011-01-21 10:23:57.497841 [ERR] ftdm_io.c:5741 FreeTDM global configuration failed!
2011-01-21 10:23:57.497841 [ERR] mod_freetdm.c:4181 Error configuring FreeTDM
2011-01-21 10:23:57.497841 [CRIT] switch_loadable_module.c:928 Error Loading module /usr/local/freeswitch-107/mod/mod_freetdm.so
**Module load routine returned an error**
2011-01-21 10:23:57.597841 [NOTICE] ftdm_sched.c:147 Main scheduling thread going out ...
[freeswitch107 at sipx4 log]$

Can anybody please help with the problem?
I'm ready to provide additional info if needed.
 
Thanks and regards,
Nikolay.
 
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From steveayre at gmail.com  Fri Jan 21 10:54:25 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 21 Jan 2011 07:54:25 +0000
Subject: [Freeswitch-users] call barging
In-Reply-To: <AANLkTimhjr_fm1ZyWRchXjL_JGBjBgknZoqQsXSp3M6f@mail.gmail.com>
References: <AANLkTinMqT+Qs1guPphYWqN5okMAD9c1K9=Jey6uepRw@mail.gmail.com>
	<AANLkTimrbwUS7s1G+catXL-XVTr_pg9yOWorR04MNFY=@mail.gmail.com>
	<AANLkTinNrHFKY22Amsv-mMw=wMcxmKheo-WxT8kYTrqC@mail.gmail.com>
	<AANLkTimeyiq92cGvtoxYuUMEJGyu-Oo2sWAGB6L3Hu1n@mail.gmail.com>
	<AANLkTimhjr_fm1ZyWRchXjL_JGBjBgknZoqQsXSp3M6f@mail.gmail.com>
Message-ID: <BFDAC4F1-393B-4CD5-B8C3-24D61AD6A12B@gmail.com>

Eavesdropping on a call requires transcoding. If you're using g729 you'll need a licensed version.

Steve on iPhone

On 21 Jan 2011, at 06:46, Sam <u2nsam at gmail.com> wrote:

> But i gives error for g729 codec !
>  
> 2011-01-21 12:04:44.817605 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode!
> 2011-01-21 12:04:44.817605 [ERR] switch_core_io.c:1042 Codec RAW Signed Linear (16 bit) encoder error!
> 2011-01-21 12:04:44.838101 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode!
> 2011-01-21 12:04:44.838101 [ERR] switch_core_io.c:1042 Codec RAW Signed Linear (16 bit) encoder error!
> 2011-01-21 12:04:44.858131 [ERR] mod_g729.c:145 This codec is only usable in passthrough mode!
> 2011-01-21 12:04:44.858131 [ERR] switch_core_io.c:433 Codec RAW Signed Linear (16 bit) decoder error!
>  
> Regds
> Sam
> 
> On Fri, Jan 21, 2011 at 10:37 AM, Sam <u2nsam at gmail.com> wrote:
> kool, it works.
> 
> Regds
> Sam
> 
> On 1/20/11, Steven Ayre <steveayre at gmail.com> wrote:
> > Yes, that will work. You can set the group either within the extension or in
> > the user config.
> >
> > For example:
> >
> >  <extension name="bridge">
> >      <condition field="destination_number" expression="^9(\d+)$">
> >        <action application="set" data="eavesdrop_group=spy1"/>
> >        <action application="bridge" data="sofia/gateway/gw1/$1"/>
> >      </condition>
> >    </extension>
> >   <extension name="eavesdrop">
> >      <condition field="destination_number" expression="^666$">
> >        <action application="answer"/>
> >        <action application="set" data="eavesdrop_require_group=spy1"/>
> >        <action application="eavesdrop" data="all"/>
> >      </condition>
> >    </extension>
> >
> >
> > -Steve
> >
> >
> > On 20 January 2011 10:11, Sam <u2nsam at gmail.com> wrote:
> >
> >> If i use
> >>
> >> <variable name="eavesdrop_group" value="spy1"/>
> >>
> >> at the user config, will the  dialplan barge the above extension where the
> >> statement is stated ?
> >>
> >> <extension name="eavesdrop">
> >>      <condition field="destination_number" expression="^779$">
> >>        <action application="answer"/>
> >>        <action application="set"
> >> data="eavesdrop_require_group=sales_call_eavesdrop"/>
> >>
> >>        <action application="eavesdrop" data="all"/>
> >>      </condition>
> >>    </extension>
> >>
> >>
> >> Regds
> >> Sam
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >> On Thu, Jan 20, 2011 at 3:13 PM, Sam <u2nsam at gmail.com> wrote:
> >>
> >>> hello ,
> >>>
> >>> I am using,
> >>>
> >>> <extension name="global" continue="true">
> >>>      <condition>
> >>>        <action application="info"/>
> >>>        <action application="db"
> >>> data="insert/spymap/${caller_id_number}/${uuid}"/>
> >>>
> >>>
> >>>        <action application="db"
> >>> data="insert/last_dial/${caller_id_number}/${destination_number}"/>
> >>>        <action application="db" data="insert/last_dial/global/${uuid}"/>
> >>>
> >>>
> >>>      </condition>
> >>>    </extension>
> >>>
> >>>    <extension name="eavesdrop">
> >>>
> >>>
> >>>      <condition field="destination_number"
> >>> expression="^88(.*)$|^\*0(.*)$">
> >>>        <action application="answer"/>
> >>>        <action application="eavesdrop" data="${db(select/spymap/$1)}"/>
> >>>
> >>>
> >>>      </condition>
> >>>    </extension>
> >>>
> >>>
> >>>
> >>> i am getting the below error:-
> >>>
> >>> 2011-01-20 14:54:01.581151 [NOTICE] mod_dptools.c:920 Channel
> >>> [sofia/internal/7006 at 192.168.2.190] has been answered
> >>> 2011-01-20 14:54:01.581151 [DEBUG] sofia.c:4646 Channel sofia/internal/
> >>> 7006 at 192.168.2.190 entering state [completed][200]
> >>> EXECUTE sofia/internal/7006 at 192.168.2.190 eavesdrop()
> >>> 2011-01-20 14:54:01.582119 [ERR] mod_dptools.c:529 Usage: [all | <uuid>]
> >>> 2011-01-20 14:54:01.582119 [NOTICE] switch_core_state_machine.c:189
> >>> sofia/internal/7006 at 192.168.2.190 has executed the last dialplan
> >>> instruction, hanging up.
> >>> 2011-01-20 14:54:01.582119 [DEBUG] switch_channel.c:2535 (sofia/internal/
> >>> 7006 at 192.168.2.190) Callstate Change ACTIVE -> HANGUP
> >>> 2011-01-20 14:54:01.582119 [NOTICE] switch_core_state_machine.c:191
> >>> Hangup
> >>> sofia/internal/7006 at 192.168.2.190 [CS_EXECUTE] [NORMAL_CLEARING]
> >>>
> >>>
> >>> http://pastebin.freeswitch.org/15076
> >>>
> >>>
> >>> Regds
> >>> Sam
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From bektas at yahoo.com  Fri Jan 21 10:32:23 2011
From: bektas at yahoo.com (Kenan BEKTAS)
Date: Thu, 20 Jan 2011 23:32:23 -0800 (PST)
Subject: [Freeswitch-users] Transfer to extension
Message-ID: <728475.83771.qm@web114717.mail.gq1.yahoo.com>

Folks,

Have an issue here. I need to have an extension to be answered by/like IVR., i.e, if the extension is rung, then, the IVR should pick up.
Calling 23 from 45, and IVR should take the call to 23.

Could anybody provide me some pointers or dialplan examples, please?

Thanks a bunch,

-Kenan


      


From marcdecorny at gmail.com  Fri Jan 21 11:14:56 2011
From: marcdecorny at gmail.com (Marc de Corny)
Date: Fri, 21 Jan 2011 08:14:56 +0000
Subject: [Freeswitch-users] Send email on successful fax sending.
In-Reply-To: <AANLkTinJ3N+zwg+ZK2RmdAZECEkLgcvOSKT0hf_mbpix@mail.gmail.com>
References: <AANLkTinJ3N+zwg+ZK2RmdAZECEkLgcvOSKT0hf_mbpix@mail.gmail.com>
Message-ID: <AANLkTi=5g-rOZtS9Dy4wq_J6OQU2roLzBDwhzhsuhQSG@mail.gmail.com>

Hi There,

Has anybody had any ideas on this ? I imagine you must all have the same
requirement in the Email to Fax scenario ?

Very grateful for any pointers
thanks
Marc

On Thu, Dec 30, 2010 at 3:29 PM, Marc de Corny <marcdecorny at gmail.com>wrote:

> Hi all,
>
> I have got all the inbound fax working and can get FS to send outbound fax
> from the shell by using the commands :
> /opt/freeswitch/bin/fs_cli \
> --execute="originate {fax_verbose=true}$DEST &txfax($TMPFAX)" as in the
> wiki
>
> However I'm looking for a way of notifying the sender on the success or
> failure of the fax emission. Is there a way of getting a result back from
> that command like fax_success 0/1 that will allow me then to send the
> relevant emails out ?
>
> Any help is much appreciated.
>
> thanks
> Marc
>
>
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From david.ponzone at ipeva.fr  Fri Jan 21 11:16:02 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 21 Jan 2011 09:16:02 +0100
Subject: [Freeswitch-users] Transfer to extension
In-Reply-To: <728475.83771.qm@web114717.mail.gq1.yahoo.com>
References: <728475.83771.qm@web114717.mail.gq1.yahoo.com>
Message-ID: <04FAB8FB-DE6B-4F74-8AA1-BAE660514F73@ipeva.fr>

Kenan,

did you take some time to check the default conf or the book ?

What you ask is fairly easy.
Here is from the default conf:

    <extension name="ivr_demo">
      <condition field="destination_number" expression="^5000$">
        <action application="answer"/>
        <action application="sleep" data="2000"/>
        <action application="ivr" data="demo_ivr"/>
      </condition>
    </extension>

Check conf/autoload_configs/ivr.conf.xml for the IVR configuration.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 21/01/2011 ? 08:32, Kenan BEKTAS a ?crit :

> Folks,
> 
> Have an issue here. I need to have an extension to be answered by/like IVR., i.e, if the extension is rung, then, the IVR should pick up.
> Calling 23 from 45, and IVR should take the call to 23.
> 
> Could anybody provide me some pointers or dialplan examples, please?
> 
> Thanks a bunch,
> 
> -Kenan
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at ipeva.fr  Fri Jan 21 11:19:29 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 21 Jan 2011 09:19:29 +0100
Subject: [Freeswitch-users] Send email on successful fax sending.
In-Reply-To: <AANLkTi=5g-rOZtS9Dy4wq_J6OQU2roLzBDwhzhsuhQSG@mail.gmail.com>
References: <AANLkTinJ3N+zwg+ZK2RmdAZECEkLgcvOSKT0hf_mbpix@mail.gmail.com>
	<AANLkTi=5g-rOZtS9Dy4wq_J6OQU2roLzBDwhzhsuhQSG@mail.gmail.com>
Message-ID: <EC688318-3BC6-491F-B33C-4A569EF34D9D@ipeva.fr>

I dont think you will manage it this way.
AFAIR, after the fax is sent, some channel variables are set, but I am not sure if there is a way to read them before the call is closed.
If not, you need to get them from the CDR.
 
David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 21/01/2011 ? 09:14, Marc de Corny a ?crit :

> Hi There,
>  
> Has anybody had any ideas on this ? I imagine you must all have the same requirement in the Email to Fax scenario ?
>  
> Very grateful for any pointers
> thanks
> Marc
> 
> On Thu, Dec 30, 2010 at 3:29 PM, Marc de Corny <marcdecorny at gmail.com> wrote:
> Hi all,
>  
> I have got all the inbound fax working and can get FS to send outbound fax from the shell by using the commands :
> /opt/freeswitch/bin/fs_cli \
> --execute="originate {fax_verbose=true}$DEST &txfax($TMPFAX)" as in the wiki
>  
> However I'm looking for a way of notifying the sender on the success or failure of the fax emission. Is there a way of getting a result back from that command like fax_success 0/1 that will allow me then to send the relevant emails out ?
>  
> Any help is much appreciated.
>  
> thanks
> Marc
>  
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From u2nsam at gmail.com  Fri Jan 21 11:36:53 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 21 Jan 2011 14:06:53 +0530
Subject: [Freeswitch-users] call barging
In-Reply-To: <BFDAC4F1-393B-4CD5-B8C3-24D61AD6A12B@gmail.com>
References: <AANLkTinMqT+Qs1guPphYWqN5okMAD9c1K9=Jey6uepRw@mail.gmail.com>
	<AANLkTimrbwUS7s1G+catXL-XVTr_pg9yOWorR04MNFY=@mail.gmail.com>
	<AANLkTinNrHFKY22Amsv-mMw=wMcxmKheo-WxT8kYTrqC@mail.gmail.com>
	<AANLkTimeyiq92cGvtoxYuUMEJGyu-Oo2sWAGB6L3Hu1n@mail.gmail.com>
	<AANLkTimhjr_fm1ZyWRchXjL_JGBjBgknZoqQsXSp3M6f@mail.gmail.com>
	<BFDAC4F1-393B-4CD5-B8C3-24D61AD6A12B@gmail.com>
Message-ID: <AANLkTinorChkZUNYCkB_uqw8mL=cYyF6tnmRG6SO+4YX@mail.gmail.com>

Is there a method to make a evesdrop master such that only he can use the
aplication and no one else ?

Regds
Sam

On Fri, Jan 21, 2011 at 1:24 PM, Steven Ayre <steveayre at gmail.com> wrote:

>  Eavesdropping on a call requires transcoding. If you're using g729 you'll
> need a licensed version.
>
> Steve on iPhone
>
> On 21 Jan 2011, at 06:46, Sam <u2nsam at gmail.com> wrote:
>
>   But i gives error for g729 codec !
>
> 2011-01-21 12:04:44.817605 [ERR] mod_g729.c:102 This codec is only usable
> in passthrough mode!
> 2011-01-21 12:04:44.817605 [ERR] switch_core_io.c:1042 Codec RAW Signed
> Linear (16 bit) encoder error!
> 2011-01-21 12:04:44.838101 [ERR] mod_g729.c:102 This codec is only usable
> in passthrough mode!
> 2011-01-21 12:04:44.838101 [ERR] switch_core_io.c:1042 Codec RAW Signed
> Linear (16 bit) encoder error!
> 2011-01-21 12:04:44.858131 [ERR] mod_g729.c:145 This codec is only usable
> in passthrough mode!
> 2011-01-21 12:04:44.858131 [ERR] switch_core_io.c:433 Codec RAW Signed
> Linear (16 bit) decoder error!
>
> Regds
> Sam
>
> On Fri, Jan 21, 2011 at 10:37 AM, Sam <u2nsam at gmail.com> wrote:
>
>> kool, it works.
>>
>> Regds
>> Sam
>>
>> On 1/20/11, Steven Ayre <steveayre at gmail.com> wrote:
>> > Yes, that will work. You can set the group either within the extension
>> or in
>> > the user config.
>> >
>> > For example:
>> >
>> >  <extension name="bridge">
>> >      <condition field="destination_number" expression="^9(\d+)$">
>> >        <action application="set" data="eavesdrop_group=spy1"/>
>> >        <action application="bridge" data="sofia/gateway/gw1/$1"/>
>> >      </condition>
>> >    </extension>
>> >   <extension name="eavesdrop">
>> >      <condition field="destination_number" expression="^666$">
>> >        <action application="answer"/>
>> >        <action application="set" data="eavesdrop_require_group=spy1"/>
>> >        <action application="eavesdrop" data="all"/>
>> >      </condition>
>> >    </extension>
>> >
>> >
>> > -Steve
>> >
>> >
>> > On 20 January 2011 10:11, Sam <u2nsam at gmail.com> wrote:
>> >
>> >> If i use
>> >>
>> >> <variable name="eavesdrop_group" value="spy1"/>
>> >>
>> >> at the user config, will the  dialplan barge the above extension where
>> the
>> >> statement is stated ?
>> >>
>> >> <extension name="eavesdrop">
>> >>      <condition field="destination_number" expression="^779$">
>> >>        <action application="answer"/>
>> >>        <action application="set"
>> >> data="eavesdrop_require_group=sales_call_eavesdrop"/>
>> >>
>> >>        <action application="eavesdrop" data="all"/>
>> >>      </condition>
>> >>    </extension>
>> >>
>> >>
>> >> Regds
>> >> Sam
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >> On Thu, Jan 20, 2011 at 3:13 PM, Sam <u2nsam at gmail.com> wrote:
>> >>
>> >>> hello ,
>> >>>
>> >>> I am using,
>> >>>
>> >>> <extension name="global" continue="true">
>> >>>      <condition>
>> >>>        <action application="info"/>
>> >>>        <action application="db"
>> >>> data="insert/spymap/${caller_id_number}/${uuid}"/>
>> >>>
>> >>>
>> >>>        <action application="db"
>> >>> data="insert/last_dial/${caller_id_number}/${destination_number}"/>
>> >>>        <action application="db"
>> data="insert/last_dial/global/${uuid}"/>
>> >>>
>> >>>
>> >>>      </condition>
>> >>>    </extension>
>> >>>
>> >>>    <extension name="eavesdrop">
>> >>>
>> >>>
>> >>>      <condition field="destination_number"
>> >>> expression="^88(.*)$|^\*0(.*)$">
>> >>>        <action application="answer"/>
>> >>>        <action application="eavesdrop"
>> data="${db(select/spymap/$1)}"/>
>> >>>
>> >>>
>> >>>      </condition>
>> >>>    </extension>
>> >>>
>> >>>
>> >>>
>> >>> i am getting the below error:-
>> >>>
>> >>> 2011-01-20 14:54:01.581151 [NOTICE] mod_dptools.c:920 Channel
>> >>> [sofia/internal/7006 at 192.168.2.190] has been answered
>> >>> 2011-01-20 14:54:01.581151 [DEBUG] sofia.c:4646 Channel
>> sofia/internal/
>> >>> 7006 at 192.168.2.190 entering state [completed][200]
>> >>> EXECUTE sofia/internal/7006 at 192.168.2.190 eavesdrop()
>> >>> 2011-01-20 14:54:01.582119 [ERR] mod_dptools.c:529 Usage: [all |
>> <uuid>]
>> >>> 2011-01-20 14:54:01.582119 [NOTICE] switch_core_state_machine.c:189
>> >>> sofia/internal/7006 at 192.168.2.190 has executed the last dialplan
>> >>> instruction, hanging up.
>> >>> 2011-01-20 14:54:01.582119 [DEBUG] switch_channel.c:2535
>> (sofia/internal/
>> >>> 7006 at 192.168.2.190) Callstate Change ACTIVE -> HANGUP
>> >>> 2011-01-20 14:54:01.582119 [NOTICE] switch_core_state_machine.c:191
>> >>> Hangup
>> >>> sofia/internal/7006 at 192.168.2.190 [CS_EXECUTE] [NORMAL_CLEARING]
>> >>>
>> >>>
>> >>> http://pastebin.freeswitch.org/15076
>> >>>
>> >>>
>> >>> Regds
>> >>> Sam
>> >>>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>>
>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Fri Jan 21 11:43:15 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 21 Jan 2011 08:43:15 +0000
Subject: [Freeswitch-users] call barging
In-Reply-To: <AANLkTinorChkZUNYCkB_uqw8mL=cYyF6tnmRG6SO+4YX@mail.gmail.com>
References: <AANLkTinMqT+Qs1guPphYWqN5okMAD9c1K9=Jey6uepRw@mail.gmail.com>
	<AANLkTimrbwUS7s1G+catXL-XVTr_pg9yOWorR04MNFY=@mail.gmail.com>
	<AANLkTinNrHFKY22Amsv-mMw=wMcxmKheo-WxT8kYTrqC@mail.gmail.com>
	<AANLkTimeyiq92cGvtoxYuUMEJGyu-Oo2sWAGB6L3Hu1n@mail.gmail.com>
	<AANLkTimhjr_fm1ZyWRchXjL_JGBjBgknZoqQsXSp3M6f@mail.gmail.com>
	<BFDAC4F1-393B-4CD5-B8C3-24D61AD6A12B@gmail.com>
	<AANLkTinorChkZUNYCkB_uqw8mL=cYyF6tnmRG6SO+4YX@mail.gmail.com>
Message-ID: <AANLkTimfs3yWxEnOQGe699Zv0TChfC2VPEDM9bwNtOcj@mail.gmail.com>

You mean so that only he can dial 666?

There're a few ways. You can create a 2nd context, set it in the user's
directory config and only place eavesdrop in that context.

You can also add a 2nd <condition> to the extension that checks either the
username or a variable set in the user's directory config.

-Steve



On 21 January 2011 08:36, Sam <u2nsam at gmail.com> wrote:

> Is there a method to make a evesdrop master such that only he can use the
> aplication and no one else ?
>
> Regds
> Sam
>
> On Fri, Jan 21, 2011 at 1:24 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>>  Eavesdropping on a call requires transcoding. If you're using g729
>> you'll need a licensed version.
>>
>> Steve on iPhone
>>
>> On 21 Jan 2011, at 06:46, Sam <u2nsam at gmail.com> wrote:
>>
>>   But i gives error for g729 codec !
>>
>> 2011-01-21 12:04:44.817605 [ERR] mod_g729.c:102 This codec is only usable
>> in passthrough mode!
>> 2011-01-21 12:04:44.817605 [ERR] switch_core_io.c:1042 Codec RAW Signed
>> Linear (16 bit) encoder error!
>> 2011-01-21 12:04:44.838101 [ERR] mod_g729.c:102 This codec is only usable
>> in passthrough mode!
>> 2011-01-21 12:04:44.838101 [ERR] switch_core_io.c:1042 Codec RAW Signed
>> Linear (16 bit) encoder error!
>> 2011-01-21 12:04:44.858131 [ERR] mod_g729.c:145 This codec is only usable
>> in passthrough mode!
>> 2011-01-21 12:04:44.858131 [ERR] switch_core_io.c:433 Codec RAW Signed
>> Linear (16 bit) decoder error!
>>
>> Regds
>> Sam
>>
>> On Fri, Jan 21, 2011 at 10:37 AM, Sam < <u2nsam at gmail.com>
>> u2nsam at gmail.com> wrote:
>>
>>> kool, it works.
>>>
>>> Regds
>>> Sam
>>>
>>> On 1/20/11, Steven Ayre < <steveayre at gmail.com>steveayre at gmail.com>
>>> wrote:
>>> > Yes, that will work. You can set the group either within the extension
>>> or in
>>> > the user config.
>>> >
>>> > For example:
>>> >
>>> >  <extension name="bridge">
>>> >      <condition field="destination_number" expression="^9(\d+)$">
>>> >        <action application="set" data="eavesdrop_group=spy1"/>
>>> >        <action application="bridge" data="sofia/gateway/gw1/$1"/>
>>> >      </condition>
>>> >    </extension>
>>> >   <extension name="eavesdrop">
>>> >      <condition field="destination_number" expression="^666$">
>>> >        <action application="answer"/>
>>> >        <action application="set" data="eavesdrop_require_group=spy1"/>
>>> >        <action application="eavesdrop" data="all"/>
>>> >      </condition>
>>> >    </extension>
>>> >
>>> >
>>> > -Steve
>>> >
>>> >
>>> > On 20 January 2011 10:11, Sam < <u2nsam at gmail.com>u2nsam at gmail.com>
>>> wrote:
>>> >
>>> >> If i use
>>> >>
>>> >> <variable name="eavesdrop_group" value="spy1"/>
>>> >>
>>> >> at the user config, will the  dialplan barge the above extension where
>>> the
>>> >> statement is stated ?
>>> >>
>>> >> <extension name="eavesdrop">
>>> >>      <condition field="destination_number" expression="^779$">
>>> >>        <action application="answer"/>
>>> >>        <action application="set"
>>> >> data="eavesdrop_require_group=sales_call_eavesdrop"/>
>>> >>
>>> >>        <action application="eavesdrop" data="all"/>
>>> >>      </condition>
>>> >>    </extension>
>>> >>
>>> >>
>>> >> Regds
>>> >> Sam
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> >> On Thu, Jan 20, 2011 at 3:13 PM, Sam < <u2nsam at gmail.com>
>>> u2nsam at gmail.com> wrote:
>>> >>
>>> >>> hello ,
>>> >>>
>>> >>> I am using,
>>> >>>
>>> >>> <extension name="global" continue="true">
>>> >>>      <condition>
>>> >>>        <action application="info"/>
>>> >>>        <action application="db"
>>> >>> data="insert/spymap/${caller_id_number}/${uuid}"/>
>>> >>>
>>> >>>
>>> >>>        <action application="db"
>>> >>> data="insert/last_dial/${caller_id_number}/${destination_number}"/>
>>> >>>        <action application="db"
>>> data="insert/last_dial/global/${uuid}"/>
>>> >>>
>>> >>>
>>> >>>      </condition>
>>> >>>    </extension>
>>> >>>
>>> >>>    <extension name="eavesdrop">
>>> >>>
>>> >>>
>>> >>>      <condition field="destination_number"
>>> >>> expression="^88(.*)$|^\*0(.*)$">
>>> >>>        <action application="answer"/>
>>> >>>        <action application="eavesdrop"
>>> data="${db(select/spymap/$1)}"/>
>>> >>>
>>> >>>
>>> >>>      </condition>
>>> >>>    </extension>
>>> >>>
>>> >>>
>>> >>>
>>> >>> i am getting the below error:-
>>> >>>
>>> >>> 2011-01-20 14:54:01.581151 [NOTICE] mod_dptools.c:920 Channel
>>> >>> [sofia/internal/ <7006 at 192.168.2.190>7006 at 192.168.2.190] has been
>>> answered
>>> >>> 2011-01-20 14:54:01.581151 [DEBUG] sofia.c:4646 Channel
>>> sofia/internal/
>>> >>> <7006 at 192.168.2.190>7006 at 192.168.2.190 entering state
>>> [completed][200]
>>> >>> EXECUTE sofia/internal/ <7006 at 192.168.2.190>7006 at 192.168.2.190eavesdrop()
>>> >>> 2011-01-20 14:54:01.582119 [ERR] mod_dptools.c:529 Usage: [all |
>>> <uuid>]
>>> >>> 2011-01-20 14:54:01.582119 [NOTICE] switch_core_state_machine.c:189
>>> >>> sofia/internal/ <7006 at 192.168.2.190>7006 at 192.168.2.190 has executed
>>> the last dialplan
>>> >>> instruction, hanging up.
>>> >>> 2011-01-20 14:54:01.582119 [DEBUG] switch_channel.c:2535
>>> (sofia/internal/
>>> >>> <7006 at 192.168.2.190>7006 at 192.168.2.190) Callstate Change ACTIVE ->
>>> HANGUP
>>> >>> 2011-01-20 14:54:01.582119 [NOTICE] switch_core_state_machine.c:191
>>> >>> Hangup
>>> >>> sofia/internal/ <7006 at 192.168.2.190>7006 at 192.168.2.190 [CS_EXECUTE]
>>> [NORMAL_CLEARING]
>>> >>>
>>> >>>
>>> >>> <http://pastebin.freeswitch.org/15076>
>>> http://pastebin.freeswitch.org/15076
>>> >>>
>>> >>>
>>> >>> Regds
>>> >>> Sam
>>> >>>
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>> >> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> <http://www.freeswitch.org/>http://www.freeswitch.org
>>> >>
>>> >>
>>> >
>>>
>>
>>  _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From avi at avimarcus.net  Fri Jan 21 12:13:06 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 21 Jan 2011 11:13:06 +0200
Subject: [Freeswitch-users] moderator conference flag and nibblebill
In-Reply-To: <3B561E18A1594FE180422EA2FC679EA6@e1705>
References: <BEC4FD4A5E3A474EA4E9A1B6DB7D33F1@e1705>
	<AANLkTimz_Ro8Tk1Qsrjy-DoF-qrjedVEbc=HnG+h4XK9@mail.gmail.com>
	<3B561E18A1594FE180422EA2FC679EA6@e1705>
Message-ID: <AANLkTik8mK=38PrUo9P+pNZFbRPJ=DicVbg7wt2+PD1O@mail.gmail.com>

Whaat about... sroring the uuid via db, and setting it in the dialplan that
when the moderator joins it unpauses the others + sets a db entry so
newcomers can be started automatically?
-Avi
On Jan 21, 2011 8:16 AM, "Madovsky" <infos at madovsky.org> wrote:
> ok thanks.
> interesting solution.
> I really need to have time to learn esl I think....
>
> ----- Original Message -----
> From: "Rupa Schomaker" <rupa at rupa.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Friday, January 21, 2011 1:03 AM
> Subject: Re: [Freeswitch-users] moderator conference flag and nibblebill
>
>
>> Not sure from the dialplan. I'd approach it like this:
>>
>> have the callers connect to the conference with nibblebill paused.
>>
>> Have a process connected to FS via ESL monitoring conference events.
>> I assume we get enough information via the ESL events to determine
>> when the moderator connects (if not, it would be a good addition so
>> suggest it). When the moderator joins, then iterate through the
>> currently joined members and unpause each one. Ensure any new members
>> that join also get unpaused on join.
>>
>> Should be pretty straightforward via ESL. Strictly from dialplan? No
>> idea.
>>
>> On Thu, Jan 20, 2011 at 7:03 PM, Madovsky <infos at madovsky.org> wrote:
>>> if a caller in waste mode go to a conference
>>> where the moderator isn't into nibblebill starts
>>> to debit his account. I can't guess hwo I can manage
>>> that in a dialplan to start debit once the conference really starts
>>>
>>> Any ?
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From kbdfck at gmail.com  Fri Jan 21 12:48:49 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Fri, 21 Jan 2011 12:48:49 +0300
Subject: [Freeswitch-users] Incorrect Loopback channel variables?
Message-ID: <AANLkTiknGMzXG9jVr7YUsPUFsg-dbOjxsVABvZG8jVtk@mail.gmail.com>

Hi all.

Seems some channel variables are set incorrectly on loopback channels.
When we do info() or uuid_dump() on loopback/-b, Channel-Name says it
is "loopback/-b", loopback leg name is B, but variable_channel_name is
set to "loopback-a".  other_loopback_leg is correctly set to uuid of
loopback/-a.

On loopback-a, there are correct "loopback/-a" only channel names and
variables...

Is this a normal behavior? Or how can I get variables not prefixed
with variable_ to deal with this inconsistence?



-- 
Best regards,

Dmitry Sytchev,
IT Engineer


From peter.olsson at visionutveckling.se  Fri Jan 21 13:06:30 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 21 Jan 2011 11:06:30 +0100
Subject: [Freeswitch-users] RFC 2833 DTMF detection - has
	something	changed?
In-Reply-To: <AANLkTimXXzVCFweC6x18yrNCf8_st9a_eRD8mxAmyega@mail.gmail.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C57ED00ADDA@cooper>
	<9CCA8E46-03D7-4F13-B6BD-8AE8D2564006@freeswitch.org>
	<AANLkTi=yhXDV7yE-BRpGuYU8rQwwMZU85VYBd87Cbo-+@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57EC81C52A@cooper>
	<AANLkTimXXzVCFweC6x18yrNCf8_st9a_eRD8mxAmyega@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57ED00B44B@cooper>

Thanks again, I tried the change below but it didn't help - however, it gave me a clue for what's going on :)

The problem seems to be that do_flush() will flush valid DTMF packets, and since do_flush() is called every time a file start/stops playing (because of rtp break is called) it is quite possible to be unlucky and miss DTMF's during this short period of time.

I've created a patch for this in jira FS-3002, it refactors the RFC2833 detection code, and after trying this out for the last 24 hours, the problems seems to be solved. Please review and apply this patch if it seems ok to you (or get back to me if you have any thoughts about the implemenation). More detailed information can be found in the jira case.

While you're at it - I have a few other patches laying around in jira - if you could have a look at those also I would really appreciate it :)

FS-2917, FS-2973 and FS-2971

Thanks,

Peter Olsson

-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale
Skickat: den 19 januari 2011 19:22
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] RFC 2833 DTMF detection - has something changed?

comment line 2993 and see if its better.
If it works better, what is on the other end of the call?, I hate it already.


On Wed, Jan 19, 2011 at 12:12 PM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> Yep, I'm on today's git, so I do have the last change/fix that was made for this in switch.rtp.c.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Steven Ayre [steveayre at gmail.com]
> Skickat: den 19 januari 2011 18:33
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] RFC 2833 DTMF detection - has something ? ?changed?
>
> Brian, he says he's on Git from today...
>
>
> On 19 January 2011 17:10, Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
> Update we did have one day where it was messed up.
>
> /b
>
> On Jan 19, 2011, at 11:02 AM, Peter Olsson wrote:
>
> Hi,
>
> I have some problems with DTMF detection, which I've never seen earlier versions. I know that there was a bug in the rtp code a few days (or week(s)?) back, but I'm running on latest git (as of today). I'm also running on the same machine as before, and more or less no config changes (except for a few dialplan changes, but nothing that changes DTMF detection).
>
> Anyway, it seems to me that the 2833-detection is not as accurate as it was before. When using a couple of months old FS version I rarely missed any DTMF's (I can't remember I ever did..:)), but now it seems to happen once in a while. Also, I decided today to get wireshark up and running, and after 5 DTMF's, FS missed the last one. I looked inside my wireshark dump, and I could clearly see all DTMF packets in there, but FS somehow missed this.
>
> Is there some kind of debugging I could enable, for instance DEBUG_2833 directive?
>
> What's the best way to move this forward? I can send my wireshark dump if that helps, but I guess you will need some more debugging info from FS as well?
>
> Thanks,
>
> Peter Olsson
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> 
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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!DSPAM:4d372d0d32761021210236!



From Stefan.Weigel at allianz-warranty.com  Fri Jan 21 13:13:12 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Fri, 21 Jan 2011 11:13:12 +0100
Subject: [Freeswitch-users] dialplan - expression matching and variable -
	2nd try
Message-ID: <5003D7D3E06F514E8C682F18D223265C046D61E5C3@AZWSMS03.azwarranty.int>

Hi all,

sorry for the first message, I hit the wrong keys an Outlook sent the message...once again:


why is the following not working:


  <condition field="destination_number" expression="^(500)$|^(600)$">
    <action application="set" data="dialed_number=$1"/> <-- $1 is empty

when using the above expression, $1 is empty.

If I do it like this


  <condition field="destination_number" expression="^(5\d\d)$|^(6\d\d)$">
    <action application="set" data="dialed_number=$1"/> <-- $1 is empty

It's working. I don't understand why.


Thanks in advance and best regards,

Stefan
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From u2nsam at gmail.com  Fri Jan 21 13:25:56 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 21 Jan 2011 15:55:56 +0530
Subject: [Freeswitch-users] call barging
In-Reply-To: <AANLkTimfs3yWxEnOQGe699Zv0TChfC2VPEDM9bwNtOcj@mail.gmail.com>
References: <AANLkTinMqT+Qs1guPphYWqN5okMAD9c1K9=Jey6uepRw@mail.gmail.com>
	<AANLkTimrbwUS7s1G+catXL-XVTr_pg9yOWorR04MNFY=@mail.gmail.com>
	<AANLkTinNrHFKY22Amsv-mMw=wMcxmKheo-WxT8kYTrqC@mail.gmail.com>
	<AANLkTimeyiq92cGvtoxYuUMEJGyu-Oo2sWAGB6L3Hu1n@mail.gmail.com>
	<AANLkTimhjr_fm1ZyWRchXjL_JGBjBgknZoqQsXSp3M6f@mail.gmail.com>
	<BFDAC4F1-393B-4CD5-B8C3-24D61AD6A12B@gmail.com>
	<AANLkTinorChkZUNYCkB_uqw8mL=cYyF6tnmRG6SO+4YX@mail.gmail.com>
	<AANLkTimfs3yWxEnOQGe699Zv0TChfC2VPEDM9bwNtOcj@mail.gmail.com>
Message-ID: <AANLkTinL3kVDSTqb=ohNo1B7hmROSH1ZiL7Ag9W-X99j@mail.gmail.com>

The second condition option is good one !

regds
Sam

On Fri, Jan 21, 2011 at 2:13 PM, Steven Ayre <steveayre at gmail.com> wrote:

> You mean so that only he can dial 666?
>
> There're a few ways. You can create a 2nd context, set it in the user's
> directory config and only place eavesdrop in that context.
>
> You can also add a 2nd <condition> to the extension that checks either the
> username or a variable set in the user's directory config.
>
> -Steve
>
>
>
>
> On 21 January 2011 08:36, Sam <u2nsam at gmail.com> wrote:
>
>> Is there a method to make a evesdrop master such that only he can use the
>> aplication and no one else ?
>>
>> Regds
>> Sam
>>
>>   On Fri, Jan 21, 2011 at 1:24 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>
>>>  Eavesdropping on a call requires transcoding. If you're using g729
>>> you'll need a licensed version.
>>>
>>> Steve on iPhone
>>>
>>> On 21 Jan 2011, at 06:46, Sam <u2nsam at gmail.com> wrote:
>>>
>>>   But i gives error for g729 codec !
>>>
>>> 2011-01-21 12:04:44.817605 [ERR] mod_g729.c:102 This codec is only usable
>>> in passthrough mode!
>>> 2011-01-21 12:04:44.817605 [ERR] switch_core_io.c:1042 Codec RAW Signed
>>> Linear (16 bit) encoder error!
>>> 2011-01-21 12:04:44.838101 [ERR] mod_g729.c:102 This codec is only usable
>>> in passthrough mode!
>>> 2011-01-21 12:04:44.838101 [ERR] switch_core_io.c:1042 Codec RAW Signed
>>> Linear (16 bit) encoder error!
>>> 2011-01-21 12:04:44.858131 [ERR] mod_g729.c:145 This codec is only usable
>>> in passthrough mode!
>>> 2011-01-21 12:04:44.858131 [ERR] switch_core_io.c:433 Codec RAW Signed
>>> Linear (16 bit) decoder error!
>>>
>>> Regds
>>> Sam
>>>
>>> On Fri, Jan 21, 2011 at 10:37 AM, Sam < <u2nsam at gmail.com>
>>> u2nsam at gmail.com> wrote:
>>>
>>>> kool, it works.
>>>>
>>>> Regds
>>>> Sam
>>>>
>>>> On 1/20/11, Steven Ayre < <steveayre at gmail.com>steveayre at gmail.com>
>>>> wrote:
>>>> > Yes, that will work. You can set the group either within the extension
>>>> or in
>>>> > the user config.
>>>> >
>>>> > For example:
>>>> >
>>>> >  <extension name="bridge">
>>>> >      <condition field="destination_number" expression="^9(\d+)$">
>>>> >        <action application="set" data="eavesdrop_group=spy1"/>
>>>> >        <action application="bridge" data="sofia/gateway/gw1/$1"/>
>>>> >      </condition>
>>>> >    </extension>
>>>> >   <extension name="eavesdrop">
>>>> >      <condition field="destination_number" expression="^666$">
>>>> >        <action application="answer"/>
>>>> >        <action application="set" data="eavesdrop_require_group=spy1"/>
>>>> >        <action application="eavesdrop" data="all"/>
>>>> >      </condition>
>>>> >    </extension>
>>>> >
>>>> >
>>>> > -Steve
>>>> >
>>>> >
>>>> > On 20 January 2011 10:11, Sam < <u2nsam at gmail.com>u2nsam at gmail.com>
>>>> wrote:
>>>> >
>>>> >> If i use
>>>> >>
>>>> >> <variable name="eavesdrop_group" value="spy1"/>
>>>> >>
>>>> >> at the user config, will the  dialplan barge the above extension
>>>> where the
>>>> >> statement is stated ?
>>>> >>
>>>> >> <extension name="eavesdrop">
>>>> >>      <condition field="destination_number" expression="^779$">
>>>> >>        <action application="answer"/>
>>>> >>        <action application="set"
>>>> >> data="eavesdrop_require_group=sales_call_eavesdrop"/>
>>>> >>
>>>> >>        <action application="eavesdrop" data="all"/>
>>>> >>      </condition>
>>>> >>    </extension>
>>>> >>
>>>> >>
>>>> >> Regds
>>>> >> Sam
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> >> On Thu, Jan 20, 2011 at 3:13 PM, Sam < <u2nsam at gmail.com>
>>>> u2nsam at gmail.com> wrote:
>>>> >>
>>>> >>> hello ,
>>>> >>>
>>>> >>> I am using,
>>>> >>>
>>>> >>> <extension name="global" continue="true">
>>>> >>>      <condition>
>>>> >>>        <action application="info"/>
>>>> >>>        <action application="db"
>>>> >>> data="insert/spymap/${caller_id_number}/${uuid}"/>
>>>> >>>
>>>> >>>
>>>> >>>        <action application="db"
>>>> >>> data="insert/last_dial/${caller_id_number}/${destination_number}"/>
>>>> >>>        <action application="db"
>>>> data="insert/last_dial/global/${uuid}"/>
>>>> >>>
>>>> >>>
>>>> >>>      </condition>
>>>> >>>    </extension>
>>>> >>>
>>>> >>>    <extension name="eavesdrop">
>>>> >>>
>>>> >>>
>>>> >>>      <condition field="destination_number"
>>>> >>> expression="^88(.*)$|^\*0(.*)$">
>>>> >>>        <action application="answer"/>
>>>> >>>        <action application="eavesdrop"
>>>> data="${db(select/spymap/$1)}"/>
>>>> >>>
>>>> >>>
>>>> >>>      </condition>
>>>> >>>    </extension>
>>>> >>>
>>>> >>>
>>>> >>>
>>>> >>> i am getting the below error:-
>>>> >>>
>>>> >>> 2011-01-20 14:54:01.581151 [NOTICE] mod_dptools.c:920 Channel
>>>> >>> [sofia/internal/ <7006 at 192.168.2.190>7006 at 192.168.2.190] has been
>>>> answered
>>>> >>> 2011-01-20 14:54:01.581151 [DEBUG] sofia.c:4646 Channel
>>>> sofia/internal/
>>>> >>> <7006 at 192.168.2.190>7006 at 192.168.2.190 entering state
>>>> [completed][200]
>>>> >>> EXECUTE sofia/internal/ <7006 at 192.168.2.190>7006 at 192.168.2.190eavesdrop()
>>>> >>> 2011-01-20 14:54:01.582119 [ERR] mod_dptools.c:529 Usage: [all |
>>>> <uuid>]
>>>> >>> 2011-01-20 14:54:01.582119 [NOTICE] switch_core_state_machine.c:189
>>>> >>> sofia/internal/ <7006 at 192.168.2.190>7006 at 192.168.2.190 has executed
>>>> the last dialplan
>>>> >>> instruction, hanging up.
>>>> >>> 2011-01-20 14:54:01.582119 [DEBUG] switch_channel.c:2535
>>>> (sofia/internal/
>>>> >>> <7006 at 192.168.2.190>7006 at 192.168.2.190) Callstate Change ACTIVE ->
>>>> HANGUP
>>>> >>> 2011-01-20 14:54:01.582119 [NOTICE] switch_core_state_machine.c:191
>>>> >>> Hangup
>>>> >>> sofia/internal/ <7006 at 192.168.2.190>7006 at 192.168.2.190 [CS_EXECUTE]
>>>> [NORMAL_CLEARING]
>>>> >>>
>>>> >>>
>>>> >>> <http://pastebin.freeswitch.org/15076>
>>>> http://pastebin.freeswitch.org/15076
>>>> >>>
>>>> >>>
>>>> >>> Regds
>>>> >>> Sam
>>>> >>>
>>>> >>
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> >> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> <http://www.freeswitch.org/>http://www.freeswitch.org
>>>> >>
>>>> >>
>>>> >
>>>>
>>>
>>>  _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
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>
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From david.ponzone at ipeva.fr  Fri Jan 21 13:56:42 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 21 Jan 2011 11:56:42 +0100
Subject: [Freeswitch-users] dialplan - expression matching and variable
	- 2nd try
In-Reply-To: <5003D7D3E06F514E8C682F18D223265C046D61E5C3@AZWSMS03.azwarranty.int>
References: <5003D7D3E06F514E8C682F18D223265C046D61E5C3@AZWSMS03.azwarranty.int>
Message-ID: <6356CA3A-7950-47B1-BB12-4D26B5E59631@ipeva.fr>

Stefan,

I am not sure to understand the difference between your 2 trials.
In both cases, you say $1 is empty, so I am not sure what is working in test 2.
Also, you should tell us what number you call to test (500 or 600).

Finally, if you right a regexp, every () group you define will be used to populate $1, $2, etc...
So in:
^(500)$|^(600)$
$1 will contain 500 if you call 500 and then $2 will be empty
$1 will be empty and $2 will contain 600 if it matches

You should write:
^(500|600)$

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 21/01/2011 ? 11:13, Weigel, Stefan a ?crit :

> Hi all,
>  
> sorry for the first message, I hit the wrong keys an Outlook sent the message?once again:
>  
>  
> why is the following not working:
>  
>   <condition field="destination_number" expression="^(500)$|^(600)$">
>     <action application="set" data="dialed_number=$1"/> <-- $1 is empty
>  
> when using the above expression, $1 is empty.
>  
> If I do it like this
>  
>   <condition field="destination_number" expression="^(5\d\d)$|^(6\d\d)$">
>     <action application="set" data="dialed_number=$1"/> <-- $1 is empty
>  
> It?s working. I don?t understand why.
>  
>  
> Thanks in advance and best regards,
>  
> Stefan
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From kond at nstel.ru  Fri Jan 21 13:58:54 2011
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Fri, 21 Jan 2011 13:58:54 +0300
Subject: [Freeswitch-users] sangoma a101: can't load freetdm module
In-Reply-To: <20110121074428.CB3FA12330@mail.nstel.ru>
Message-ID: <20110121105854.B431611434@mail.nstel.ru>

I loaded mod_freetdm.
The problem was in permissions on /dev/wanpipe* files. 
I run frreeswitch as non root user. When i gave him rw access to /dev/wanpipe* files module is loaded.
Thanks to google and list archive - http://lists.freeswitch.org/pipermail/freeswitch-users/2010-February/054359.html.
Rgds, Nikolay.


  _____  

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nikolay Kondratyev
Sent: Friday, January 21, 2011 10:44 AM
To: 'FreeSWITCH Users Help'
Subject: [Freeswitch-users] sangoma a101: can't load freetdm module


Hi all, 
 
i have a problem with A101 sangoma pri card.
I spent couple of days trying to make it work but alas...
 
My problem is that i can't load mod_freetdm module.
I'm sure i'm doing something wrong, but i can't find out what is my mistake...
So.. please help to solve the problem...
 
I setup my test system according to http://wiki.freeswitch.org/wiki/FreeTDM
I use freeswitch 1.0.7, wanpipe 3.5.18 and libsng_isdn-7.0.0.
 
 
wanrouter can see the card:
 
[root at sipx4 conf]# wanrouter hwprobe
-------------------------------
| Wanpipe Hardware Probe Info |
-------------------------------
1 . AFT-A101-SH : SLOT=4 : BUS=8 : IRQ=177 : CPU=A : PORT=1 : HWEC=32 : V=37
Card Cnt: A101-2=1
 
[root at sipx4 conf]# wanrouter status
Devices currently active:
        wanpipe1
Wanpipe Config:
Device name | Protocol Map | Adapter  | IRQ | Slot/IO | If's | CLK | Baud rate |
wanpipe1    | N/A          | A101/1D/A102/2D/4/4D/8| 177 | 4       | 1    | N/A | 0         |
Wanrouter Status:
Device name | Protocol | Station | Status        |
wanpipe1    | AFT TE1  | N/A     | Connected     |
 
Here is my config files:
 
[root at sipx4 conf]# cat /etc/wanpipe/wanpipe1.conf
#================================================
# WANPIPE1 Configuration File
#================================================
#
# Date: Wed Dec  6 20:29:03 UTC 2006
#
# Note: This file was generated automatically
#       by /usr/local/sbin/setup-sangoma program.
#
#       If you want to edit this file, it is
#       recommended that you use wancfg program
#       to do so.
#================================================
# Sangoma Technologies Inc.
#================================================
 
[devices]
wanpipe1 = WAN_AFT_TE1, Comment
 
[interfaces]
w1g1 = wanpipe1, , TDM_VOICE_API, Comment
 
[wanpipe1]
CARD_TYPE       = AFT
S514CPU         = A
CommPort        = PRI
AUTO_PCISLOT    = NO
PCISLOT         = 4
PCIBUS          = 8
FE_MEDIA        = E1
FE_LCODE        = HDB3
FE_FRAME        = CRC4
FE_LINE         = 1
TE_CLOCK        = MASTER
TE_REF_CLOCK    = 0
TE_SIG_MODE     = CCS
TE_HIGHIMPEDANCE        = NO
TE_RX_SLEVEL    = 430
LBO             = 120OH
FE_TXTRISTATE   = NO
MTU             = 1500
UDPPORT         = 9000
TTL             = 255
IGNORE_FRONT_END        = NO
TDMV_SPAN               = 1
TDMV_DCHAN              = 16
TE_AIS_MAINTENANCE = NO         #NO: defualt  YES: Start port in AIS Blue Alarm and keep line down
                                #wanpipemon -i w1g1 -c Ttx_ais_off to disable AIS maintenance mode
                                                                #wanpipemon -i w1g1 -c Ttx_ais_on to enable AIS maintenance mode
TDMV_HW_DTMF            = YES           # YES: receive dtmf events from hardware
TDMV_HW_FAX_DETECT              = YES           # YES: receive fax 1100hz events from hardware
HWEC_OPERATION_MODE     = OCT_NORMAL    # OCT_NORMAL: echo cancelation enabled with nlp (default)
                                                                                # OCT_SPEECH: improves software tone detection by disabling NLP (echo possible)
                                                                                # OCT_NO_ECHO:disables echo cancelation but allows VQE/tone functions.
HWEC_DTMF_REMOVAL       = NO    # NO: default  YES: remove dtmf out of incoming media (must have hwdtmf enabled)
HWEC_NOISE_REDUCTION    = NO    # NO: default  YES: reduces noise on the line - could break fax
HWEC_ACUSTIC_ECHO       = NO    # NO: default  YES: enables acustic echo cancelation
HWEC_NLP_DISABLE        = NO    # NO: default  YES: guarantees software tone detection (possible echo)
HWEC_TX_AUTO_GAIN       = 0     # 0: disable   -40-0: default tx audio level to be maintained (-20 default)
HWEC_RX_AUTO_GAIN       = 0     # 0: disable   -40-0: default tx audio level to be maintained (-20 default)
HWEC_TX_GAIN            = 0             # 0: disable   -24-24: db values to be applied to tx signal
HWEC_RX_GAIN            = 0             # 0: disable   -24-24: db values to be applied to tx signal
 
[w1g1]
ACTIVE_CH       = ALL
TDMV_HWEC       = YES
MTU             = 80
[root at sipx4 conf]#

[root at sipx4 conf]# cat /usr/local/freeswitch-107/conf/wanpipe.conf  (i installed FS into /usr/local/freeswitch-107)
[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750

[root at sipx4 conf]# cat /usr/local/freeswitch-107/conf/freetdm.conf
[span wanpipe wp1]
trunk_type => e1
group=1
b-channel => 1:1-15
b-channel => 1:17-31
d-channel => 1:16

[root at sipx4 conf]# cat /usr/local/freeswitch-107/conf/autoload_configs/freetdm.conf.xml
<configuration name="freetdm.conf" description="Freetdm Configuration">
  <settings>
    <param name="debug" value="0"/>
    <!--<param name="hold-music" value="$${moh_uri}"/>-->
    <!--<param name="enable-analog-option" value="call-swap"/>-->
    <!--<param name="enable-analog-option" value="3-way"/>-->
  </settings>
                <config_profiles>
                <profile name="my_pri_nt_1">
                        <param name="switchtype" value="euroisdn" />
                        <param name="interface" value="net"/>
                </profile>
                </config_profiles>
        <sangoma_pri_spans>
                <span name="wp1" cfgprofile="my_pri_nt_1">
                        <param name="dialplan" value="XML"/>
                        <param name="context" value="default"/>
                </span>
        </sangoma_pri_spans>
        <analog_spans>
        </analog_spans>
</configuration>
[root at sipx4 conf]#

The other side of the E1  connection is mediant 2000 E1 port. Mediant shows d-channel alarm.
 
And here is what i got in the log, when tried to load module:
[freeswitch107 at sipx4 log]$ cat freeswitch.log.2011-01-21-10-24-01.1
2011-01-21 10:23:15.775934 [NOTICE] mod_logfile.c:158 New log started.
2011-01-21 10:23:57.495838 [INFO] mod_enum.c:808 ENUM Reloaded
2011-01-21 10:23:57.495838 [INFO] switch_time.c:954 Timezone reloaded 530 definitions
2011-01-21 10:23:57.495838 [DEBUG] ftdm_config.c:52 New mod directory: /usr/local/freeswitch-107/mod
2011-01-21 10:23:57.495838 [DEBUG] ftdm_config.c:58 New config directory: /usr/local/freeswitch-107/conf
2011-01-21 10:23:57.495838 [DEBUG] ftdm_sched.c:154 Initializing scheduling API
2011-01-21 10:23:57.495838 [DEBUG] ftdm_sched.c:251 Created schedule freetdm-master
2011-01-21 10:23:57.495838 [NOTICE] ftdm_sched.c:178 Launching main schedule thread
2011-01-21 10:23:57.495838 [DEBUG] ftdm_sched.c:187 Running schedule freetdm-master in the main schedule thread
2011-01-21 10:23:57.495838 [DEBUG] ftdm_config.c:80 Configuration file is /usr/local/freeswitch-107/conf/modules.conf.
2011-01-21 10:23:57.495838 [NOTICE] ftdm_io.c:5731 Modules configured: 1
2011-01-21 10:23:57.495838 [DEBUG] ftdm_config.c:80 Configuration file is /usr/local/freeswitch-107/conf/freetdm.conf.
2011-01-21 10:23:57.495838 [DEBUG] ftdm_io.c:4676 Reading FreeTDM configuration file
2011-01-21 10:23:57.495838 [DEBUG] ftdm_io.c:4692 found config for span
2011-01-21 10:23:57.495838 [INFO] ftdm_io.c:4976 Loading IO from /usr/local/freeswitch-107/mod/ftmod_wanpipe.so [wanpipe]
2011-01-21 10:23:57.495838 [DEBUG] ftdm_config.c:80 Configuration file is /usr/local/freeswitch-107/conf/wanpipe.conf.
2011-01-21 10:23:57.495838 [INFO] ftdm_io.c:800 Auto-loaded I/O module 'wanpipe'
2011-01-21 10:23:57.495838 [DEBUG] ftdm_io.c:4706 created span 1 (wp1) of type wanpipe
2011-01-21 10:23:57.495838 [DEBUG] ftdm_io.c:4722 span 1 [trunk_type]=[e1]
2011-01-21 10:23:57.495838 [DEBUG] ftdm_io.c:4727 setting trunk type to 'E1'
2011-01-21 10:23:57.495838 [DEBUG] ftdm_io.c:4722 span 1 [group]=[1]
2011-01-21 10:23:57.495838 [DEBUG] ftdm_io.c:4722 span 1 [b-channel]=[1:1-15]
2011-01-21 10:23:57.495838 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 1
2011-01-21 10:23:57.495838 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 2
2011-01-21 10:23:57.495838 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 3
2011-01-21 10:23:57.495838 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 4
2011-01-21 10:23:57.495838 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 5
2011-01-21 10:23:57.495838 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 6
2011-01-21 10:23:57.495838 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 7
2011-01-21 10:23:57.495838 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 8
2011-01-21 10:23:57.495838 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 9
2011-01-21 10:23:57.495838 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 10
2011-01-21 10:23:57.495838 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 11
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 12
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 13
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 14
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 15
2011-01-21 10:23:57.497841 [ERR] ftdm_io.c:4623 1:Failed to configure span2011-01-21 10:23:57.497841 [DEBUG] ftdm_io.c:4722 span 1 [b-channel]=[1:17-31]
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 17
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 18
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 19
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 20
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 21
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 22
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 23
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 24
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 25
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 26
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 27
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 28
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 29
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 30
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 31
2011-01-21 10:23:57.497841 [ERR] ftdm_io.c:4623 1:Failed to configure span2011-01-21 10:23:57.497841 [DEBUG] ftdm_io.c:4722 span 1 [d-channel]=[1:16]
2011-01-21 10:23:57.497841 [ERR] ftmod_wanpipe.c:246 Failed to open wanpipe device span 1 channel 16
2011-01-21 10:23:57.497841 [ERR] ftdm_io.c:4623 1:Failed to configure span2011-01-21 10:23:57.497841 [INFO] ftdm_io.c:4901 Configured 0 channel(s)
2011-01-21 10:23:57.497841 [ERR] ftdm_io.c:5741 FreeTDM global configuration failed!
2011-01-21 10:23:57.497841 [ERR] mod_freetdm.c:4181 Error configuring FreeTDM
2011-01-21 10:23:57.497841 [CRIT] switch_loadable_module.c:928 Error Loading module /usr/local/freeswitch-107/mod/mod_freetdm.so
**Module load routine returned an error**
2011-01-21 10:23:57.597841 [NOTICE] ftdm_sched.c:147 Main scheduling thread going out ...
[freeswitch107 at sipx4 log]$

Can anybody please help with the problem?
I'm ready to provide additional info if needed.
 
Thanks and regards,
Nikolay.
 

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From marcdecorny at gmail.com  Fri Jan 21 14:32:26 2011
From: marcdecorny at gmail.com (Marc de Corny)
Date: Fri, 21 Jan 2011 11:32:26 +0000
Subject: [Freeswitch-users] Send email on successful fax sending.
In-Reply-To: <EC688318-3BC6-491F-B33C-4A569EF34D9D@ipeva.fr>
References: <AANLkTinJ3N+zwg+ZK2RmdAZECEkLgcvOSKT0hf_mbpix@mail.gmail.com>
	<AANLkTi=5g-rOZtS9Dy4wq_J6OQU2roLzBDwhzhsuhQSG@mail.gmail.com>
	<EC688318-3BC6-491F-B33C-4A569EF34D9D@ipeva.fr>
Message-ID: <AANLkTimFs3yZxROM7K8UzoeO6nn0ROxYNMwc-gEqaeo2@mail.gmail.com>

Thanks David.
I will see what I can find
Marc

On Fri, Jan 21, 2011 at 8:19 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> I dont think you will manage it this way.
> AFAIR, after the fax is sent, some channel variables are set, but I am not
> sure if there is a way to read them before the call is closed.
> If not, you need to get them from the CDR.
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
>  tel:      0811 46 26 26
>  www.ipeva.fr  -   www.ipeva-studio.com
>
>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
>  Le 21/01/2011 ? 09:14, Marc de Corny a ?crit :
>
>   Hi There,
>
> Has anybody had any ideas on this ? I imagine you must all have the same
> requirement in the Email to Fax scenario ?
>
> Very grateful for any pointers
> thanks
> Marc
>
> On Thu, Dec 30, 2010 at 3:29 PM, Marc de Corny <marcdecorny at gmail.com>wrote:
>
>> Hi all,
>>
>> I have got all the inbound fax working and can get FS to send outbound fax
>> from the shell by using the commands :
>> /opt/freeswitch/bin/fs_cli \
>> --execute="originate {fax_verbose=true}$DEST &txfax($TMPFAX)" as in the
>> wiki
>>
>> However I'm looking for a way of notifying the sender on the success or
>> failure of the fax emission. Is there a way of getting a result back from
>> that command like fax_success 0/1 that will allow me then to send the
>> relevant emails out ?
>>
>> Any help is much appreciated.
>>
>> thanks
>> Marc
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Fri Jan 21 14:47:46 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 21 Jan 2011 11:47:46 +0000
Subject: [Freeswitch-users] dialplan - expression matching and variable
 - 2nd try
In-Reply-To: <5003D7D3E06F514E8C682F18D223265C046D61E5C3@AZWSMS03.azwarranty.int>
References: <5003D7D3E06F514E8C682F18D223265C046D61E5C3@AZWSMS03.azwarranty.int>
Message-ID: <AANLkTi=CqO=2P8AG7xT7JkRHc4fnZ+uKrtXnLc_BMH2D@mail.gmail.com>

$1 is the first brackets
$2 is the second brackets

If you dial 5xx I would expect $1 to be 5xx and $2 empty
If you dial 6xx I would expect $1 empty and $2 to be 6xx

Try:
<condition field="destination_number" expression="^(500|600)$">

-Steve


On 21 January 2011 10:13, Weigel, Stefan <Stefan.Weigel at allianz-warranty.com
> wrote:

>  Hi all,
>
>
>
> sorry for the first message, I hit the wrong keys an Outlook sent the
> message?once again:
>
>
>
>
>
> why is the following not working:
>
>
>
>   <condition field="destination_number" expression="^(500)$|^(600)$">
>
>     <action application="set" data="dialed_number=$1"/> <-- $1 is empty
>
>
>
> when using the above expression, $1 is empty.
>
>
>
> If I do it like this
>
>
>
>   <condition field="destination_number" expression="^(5\d\d)$|^(6\d\d)$">
>
>     <action application="set" data="dialed_number=$1"/> <-- $1 is empty
>
>
>
> It?s working. I don?t understand why.
>
>
>
>
>
> Thanks in advance and best regards,
>
>
>
> Stefan
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From thomas at chaschperli.ch  Fri Jan 21 15:07:25 2011
From: thomas at chaschperli.ch (Thomas Mueller)
Date: Fri, 21 Jan 2011 13:07:25 +0100
Subject: [Freeswitch-users] Send email on successful fax sending.
In-Reply-To: <AANLkTi=5g-rOZtS9Dy4wq_J6OQU2roLzBDwhzhsuhQSG@mail.gmail.com>
References: <AANLkTinJ3N+zwg+ZK2RmdAZECEkLgcvOSKT0hf_mbpix@mail.gmail.com>
	<AANLkTi=5g-rOZtS9Dy4wq_J6OQU2roLzBDwhzhsuhQSG@mail.gmail.com>
Message-ID: <4D3976FD.4090805@chaschperli.ch>

On 21.01.2011 09:14, Marc de Corny wrote:
> Hi There,
> Has anybody had any ideas on this ? I imagine you must all have the 
> same requirement in the Email to Fax scenario ?


reading the sourcecode the channel var "fax_success" gets set to 1 for 
OK, to 0 for fail.

- Thomas



From steveayre at gmail.com  Fri Jan 21 15:33:42 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 21 Jan 2011 12:33:42 +0000
Subject: [Freeswitch-users] Send email on successful fax sending.
In-Reply-To: <4D3976FD.4090805@chaschperli.ch>
References: <AANLkTinJ3N+zwg+ZK2RmdAZECEkLgcvOSKT0hf_mbpix@mail.gmail.com>
	<AANLkTi=5g-rOZtS9Dy4wq_J6OQU2roLzBDwhzhsuhQSG@mail.gmail.com>
	<4D3976FD.4090805@chaschperli.ch>
Message-ID: <AANLkTimo0n3UP2c09F3AmgQdkRdU+DVAMxjYrw7cP9gm@mail.gmail.com>

But since the channel will hangup, you'd need to wait for and check the CDR
to see that variable.

mod_spandsp_fax.c:412 is "TODO Fire events"

It looks like that it's planned for mod_spandsp to fire an event that'll
indicate that the fax has finished sending and would contain the status.

If that was implemented, you could connect via ESL, do the originate and
wait for the event to tell you whether it worked or not.

-Steve



On 21 January 2011 12:07, Thomas Mueller <thomas at chaschperli.ch> wrote:

> On 21.01.2011 09:14, Marc de Corny wrote:
> > Hi There,
> > Has anybody had any ideas on this ? I imagine you must all have the
> > same requirement in the Email to Fax scenario ?
>
>
> reading the sourcecode the channel var "fax_success" gets set to 1 for
> OK, to 0 for fail.
>
> - Thomas
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From marcdecorny at gmail.com  Fri Jan 21 15:40:14 2011
From: marcdecorny at gmail.com (Marc de Corny)
Date: Fri, 21 Jan 2011 12:40:14 +0000
Subject: [Freeswitch-users] Send email on successful fax sending.
In-Reply-To: <AANLkTimo0n3UP2c09F3AmgQdkRdU+DVAMxjYrw7cP9gm@mail.gmail.com>
References: <AANLkTinJ3N+zwg+ZK2RmdAZECEkLgcvOSKT0hf_mbpix@mail.gmail.com>
	<AANLkTi=5g-rOZtS9Dy4wq_J6OQU2roLzBDwhzhsuhQSG@mail.gmail.com>
	<4D3976FD.4090805@chaschperli.ch>
	<AANLkTimo0n3UP2c09F3AmgQdkRdU+DVAMxjYrw7cP9gm@mail.gmail.com>
Message-ID: <AANLkTi=zBgX-trHwVUUgfGNZh3SmcJfru1tD=5-aE=MY@mail.gmail.com>

Thanks, all I have not investigated the ESL options yet, I will have a look
and see,


many thanks for the responses,

Marc

On Fri, Jan 21, 2011 at 12:33 PM, Steven Ayre <steveayre at gmail.com> wrote:

> But since the channel will hangup, you'd need to wait for and check the CDR
> to see that variable.
>
> mod_spandsp_fax.c:412 is "TODO Fire events"
>
> It looks like that it's planned for mod_spandsp to fire an event that'll
> indicate that the fax has finished sending and would contain the status.
>
> If that was implemented, you could connect via ESL, do the originate and
> wait for the event to tell you whether it worked or not.
>
> -Steve
>
>
>
>   On 21 January 2011 12:07, Thomas Mueller <thomas at chaschperli.ch> wrote:
>
>>   On 21.01.2011 09:14, Marc de Corny wrote:
>> > Hi There,
>> > Has anybody had any ideas on this ? I imagine you must all have the
>> > same requirement in the Email to Fax scenario ?
>>
>>
>> reading the sourcecode the channel var "fax_success" gets set to 1 for
>> OK, to 0 for fail.
>>
>> - Thomas
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Fri Jan 21 16:05:16 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 21 Jan 2011 10:05:16 -0300
Subject: [Freeswitch-users] Run dialplan tools from event socket
In-Reply-To: <AANLkTin84SkDdDsBE0wfU2QnG025bNK8BhcW=JDvz9gv@mail.gmail.com>
References: <AANLkTikg5+1aMoWxYr3UYO13iO2qA6FsG9FFi-Zg3_bD@mail.gmail.com>
	<AANLkTin5fn9qMre-8xFAZD5_SUzbLbp_YueTRPPyh-=K@mail.gmail.com>
	<AANLkTin84SkDdDsBE0wfU2QnG025bNK8BhcW=JDvz9gv@mail.gmail.com>
Message-ID: <AANLkTi=TX6BURrxR38X9bwyUai_52xLK80kMvm9CmZ8T@mail.gmail.com>

Actually, Fraser, I think this won't work...

att_xfer uses the signal_bond variable to get the leg connected to the party
being transferred. This variable is unset when you park the legs or break
the bridge in any way. Maybe I can make a API command that does att_xfer
taking the trasferred leg UUID as the transferred party? Do me a favor, make
some tests and I will take a deeper look at the att_xfer application code.

Regards,
Jo?o Mesquita


On Fri, Jan 21, 2011 at 1:37 AM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> Thanks Jo?o.
>
> My transfer_call extension runs a couple of js scripts to get and validate
> the number to transfer to, then does
>                 <action application="att_xfer" data="${dial_string}"/>
> (and it has a couple of steps after that to handle failed transfers.)
>
> So could I use ESL's execute command to run the execute_extension? Not sure
> how I missed that option in the wiki. I"ll give it a try, see what happens.
>
>
> I forgot to say in the original post, but execute_extension seems to be
> particularly nice for this use-case, as it falls back through the dialplan
> gracefully if there's a problem.
>
> Cheers,
> Fraser
>
>
>
>
> 2011/1/20 Jo?o Mesquita <jmesquita at freeswitch.org>
>
> Lucky for you I have been working on this lately and the bad news is ...
>> there's no easy way to do it....
>>
>> You can execute an extension like you said, but you have to park the legs
>> first... It would help to know how's the transfer_call extension so that I
>> can try to help you out, but maybe it is easier if you think of it this way:
>>
>> When you use an app like att_xfer, the core already knows what to do next
>> with a call and parks the legs for you. If you do it on ESL, you've done it
>> half way and you didn't really park anything before you transfered the call.
>> When the bridge is undone, the leg that was not transfered doesn't know what
>> to do, has no applications to be run at this moment and so all it's left for
>> it is to let go.
>>
>> A bit clearer? Att_xfer is a bit of a pain in the butt and it kinda
>> requires you to know a bit more of the inner workings of the state machine.
>>
>> You can always execute att_xfer using ESL's execute (
>> http://wiki.freeswitch.org/wiki/Event_Socket_Library#execute) if you
>> don't care what happens to the legs afterwards but if you want to have
>> control over all 3 legs, no luck for you...
>>
>> Regards,
>> Jo?o Mesquita
>>
>>
>> On Fri, Jan 21, 2011 at 12:13 AM, Fraser Redmond <fraserredmond at gmail.com
>> > wrote:
>>
>>> Is there any way to run a dialplan tool from the event socket?
>>>
>>> I have a dialplan that uses a dtmf to set up and perform an attended
>>> transfer, like so:
>>> <action application="bind_meta_app" data="8 a s
>>> execute_extension::TransferCall XML transfer_call"/>
>>>
>>> But I can't see any way to run the same thing from the event socket. I
>>> thought doing an  "api uuid_transfer"  might do it, but that hangs up one of
>>> the legs (no good for attended transfer.)
>>>
>>> api uuid_transfer Uuid -bleg TransferCall XML transfer_call
>>>
>>> As far as I can see, the closest thing is "sendmsg execute", but it looks
>>> like you have to park a call/channel first to use that, so I'm not sure that
>>> that is much use for attended transfer either.
>>>
>>> Or should I be lame and do "api uuid_send_dtmf" to send * 8.
>>>
>>> Cheers,
>>> Fraser
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From lakindia89 at gmail.com  Fri Jan 21 16:07:11 2011
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 21 Jan 2011 18:37:11 +0530
Subject: [Freeswitch-users] Help regard to park_timeout
Message-ID: <AANLkTi=edjFv90+q3G2kkzoJ76kWgx+NBe=Qx3CYA2Jd@mail.gmail.com>

Dear all,
I was using park_timeout and I come across the following scenario which I
felt something is missing.
I've originated a call as follows.

originate
{ignore_early_media=true,exec_after_bridge_app=park,park_timeout=60,api_hangup_hook='perl
/root/a.pl'}freetdm/grp1/a/9952248266 &park()

Once the call is answered I originated another call.
originate {ignore_early_media=true,park_timeout=60,api_hangup_hook='perl
/root/a.pl'}freetdm/grp1/a/9843171457 &park()

Once this call is also answered, I said "uuid_bridge <uuid1> <uuid2>". Both
call gets bridged. After some time, I hangup the second call (9843171457).
Now the first call goes into park().

I expect that the first call will hangup after 60 seconds, but it didn't.

The freeswitch log is here
http://pastebin.freeswitch.org/15099

When I start to use the park_timeout, I thought once a leg is in park, then
the timer will start, and once it is unparked for various reason the timer
will be reseted. After sometime, when the leg again comes in park, the timer
will start. Is this correct?
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From steveayre at gmail.com  Fri Jan 21 16:10:28 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 21 Jan 2011 13:10:28 +0000
Subject: [Freeswitch-users] Send email on successful fax sending.
In-Reply-To: <AANLkTi=zBgX-trHwVUUgfGNZh3SmcJfru1tD=5-aE=MY@mail.gmail.com>
References: <AANLkTinJ3N+zwg+ZK2RmdAZECEkLgcvOSKT0hf_mbpix@mail.gmail.com>
	<AANLkTi=5g-rOZtS9Dy4wq_J6OQU2roLzBDwhzhsuhQSG@mail.gmail.com>
	<4D3976FD.4090805@chaschperli.ch>
	<AANLkTimo0n3UP2c09F3AmgQdkRdU+DVAMxjYrw7cP9gm@mail.gmail.com>
	<AANLkTi=zBgX-trHwVUUgfGNZh3SmcJfru1tD=5-aE=MY@mail.gmail.com>
Message-ID: <AANLkTimKgZDpuDUgNaqT1=61mqwOoJ66u4m-u4LBjxXM@mail.gmail.com>

I've just put a patch together for mod_spandsp that should add the required
event.

It's on Jira, FS-3004.
http://jira.freeswitch.org/browse/FS-3004

I've checked it compiles, but I have no T38 capable gateways so someone else
will need to test it out. Does anyone fancy trying it out?

-Steve



On 21 January 2011 12:40, Marc de Corny <marcdecorny at gmail.com> wrote:

> Thanks, all I have not investigated the ESL options yet, I will have a look
> and see,
>
>
> many thanks for the responses,
>
> Marc
>
> On Fri, Jan 21, 2011 at 12:33 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> But since the channel will hangup, you'd need to wait for and check the
>> CDR to see that variable.
>>
>> mod_spandsp_fax.c:412 is "TODO Fire events"
>>
>> It looks like that it's planned for mod_spandsp to fire an event that'll
>> indicate that the fax has finished sending and would contain the status.
>>
>> If that was implemented, you could connect via ESL, do the originate and
>> wait for the event to tell you whether it worked or not.
>>
>> -Steve
>>
>>
>>
>>   On 21 January 2011 12:07, Thomas Mueller <thomas at chaschperli.ch> wrote:
>>
>>>   On 21.01.2011 09:14, Marc de Corny wrote:
>>> > Hi There,
>>> > Has anybody had any ideas on this ? I imagine you must all have the
>>> > same requirement in the Email to Fax scenario ?
>>>
>>>
>>> reading the sourcecode the channel var "fax_success" gets set to 1 for
>>> OK, to 0 for fail.
>>>
>>> - Thomas
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From steveayre at gmail.com  Fri Jan 21 16:16:07 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 21 Jan 2011 13:16:07 +0000
Subject: [Freeswitch-users] Run dialplan tools from event socket
In-Reply-To: <AANLkTi=TX6BURrxR38X9bwyUai_52xLK80kMvm9CmZ8T@mail.gmail.com>
References: <AANLkTikg5+1aMoWxYr3UYO13iO2qA6FsG9FFi-Zg3_bD@mail.gmail.com>
	<AANLkTin5fn9qMre-8xFAZD5_SUzbLbp_YueTRPPyh-=K@mail.gmail.com>
	<AANLkTin84SkDdDsBE0wfU2QnG025bNK8BhcW=JDvz9gv@mail.gmail.com>
	<AANLkTi=TX6BURrxR38X9bwyUai_52xLK80kMvm9CmZ8T@mail.gmail.com>
Message-ID: <AANLkTikDBWtCTp=h0t_Qxtm1S+ZTX_UUtvj5RgbFdt+g@mail.gmail.com>

uuid_broadcast appears to have support for executing a dialplan app from the
api (it uses say as an example):

uuid_broadcast <uuid> app!::args [aleg|bleg|both]

http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast

I haven't tried it myself though.

-Steve


2011/1/21 Jo?o Mesquita <jmesquita at freeswitch.org>

> Actually, Fraser, I think this won't work...
>
> att_xfer uses the signal_bond variable to get the leg connected to the
> party being transferred. This variable is unset when you park the legs or
> break the bridge in any way. Maybe I can make a API command that does
> att_xfer taking the trasferred leg UUID as the transferred party? Do me a
> favor, make some tests and I will take a deeper look at the att_xfer
> application code.
>
> Regards,
> Jo?o Mesquita
>
>
>
> On Fri, Jan 21, 2011 at 1:37 AM, Fraser Redmond <fraserredmond at gmail.com>wrote:
>
>> Thanks Jo?o.
>>
>> My transfer_call extension runs a couple of js scripts to get and validate
>> the number to transfer to, then does
>>                 <action application="att_xfer" data="${dial_string}"/>
>> (and it has a couple of steps after that to handle failed transfers.)
>>
>> So could I use ESL's execute command to run the execute_extension? Not
>> sure how I missed that option in the wiki. I"ll give it a try, see what
>> happens.
>>
>>
>> I forgot to say in the original post, but execute_extension seems to be
>> particularly nice for this use-case, as it falls back through the dialplan
>> gracefully if there's a problem.
>>
>> Cheers,
>> Fraser
>>
>>
>>
>>
>> 2011/1/20 Jo?o Mesquita <jmesquita at freeswitch.org>
>>
>> Lucky for you I have been working on this lately and the bad news is ...
>>> there's no easy way to do it....
>>>
>>> You can execute an extension like you said, but you have to park the legs
>>> first... It would help to know how's the transfer_call extension so that I
>>> can try to help you out, but maybe it is easier if you think of it this way:
>>>
>>> When you use an app like att_xfer, the core already knows what to do next
>>> with a call and parks the legs for you. If you do it on ESL, you've done it
>>> half way and you didn't really park anything before you transfered the call.
>>> When the bridge is undone, the leg that was not transfered doesn't know what
>>> to do, has no applications to be run at this moment and so all it's left for
>>> it is to let go.
>>>
>>> A bit clearer? Att_xfer is a bit of a pain in the butt and it kinda
>>> requires you to know a bit more of the inner workings of the state machine.
>>>
>>> You can always execute att_xfer using ESL's execute (
>>> http://wiki.freeswitch.org/wiki/Event_Socket_Library#execute) if you
>>> don't care what happens to the legs afterwards but if you want to have
>>> control over all 3 legs, no luck for you...
>>>
>>> Regards,
>>> Jo?o Mesquita
>>>
>>>
>>> On Fri, Jan 21, 2011 at 12:13 AM, Fraser Redmond <
>>> fraserredmond at gmail.com> wrote:
>>>
>>>> Is there any way to run a dialplan tool from the event socket?
>>>>
>>>> I have a dialplan that uses a dtmf to set up and perform an attended
>>>> transfer, like so:
>>>> <action application="bind_meta_app" data="8 a s
>>>> execute_extension::TransferCall XML transfer_call"/>
>>>>
>>>> But I can't see any way to run the same thing from the event socket. I
>>>> thought doing an  "api uuid_transfer"  might do it, but that hangs up one of
>>>> the legs (no good for attended transfer.)
>>>>
>>>> api uuid_transfer Uuid -bleg TransferCall XML transfer_call
>>>>
>>>> As far as I can see, the closest thing is "sendmsg execute", but it
>>>> looks like you have to park a call/channel first to use that, so I'm not
>>>> sure that that is much use for attended transfer either.
>>>>
>>>> Or should I be lame and do "api uuid_send_dtmf" to send * 8.
>>>>
>>>> Cheers,
>>>> Fraser
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Fri Jan 21 16:17:24 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 21 Jan 2011 10:17:24 -0300
Subject: [Freeswitch-users] Run dialplan tools from event socket
In-Reply-To: <AANLkTi=TX6BURrxR38X9bwyUai_52xLK80kMvm9CmZ8T@mail.gmail.com>
References: <AANLkTikg5+1aMoWxYr3UYO13iO2qA6FsG9FFi-Zg3_bD@mail.gmail.com>
	<AANLkTin5fn9qMre-8xFAZD5_SUzbLbp_YueTRPPyh-=K@mail.gmail.com>
	<AANLkTin84SkDdDsBE0wfU2QnG025bNK8BhcW=JDvz9gv@mail.gmail.com>
	<AANLkTi=TX6BURrxR38X9bwyUai_52xLK80kMvm9CmZ8T@mail.gmail.com>
Message-ID: <AANLkTin0zNpex_qcsbzfOn1AM2jOvvxCWmfpGrQT3T=3@mail.gmail.com>

Yep, I just made a test with the default dialplan and when you do this on
the CLI:

uuid_transfer <uuid> execute_extension:att_xfer\sXML\sfeatures inline

the leg to be transfered is lost because you att_xfer can't clear the
CF_INNER_BRIDGE on the leg, which, by the looks of the bridge code, will
park the leg for you...

I am not a very experienced programmer, but that's what it looks like to be
happening. Maybe an API command for att_xfer would make most ppl life's
easier? It seems to be basically the same function as the application but
instead of using the signal_bond variable, we could use a var provided on
the command line? Anyone?

The other option is to do what Edwin said, do it all "by hand".

Regards,
Jo?o Mesquita


2011/1/21 Jo?o Mesquita <jmesquita at freeswitch.org>

> Actually, Fraser, I think this won't work...
>
> att_xfer uses the signal_bond variable to get the leg connected to the
> party being transferred. This variable is unset when you park the legs or
> break the bridge in any way. Maybe I can make a API command that does
> att_xfer taking the trasferred leg UUID as the transferred party? Do me a
> favor, make some tests and I will take a deeper look at the att_xfer
> application code.
>
> Regards,
> Jo?o Mesquita
>
>
>
> On Fri, Jan 21, 2011 at 1:37 AM, Fraser Redmond <fraserredmond at gmail.com>wrote:
>
>> Thanks Jo?o.
>>
>> My transfer_call extension runs a couple of js scripts to get and validate
>> the number to transfer to, then does
>>                 <action application="att_xfer" data="${dial_string}"/>
>> (and it has a couple of steps after that to handle failed transfers.)
>>
>> So could I use ESL's execute command to run the execute_extension? Not
>> sure how I missed that option in the wiki. I"ll give it a try, see what
>> happens.
>>
>>
>> I forgot to say in the original post, but execute_extension seems to be
>> particularly nice for this use-case, as it falls back through the dialplan
>> gracefully if there's a problem.
>>
>> Cheers,
>> Fraser
>>
>>
>>
>>
>> 2011/1/20 Jo?o Mesquita <jmesquita at freeswitch.org>
>>
>> Lucky for you I have been working on this lately and the bad news is ...
>>> there's no easy way to do it....
>>>
>>> You can execute an extension like you said, but you have to park the legs
>>> first... It would help to know how's the transfer_call extension so that I
>>> can try to help you out, but maybe it is easier if you think of it this way:
>>>
>>> When you use an app like att_xfer, the core already knows what to do next
>>> with a call and parks the legs for you. If you do it on ESL, you've done it
>>> half way and you didn't really park anything before you transfered the call.
>>> When the bridge is undone, the leg that was not transfered doesn't know what
>>> to do, has no applications to be run at this moment and so all it's left for
>>> it is to let go.
>>>
>>> A bit clearer? Att_xfer is a bit of a pain in the butt and it kinda
>>> requires you to know a bit more of the inner workings of the state machine.
>>>
>>> You can always execute att_xfer using ESL's execute (
>>> http://wiki.freeswitch.org/wiki/Event_Socket_Library#execute) if you
>>> don't care what happens to the legs afterwards but if you want to have
>>> control over all 3 legs, no luck for you...
>>>
>>> Regards,
>>> Jo?o Mesquita
>>>
>>>
>>> On Fri, Jan 21, 2011 at 12:13 AM, Fraser Redmond <
>>> fraserredmond at gmail.com> wrote:
>>>
>>>> Is there any way to run a dialplan tool from the event socket?
>>>>
>>>> I have a dialplan that uses a dtmf to set up and perform an attended
>>>> transfer, like so:
>>>> <action application="bind_meta_app" data="8 a s
>>>> execute_extension::TransferCall XML transfer_call"/>
>>>>
>>>> But I can't see any way to run the same thing from the event socket. I
>>>> thought doing an  "api uuid_transfer"  might do it, but that hangs up one of
>>>> the legs (no good for attended transfer.)
>>>>
>>>> api uuid_transfer Uuid -bleg TransferCall XML transfer_call
>>>>
>>>> As far as I can see, the closest thing is "sendmsg execute", but it
>>>> looks like you have to park a call/channel first to use that, so I'm not
>>>> sure that that is much use for attended transfer either.
>>>>
>>>> Or should I be lame and do "api uuid_send_dtmf" to send * 8.
>>>>
>>>> Cheers,
>>>> Fraser
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From bektas at yahoo.com  Fri Jan 21 11:29:17 2011
From: bektas at yahoo.com (Kenan BEKTAS)
Date: Fri, 21 Jan 2011 00:29:17 -0800 (PST)
Subject: [Freeswitch-users] Transfer to extension
In-Reply-To: <04FAB8FB-DE6B-4F74-8AA1-BAE660514F73@ipeva.fr>
Message-ID: <402127.1648.qm@web114701.mail.gq1.yahoo.com>

Hi David,
Thanks for your prompt reply. Really appreciate that. Yes, I have checked those conf files and tried many things but does not seem to do what I want.
I tried the following code in?"conf/autoload_configs/ivr.conf.xml" and "conf/dialplan/public.xml" one at a time?but no success, yet.
<extension name="ivr_demo">?? ? ?<condition field="destination_number" expression="23">?? ? ? ?<action application="answer"/>?? ? ? ?<action application="sleep" data="2000"/>?? ? ? ?<action application="ivr" data="demo_ivr"/>?? ? ?</condition>?? ?</extension>
All I need is that all the calls to 23 to be responded by IVR no matter what happens. Kind a call forwarding.
Thanks a bunch,
---

KenanToronto, Canada


--- On Fri, 1/21/11, David Ponzone <david.ponzone at ipeva.fr> wrote:

From: David Ponzone <david.ponzone at ipeva.fr>
Subject: Re: [Freeswitch-users] Transfer to extension
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Friday, January 21, 2011, 3:16 AM

Kenan,
did you take some time to check the default conf or the book ?
What you ask is fairly easy.Here is from the default conf:
?? ?<extension name="ivr_demo">?? ? ?<condition field="destination_number" expression="^5000$">?? ? ? ?<action application="answer"/>?? ? ? ?<action application="sleep" data="2000"/>?? ? ? ?<action application="ivr" data="demo_ivr"/>?? ? ?</condition>?? ?</extension>
Check conf/autoload_configs/ivr.conf.xml for the IVR configuration.

David Ponzone ?Direction Techniqueemail: david.ponzone at ipeva.frtel: ? ? ?01 74 03 18 97gsm: ? 06 66 98 76 34
Service Client?IPevatel: ? ? ?0811 46 26 26www.ipeva.fr? -? ?www.ipeva-studio.com
Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.





Le 21/01/2011 ? 08:32, Kenan BEKTAS a ?crit :
Folks,

Have an issue here. I need to have an extension to be answered by/like IVR., i.e, if the extension is rung, then, the IVR should pick up.
Calling 23 from 45, and IVR should take the call to 23.

Could anybody provide me some pointers or dialplan examples, please?

Thanks a bunch,

-Kenan




_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From Stefan.Weigel at allianz-warranty.com  Fri Jan 21 13:10:21 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Fri, 21 Jan 2011 11:10:21 +0100
Subject: [Freeswitch-users] dialplan - expression matching and variable
Message-ID: <5003D7D3E06F514E8C682F18D223265C046D61E5C2@AZWSMS03.azwarranty.int>

Hi all,

why is the following not working:


  <condition field="destination_number" expression="^(500)$|^(600)$">
    <action application="set" data="dialed_number=$1"/> <-- $1 is empty

when using the following not working:



Stefan Weigel
Advanced IT-Professional
Tel.: +49 89 2000 48 975
Fax: +49 89 2000 48 566
eMail: Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>
Allianz Automotive Services GmbH
Einsteinring 28
85609 Aschheim
Germany

http://www.allianz-warranty.com
Gesch?ftsf?hrung: Andreas R?sing, Horst Ziegler
Amtsgericht M?nchen, HRB 175682
F?r Umsatzsteuerzwecke: Ust-ID-Nr.: DE 262 617 720

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From Stefan.Weigel at allianz-warranty.com  Fri Jan 21 16:45:27 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Fri, 21 Jan 2011 14:45:27 +0100
Subject: [Freeswitch-users] dialplan - expression matching and variable
 - 2nd try
In-Reply-To: <6356CA3A-7950-47B1-BB12-4D26B5E59631@ipeva.fr>
References: <5003D7D3E06F514E8C682F18D223265C046D61E5C3@AZWSMS03.azwarranty.int>
	<6356CA3A-7950-47B1-BB12-4D26B5E59631@ipeva.fr>
Message-ID: <5003D7D3E06F514E8C682F18D223265C046D61E5C4@AZWSMS03.azwarranty.int>

Hi David,

got it! I forgot about the multiple ?()? leading to $1, $2 .. $n. I was always using $1, so in some cases it was empty. Dooh..


Thanks!


Best regards

Stefan

Stefan Weigel
Advanced IT-Professional
Tel.: +49 89 2000 48 975
Fax: +49 89 2000 48 566
eMail: Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>
Allianz Automotive Services GmbH
Einsteinring 28
85609 Aschheim
Germany

http://www.allianz-warranty.com
Gesch?ftsf?hrung: Andreas R?sing, Horst Ziegler
Amtsgericht M?nchen, HRB 175682
F?r Umsatzsteuerzwecke: Ust-ID-Nr.: DE 262 617 720

Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von David Ponzone
Gesendet: Freitag, 21. Januar 2011 11:57
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] dialplan - expression matching and variable - 2nd try

Stefan,

I am not sure to understand the difference between your 2 trials.
In both cases, you say $1 is empty, so I am not sure what is working in test 2.
Also, you should tell us what number you call to test (500 or 600).

Finally, if you right a regexp, every () group you define will be used to populate $1, $2, etc...
So in:
^(500)$|^(600)$
$1 will contain 500 if you call 500 and then $2 will be empty
$1 will be empty and $2 will contain 600 if it matches

You should write:
^(500|600)$

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr<mailto:david.ponzone at ipeva.fr>
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 21/01/2011 ? 11:13, Weigel, Stefan a ?crit :


Hi all,

sorry for the first message, I hit the wrong keys an Outlook sent the message?once again:


why is the following not working:


  <condition field="destination_number" expression="^(500)$|^(600)$">
    <action application="set" data="dialed_number=$1"/> <-- $1 is empty

when using the above expression, $1 is empty.

If I do it like this


  <condition field="destination_number" expression="^(5\d\d)$|^(6\d\d)$">
    <action application="set" data="dialed_number=$1"/> <-- $1 is empty

It?s working. I don?t understand why.


Thanks in advance and best regards,

Stefan
_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From jmesquita at freeswitch.org  Fri Jan 21 17:06:58 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 21 Jan 2011 11:06:58 -0300
Subject: [Freeswitch-users] Send email on successful fax sending.
In-Reply-To: <AANLkTimKgZDpuDUgNaqT1=61mqwOoJ66u4m-u4LBjxXM@mail.gmail.com>
References: <AANLkTinJ3N+zwg+ZK2RmdAZECEkLgcvOSKT0hf_mbpix@mail.gmail.com>
	<AANLkTi=5g-rOZtS9Dy4wq_J6OQU2roLzBDwhzhsuhQSG@mail.gmail.com>
	<4D3976FD.4090805@chaschperli.ch>
	<AANLkTimo0n3UP2c09F3AmgQdkRdU+DVAMxjYrw7cP9gm@mail.gmail.com>
	<AANLkTi=zBgX-trHwVUUgfGNZh3SmcJfru1tD=5-aE=MY@mail.gmail.com>
	<AANLkTimKgZDpuDUgNaqT1=61mqwOoJ66u4m-u4LBjxXM@mail.gmail.com>
Message-ID: <AANLkTin3GQ8YoG=YjWdTChuqdPrZNt4AxfSrRFXHhaSz@mail.gmail.com>

One other thing you could do is use api_reporting_hook. There, you will have
variables set and you can run a script that you like. Beware that in this
script, you won't have session.

I think api_reporting_hook is still undocumented...

Jo?o Mesquita


On Fri, Jan 21, 2011 at 10:10 AM, Steven Ayre <steveayre at gmail.com> wrote:

> I've just put a patch together for mod_spandsp that should add the required
> event.
>
> It's on Jira, FS-3004.
> http://jira.freeswitch.org/browse/FS-3004
>
> I've checked it compiles, but I have no T38 capable gateways so someone
> else will need to test it out. Does anyone fancy trying it out?
>
> -Steve
>
>
>
>
> On 21 January 2011 12:40, Marc de Corny <marcdecorny at gmail.com> wrote:
>
>> Thanks, all I have not investigated the ESL options yet, I will have a
>> look and see,
>>
>>
>> many thanks for the responses,
>>
>> Marc
>>
>> On Fri, Jan 21, 2011 at 12:33 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>
>>> But since the channel will hangup, you'd need to wait for and check the
>>> CDR to see that variable.
>>>
>>> mod_spandsp_fax.c:412 is "TODO Fire events"
>>>
>>> It looks like that it's planned for mod_spandsp to fire an event that'll
>>> indicate that the fax has finished sending and would contain the status.
>>>
>>> If that was implemented, you could connect via ESL, do the originate and
>>> wait for the event to tell you whether it worked or not.
>>>
>>> -Steve
>>>
>>>
>>>
>>>   On 21 January 2011 12:07, Thomas Mueller <thomas at chaschperli.ch>wrote:
>>>
>>>>   On 21.01.2011 09:14, Marc de Corny wrote:
>>>> > Hi There,
>>>> > Has anybody had any ideas on this ? I imagine you must all have the
>>>> > same requirement in the Email to Fax scenario ?
>>>>
>>>>
>>>> reading the sourcecode the channel var "fax_success" gets set to 1 for
>>>> OK, to 0 for fail.
>>>>
>>>> - Thomas
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From erik.dekkers at wvds.nl  Fri Jan 21 17:14:30 2011
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Fri, 21 Jan 2011 15:14:30 +0100
Subject: [Freeswitch-users] dialplan - expression matching and variable
In-Reply-To: <5003D7D3E06F514E8C682F18D223265C046D61E5C2@AZWSMS03.azwarranty.int>
References: <5003D7D3E06F514E8C682F18D223265C046D61E5C2@AZWSMS03.azwarranty.int>
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE0782B278DC@FS1.wvds.local>

Regex should be:


<condition field="destination_number" expression="^(500|600)$">

Now it's saying: IF destination number is 500 OR 600. For more info on regular expressions see http://www.addedbytes.com/cheat-sheets/download/regular-expressions-cheat-sheet-v2.pdf

Regards,

Erik

Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Weigel, Stefan
Verzonden: vrijdag 21 januari 2011 11:10
Aan: 'freeswitch-users at lists.freeswitch.org'
Onderwerp: [Freeswitch-users] dialplan - expression matching and variable

Hi all,

why is the following not working:


  <condition field="destination_number" expression="^(500)$|^(600)$">
    <action application="set" data="dialed_number=$1"/> <-- $1 is empty

when using the following not working:



Stefan Weigel
Advanced IT-Professional
Tel.: +49 89 2000 48 975
Fax: +49 89 2000 48 566
eMail: Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>
Allianz Automotive Services GmbH
Einsteinring 28
85609 Aschheim
Germany

http://www.allianz-warranty.com
Gesch?ftsf?hrung: Andreas R?sing, Horst Ziegler
Amtsgericht M?nchen, HRB 175682
F?r Umsatzsteuerzwecke: Ust-ID-Nr.: DE 262 617 720

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From kris at kriskinc.com  Fri Jan 21 17:27:40 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Fri, 21 Jan 2011 09:27:40 -0500
Subject: [Freeswitch-users] Send email on successful fax sending.
In-Reply-To: <AANLkTi=5g-rOZtS9Dy4wq_J6OQU2roLzBDwhzhsuhQSG@mail.gmail.com>
References: <AANLkTinJ3N+zwg+ZK2RmdAZECEkLgcvOSKT0hf_mbpix@mail.gmail.com>
	<AANLkTi=5g-rOZtS9Dy4wq_J6OQU2roLzBDwhzhsuhQSG@mail.gmail.com>
Message-ID: <AANLkTi=fWOPE8h+cadpK96FvV0mZ--qKVEhi-jQR5Aw8@mail.gmail.com>

You can use api_hangup_hook to call a lua script to do this right now:

http://wiki.freeswitch.org/wiki/Mod_lua#Special_Case:_env_object

On Fri, Jan 21, 2011 at 3:14 AM, Marc de Corny <marcdecorny at gmail.com> wrote:
> Hi There,
>
> Has anybody had any ideas on this ? I imagine you must all have the same
> requirement in the Email to Fax scenario ?
>
> Very grateful for any pointers
> thanks
> Marc
>
> On Thu, Dec 30, 2010 at 3:29 PM, Marc de Corny <marcdecorny at gmail.com>
> wrote:
>>
>> Hi all,
>>
>> I have got all the inbound fax working and can get FS to send outbound fax
>> from the shell by using the commands :
>> /opt/freeswitch/bin/fs_cli \
>> --execute="originate {fax_verbose=true}$DEST &txfax($TMPFAX)" as in the
>> wiki
>>
>> However I'm looking for a way of notifying the sender on the success or
>> failure of the fax emission. Is there a way of getting a result back from
>> that command like fax_success 0/1 that will allow me then to send the
>> relevant emails out ?
>>
>> Any help is much appreciated.
>>
>> thanks
>> Marc
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner


From raul at etellicom.com  Fri Jan 21 17:52:33 2011
From: raul at etellicom.com (Raul Fragoso)
Date: Fri, 21 Jan 2011 12:52:33 -0200
Subject: [Freeswitch-users] Run dialplan tools from event socket
In-Reply-To: <AANLkTikg5+1aMoWxYr3UYO13iO2qA6FsG9FFi-Zg3_bD@mail.gmail.com>
References: <AANLkTikg5+1aMoWxYr3UYO13iO2qA6FsG9FFi-Zg3_bD@mail.gmail.com>
Message-ID: <1295621553.21875.8.camel@raul-laptop>

Be lame :) That's what I use as well, along with bind_meta_app to
execute an intermediary extension just to att_xfer the call, and then
use uuid_recv_dtmf (not uuid_send_dtmf) to initiate a consultation
transfer from event socket. I do that only for 2-legged calls though,
otherwise, for parked or other 1-legged calls, I simply use
uuid_transfer to send the call to another intermediary extension in the
dial plan which performs a consultation transfer with apps other than
att_xfer.

Regards,

Raul

On Thu, 2011-01-20 at 22:13 -0500, Fraser Redmond wrote:
> Is there any way to run a dialplan tool from the event socket?
> 
> I have a dialplan that uses a dtmf to set up and perform an attended
> transfer, like so:
> <action application="bind_meta_app" data="8 a s
> execute_extension::TransferCall XML transfer_call"/>
> 
> But I can't see any way to run the same thing from the event socket. I
> thought doing an  "api uuid_transfer"  might do it, but that hangs up
> one of the legs (no good for attended transfer.)
> 
> api uuid_transfer Uuid -bleg TransferCall XML transfer_call
> 
> As far as I can see, the closest thing is "sendmsg execute", but it
> looks like you have to park a call/channel first to use that, so I'm
> not sure that that is much use for attended transfer either.
> 
> Or should I be lame and do "api uuid_send_dtmf" to send * 8.
> 
> Cheers,
> Fraser
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From infos at madovsky.org  Fri Jan 21 19:03:51 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 21 Jan 2011 11:03:51 -0500
Subject: [Freeswitch-users] moderator conference flag and nibblebill
References: <BEC4FD4A5E3A474EA4E9A1B6DB7D33F1@e1705><AANLkTimz_Ro8Tk1Qsrjy-DoF-qrjedVEbc=HnG+h4XK9@mail.gmail.com><3B561E18A1594FE180422EA2FC679EA6@e1705>
	<AANLkTik8mK=38PrUo9P+pNZFbRPJ=DicVbg7wt2+PD1O@mail.gmail.com>
Message-ID: <8DB21ABA698745DC89ADF180019369A0@e1705>

also...
but I decided to not allow anyone to enter in conf
unless if the moderator is already in with a little db check 
so as I know who is moderator in dialplan I can easily filter the conference ;)
  ----- Original Message ----- 
  From: Avi Marcus 
  To: FreeSWITCH Users Help 
  Sent: Friday, January 21, 2011 4:13 AM
  Subject: Re: [Freeswitch-users] moderator conference flag and nibblebill


  Whaat about... sroring the uuid via db, and setting it in the dialplan that when the moderator joins it unpauses the others + sets a db entry so newcomers can be started automatically?
  -Avi

  On Jan 21, 2011 8:16 AM, "Madovsky" <infos at madovsky.org> wrote:
  > ok thanks.
  > interesting solution.
  > I really need to have time to learn esl I think....
  > 
  > ----- Original Message ----- 
  > From: "Rupa Schomaker" <rupa at rupa.com>
  > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
  > Sent: Friday, January 21, 2011 1:03 AM
  > Subject: Re: [Freeswitch-users] moderator conference flag and nibblebill
  > 
  > 
  >> Not sure from the dialplan. I'd approach it like this:
  >>
  >> have the callers connect to the conference with nibblebill paused.
  >>
  >> Have a process connected to FS via ESL monitoring conference events.
  >> I assume we get enough information via the ESL events to determine
  >> when the moderator connects (if not, it would be a good addition so
  >> suggest it). When the moderator joins, then iterate through the
  >> currently joined members and unpause each one. Ensure any new members
  >> that join also get unpaused on join.
  >>
  >> Should be pretty straightforward via ESL. Strictly from dialplan? No 
  >> idea.
  >>
  >> On Thu, Jan 20, 2011 at 7:03 PM, Madovsky <infos at madovsky.org> wrote:
  >>> if a caller in waste mode go to a conference
  >>> where the moderator isn't into nibblebill starts
  >>> to debit his account. I can't guess hwo I can manage
  >>> that in a dialplan to start debit once the conference really starts
  >>>
  >>> Any ?
  >>>
  >>> _______________________________________________
  >>> FreeSWITCH-users mailing list
  >>> FreeSWITCH-users at lists.freeswitch.org
  >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  >>> http://www.freeswitch.org
  >>>
  >>>
  >>
  >>
  >>
  >> -- 
  >> -Rupa
  >>
  >> _______________________________________________
  >> FreeSWITCH-users mailing list
  >> FreeSWITCH-users at lists.freeswitch.org
  >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  >> http://www.freeswitch.org 
  > 
  > 
  > _______________________________________________
  > FreeSWITCH-users mailing list
  > FreeSWITCH-users at lists.freeswitch.org
  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  > http://www.freeswitch.org



------------------------------------------------------------------------------


  _______________________________________________
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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Fri Jan 21 19:13:54 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 21 Jan 2011 11:13:54 -0500
Subject: [Freeswitch-users] praise of freeSWITCH developers
Message-ID: <9FE96265969E4BAEB801002D975CC50F@e1705>

I would like to remember to all people on this
emailist that all freeSWITCH guys work very hard
to update/upgrade and make FS more powerful, and
also to answer to hundreds of user emails every day.
Donations are welcome and appreciated.
It helps the humans behind the machine
who give to you the opportunity to create and use amazing projects 
and to continue to make this open source project alive and active.

Regards
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From rmartinez at redvoiss.net  Fri Jan 21 18:06:49 2011
From: rmartinez at redvoiss.net (Ricardo Martinez)
Date: Fri, 21 Jan 2011 12:06:49 -0300
Subject: [Freeswitch-users] Question aboutCPU usage.
Message-ID: <57f59d718411e237cd08c17c7d93d03c@mail.gmail.com>

Hello.

I have the opportunity to use an even more powerful server.  I have just
installed CentOS 5.5 64 bit, and compile freeswitch with 64 bits.

This is the output with 100 simm calls using transcoding from G711 to G729



top - 16:57:56 up  6:04,  2 users,  load average: 11.60, 12.49, 12.58

Tasks: 168 total,   1 running, 167 sleeping,   0 stopped,   0 zombie

Cpu0  : 35.4%us,  0.6%sy,  0.0%ni, 63.6%id,  0.3%wa,  0.0%hi,  0.1%si,
0.0%st

Cpu1  : 34.9%us,  0.6%sy,  0.0%ni, 58.8%id,  4.9%wa,  0.1%hi,  0.7%si,
0.0%st

Cpu2  : 34.9%us,  0.6%sy,  0.0%ni, 59.6%id,  4.1%wa,  0.1%hi,  0.7%si,
0.0%st

Cpu3  : 35.0%us,  0.7%sy,  0.0%ni, 62.0%id,  1.8%wa,  0.2%hi,  0.3%si,
0.0%st

Cpu4  : 35.8%us,  1.3%sy,  0.0%ni, 62.5%id,  0.2%wa,  0.0%hi,  0.1%si,
0.0%st

Cpu5  : 35.6%us,  1.6%sy,  0.0%ni, 60.1%id,  1.8%wa,  0.1%hi,  0.8%si,
0.0%st

Cpu6  : 35.8%us,  1.9%sy,  0.0%ni, 59.9%id,  1.5%wa,  0.1%hi,  0.8%si,
0.0%st

Cpu7  : 36.5%us,  2.0%sy,  0.0%ni, 60.6%id,  0.7%wa,  0.0%hi,  0.2%si,
 0.0%st

Mem:   8165808k total,  1208292k used,  6957516k free,   150228k buffers

Swap: 10223608k total,        0k used, 10223608k free,   772980k cached



  PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
COMMAND

 5265 root      18   0  541m  79m 5796 S 469.5  1.0   1077:32
freeswitch

    1 root      15   0 10348  684  572 S  0.0  0.0   0:01.24
init

    2 root      RT  -5     0    0    0 S  0.0  0.0   0:00.00
migration/0

    3 root      34  19     0    0    0 S  0.0  0.0   0:00.00
ksoftirqd/0

    4 root      RT  -5     0    0    0 S  0.0  0.0   0:00.00
watchdog/0

    5 root      RT  -5     0    0    0 S  0.0  0.0   0:00.00
migration/1

    6 root      34  19     0    0    0 S  0.0  0.0   0:00



Is normal the load average?



This is the output with uname ?a



[root at siptrcrv2 snmp]# uname -a

Linux siptrcrv2 2.6.18-194.el5 #1 SMP Fri Apr 2 14:58:14 EDT 2010 x86_64
x86_64 x86_64 GNU/Linux



Regards,

Ricardo.-







*De:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *En nombre de *Brian West
*Enviado el:* mi?rcoles, 19 de enero de 2011 16:22
*Para:* FreeSWITCH Users Help
*Asunto:* Re: [Freeswitch-users] Question aboutCPU usage.



Chances are if you would install a 64bit OS on that there NICE 64bit CPU it
would work much better.



/b



On Jan 19, 2011, at 12:43 PM, Ricardo Martinez wrote:



Hello.



[root at ser-ng bin]# uname -a

Linux 2.6.33.5-124.fc13.i686.PAE #1 SMP Fri Jun 11 09:42:24 UTC 2010 i686
i686 i386 GNU/Linux



Ricardo.-



*De:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *En nombre de *Brian West
*Enviado el:* mi?rcoles, 19 de enero de 2011 15:41
*Para:* FreeSWITCH Users Help
*Asunto:* Re: [Freeswitch-users] Question aboutCPU usage.



Depend what does uname -a say?



/b
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From fdelawarde at wirelessmundi.com  Fri Jan 21 19:28:26 2011
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Fri, 21 Jan 2011 17:28:26 +0100
Subject: [Freeswitch-users] praise of freeSWITCH developers
In-Reply-To: <9FE96265969E4BAEB801002D975CC50F@e1705>
References: <9FE96265969E4BAEB801002D975CC50F@e1705>
Message-ID: <1295627306.30863.188.camel@luna.tc.commsmundi.com>

On Fri, 2011-01-21 at 11:13 -0500, Madovsky wrote:
> Donations are welcome and appreciated.

You're right, not much left in my bank account but here goes $20.

Everyone follow me on this! Together we can make this thread bigger than
2010's "freeswitch CPU usage".

Fran?ois.



From infos at madovsky.org  Fri Jan 21 19:36:55 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 21 Jan 2011 11:36:55 -0500
Subject: [Freeswitch-users] praise of freeSWITCH developers
References: <9FE96265969E4BAEB801002D975CC50F@e1705>
	<1295627306.30863.188.camel@luna.tc.commsmundi.com>
Message-ID: <E4DFCF0A44324A7786C03C1E3229BEEA@e1705>

Yes,
even $1 can help.

Regards

----- Original Message ----- 
From: "Fran?ois Delawarde" <fdelawarde at wirelessmundi.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, January 21, 2011 11:28 AM
Subject: Re: [Freeswitch-users] praise of freeSWITCH developers


> On Fri, 2011-01-21 at 11:13 -0500, Madovsky wrote:
>> Donations are welcome and appreciated.
>
> You're right, not much left in my bank account but here goes $20.
>
> Everyone follow me on this! Together we can make this thread bigger than
> 2010's "freeswitch CPU usage".
>
> Fran?ois.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From rajesh.npnr at yahoo.com  Fri Jan 21 19:38:34 2011
From: rajesh.npnr at yahoo.com (rex.alex)
Date: Fri, 21 Jan 2011 08:38:34 -0800 (PST)
Subject: [Freeswitch-users] Send email on successful fax sending.
In-Reply-To: <AANLkTimKgZDpuDUgNaqT1=61mqwOoJ66u4m-u4LBjxXM@mail.gmail.com>
References: <AANLkTinJ3N+zwg+ZK2RmdAZECEkLgcvOSKT0hf_mbpix@mail.gmail.com>
	<AANLkTi=5g-rOZtS9Dy4wq_J6OQU2roLzBDwhzhsuhQSG@mail.gmail.com>
	<4D3976FD.4090805@chaschperli.ch>
	<AANLkTimo0n3UP2c09F3AmgQdkRdU+DVAMxjYrw7cP9gm@mail.gmail.com>
	<AANLkTi=zBgX-trHwVUUgfGNZh3SmcJfru1tD=5-aE=MY@mail.gmail.com>
	<AANLkTimKgZDpuDUgNaqT1=61mqwOoJ66u4m-u4LBjxXM@mail.gmail.com>
Message-ID: <1295627914599-5948386.post@n2.nabble.com>


Hello Steve,

I am also working around the fax module and I would like to get the fax
delivery status through "uuid_getvar uuid fax_success" api function, so I
have installed this patch. But it is still not fetching the required status.

Please assist.

Regards,

Rex 
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Send-email-on-successful-fax-sending-tp5881415p5948386.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
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From steveayre at gmail.com  Fri Jan 21 19:44:42 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 21 Jan 2011 16:44:42 +0000
Subject: [Freeswitch-users] Send email on successful fax sending.
In-Reply-To: <1295627914599-5948386.post@n2.nabble.com>
References: <AANLkTinJ3N+zwg+ZK2RmdAZECEkLgcvOSKT0hf_mbpix@mail.gmail.com>
	<AANLkTi=5g-rOZtS9Dy4wq_J6OQU2roLzBDwhzhsuhQSG@mail.gmail.com>
	<4D3976FD.4090805@chaschperli.ch>
	<AANLkTimo0n3UP2c09F3AmgQdkRdU+DVAMxjYrw7cP9gm@mail.gmail.com>
	<AANLkTi=zBgX-trHwVUUgfGNZh3SmcJfru1tD=5-aE=MY@mail.gmail.com>
	<AANLkTimKgZDpuDUgNaqT1=61mqwOoJ66u4m-u4LBjxXM@mail.gmail.com>
	<1295627914599-5948386.post@n2.nabble.com>
Message-ID: <AANLkTi==DN40Ebdi0A0bqX95Ct7Ra_Px+uSX8i4acf0i@mail.gmail.com>

That won't work if the channel no longer exists (it accesses the channel's
object and gets the variable stored there).

You can either use a hangup hook as someone else suggested (runs after
hangup but before the session is destroyed so the channels are still
available) or look at my event patch which raises an event containing a
header with that variable's value.

-Steve


On 21 January 2011 16:38, rex.alex <rajesh.npnr at yahoo.com> wrote:

> Hello Steve,
>
> I am also working around the fax module and I would like to get the fax
> delivery status through "uuid_getvar uuid fax_success" api function, so I
> have installed this patch. But it is still not fetching the required status.
>
>
> Please assist.
>
> Regards,
> Rex
> ------------------------------
> View this message in context: Re: Send email on successful fax sending.<http://freeswitch-users.2379917.n2.nabble.com/Send-email-on-successful-fax-sending-tp5881415p5948386.html>
> Sent from the freeswitch-users mailing list archive<http://freeswitch-users.2379917.n2.nabble.com/>at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Fri Jan 21 19:50:42 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 21 Jan 2011 13:50:42 -0300
Subject: [Freeswitch-users] Send email on successful fax sending.
In-Reply-To: <AANLkTi==DN40Ebdi0A0bqX95Ct7Ra_Px+uSX8i4acf0i@mail.gmail.com>
References: <AANLkTinJ3N+zwg+ZK2RmdAZECEkLgcvOSKT0hf_mbpix@mail.gmail.com>
	<AANLkTi=5g-rOZtS9Dy4wq_J6OQU2roLzBDwhzhsuhQSG@mail.gmail.com>
	<4D3976FD.4090805@chaschperli.ch>
	<AANLkTimo0n3UP2c09F3AmgQdkRdU+DVAMxjYrw7cP9gm@mail.gmail.com>
	<AANLkTi=zBgX-trHwVUUgfGNZh3SmcJfru1tD=5-aE=MY@mail.gmail.com>
	<AANLkTimKgZDpuDUgNaqT1=61mqwOoJ66u4m-u4LBjxXM@mail.gmail.com>
	<1295627914599-5948386.post@n2.nabble.com>
	<AANLkTi==DN40Ebdi0A0bqX95Ct7Ra_Px+uSX8i4acf0i@mail.gmail.com>
Message-ID: <AANLkTimpPuf51m3VA9UTH2iyL0JrU-MdL8pnGa8chbcx@mail.gmail.com>

The api_reporting_hook is called on the same state where the CDRs are
available so you are 100% sure that you have all vars that go to CDR
available.

Regards,
Jo?o Mesquita


On Fri, Jan 21, 2011 at 1:44 PM, Steven Ayre <steveayre at gmail.com> wrote:

> That won't work if the channel no longer exists (it accesses the channel's
> object and gets the variable stored there).
>
> You can either use a hangup hook as someone else suggested (runs after
> hangup but before the session is destroyed so the channels are still
> available) or look at my event patch which raises an event containing a
> header with that variable's value.
>
> -Steve
>
>
> On 21 January 2011 16:38, rex.alex <rajesh.npnr at yahoo.com> wrote:
>
>> Hello Steve,
>>
>> I am also working around the fax module and I would like to get the fax
>> delivery status through "uuid_getvar uuid fax_success" api function, so I
>> have installed this patch. But it is still not fetching the required status.
>>
>>
>> Please assist.
>>
>> Regards,
>> Rex
>> ------------------------------
>> View this message in context: Re: Send email on successful fax sending.<http://freeswitch-users.2379917.n2.nabble.com/Send-email-on-successful-fax-sending-tp5881415p5948386.html>
>> Sent from the freeswitch-users mailing list archive<http://freeswitch-users.2379917.n2.nabble.com/>at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mitch.capper at gmail.com  Fri Jan 21 20:00:19 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Fri, 21 Jan 2011 09:00:19 -0800
Subject: [Freeswitch-users] Portaudio Improvements
Message-ID: <AANLkTim9AmtEGZoK0NyG1yKVsmJ_MSwXAppi-6AcCqnz@mail.gmail.com>

I have submitted a ticket with a patch for portaudio and my improvements.
This includes the improvements I had discussed previously during my call for
input on the mailing list and is available at:
http://jira.freeswitch.org/browse/FS-3006

It was only tested in windows, however most of the changes should not effect
default behavior (but please test if you can).

To test you will want to enable some things in the portaudio config
including:
                    <param name="live-stream-switch" value="true" />
                    <param name="no-auto-resume-call" value="true" />
                    <param name="no-ring-during-call" value="true" />

you will then be able to use the new features fully.

Many of these changes were made to add better support for softphone's using
freeswitch, I will be releasing my embedded freeswitch phone (FSClient) for
the conference call in a week and a half  it is .net/c# but WPF so sadly
windows only currently.  jlink and drk are helping so we should end up with
a nice installer also for it.     If you would like to test it please let me
know.

~Mitch
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From jmesquita at freeswitch.org  Fri Jan 21 20:07:54 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 21 Jan 2011 14:07:54 -0300
Subject: [Freeswitch-users] Portaudio Improvements
In-Reply-To: <AANLkTim9AmtEGZoK0NyG1yKVsmJ_MSwXAppi-6AcCqnz@mail.gmail.com>
References: <AANLkTim9AmtEGZoK0NyG1yKVsmJ_MSwXAppi-6AcCqnz@mail.gmail.com>
Message-ID: <AANLkTimm69f6adgkgDz03GUOq8xJw_oaVvrkAGxd1yjG@mail.gmail.com>

Mitch, I will work on continuing FSComm for a thick client multi-platform
solution but one of the big show stoppers for me was the lack of AEC on
mod_pa.

I tried the preprocessors embedded on the core by Tony that are using the
speexdsp but I got no luck. The main tests I made were on a mac using the
speakerphone. Do you have any experience with this type of technology? It
seems to me that the only AEC available is the one implemented on speexdsp
but I am completely new to this.

Regards,
Jo?o Mesquita


On Fri, Jan 21, 2011 at 2:00 PM, Mitch Capper <mitch.capper at gmail.com>wrote:

> I have submitted a ticket with a patch for portaudio and my improvements.
> This includes the improvements I had discussed previously during my call for
> input on the mailing list and is available at:
> http://jira.freeswitch.org/browse/FS-3006
>
> It was only tested in windows, however most of the changes should not
> effect default behavior (but please test if you can).
>
> To test you will want to enable some things in the portaudio config
> including:
>                     <param name="live-stream-switch" value="true" />
>                     <param name="no-auto-resume-call" value="true" />
>                     <param name="no-ring-during-call" value="true" />
>
> you will then be able to use the new features fully.
>
> Many of these changes were made to add better support for softphone's using
> freeswitch, I will be releasing my embedded freeswitch phone (FSClient) for
> the conference call in a week and a half  it is .net/c# but WPF so sadly
> windows only currently.  jlink and drk are helping so we should end up with
> a nice installer also for it.     If you would like to test it please let me
> know.
>
> ~Mitch
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Fri Jan 21 20:08:04 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 21 Jan 2011 14:08:04 -0300
Subject: [Freeswitch-users] Portaudio Improvements
In-Reply-To: <AANLkTim9AmtEGZoK0NyG1yKVsmJ_MSwXAppi-6AcCqnz@mail.gmail.com>
References: <AANLkTim9AmtEGZoK0NyG1yKVsmJ_MSwXAppi-6AcCqnz@mail.gmail.com>
Message-ID: <AANLkTinTemTaXbRNjc8pWk7HBMqbULrbK18c9Y8F9EU_@mail.gmail.com>

Mitch, I will work on continuing FSComm for a thick client multi-platform
solution but one of the big show stoppers for me was the lack of AEC on
mod_pa.

I tried the preprocessors embedded on the core by Tony that are using the
speexdsp but I got no luck. The main tests I made were on a mac using the
speakerphone. Do you have any experience with this type of technology? It
seems to me that the only AEC available is the one implemented on speexdsp
but I am completely new to this.

Regards,
Jo?o Mesquita


On Fri, Jan 21, 2011 at 2:00 PM, Mitch Capper <mitch.capper at gmail.com>wrote:

> I have submitted a ticket with a patch for portaudio and my improvements.
> This includes the improvements I had discussed previously during my call for
> input on the mailing list and is available at:
> http://jira.freeswitch.org/browse/FS-3006
>
> It was only tested in windows, however most of the changes should not
> effect default behavior (but please test if you can).
>
> To test you will want to enable some things in the portaudio config
> including:
>                     <param name="live-stream-switch" value="true" />
>                     <param name="no-auto-resume-call" value="true" />
>                     <param name="no-ring-during-call" value="true" />
>
> you will then be able to use the new features fully.
>
> Many of these changes were made to add better support for softphone's using
> freeswitch, I will be releasing my embedded freeswitch phone (FSClient) for
> the conference call in a week and a half  it is .net/c# but WPF so sadly
> windows only currently.  jlink and drk are helping so we should end up with
> a nice installer also for it.     If you would like to test it please let me
> know.
>
> ~Mitch
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Fri Jan 21 20:23:57 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 21 Jan 2011 14:23:57 -0300
Subject: [Freeswitch-users] Portaudio Improvements
In-Reply-To: <AANLkTim9AmtEGZoK0NyG1yKVsmJ_MSwXAppi-6AcCqnz@mail.gmail.com>
References: <AANLkTim9AmtEGZoK0NyG1yKVsmJ_MSwXAppi-6AcCqnz@mail.gmail.com>
Message-ID: <AANLkTim0=S4kD3YGoLOJWQpfSD60CCTh=6utXFOfQ4EE@mail.gmail.com>

Sorry, Gmail went daft on me.

Jo?o Mesquita


On Fri, Jan 21, 2011 at 2:00 PM, Mitch Capper <mitch.capper at gmail.com>wrote:

> I have submitted a ticket with a patch for portaudio and my improvements.
> This includes the improvements I had discussed previously during my call for
> input on the mailing list and is available at:
> http://jira.freeswitch.org/browse/FS-3006
>
> It was only tested in windows, however most of the changes should not
> effect default behavior (but please test if you can).
>
> To test you will want to enable some things in the portaudio config
> including:
>                     <param name="live-stream-switch" value="true" />
>                     <param name="no-auto-resume-call" value="true" />
>                     <param name="no-ring-during-call" value="true" />
>
> you will then be able to use the new features fully.
>
> Many of these changes were made to add better support for softphone's using
> freeswitch, I will be releasing my embedded freeswitch phone (FSClient) for
> the conference call in a week and a half  it is .net/c# but WPF so sadly
> windows only currently.  jlink and drk are helping so we should end up with
> a nice installer also for it.     If you would like to test it please let me
> know.
>
> ~Mitch
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From george.niculae79 at gmail.com  Fri Jan 21 20:45:38 2011
From: george.niculae79 at gmail.com (George Niculae)
Date: Fri, 21 Jan 2011 19:45:38 +0200
Subject: [Freeswitch-users] Send email on successful fax sending.
In-Reply-To: <AANLkTi=fWOPE8h+cadpK96FvV0mZ--qKVEhi-jQR5Aw8@mail.gmail.com>
References: <AANLkTinJ3N+zwg+ZK2RmdAZECEkLgcvOSKT0hf_mbpix@mail.gmail.com>
	<AANLkTi=5g-rOZtS9Dy4wq_J6OQU2roLzBDwhzhsuhQSG@mail.gmail.com>
	<AANLkTi=fWOPE8h+cadpK96FvV0mZ--qKVEhi-jQR5Aw8@mail.gmail.com>
Message-ID: <AANLkTik5dEFXM0wuvm9SkhmhFzuj-_5gheShv4=b=+8A@mail.gmail.com>

Could be off topic here but we are using linger for receiving all the events
and sending email on fax -
http://wiki.freeswitch.org/wiki/Event_socket_outbound#Events

<http://wiki.freeswitch.org/wiki/Event_socket_outbound#Events>George

2011/1/21 Kristian Kielhofner <kris at kriskinc.com>

> You can use api_hangup_hook to call a lua script to do this right now:
>
> http://wiki.freeswitch.org/wiki/Mod_lua#Special_Case:_env_object
>
> On Fri, Jan 21, 2011 at 3:14 AM, Marc de Corny <marcdecorny at gmail.com>
> wrote:
> > Hi There,
> >
> > Has anybody had any ideas on this ? I imagine you must all have the same
> > requirement in the Email to Fax scenario ?
> >
> > Very grateful for any pointers
> > thanks
> > Marc
> >
> > On Thu, Dec 30, 2010 at 3:29 PM, Marc de Corny <marcdecorny at gmail.com>
> > wrote:
> >>
> >> Hi all,
> >>
> >> I have got all the inbound fax working and can get FS to send outbound
> fax
> >> from the shell by using the commands :
> >> /opt/freeswitch/bin/fs_cli \
> >> --execute="originate {fax_verbose=true}$DEST &txfax($TMPFAX)" as in the
> >> wiki
> >>
> >> However I'm looking for a way of notifying the sender on the success or
> >> failure of the fax emission. Is there a way of getting a result back
> from
> >> that command like fax_success 0/1 that will allow me then to send the
> >> relevant emails out ?
> >>
> >> Any help is much appreciated.
> >>
> >> thanks
> >> Marc
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Kristian Kielhofner
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.ponzone at ipeva.fr  Fri Jan 21 20:51:10 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 21 Jan 2011 18:51:10 +0100
Subject: [Freeswitch-users] Transfer to extension
In-Reply-To: <402127.1648.qm@web114701.mail.gq1.yahoo.com>
References: <402127.1648.qm@web114701.mail.gq1.yahoo.com>
Message-ID: <04789156-0B38-44D5-B539-3357D00720B4@ipeva.fr>

Please read about Freeswitch first, the wiki or the book, as you got it wrong.
Your extension should go in default.xml if you call from a registered phone with the default conf.
The code I gave you is already in default.conf.

In ivr.conf.xml, it's where you define your IVRs prompts (demo_ivr in this example).

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 21/01/2011 ? 09:29, Kenan BEKTAS a ?crit :

> Hi David,
> 
> Thanks for your prompt reply. Really appreciate that. Yes, I have checked those conf files and tried many things but does not seem to do what I want.
> 
> I tried the following code in "conf/autoload_configs/ivr.conf.xml" and "conf/dialplan/public.xml" one at a time but no success, yet.
> 
> <extension name="ivr_demo">
>       <condition field="destination_number" expression="23">
>         <action application="answer"/>
>         <action application="sleep" data="2000"/>
>         <action application="ivr" data="demo_ivr"/>
>       </condition>
>     </extension>
> 
> All I need is that all the calls to 23 to be responded by IVR no matter what happens. Kind a call forwarding.
> 
> Thanks a bunch,
> 
> ---
> Kenan
> Toronto, Canada
> 
> 
> 
> --- On Fri, 1/21/11, David Ponzone <david.ponzone at ipeva.fr> wrote:
> 
> From: David Ponzone <david.ponzone at ipeva.fr>
> Subject: Re: [Freeswitch-users] Transfer to extension
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Friday, January 21, 2011, 3:16 AM
> 
> Kenan,
> 
> did you take some time to check the default conf or the book ?
> 
> What you ask is fairly easy.
> Here is from the default conf:
> 
>     <extension name="ivr_demo">
>       <condition field="destination_number" expression="^5000$">
>         <action application="answer"/>
>         <action application="sleep" data="2000"/>
>         <action application="ivr" data="demo_ivr"/>
>       </condition>
>     </extension>
> 
> Check conf/autoload_configs/ivr.conf.xml for the IVR configuration.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 21/01/2011 ? 08:32, Kenan BEKTAS a ?crit :
> 
>> Folks,
>> 
>> Have an issue here. I need to have an extension to be answered by/like IVR., i.e, if the extension is rung, then, the IVR should pick up.
>> Calling 23 from 45, and IVR should take the call to 23.
>> 
>> Could anybody provide me some pointers or dialplan examples, please?
>> 
>> Thanks a bunch,
>> 
>> -Kenan
>> 
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> -----Inline Attachment Follows-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From rupa at rupa.com  Fri Jan 21 21:07:39 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 21 Jan 2011 12:07:39 -0600
Subject: [Freeswitch-users] conference list
In-Reply-To: <021F1447C64046159BEE423C61E947A0@e1705>
References: <DBCC446AC5A943F1AEB555BF1707ED6C@e1705>
	<AANLkTi=zp_5gMY51Puc-4bpmJ0duZtAZnYM9Vw0t_xF1@mail.gmail.com>
	<021F1447C64046159BEE423C61E947A0@e1705>
Message-ID: <AANLkTinYCtPT8wzgGXaV8GUScGwCdRv9O4OWJNc7yi+=@mail.gmail.com>

does the xml list have what you want?

On Fri, Jan 21, 2011 at 12:13 AM, Madovsky <infos at madovsky.org> wrote:
> sorry when I say csv it's the csv like list when
> you type api conference list
> but there is only flags as speak|hear|floor,
> the flags-member is not listed
>
> ----- Original Message -----
> From: "Rupa Schomaker" <rupa at rupa.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Friday, January 21, 2011 1:08 AM
> Subject: Re: [Freeswitch-users] conference list
>
>
>> Dunno about csv, but xml_list has a flags node for flags for each member.
>>
>> On Thu, Jan 20, 2011 at 8:50 PM, Madovsky <infos at madovsky.org> wrote:
>>> Sorry Brian, it's my last question before next week ;)
>>>
>>> it's more a suggestion that a question,
>>> when use api conference myconf list command
>>> maybe it would be useful to know in the csv list
>>> which channel is moderator etc...
>>>
>>> Thanks
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa


From mitch.capper at gmail.com  Fri Jan 21 21:22:09 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Fri, 21 Jan 2011 10:22:09 -0800
Subject: [Freeswitch-users] Portaudio Improvements
In-Reply-To: <AANLkTimm69f6adgkgDz03GUOq8xJw_oaVvrkAGxd1yjG@mail.gmail.com>
References: <AANLkTim9AmtEGZoK0NyG1yKVsmJ_MSwXAppi-6AcCqnz@mail.gmail.com>
	<AANLkTimm69f6adgkgDz03GUOq8xJw_oaVvrkAGxd1yjG@mail.gmail.com>
Message-ID: <AANLkTikk==DKhXft67HWcj0ytya030bS=4vtt2o7+tD8@mail.gmail.com>

I did look into AEC a bit, although one of the best options for me looked
like the DirectX AEC support, but obviously this would be windows only.
There is oslec for linux which tries to do AEC at a driver level and is in
the kernel.   The speex AEC processing does not look very complex (
http://www.speex.org/docs/manual/speex-manual/node7.html using
speex_echo_playback/speex_echo_capture).   Of course easier said then done,
and does require passing it the audio in the structure it expects.  One of
my main concerns is I do not also know about how good the speex AEC
processing is for it to be worthwhile and would be curious if it has been
found to give good results in voip settings (as I only saw to the contrary).

Finally, it probably hasn't been too high on my list due to the fact that I
have been using my client with a headset so the need is not really there.

~Mitch

2011/1/21 Jo?o Mesquita <jmesquita at freeswitch.org>

> Mitch, I will work on continuing FSComm for a thick client multi-platform
> solution but one of the big show stoppers for me was the lack of AEC on
> mod_pa.
>
> I tried the preprocessors embedded on the core by Tony that are using the
> speexdsp but I got no luck. The main tests I made were on a mac using the
> speakerphone. Do you have any experience with this type of technology? It
> seems to me that the only AEC available is the one implemented on speexdsp
> but I am completely new to this.
>
> Regards,
> Jo?o Mesquita
>
>
>
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From infos at madovsky.org  Fri Jan 21 21:27:06 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 21 Jan 2011 13:27:06 -0500
Subject: [Freeswitch-users] conference list
References: <DBCC446AC5A943F1AEB555BF1707ED6C@e1705><AANLkTi=zp_5gMY51Puc-4bpmJ0duZtAZnYM9Vw0t_xF1@mail.gmail.com><021F1447C64046159BEE423C61E947A0@e1705>
	<AANLkTinYCtPT8wzgGXaV8GUScGwCdRv9O4OWJNc7yi+=@mail.gmail.com>
Message-ID: <4FE385B18ADC4C97B67EC94E8CEA278B@e1705>

Rupa,

sorry I don't understand how to use xml_list.
it isn't exist as api and as dialplan application

Thanks



----- Original Message ----- 
From: "Rupa Schomaker" <rupa at rupa.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, January 21, 2011 1:07 PM
Subject: Re: [Freeswitch-users] conference list


> does the xml list have what you want?
>
> On Fri, Jan 21, 2011 at 12:13 AM, Madovsky <infos at madovsky.org> wrote:
>> sorry when I say csv it's the csv like list when
>> you type api conference list
>> but there is only flags as speak|hear|floor,
>> the flags-member is not listed
>>
>> ----- Original Message -----
>> From: "Rupa Schomaker" <rupa at rupa.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Friday, January 21, 2011 1:08 AM
>> Subject: Re: [Freeswitch-users] conference list
>>
>>
>>> Dunno about csv, but xml_list has a flags node for flags for each 
>>> member.
>>>
>>> On Thu, Jan 20, 2011 at 8:50 PM, Madovsky <infos at madovsky.org> wrote:
>>>> Sorry Brian, it's my last question before next week ;)
>>>>
>>>> it's more a suggestion that a question,
>>>> when use api conference myconf list command
>>>> maybe it would be useful to know in the csv list
>>>> which channel is moderator etc...
>>>>
>>>> Thanks
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From rupa at rupa.com  Fri Jan 21 21:38:48 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 21 Jan 2011 12:38:48 -0600
Subject: [Freeswitch-users] conference list
In-Reply-To: <4FE385B18ADC4C97B67EC94E8CEA278B@e1705>
References: <DBCC446AC5A943F1AEB555BF1707ED6C@e1705>
	<AANLkTi=zp_5gMY51Puc-4bpmJ0duZtAZnYM9Vw0t_xF1@mail.gmail.com>
	<021F1447C64046159BEE423C61E947A0@e1705>
	<AANLkTinYCtPT8wzgGXaV8GUScGwCdRv9O4OWJNc7yi+=@mail.gmail.com>
	<4FE385B18ADC4C97B67EC94E8CEA278B@e1705>
Message-ID: <AANLkTi=qmqiTUrkiqzhFqz2gh568Dq=YscEmXguZgaQh@mail.gmail.com>

conference,             <confname> list [delim <string>]
---->                <confname> xml_list

use xml_list instead of list

On Fri, Jan 21, 2011 at 12:27 PM, Madovsky <infos at madovsky.org> wrote:
> Rupa,
>
> sorry I don't understand how to use xml_list.
> it isn't exist as api and as dialplan application
>
> Thanks
>
>
>
> ----- Original Message -----
> From: "Rupa Schomaker" <rupa at rupa.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Friday, January 21, 2011 1:07 PM
> Subject: Re: [Freeswitch-users] conference list
>
>
>> does the xml list have what you want?
>>
>> On Fri, Jan 21, 2011 at 12:13 AM, Madovsky <infos at madovsky.org> wrote:
>>> sorry when I say csv it's the csv like list when
>>> you type api conference list
>>> but there is only flags as speak|hear|floor,
>>> the flags-member is not listed
>>>
>>> ----- Original Message -----
>>> From: "Rupa Schomaker" <rupa at rupa.com>
>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>> Sent: Friday, January 21, 2011 1:08 AM
>>> Subject: Re: [Freeswitch-users] conference list
>>>
>>>
>>>> Dunno about csv, but xml_list has a flags node for flags for each
>>>> member.
>>>>
>>>> On Thu, Jan 20, 2011 at 8:50 PM, Madovsky <infos at madovsky.org> wrote:
>>>>> Sorry Brian, it's my last question before next week ;)
>>>>>
>>>>> it's more a suggestion that a question,
>>>>> when use api conference myconf list command
>>>>> maybe it would be useful to know in the csv list
>>>>> which channel is moderator etc...
>>>>>
>>>>> Thanks
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa


From infos at madovsky.org  Fri Jan 21 21:59:30 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 21 Jan 2011 13:59:30 -0500
Subject: [Freeswitch-users] conference list
References: <DBCC446AC5A943F1AEB555BF1707ED6C@e1705><AANLkTi=zp_5gMY51Puc-4bpmJ0duZtAZnYM9Vw0t_xF1@mail.gmail.com><021F1447C64046159BEE423C61E947A0@e1705><AANLkTinYCtPT8wzgGXaV8GUScGwCdRv9O4OWJNc7yi+=@mail.gmail.com><4FE385B18ADC4C97B67EC94E8CEA278B@e1705>
	<AANLkTi=qmqiTUrkiqzhFqz2gh568Dq=YscEmXguZgaQh@mail.gmail.com>
Message-ID: <A4B3B1FD7AFE4809A32A69A49F37943E@e1705>

Ah ok right,
but if I call conference conf_name list xml_list
from dialplan I haave to parse all the xml code to get the var is_moderator 
isn't it ?

Thanks

----- Original Message ----- 
From: "Rupa Schomaker" <rupa at rupa.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, January 21, 2011 1:38 PM
Subject: Re: [Freeswitch-users] conference list


> conference,             <confname> list [delim <string>]
> ---->                <confname> xml_list
>
> use xml_list instead of list
>
> On Fri, Jan 21, 2011 at 12:27 PM, Madovsky <infos at madovsky.org> wrote:
>> Rupa,
>>
>> sorry I don't understand how to use xml_list.
>> it isn't exist as api and as dialplan application
>>
>> Thanks
>>
>>
>>
>> ----- Original Message -----
>> From: "Rupa Schomaker" <rupa at rupa.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Friday, January 21, 2011 1:07 PM
>> Subject: Re: [Freeswitch-users] conference list
>>
>>
>>> does the xml list have what you want?
>>>
>>> On Fri, Jan 21, 2011 at 12:13 AM, Madovsky <infos at madovsky.org> wrote:
>>>> sorry when I say csv it's the csv like list when
>>>> you type api conference list
>>>> but there is only flags as speak|hear|floor,
>>>> the flags-member is not listed
>>>>
>>>> ----- Original Message -----
>>>> From: "Rupa Schomaker" <rupa at rupa.com>
>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Friday, January 21, 2011 1:08 AM
>>>> Subject: Re: [Freeswitch-users] conference list
>>>>
>>>>
>>>>> Dunno about csv, but xml_list has a flags node for flags for each
>>>>> member.
>>>>>
>>>>> On Thu, Jan 20, 2011 at 8:50 PM, Madovsky <infos at madovsky.org> wrote:
>>>>>> Sorry Brian, it's my last question before next week ;)
>>>>>>
>>>>>> it's more a suggestion that a question,
>>>>>> when use api conference myconf list command
>>>>>> maybe it would be useful to know in the csv list
>>>>>> which channel is moderator etc...
>>>>>>
>>>>>> Thanks
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> -Rupa
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From anthony.minessale at gmail.com  Fri Jan 21 22:13:21 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 21 Jan 2011 13:13:21 -0600
Subject: [Freeswitch-users] Question aboutCPU usage.
In-Reply-To: <57f59d718411e237cd08c17c7d93d03c@mail.gmail.com>
References: <Acu4Dj/SgL9TknUcQLyhSmvOooZKqgAzlclAACgGYUA=>
	<57f59d718411e237cd08c17c7d93d03c@mail.gmail.com>
Message-ID: <AANLkTimK8wXJ3LMntpGZ2xhfqt80XVLM+jRrAkCM0Q_X@mail.gmail.com>

Yes, G729 is a CPU hog, especially the illegal one you are testing with.
The commercial one that you can purchase from
http://www.freeswitch.org probably will do slightly better but you
will always get much less milage out of cpu intensive codecs.


On Fri, Jan 21, 2011 at 9:06 AM, Ricardo Martinez
<rmartinez at redvoiss.net> wrote:
> Hello.
>
> I have the opportunity to use an even more powerful server. ?I have just
> installed CentOS 5.5 64 bit, and compile freeswitch with 64 bits.
>
> This is the output with 100 simm calls using transcoding from G711 to G729
>
>
>
> top - 16:57:56 up? 6:04,? 2 users,? load average: 11.60, 12.49, 12.58
>
> Tasks: 168 total,?? 1 running, 167 sleeping,?? 0 stopped,?? 0 zombie
>
> Cpu0? : 35.4%us,? 0.6%sy,? 0.0%ni, 63.6%id,? 0.3%wa,? 0.0%hi,? 0.1%si,
> 0.0%st
>
> Cpu1? : 34.9%us,? 0.6%sy,? 0.0%ni, 58.8%id,? 4.9%wa,? 0.1%hi,? 0.7%si,
> 0.0%st
>
> Cpu2? : 34.9%us,? 0.6%sy,? 0.0%ni, 59.6%id,? 4.1%wa,? 0.1%hi,? 0.7%si,
> 0.0%st
>
> Cpu3? : 35.0%us,? 0.7%sy,? 0.0%ni, 62.0%id,? 1.8%wa,? 0.2%hi,? 0.3%si,
> 0.0%st
>
> Cpu4? : 35.8%us,? 1.3%sy,? 0.0%ni, 62.5%id,? 0.2%wa,? 0.0%hi,? 0.1%si,
> 0.0%st
>
> Cpu5? : 35.6%us,? 1.6%sy,? 0.0%ni, 60.1%id,? 1.8%wa,? 0.1%hi,? 0.8%si,
> 0.0%st
>
> Cpu6? : 35.8%us,? 1.9%sy,? 0.0%ni, 59.9%id,? 1.5%wa,? 0.1%hi,? 0.8%si,
> 0.0%st
>
> Cpu7? : 36.5%us,? 2.0%sy,? 0.0%ni, 60.6%id,? 0.7%wa,? 0.0%hi,? 0.2%si,
> ?0.0%st
>
> Mem:?? 8165808k total,? 1208292k used,? 6957516k free,?? 150228k buffers
>
> Swap: 10223608k total,??????? 0k used, 10223608k free,?? 772980k cached
>
>
>
> ? PID USER????? PR? NI? VIRT? RES? SHR S %CPU %MEM??? TIME+
> COMMAND
>
> ?5265 root????? 18?? 0? 541m? 79m 5796 S 469.5? 1.0?? 1077:32
> freeswitch
>
> ????1 root????? 15?? 0 10348? 684? 572 S? 0.0? 0.0?? 0:01.24
> init
>
> ????2 root????? RT? -5???? 0??? 0??? 0 S? 0.0? 0.0?? 0:00.00
> migration/0
>
> ????3 root????? 34? 19???? 0??? 0??? 0 S? 0.0? 0.0?? 0:00.00
> ksoftirqd/0
>
> ????4 root????? RT? -5???? 0??? 0??? 0 S? 0.0? 0.0?? 0:00.00
> watchdog/0
>
> ????5 root????? RT? -5???? 0??? 0??? 0 S? 0.0? 0.0?? 0:00.00
> migration/1
>
> ????6 root????? 34? 19???? 0??? 0??? 0 S? 0.0? 0.0?? 0:00
>
>
>
> Is normal the load average?
>
>
>
> This is the output with uname ?a
>
>
>
> [root at siptrcrv2 snmp]# uname -a
>
> Linux siptrcrv2 2.6.18-194.el5 #1 SMP Fri Apr 2 14:58:14 EDT 2010 x86_64
> x86_64 x86_64 GNU/Linux
>
>
>
> Regards,
>
> Ricardo.-
>
>
>
>
>
>
>
> De: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de Brian
> West
> Enviado el: mi?rcoles, 19 de enero de 2011 16:22
> Para: FreeSWITCH Users Help
>
> Asunto: Re: [Freeswitch-users] Question aboutCPU usage.
>
>
>
> Chances are if you would install a 64bit OS on that there NICE 64bit CPU it
> would work much better.
>
>
>
> /b
>
>
>
> On Jan 19, 2011, at 12:43 PM, Ricardo Martinez wrote:
>
>
>
> Hello.
>
>
>
> [root at ser-ng bin]# uname -a
>
> Linux 2.6.33.5-124.fc13.i686.PAE #1 SMP Fri Jun 11 09:42:24 UTC 2010 i686
> i686 i386 GNU/Linux
>
>
>
> Ricardo.-
>
>
>
> De:?freeswitch-users-bounces at lists.freeswitch.org?[mailto:freeswitch-users-bounces at lists.freeswitch.org]?En
> nombre de?Brian West
> Enviado el:?mi?rcoles, 19 de enero de 2011 15:41
> Para:?FreeSWITCH Users Help
> Asunto:?Re: [Freeswitch-users] Question aboutCPU usage.
>
>
>
> Depend what does uname -a say?
>
>
>
> /b
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From steveayre at gmail.com  Fri Jan 21 22:30:10 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 21 Jan 2011 19:30:10 +0000
Subject: [Freeswitch-users] Send email on successful fax sending.
In-Reply-To: <AANLkTik5dEFXM0wuvm9SkhmhFzuj-_5gheShv4=b=+8A@mail.gmail.com>
References: <AANLkTinJ3N+zwg+ZK2RmdAZECEkLgcvOSKT0hf_mbpix@mail.gmail.com>
	<AANLkTi=5g-rOZtS9Dy4wq_J6OQU2roLzBDwhzhsuhQSG@mail.gmail.com>
	<AANLkTi=fWOPE8h+cadpK96FvV0mZ--qKVEhi-jQR5Aw8@mail.gmail.com>
	<AANLkTik5dEFXM0wuvm9SkhmhFzuj-_5gheShv4=b=+8A@mail.gmail.com>
Message-ID: <AANLkTi==eyaCiRMmsUueOQOCVARqfQ+6t8GRrrkyz5oD@mail.gmail.com>

That's only for outbound sockets though, isn't it?

Looks like you're talking about sending email on received fax, while he's
looking for email confirmation on transmitted fax.

-Steve


On 21 January 2011 17:45, George Niculae <george.niculae79 at gmail.com> wrote:

> Could be off topic here but we are using linger for receiving all the
> events and sending email on fax -
> http://wiki.freeswitch.org/wiki/Event_socket_outbound#Events
>
> <http://wiki.freeswitch.org/wiki/Event_socket_outbound#Events>George
>
> 2011/1/21 Kristian Kielhofner <kris at kriskinc.com>
>
> You can use api_hangup_hook to call a lua script to do this right now:
>>
>> http://wiki.freeswitch.org/wiki/Mod_lua#Special_Case:_env_object
>>
>> On Fri, Jan 21, 2011 at 3:14 AM, Marc de Corny <marcdecorny at gmail.com>
>> wrote:
>> > Hi There,
>> >
>> > Has anybody had any ideas on this ? I imagine you must all have the same
>> > requirement in the Email to Fax scenario ?
>> >
>> > Very grateful for any pointers
>> > thanks
>> > Marc
>> >
>> > On Thu, Dec 30, 2010 at 3:29 PM, Marc de Corny <marcdecorny at gmail.com>
>> > wrote:
>> >>
>> >> Hi all,
>> >>
>> >> I have got all the inbound fax working and can get FS to send outbound
>> fax
>> >> from the shell by using the commands :
>> >> /opt/freeswitch/bin/fs_cli \
>> >> --execute="originate {fax_verbose=true}$DEST &txfax($TMPFAX)" as in the
>> >> wiki
>> >>
>> >> However I'm looking for a way of notifying the sender on the success or
>> >> failure of the fax emission. Is there a way of getting a result back
>> from
>> >> that command like fax_success 0/1 that will allow me then to send the
>> >> relevant emails out ?
>> >>
>> >> Any help is much appreciated.
>> >>
>> >> thanks
>> >> Marc
>> >>
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Kristian Kielhofner
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Fri Jan 21 22:35:21 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 21 Jan 2011 19:35:21 +0000
Subject: [Freeswitch-users] Question aboutCPU usage.
In-Reply-To: <AANLkTimK8wXJ3LMntpGZ2xhfqt80XVLM+jRrAkCM0Q_X@mail.gmail.com>
References: <57f59d718411e237cd08c17c7d93d03c@mail.gmail.com>
	<AANLkTimK8wXJ3LMntpGZ2xhfqt80XVLM+jRrAkCM0Q_X@mail.gmail.com>
Message-ID: <AANLkTim78EOCHuFrYzhc1Z8U37yfQffN_0p1tgC9tvgy@mail.gmail.com>

Ricardo,


"I also installed the codec g729 for transcoding from g711 to g729  (for
testing purposes only)."
We're assuming you're using deepwalker's fs_itu_g729 codec?

If you're planning to transcode G729 it's always better to test with the one
you're actually going to be using. You can get completely different results
since they're different code bases.

fs_itu_g729 will use more CPU and memory than mod_com_g729 so you'll get
different load testing results.
It also has no concept of licenses, so there's no way of testing that you're
handling the conditions where mod_com_g729 has reached the license limit.

-Steve



On 21 January 2011 19:13, Anthony Minessale <anthony.minessale at gmail.com>wrote:

> Yes, G729 is a CPU hog, especially the illegal one you are testing with.
> The commercial one that you can purchase from
> http://www.freeswitch.org probably will do slightly better but you
> will always get much less milage out of cpu intensive codecs.
>
>
> On Fri, Jan 21, 2011 at 9:06 AM, Ricardo Martinez
> <rmartinez at redvoiss.net> wrote:
> > Hello.
> >
> > I have the opportunity to use an even more powerful server.  I have just
> > installed CentOS 5.5 64 bit, and compile freeswitch with 64 bits.
> >
> > This is the output with 100 simm calls using transcoding from G711 to
> G729
> >
> >
> >
> > top - 16:57:56 up  6:04,  2 users,  load average: 11.60, 12.49, 12.58
> >
> > Tasks: 168 total,   1 running, 167 sleeping,   0 stopped,   0 zombie
> >
> > Cpu0  : 35.4%us,  0.6%sy,  0.0%ni, 63.6%id,  0.3%wa,  0.0%hi,  0.1%si,
> > 0.0%st
> >
> > Cpu1  : 34.9%us,  0.6%sy,  0.0%ni, 58.8%id,  4.9%wa,  0.1%hi,  0.7%si,
> > 0.0%st
> >
> > Cpu2  : 34.9%us,  0.6%sy,  0.0%ni, 59.6%id,  4.1%wa,  0.1%hi,  0.7%si,
> > 0.0%st
> >
> > Cpu3  : 35.0%us,  0.7%sy,  0.0%ni, 62.0%id,  1.8%wa,  0.2%hi,  0.3%si,
> > 0.0%st
> >
> > Cpu4  : 35.8%us,  1.3%sy,  0.0%ni, 62.5%id,  0.2%wa,  0.0%hi,  0.1%si,
> > 0.0%st
> >
> > Cpu5  : 35.6%us,  1.6%sy,  0.0%ni, 60.1%id,  1.8%wa,  0.1%hi,  0.8%si,
> > 0.0%st
> >
> > Cpu6  : 35.8%us,  1.9%sy,  0.0%ni, 59.9%id,  1.5%wa,  0.1%hi,  0.8%si,
> > 0.0%st
> >
> > Cpu7  : 36.5%us,  2.0%sy,  0.0%ni, 60.6%id,  0.7%wa,  0.0%hi,  0.2%si,
> >  0.0%st
> >
> > Mem:   8165808k total,  1208292k used,  6957516k free,   150228k buffers
> >
> > Swap: 10223608k total,        0k used, 10223608k free,   772980k cached
> >
> >
> >
> >   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
> > COMMAND
> >
> >  5265 root      18   0  541m  79m 5796 S 469.5  1.0   1077:32
> > freeswitch
> >
> >     1 root      15   0 10348  684  572 S  0.0  0.0   0:01.24
> > init
> >
> >     2 root      RT  -5     0    0    0 S  0.0  0.0   0:00.00
> > migration/0
> >
> >     3 root      34  19     0    0    0 S  0.0  0.0   0:00.00
> > ksoftirqd/0
> >
> >     4 root      RT  -5     0    0    0 S  0.0  0.0   0:00.00
> > watchdog/0
> >
> >     5 root      RT  -5     0    0    0 S  0.0  0.0   0:00.00
> > migration/1
> >
> >     6 root      34  19     0    0    0 S  0.0  0.0   0:00
> >
> >
> >
> > Is normal the load average?
> >
> >
> >
> > This is the output with uname ?a
> >
> >
> >
> > [root at siptrcrv2 snmp]# uname -a
> >
> > Linux siptrcrv2 2.6.18-194.el5 #1 SMP Fri Apr 2 14:58:14 EDT 2010 x86_64
> > x86_64 x86_64 GNU/Linux
> >
> >
> >
> > Regards,
> >
> > Ricardo.-
> >
> >
> >
> >
> >
> >
> >
> > De: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de
> Brian
> > West
> > Enviado el: mi?rcoles, 19 de enero de 2011 16:22
> > Para: FreeSWITCH Users Help
> >
> > Asunto: Re: [Freeswitch-users] Question aboutCPU usage.
> >
> >
> >
> > Chances are if you would install a 64bit OS on that there NICE 64bit CPU
> it
> > would work much better.
> >
> >
> >
> > /b
> >
> >
> >
> > On Jan 19, 2011, at 12:43 PM, Ricardo Martinez wrote:
> >
> >
> >
> > Hello.
> >
> >
> >
> > [root at ser-ng bin]# uname -a
> >
> > Linux 2.6.33.5-124.fc13.i686.PAE #1 SMP Fri Jun 11 09:42:24 UTC 2010 i686
> > i686 i386 GNU/Linux
> >
> >
> >
> > Ricardo.-
> >
> >
> >
> > De: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] En
> > nombre de Brian West
> > Enviado el: mi?rcoles, 19 de enero de 2011 15:41
> > Para: FreeSWITCH Users Help
> > Asunto: Re: [Freeswitch-users] Question aboutCPU usage.
> >
> >
> >
> > Depend what does uname -a say?
> >
> >
> >
> > /b
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Fri Jan 21 22:37:29 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 21 Jan 2011 19:37:29 +0000
Subject: [Freeswitch-users] Transfer to extension
In-Reply-To: <402127.1648.qm@web114701.mail.gq1.yahoo.com>
References: <04FAB8FB-DE6B-4F74-8AA1-BAE660514F73@ipeva.fr>
	<402127.1648.qm@web114701.mail.gq1.yahoo.com>
Message-ID: <AANLkTinhuOT1Vrx97vNNt7x7PNt1+Ossb79=Ec5imMwi@mail.gmail.com>

<condition field="destination_number" expression="23">

Expression is a regular expression. That will match any number containing 23
- including 123, 234, 12345 etc. You need ^23$ to match it as the exact
string, ^ matches the start of the string and $ matches the end.

-Steve



On 21 January 2011 08:29, Kenan BEKTAS <bektas at yahoo.com> wrote:

> Hi David,
>
> Thanks for your prompt reply. Really appreciate that. Yes, I have checked
> those conf files and tried many things but does not seem to do what I want.
>
> I tried the following code in "conf/autoload_configs/ivr.conf.xml" and
> "conf/dialplan/public.xml" one at a time but no success, yet.
>
> <extension name="ivr_demo">
>       <condition field="destination_number" expression="23">
>         <action application="answer"/>
>         <action application="sleep" data="2000"/>
>         <action application="ivr" data="demo_ivr"/>
>       </condition>
>     </extension>
>
> All I need is that all the calls to 23 to be responded by IVR no matter
> what happens. Kind a call forwarding.
>
> Thanks a bunch,
>
> ---
> Kenan
> Toronto, Canada
>
>
>
> --- On *Fri, 1/21/11, David Ponzone <david.ponzone at ipeva.fr>* wrote:
>
>
> From: David Ponzone <david.ponzone at ipeva.fr>
> Subject: Re: [Freeswitch-users] Transfer to extension
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Friday, January 21, 2011, 3:16 AM
>
>
> Kenan,
>
> did you take some time to check the default conf or the book ?
>
> What you ask is fairly easy.
> Here is from the default conf:
>
>     <extension name="ivr_demo">
>       <condition field="destination_number" expression="^5000$">
>         <action application="answer"/>
>         <action application="sleep" data="2000"/>
>         <action application="ivr" data="demo_ivr"/>
>       </condition>
>     </extension>
>
> Check conf/autoload_configs/ivr.conf.xml for the IVR configuration.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr<http://mc/compose?to=david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 21/01/2011 ? 08:32, Kenan BEKTAS a ?crit :
>
> Folks,
>
> Have an issue here. I need to have an extension to be answered by/like
> IVR., i.e, if the extension is rung, then, the IVR should pick up.
> Calling 23 from 45, and IVR should take the call to 23.
>
> Could anybody provide me some pointers or dialplan examples, please?
>
> Thanks a bunch,
>
> -Kenan
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> -----Inline Attachment Follows-----
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Fri Jan 21 22:38:42 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 21 Jan 2011 19:38:42 +0000
Subject: [Freeswitch-users] Transfer to extension
In-Reply-To: <402127.1648.qm@web114701.mail.gq1.yahoo.com>
References: <04FAB8FB-DE6B-4F74-8AA1-BAE660514F73@ipeva.fr>
	<402127.1648.qm@web114701.mail.gq1.yahoo.com>
Message-ID: <AANLkTi=cN3vQw3HmSZuLL9R8bKGh=0mx+JfuATBKhaFa@mail.gmail.com>

I tried the following code in "conf/autoload_configs/ivr.conf.xml" and
"conf/dialplan/public.xml" one at a time but no success, yet.

Any dialplan changes need you to do 'reloadxml' for them to take effect.

-Steve



On 21 January 2011 08:29, Kenan BEKTAS <bektas at yahoo.com> wrote:

> Hi David,
>
> Thanks for your prompt reply. Really appreciate that. Yes, I have checked
> those conf files and tried many things but does not seem to do what I want.
>
> I tried the following code in "conf/autoload_configs/ivr.conf.xml" and
> "conf/dialplan/public.xml" one at a time but no success, yet.
>
> <extension name="ivr_demo">
>       <condition field="destination_number" expression="23">
>         <action application="answer"/>
>         <action application="sleep" data="2000"/>
>         <action application="ivr" data="demo_ivr"/>
>       </condition>
>     </extension>
>
> All I need is that all the calls to 23 to be responded by IVR no matter
> what happens. Kind a call forwarding.
>
> Thanks a bunch,
>
> ---
> Kenan
> Toronto, Canada
>
>
>
> --- On *Fri, 1/21/11, David Ponzone <david.ponzone at ipeva.fr>* wrote:
>
>
> From: David Ponzone <david.ponzone at ipeva.fr>
> Subject: Re: [Freeswitch-users] Transfer to extension
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Friday, January 21, 2011, 3:16 AM
>
>
> Kenan,
>
> did you take some time to check the default conf or the book ?
>
> What you ask is fairly easy.
> Here is from the default conf:
>
>     <extension name="ivr_demo">
>       <condition field="destination_number" expression="^5000$">
>         <action application="answer"/>
>         <action application="sleep" data="2000"/>
>         <action application="ivr" data="demo_ivr"/>
>       </condition>
>     </extension>
>
> Check conf/autoload_configs/ivr.conf.xml for the IVR configuration.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr<http://mc/compose?to=david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 21/01/2011 ? 08:32, Kenan BEKTAS a ?crit :
>
> Folks,
>
> Have an issue here. I need to have an extension to be answered by/like
> IVR., i.e, if the extension is rung, then, the IVR should pick up.
> Calling 23 from 45, and IVR should take the call to 23.
>
> Could anybody provide me some pointers or dialplan examples, please?
>
> Thanks a bunch,
>
> -Kenan
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> -----Inline Attachment Follows-----
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Fri Jan 21 23:39:22 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 21 Jan 2011 14:39:22 -0600
Subject: [Freeswitch-users] RFC 2833 DTMF detection - has something
	changed?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57ED00B44B@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C57ED00ADDA@cooper>
	<9CCA8E46-03D7-4F13-B6BD-8AE8D2564006@freeswitch.org>
	<AANLkTi=yhXDV7yE-BRpGuYU8rQwwMZU85VYBd87Cbo-+@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57EC81C52A@cooper>
	<AANLkTimXXzVCFweC6x18yrNCf8_st9a_eRD8mxAmyega@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57ED00B44B@cooper>
Message-ID: <AANLkTi=-d0PnWRdjh7RHJAAaCHqj9DRGGww0+N6_1whB@mail.gmail.com>

the new one will go in today.

as for the other 2 you mentioned:

The first two are applied, the last one needs one bit more of work.



On Fri, Jan 21, 2011 at 4:06 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> Thanks again, I tried the change below but it didn't help - however, it gave me a clue for what's going on :)
>
> The problem seems to be that do_flush() will flush valid DTMF packets, and since do_flush() is called every time a file start/stops playing (because of rtp break is called) it is quite possible to be unlucky and miss DTMF's during this short period of time.
>
> I've created a patch for this in jira FS-3002, it refactors the RFC2833 detection code, and after trying this out for the last 24 hours, the problems seems to be solved. Please review and apply this patch if it seems ok to you (or get back to me if you have any thoughts about the implemenation). More detailed information can be found in the jira case.
>
> While you're at it - I have a few other patches laying around in jira - if you could have a look at those also I would really appreciate it :)
>
> FS-2917, FS-2973 and FS-2971
>
> Thanks,
>
> Peter Olsson
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale
> Skickat: den 19 januari 2011 19:22
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] RFC 2833 DTMF detection - has something changed?
>
> comment line 2993 and see if its better.
> If it works better, what is on the other end of the call?, I hate it already.
>
>
> On Wed, Jan 19, 2011 at 12:12 PM, Peter Olsson
> <peter.olsson at visionutveckling.se> wrote:
>> Yep, I'm on today's git, so I do have the last change/fix that was made for this in switch.rtp.c.
>>
>> /Peter
>> ________________________________________
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Steven Ayre [steveayre at gmail.com]
>> Skickat: den 19 januari 2011 18:33
>> Till: FreeSWITCH Users Help
>> ?mne: Re: [Freeswitch-users] RFC 2833 DTMF detection - has something ? ?changed?
>>
>> Brian, he says he's on Git from today...
>>
>>
>> On 19 January 2011 17:10, Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
>> Update we did have one day where it was messed up.
>>
>> /b
>>
>> On Jan 19, 2011, at 11:02 AM, Peter Olsson wrote:
>>
>> Hi,
>>
>> I have some problems with DTMF detection, which I've never seen earlier versions. I know that there was a bug in the rtp code a few days (or week(s)?) back, but I'm running on latest git (as of today). I'm also running on the same machine as before, and more or less no config changes (except for a few dialplan changes, but nothing that changes DTMF detection).
>>
>> Anyway, it seems to me that the 2833-detection is not as accurate as it was before. When using a couple of months old FS version I rarely missed any DTMF's (I can't remember I ever did..:)), but now it seems to happen once in a while. Also, I decided today to get wireshark up and running, and after 5 DTMF's, FS missed the last one. I looked inside my wireshark dump, and I could clearly see all DTMF packets in there, but FS somehow missed this.
>>
>> Is there some kind of debugging I could enable, for instance DEBUG_2833 directive?
>>
>> What's the best way to move this forward? I can send my wireshark dump if that helps, but I guess you will need some more debugging info from FS as well?
>>
>> Thanks,
>>
>> Peter Olsson
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4d372d0d32761021210236!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From nazim.aghabayov at gmail.com  Fri Jan 21 23:40:06 2011
From: nazim.aghabayov at gmail.com (Nazim Aghabayov)
Date: Sat, 22 Jan 2011 00:40:06 +0400
Subject: [Freeswitch-users] praise of freeSWITCH developers
In-Reply-To: <E4DFCF0A44324A7786C03C1E3229BEEA@e1705>
References: <9FE96265969E4BAEB801002D975CC50F@e1705>	<1295627306.30863.188.camel@luna.tc.commsmundi.com>
	<E4DFCF0A44324A7786C03C1E3229BEEA@e1705>
Message-ID: <4D39EF26.1080706@gmail.com>

I should have been paid 100x$20, but here it goes another $20.
When I'll become filthy reach, I'll be your next platinum sponsor )
Thank you guys! FreeSWITCH rocks.

On 01/21/2011 08:36 PM, Madovsky wrote:
> Yes,
> even $1 can help.
>
> Regards
>
> ----- Original Message -----
> From: "Fran?ois Delawarde"<fdelawarde at wirelessmundi.com>
> To: "FreeSWITCH Users Help"<freeswitch-users at lists.freeswitch.org>
> Sent: Friday, January 21, 2011 11:28 AM
> Subject: Re: [Freeswitch-users] praise of freeSWITCH developers
>
>
>> On Fri, 2011-01-21 at 11:13 -0500, Madovsky wrote:
>>> Donations are welcome and appreciated.
>> You're right, not much left in my bank account but here goes $20.
>>
>> Everyone follow me on this! Together we can make this thread bigger than
>> 2010's "freeswitch CPU usage".
>>
>> Fran?ois.
>>


From Nabble at slickdeals.endjunk.com  Sat Jan 22 00:08:55 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 21 Jan 2011 13:08:55 -0800 (PST)
Subject: [Freeswitch-users] praise of freeSWITCH developers
In-Reply-To: <9FE96265969E4BAEB801002D975CC50F@e1705>
References: <9FE96265969E4BAEB801002D975CC50F@e1705>
Message-ID: <1295644135281-5949332.post@n2.nabble.com>



Madovsky wrote:
> 
> I would like to remember to all people on this
> emailist that all freeSWITCH guys work very hard
> to update/upgrade and make FS more powerful, and
> also to answer to hundreds of user emails every day.
> Donations are welcome and appreciated.
> It helps the humans behind the machine
> who give to you the opportunity to create and use amazing projects 
> and to continue to make this open source project alive and active.
> 
> Regards
Here goes my 2?. Thanks

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/praise-of-freeSWITCH-developers-tp5948283p5949332.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From edpimentl at gmail.com  Sat Jan 22 00:11:48 2011
From: edpimentl at gmail.com (EdPimentl)
Date: Fri, 21 Jan 2011 16:11:48 -0500
Subject: [Freeswitch-users] praise of freeSWITCH developers
In-Reply-To: <1295627306.30863.188.camel@luna.tc.commsmundi.com>
References: <9FE96265969E4BAEB801002D975CC50F@e1705>
	<1295627306.30863.188.camel@luna.tc.commsmundi.com>
Message-ID: <AANLkTin-cNGJEBBjGDmmFSfPdd-B6kcMV_sZyt9H2-Xm@mail.gmail.com>

Will do $20.00 USD

-E
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From phone.bytes at gmail.com  Sat Jan 22 00:40:26 2011
From: phone.bytes at gmail.com (Phone)
Date: Fri, 21 Jan 2011 14:40:26 -0700
Subject: [Freeswitch-users] Caller ID Number not going out on PRI
In-Reply-To: <AANLkTimsZ04PiNVupqwSymVmW6j=khrLin7skAxd4icT@mail.gmail.com>
References: <4D35F532.2080802@gmail.com>
	<AANLkTimsZ04PiNVupqwSymVmW6j=khrLin7skAxd4icT@mail.gmail.com>
Message-ID: <4D39FD4A.3020108@gmail.com>

Problem solved.

Extra $ on variable reference was removed.
Works great again.

Thanks


On 01/20/2011 1:08 PM, Michael Collins wrote:
> I'd recommend pastebin of your relevant dialplan and freetdm configs. 
> It would also help to see a debug console log trace of a call that is 
> displaying the behavior.
> -MC
>
> On Tue, Jan 18, 2011 at 12:16 PM, Phone <phone.bytes at gmail.com 
> <mailto:phone.bytes at gmail.com>> wrote:
>
>     A few upgrades ago, the Caller ID Number on outbound calls stopped
>     working.  Now it always displays the extension number, where before it
>     was using the value specified in the directory entry.  Seems like
>     maybe
>     the variables are not passing from the directory entry to the dial
>     plan.
>
>     Any ideas?
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From peter.olsson at visionutveckling.se  Sat Jan 22 00:47:13 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 21 Jan 2011 22:47:13 +0100
Subject: [Freeswitch-users] RFC 2833 DTMF detection - has
	something	changed?
In-Reply-To: <AANLkTi=-d0PnWRdjh7RHJAAaCHqj9DRGGww0+N6_1whB@mail.gmail.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C57ED00ADDA@cooper>
	<9CCA8E46-03D7-4F13-B6BD-8AE8D2564006@freeswitch.org>
	<AANLkTi=yhXDV7yE-BRpGuYU8rQwwMZU85VYBd87Cbo-+@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57EC81C52A@cooper>
	<AANLkTimXXzVCFweC6x18yrNCf8_st9a_eRD8mxAmyega@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57ED00B44B@cooper>,
	<AANLkTi=-d0PnWRdjh7RHJAAaCHqj9DRGGww0+N6_1whB@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57EC81C53E@cooper>

Tony, thanks. I will have another look at at this one :)

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Anthony Minessale [anthony.minessale at gmail.com]
Skickat: den 21 januari 2011 21:39
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] RFC 2833 DTMF detection - has something    changed?

the new one will go in today.

as for the other 2 you mentioned:

The first two are applied, the last one needs one bit more of work.



On Fri, Jan 21, 2011 at 4:06 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> Thanks again, I tried the change below but it didn't help - however, it gave me a clue for what's going on :)
>
> The problem seems to be that do_flush() will flush valid DTMF packets, and since do_flush() is called every time a file start/stops playing (because of rtp break is called) it is quite possible to be unlucky and miss DTMF's during this short period of time.
>
> I've created a patch for this in jira FS-3002, it refactors the RFC2833 detection code, and after trying this out for the last 24 hours, the problems seems to be solved. Please review and apply this patch if it seems ok to you (or get back to me if you have any thoughts about the implemenation). More detailed information can be found in the jira case.
>
> While you're at it - I have a few other patches laying around in jira - if you could have a look at those also I would really appreciate it :)
>
> FS-2917, FS-2973 and FS-2971
>
> Thanks,
>
> Peter Olsson
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale
> Skickat: den 19 januari 2011 19:22
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] RFC 2833 DTMF detection - has something changed?
>
> comment line 2993 and see if its better.
> If it works better, what is on the other end of the call?, I hate it already.
>
>
> On Wed, Jan 19, 2011 at 12:12 PM, Peter Olsson
> <peter.olsson at visionutveckling.se> wrote:
>> Yep, I'm on today's git, so I do have the last change/fix that was made for this in switch.rtp.c.
>>
>> /Peter
>> ________________________________________
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Steven Ayre [steveayre at gmail.com]
>> Skickat: den 19 januari 2011 18:33
>> Till: FreeSWITCH Users Help
>> ?mne: Re: [Freeswitch-users] RFC 2833 DTMF detection - has something    changed?
>>
>> Brian, he says he's on Git from today...
>>
>>
>> On 19 January 2011 17:10, Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
>> Update we did have one day where it was messed up.
>>
>> /b
>>
>> On Jan 19, 2011, at 11:02 AM, Peter Olsson wrote:
>>
>> Hi,
>>
>> I have some problems with DTMF detection, which I've never seen earlier versions. I know that there was a bug in the rtp code a few days (or week(s)?) back, but I'm running on latest git (as of today). I'm also running on the same machine as before, and more or less no config changes (except for a few dialplan changes, but nothing that changes DTMF detection).
>>
>> Anyway, it seems to me that the 2833-detection is not as accurate as it was before. When using a couple of months old FS version I rarely missed any DTMF's (I can't remember I ever did..:)), but now it seems to happen once in a while. Also, I decided today to get wireshark up and running, and after 5 DTMF's, FS missed the last one. I looked inside my wireshark dump, and I could clearly see all DTMF packets in there, but FS somehow missed this.
>>
>> Is there some kind of debugging I could enable, for instance DEBUG_2833 directive?
>>
>> What's the best way to move this forward? I can send my wireshark dump if that helps, but I guess you will need some more debugging info from FS as well?
>>
>> Thanks,
>>
>> Peter Olsson
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4d39f06632762119121577!



From george at ezuce.com  Fri Jan 21 19:50:22 2011
From: george at ezuce.com (George Niculae)
Date: Fri, 21 Jan 2011 18:50:22 +0200
Subject: [Freeswitch-users] Send email on successful fax sending.
In-Reply-To: <AANLkTi==DN40Ebdi0A0bqX95Ct7Ra_Px+uSX8i4acf0i@mail.gmail.com>
References: <AANLkTinJ3N+zwg+ZK2RmdAZECEkLgcvOSKT0hf_mbpix@mail.gmail.com>
	<AANLkTi=5g-rOZtS9Dy4wq_J6OQU2roLzBDwhzhsuhQSG@mail.gmail.com>
	<4D3976FD.4090805@chaschperli.ch>
	<AANLkTimo0n3UP2c09F3AmgQdkRdU+DVAMxjYrw7cP9gm@mail.gmail.com>
	<AANLkTi=zBgX-trHwVUUgfGNZh3SmcJfru1tD=5-aE=MY@mail.gmail.com>
	<AANLkTimKgZDpuDUgNaqT1=61mqwOoJ66u4m-u4LBjxXM@mail.gmail.com>
	<1295627914599-5948386.post@n2.nabble.com>
	<AANLkTi==DN40Ebdi0A0bqX95Ct7Ra_Px+uSX8i4acf0i@mail.gmail.com>
Message-ID: <AANLkTin6BdJLZ6O2KydDY768U1tOU-xqxSACSFmL94bq@mail.gmail.com>

On Fri, Jan 21, 2011 at 6:44 PM, Steven Ayre <steveayre at gmail.com> wrote:
> That won't work if the channel no longer exists (it accesses the channel's
> object and gets the variable stored there).
>
> You can either use a hangup hook as someone else suggested (runs after
> hangup but before the session is destroyed so the channels are still
> available) or look at my event patch which raises an event containing a
> header with that variable's value.
>
> -Steve

Could be off topic here, but we are using linger for this:
http://wiki.freeswitch.org/wiki/Event_socket_outbound#Events

George


From george at ezuce.com  Fri Jan 21 23:29:05 2011
From: george at ezuce.com (George Niculae)
Date: Fri, 21 Jan 2011 22:29:05 +0200
Subject: [Freeswitch-users] Send email on successful fax sending.
In-Reply-To: <AANLkTi==eyaCiRMmsUueOQOCVARqfQ+6t8GRrrkyz5oD@mail.gmail.com>
References: <AANLkTinJ3N+zwg+ZK2RmdAZECEkLgcvOSKT0hf_mbpix@mail.gmail.com>
	<AANLkTi=5g-rOZtS9Dy4wq_J6OQU2roLzBDwhzhsuhQSG@mail.gmail.com>
	<AANLkTi=fWOPE8h+cadpK96FvV0mZ--qKVEhi-jQR5Aw8@mail.gmail.com>
	<AANLkTik5dEFXM0wuvm9SkhmhFzuj-_5gheShv4=b=+8A@mail.gmail.com>
	<AANLkTi==eyaCiRMmsUueOQOCVARqfQ+6t8GRrrkyz5oD@mail.gmail.com>
Message-ID: <AANLkTin0kc8n9eQHFsWnXbWrhhHMJGoDOcEPankPVAZ-@mail.gmail.com>

Yes, you are right, I really was off topic

George

On Fri, Jan 21, 2011 at 9:30 PM, Steven Ayre <steveayre at gmail.com> wrote:
> That's only for outbound sockets though, isn't it?
>
> Looks like you're talking about sending email on received fax, while he's
> looking for email confirmation on transmitted fax.
>
> -Steve
>
>
> On 21 January 2011 17:45, George Niculae <george.niculae79 at gmail.com> wrote:
>>
>> Could be off topic here but we are using linger for receiving all the
>> events and sending email on fax
>> -?http://wiki.freeswitch.org/wiki/Event_socket_outbound#Events
>> George
>>
>> 2011/1/21 Kristian Kielhofner <kris at kriskinc.com>
>>>
>>> You can use api_hangup_hook to call a lua script to do this right now:
>>>
>>> http://wiki.freeswitch.org/wiki/Mod_lua#Special_Case:_env_object
>>>
>>> On Fri, Jan 21, 2011 at 3:14 AM, Marc de Corny <marcdecorny at gmail.com>
>>> wrote:
>>> > Hi There,
>>> >
>>> > Has anybody had any ideas on this ? I imagine you must all have the
>>> > same
>>> > requirement in the Email to Fax scenario ?
>>> >
>>> > Very grateful for any pointers
>>> > thanks
>>> > Marc
>>> >
>>> > On Thu, Dec 30, 2010 at 3:29 PM, Marc de Corny <marcdecorny at gmail.com>
>>> > wrote:
>>> >>
>>> >> Hi all,
>>> >>
>>> >> I have got all the inbound fax working and can get FS to send outbound
>>> >> fax
>>> >> from the shell by using the commands :
>>> >> /opt/freeswitch/bin/fs_cli \
>>> >> --execute="originate {fax_verbose=true}$DEST &txfax($TMPFAX)" as in
>>> >> the
>>> >> wiki
>>> >>
>>> >> However I'm looking for a way of notifying the sender on the success
>>> >> or
>>> >> failure of the fax emission. Is there a way of getting a result back
>>> >> from
>>> >> that command like fax_success 0/1 that will allow me then to send the
>>> >> relevant emails out ?
>>> >>
>>> >> Any help is much appreciated.
>>> >>
>>> >> thanks
>>> >> Marc
>>> >>
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Kristian Kielhofner
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From curriegrad2004 at gmail.com  Sat Jan 22 01:41:07 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Fri, 21 Jan 2011 14:41:07 -0800
Subject: [Freeswitch-users] Question aboutCPU usage.
In-Reply-To: <AANLkTim78EOCHuFrYzhc1Z8U37yfQffN_0p1tgC9tvgy@mail.gmail.com>
References: <57f59d718411e237cd08c17c7d93d03c@mail.gmail.com>
	<AANLkTimK8wXJ3LMntpGZ2xhfqt80XVLM+jRrAkCM0Q_X@mail.gmail.com>
	<AANLkTim78EOCHuFrYzhc1Z8U37yfQffN_0p1tgC9tvgy@mail.gmail.com>
Message-ID: <AANLkTikHmFAtnuDdm19aE4QW34pZSZ5DwGAwa=RGj5V4@mail.gmail.com>

One solution is to use the GSM codec instead of the g.729 codec.
Although it may not sound as good as the g.729 codec, it uses about
the same amount of bandwidth and less CPU iirc. The GSM codec is
included in FreeSwitch by default.

On Fri, Jan 21, 2011 at 11:35 AM, Steven Ayre <steveayre at gmail.com> wrote:
> Ricardo,
>
> "I also installed the codec g729 for transcoding from g711 to g729? (for
> testing purposes only)."
>
> We're assuming you're using deepwalker's fs_itu_g729 codec?
>
> If you're planning to transcode G729 it's always better to test with the one
> you're actually going to be using. You can get completely different results
> since they're different code bases.
>
> fs_itu_g729 will use more CPU and memory than mod_com_g729 so you'll get
> different load testing results.
> It also has no concept of licenses, so there's no way of testing that you're
> handling the conditions where mod_com_g729 has reached the license limit.
>
> -Steve
>
>
>
> On 21 January 2011 19:13, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
>>
>> Yes, G729 is a CPU hog, especially the illegal one you are testing with.
>> The commercial one that you can purchase from
>> http://www.freeswitch.org probably will do slightly better but you
>> will always get much less milage out of cpu intensive codecs.
>>
>>
>> On Fri, Jan 21, 2011 at 9:06 AM, Ricardo Martinez
>> <rmartinez at redvoiss.net> wrote:
>> > Hello.
>> >
>> > I have the opportunity to use an even more powerful server. ?I have just
>> > installed CentOS 5.5 64 bit, and compile freeswitch with 64 bits.
>> >
>> > This is the output with 100 simm calls using transcoding from G711 to
>> > G729
>> >
>> >
>> >
>> > top - 16:57:56 up? 6:04,? 2 users,? load average: 11.60, 12.49, 12.58
>> >
>> > Tasks: 168 total,?? 1 running, 167 sleeping,?? 0 stopped,?? 0 zombie
>> >
>> > Cpu0? : 35.4%us,? 0.6%sy,? 0.0%ni, 63.6%id,? 0.3%wa,? 0.0%hi,? 0.1%si,
>> > 0.0%st
>> >
>> > Cpu1? : 34.9%us,? 0.6%sy,? 0.0%ni, 58.8%id,? 4.9%wa,? 0.1%hi,? 0.7%si,
>> > 0.0%st
>> >
>> > Cpu2? : 34.9%us,? 0.6%sy,? 0.0%ni, 59.6%id,? 4.1%wa,? 0.1%hi,? 0.7%si,
>> > 0.0%st
>> >
>> > Cpu3? : 35.0%us,? 0.7%sy,? 0.0%ni, 62.0%id,? 1.8%wa,? 0.2%hi,? 0.3%si,
>> > 0.0%st
>> >
>> > Cpu4? : 35.8%us,? 1.3%sy,? 0.0%ni, 62.5%id,? 0.2%wa,? 0.0%hi,? 0.1%si,
>> > 0.0%st
>> >
>> > Cpu5? : 35.6%us,? 1.6%sy,? 0.0%ni, 60.1%id,? 1.8%wa,? 0.1%hi,? 0.8%si,
>> > 0.0%st
>> >
>> > Cpu6? : 35.8%us,? 1.9%sy,? 0.0%ni, 59.9%id,? 1.5%wa,? 0.1%hi,? 0.8%si,
>> > 0.0%st
>> >
>> > Cpu7? : 36.5%us,? 2.0%sy,? 0.0%ni, 60.6%id,? 0.7%wa,? 0.0%hi,? 0.2%si,
>> > ?0.0%st
>> >
>> > Mem:?? 8165808k total,? 1208292k used,? 6957516k free,?? 150228k buffers
>> >
>> > Swap: 10223608k total,??????? 0k used, 10223608k free,?? 772980k cached
>> >
>> >
>> >
>> > ? PID USER????? PR? NI? VIRT? RES? SHR S %CPU %MEM??? TIME+
>> > COMMAND
>> >
>> > ?5265 root????? 18?? 0? 541m? 79m 5796 S 469.5? 1.0?? 1077:32
>> > freeswitch
>> >
>> > ????1 root????? 15?? 0 10348? 684? 572 S? 0.0? 0.0?? 0:01.24
>> > init
>> >
>> > ????2 root????? RT? -5???? 0??? 0??? 0 S? 0.0? 0.0?? 0:00.00
>> > migration/0
>> >
>> > ????3 root????? 34? 19???? 0??? 0??? 0 S? 0.0? 0.0?? 0:00.00
>> > ksoftirqd/0
>> >
>> > ????4 root????? RT? -5???? 0??? 0??? 0 S? 0.0? 0.0?? 0:00.00
>> > watchdog/0
>> >
>> > ????5 root????? RT? -5???? 0??? 0??? 0 S? 0.0? 0.0?? 0:00.00
>> > migration/1
>> >
>> > ????6 root????? 34? 19???? 0??? 0??? 0 S? 0.0? 0.0?? 0:00
>> >
>> >
>> >
>> > Is normal the load average?
>> >
>> >
>> >
>> > This is the output with uname ?a
>> >
>> >
>> >
>> > [root at siptrcrv2 snmp]# uname -a
>> >
>> > Linux siptrcrv2 2.6.18-194.el5 #1 SMP Fri Apr 2 14:58:14 EDT 2010 x86_64
>> > x86_64 x86_64 GNU/Linux
>> >
>> >
>> >
>> > Regards,
>> >
>> > Ricardo.-
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> > De: freeswitch-users-bounces at lists.freeswitch.org
>> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de
>> > Brian
>> > West
>> > Enviado el: mi?rcoles, 19 de enero de 2011 16:22
>> > Para: FreeSWITCH Users Help
>> >
>> > Asunto: Re: [Freeswitch-users] Question aboutCPU usage.
>> >
>> >
>> >
>> > Chances are if you would install a 64bit OS on that there NICE 64bit CPU
>> > it
>> > would work much better.
>> >
>> >
>> >
>> > /b
>> >
>> >
>> >
>> > On Jan 19, 2011, at 12:43 PM, Ricardo Martinez wrote:
>> >
>> >
>> >
>> > Hello.
>> >
>> >
>> >
>> > [root at ser-ng bin]# uname -a
>> >
>> > Linux 2.6.33.5-124.fc13.i686.PAE #1 SMP Fri Jun 11 09:42:24 UTC 2010
>> > i686
>> > i686 i386 GNU/Linux
>> >
>> >
>> >
>> > Ricardo.-
>> >
>> >
>> >
>> >
>> > De:?freeswitch-users-bounces at lists.freeswitch.org?[mailto:freeswitch-users-bounces at lists.freeswitch.org]?En
>> > nombre de?Brian West
>> > Enviado el:?mi?rcoles, 19 de enero de 2011 15:41
>> > Para:?FreeSWITCH Users Help
>> > Asunto:?Re: [Freeswitch-users] Question aboutCPU usage.
>> >
>> >
>> >
>> > Depend what does uname -a say?
>> >
>> >
>> >
>> > /b
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From motosota at gmail.com  Sat Jan 22 01:44:24 2011
From: motosota at gmail.com (Mike)
Date: Fri, 21 Jan 2011 22:44:24 +0000
Subject: [Freeswitch-users] Freeswitch Sends RTP To Private IP Address
Message-ID: <AANLkTimru1i31uSYW+bGK_3z=GiW=QNPzjvTZRRwP5sq@mail.gmail.com>

Scenario. Polycom phone, private IP address, behind Cisco router doing NAT
overload to a public IP address (SIP 'fixup' disabled, so the Cisco isn't
mangling anything).

RTP stream from FreeSWITCH is sent to the private, not public address of the
phone.

I've got this working on 1.0.6  but I've tried this with FreeSWITCH Version
1.0.head (git-7070061 2011-01-20 13-52-00 -0600) and it doesn't seem to work
for me.

Here http://pastebin.freeswitch.org/15109 is my sip profile.

Here http://pastebin.freeswitch.org/15110 is a FreeSWITCH log for a call
made from one of the Polycoms to voicemail with SIP tracing enabled.

And here http://pastebin.freeswitch.org/15111 is the wireshark trace showing
the media stream being sent by FreeSWITCH to the wrong port.

I may be (hopefully) missing something very obvious here.

Mike
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From jmesquita at freeswitch.org  Sat Jan 22 01:52:43 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 21 Jan 2011 19:52:43 -0300
Subject: [Freeswitch-users] praise of freeSWITCH developers
In-Reply-To: <AANLkTin-cNGJEBBjGDmmFSfPdd-B6kcMV_sZyt9H2-Xm@mail.gmail.com>
References: <9FE96265969E4BAEB801002D975CC50F@e1705>
	<1295627306.30863.188.camel@luna.tc.commsmundi.com>
	<AANLkTin-cNGJEBBjGDmmFSfPdd-B6kcMV_sZyt9H2-Xm@mail.gmail.com>
Message-ID: <AANLkTi=nHRE1TqGWtDR=iSnjGcWPwK8x2FeA1nkTGiCo@mail.gmail.com>

I am not sure where I stand, a developer or a donor but I donated as well.
My 20 went in a few seconds ago.

Regards,
Jo?o Mesquita


On Fri, Jan 21, 2011 at 6:11 PM, EdPimentl <edpimentl at gmail.com> wrote:

> Will do $20.00 USD
>
> -E
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From william.nishio at voicetechnology.com.br  Sat Jan 22 01:59:03 2011
From: william.nishio at voicetechnology.com.br (William Kendi ...)
Date: Fri, 21 Jan 2011 20:59:03 -0200
Subject: [Freeswitch-users] mod_fsv, record and playback options
In-Reply-To: <AANLkTink1-sZBh9zO9p8_he8XfaH+NdLHRHwZi5n_BE5@mail.gmail.com>
References: <AANLkTink1-sZBh9zO9p8_he8XfaH+NdLHRHwZi5n_BE5@mail.gmail.com>
Message-ID: <AANLkTi=e_n9T3NFaB=Y1BE6_NAtFhAAan7m96WoKOM6t@mail.gmail.com>

Ok,

After being tired of waiting for some answer from the community, I did some
modifications in the "mod_fsv" module, with help from Paulo Panhoto, the
same guy that made the "mod_mp4" module.

With the modifications, now the "mod_fsv" supports the "break" application
and also supports interrupts by DTMF. The interrupts by DTMF must be used
through the channel variable "playback_terminators".

Can someone tell me if this modification is going to be included in the next
FreeSWITCH release?

Thanks in advance.

2010/12/22 William Kendi ... <william.nishio at voicetechnology.com.br>

> Greetings,
>
> Currently, I am trying to record and play videos through FreeSWITCH using
> the "mod_fsv" module. Some options like timeouts and interrupts by dtmf
> digits seems to be lacking in the "mod_fsv" module and I cant even stop the
> video recording using the "break" application. The video recording stop when
> the user disconnects.
> Anyone has any ideas about how to stop the FSV video recording without
> disconnecting the user?
>
> Thanks in advance.
>
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From william.suffill at gmail.com  Sat Jan 22 02:17:35 2011
From: william.suffill at gmail.com (William Suffill)
Date: Fri, 21 Jan 2011 18:17:35 -0500
Subject: [Freeswitch-users] mod_fsv, record and playback options
In-Reply-To: <AANLkTi=e_n9T3NFaB=Y1BE6_NAtFhAAan7m96WoKOM6t@mail.gmail.com>
References: <AANLkTink1-sZBh9zO9p8_he8XfaH+NdLHRHwZi5n_BE5@mail.gmail.com>
	<AANLkTi=e_n9T3NFaB=Y1BE6_NAtFhAAan7m96WoKOM6t@mail.gmail.com>
Message-ID: <AANLkTim0Zm1GLpbU5q2dvsm=nguwuMr6xY4ZUQAMWGX0@mail.gmail.com>

Best to add the patches/details into Jira [http://jira.freeswitch.org] so it
can be tracked and reviewed for being added to the source tree.
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From motosota at gmail.com  Sat Jan 22 02:19:05 2011
From: motosota at gmail.com (Mike)
Date: Fri, 21 Jan 2011 23:19:05 +0000
Subject: [Freeswitch-users] Freeswitch Sends RTP To Private IP Address
In-Reply-To: <AANLkTimru1i31uSYW+bGK_3z=GiW=QNPzjvTZRRwP5sq@mail.gmail.com>
References: <AANLkTimru1i31uSYW+bGK_3z=GiW=QNPzjvTZRRwP5sq@mail.gmail.com>
Message-ID: <AANLkTinVK90fOOpCq3Nj7ihb6FZom_3k33cNowcCo8T-@mail.gmail.com>

And the Cretin-Of-The-Day award goes to .... 'me'.

Having re-looked at the Wireshark trace I realised I hadn't opened up enough
ports in the FreeSWITCH firewall to let the RTP from the Polycom phone
through - so it was unable to learn what the real public IP address and port
number.

This is what I get for breaking my own rules about working too late!

Forget I asked....move along....nothing to see here.

On Fri, Jan 21, 2011 at 10:44 PM, Mike <motosota at gmail.com> wrote:

> Scenario. Polycom phone, private IP address, behind Cisco router doing NAT
> overload to a public IP address (SIP 'fixup' disabled, so the Cisco isn't
> mangling anything).
>
> RTP stream from FreeSWITCH is sent to the private, not public address of
> the phone.
>
> I've got this working on 1.0.6  but I've tried this with FreeSWITCH Version
> 1.0.head (git-7070061 2011-01-20 13-52-00 -0600) and it doesn't seem to work
> for me.
>
> Here http://pastebin.freeswitch.org/15109 is my sip profile.
>
> Here http://pastebin.freeswitch.org/15110 is a FreeSWITCH log for a call
> made from one of the Polycoms to voicemail with SIP tracing enabled.
>
> And here http://pastebin.freeswitch.org/15111 is the wireshark trace
> showing the media stream being sent by FreeSWITCH to the wrong port.
>
> I may be (hopefully) missing something very obvious here.
>
> Mike
>
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From anthony.minessale at gmail.com  Sat Jan 22 02:36:34 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 21 Jan 2011 17:36:34 -0600
Subject: [Freeswitch-users] praise of freeSWITCH developers
In-Reply-To: <AANLkTi=nHRE1TqGWtDR=iSnjGcWPwK8x2FeA1nkTGiCo@mail.gmail.com>
References: <9FE96265969E4BAEB801002D975CC50F@e1705>
	<1295627306.30863.188.camel@luna.tc.commsmundi.com>
	<AANLkTin-cNGJEBBjGDmmFSfPdd-B6kcMV_sZyt9H2-Xm@mail.gmail.com>
	<AANLkTi=nHRE1TqGWtDR=iSnjGcWPwK8x2FeA1nkTGiCo@mail.gmail.com>
Message-ID: <AANLkTi=13Gv-DPUZ5t=Vz9OFHJ24R0+db7PRaz+_8oJK@mail.gmail.com>

That is very cool, thank you all very much for the support and the
continued use of FS.


2011/1/21 Jo?o Mesquita <jmesquita at freeswitch.org>:
> I am not sure where I stand, a developer or a donor but I donated as well.
> My 20 went in a few seconds ago.
> Regards,
> Jo?o Mesquita
>
>
> On Fri, Jan 21, 2011 at 6:11 PM, EdPimentl <edpimentl at gmail.com> wrote:
>>
>> Will do $20.00 USD
>>
>> -E
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Sat Jan 22 02:39:37 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 21 Jan 2011 17:39:37 -0600
Subject: [Freeswitch-users] mod_fsv, record and playback options
In-Reply-To: <AANLkTim0Zm1GLpbU5q2dvsm=nguwuMr6xY4ZUQAMWGX0@mail.gmail.com>
References: <AANLkTink1-sZBh9zO9p8_he8XfaH+NdLHRHwZi5n_BE5@mail.gmail.com>
	<AANLkTi=e_n9T3NFaB=Y1BE6_NAtFhAAan7m96WoKOM6t@mail.gmail.com>
	<AANLkTim0Zm1GLpbU5q2dvsm=nguwuMr6xY4ZUQAMWGX0@mail.gmail.com>
Message-ID: <AANLkTinAwkmV4R=RLrhASWkymL8Crb7MRn5aJWoTM1tY@mail.gmail.com>

Sure, send it to Jira and we'll get it in.
Though, I'm surprised you would not want to use the mod_mp4 now that
it exists =D  the FSV was sort if a hack I made up on a whim.



On Fri, Jan 21, 2011 at 5:17 PM, William Suffill
<william.suffill at gmail.com> wrote:
> Best to add the patches/details into Jira [http://jira.freeswitch.org] so it
> can be tracked and reviewed for being added to the source tree.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From david.ponzone at ipeva.fr  Sat Jan 22 03:07:07 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sat, 22 Jan 2011 01:07:07 +0100
Subject: [Freeswitch-users] Question aboutCPU usage.
In-Reply-To: <AANLkTikHmFAtnuDdm19aE4QW34pZSZ5DwGAwa=RGj5V4@mail.gmail.com>
References: <57f59d718411e237cd08c17c7d93d03c@mail.gmail.com>
	<AANLkTimK8wXJ3LMntpGZ2xhfqt80XVLM+jRrAkCM0Q_X@mail.gmail.com>
	<AANLkTim78EOCHuFrYzhc1Z8U37yfQffN_0p1tgC9tvgy@mail.gmail.com>
	<AANLkTikHmFAtnuDdm19aE4QW34pZSZ5DwGAwa=RGj5V4@mail.gmail.com>
Message-ID: <532EE09C-28AD-4FEB-8A7B-2C082858070C@ipeva.fr>

The issue he will have then is that the GSM codec is not widely supported by carriers, nor by SIP devices.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 21/01/2011 ? 23:41, curriegrad2004 a ?crit :

> One solution is to use the GSM codec instead of the g.729 codec.
> Although it may not sound as good as the g.729 codec, it uses about
> the same amount of bandwidth and less CPU iirc. The GSM codec is
> included in FreeSwitch by default.
> 
> On Fri, Jan 21, 2011 at 11:35 AM, Steven Ayre <steveayre at gmail.com> wrote:
>> Ricardo,
>> 
>> "I also installed the codec g729 for transcoding from g711 to g729  (for
>> testing purposes only)."
>> 
>> We're assuming you're using deepwalker's fs_itu_g729 codec?
>> 
>> If you're planning to transcode G729 it's always better to test with the one
>> you're actually going to be using. You can get completely different results
>> since they're different code bases.
>> 
>> fs_itu_g729 will use more CPU and memory than mod_com_g729 so you'll get
>> different load testing results.
>> It also has no concept of licenses, so there's no way of testing that you're
>> handling the conditions where mod_com_g729 has reached the license limit.
>> 
>> -Steve
>> 
>> 
>> 
>> On 21 January 2011 19:13, Anthony Minessale <anthony.minessale at gmail.com>
>> wrote:
>>> 
>>> Yes, G729 is a CPU hog, especially the illegal one you are testing with.
>>> The commercial one that you can purchase from
>>> http://www.freeswitch.org probably will do slightly better but you
>>> will always get much less milage out of cpu intensive codecs.
>>> 
>>> 
>>> On Fri, Jan 21, 2011 at 9:06 AM, Ricardo Martinez
>>> <rmartinez at redvoiss.net> wrote:
>>>> Hello.
>>>> 
>>>> I have the opportunity to use an even more powerful server.  I have just
>>>> installed CentOS 5.5 64 bit, and compile freeswitch with 64 bits.
>>>> 
>>>> This is the output with 100 simm calls using transcoding from G711 to
>>>> G729
>>>> 
>>>> 
>>>> 
>>>> top - 16:57:56 up  6:04,  2 users,  load average: 11.60, 12.49, 12.58
>>>> 
>>>> Tasks: 168 total,   1 running, 167 sleeping,   0 stopped,   0 zombie
>>>> 
>>>> Cpu0  : 35.4%us,  0.6%sy,  0.0%ni, 63.6%id,  0.3%wa,  0.0%hi,  0.1%si,
>>>> 0.0%st
>>>> 
>>>> Cpu1  : 34.9%us,  0.6%sy,  0.0%ni, 58.8%id,  4.9%wa,  0.1%hi,  0.7%si,
>>>> 0.0%st
>>>> 
>>>> Cpu2  : 34.9%us,  0.6%sy,  0.0%ni, 59.6%id,  4.1%wa,  0.1%hi,  0.7%si,
>>>> 0.0%st
>>>> 
>>>> Cpu3  : 35.0%us,  0.7%sy,  0.0%ni, 62.0%id,  1.8%wa,  0.2%hi,  0.3%si,
>>>> 0.0%st
>>>> 
>>>> Cpu4  : 35.8%us,  1.3%sy,  0.0%ni, 62.5%id,  0.2%wa,  0.0%hi,  0.1%si,
>>>> 0.0%st
>>>> 
>>>> Cpu5  : 35.6%us,  1.6%sy,  0.0%ni, 60.1%id,  1.8%wa,  0.1%hi,  0.8%si,
>>>> 0.0%st
>>>> 
>>>> Cpu6  : 35.8%us,  1.9%sy,  0.0%ni, 59.9%id,  1.5%wa,  0.1%hi,  0.8%si,
>>>> 0.0%st
>>>> 
>>>> Cpu7  : 36.5%us,  2.0%sy,  0.0%ni, 60.6%id,  0.7%wa,  0.0%hi,  0.2%si,
>>>>  0.0%st
>>>> 
>>>> Mem:   8165808k total,  1208292k used,  6957516k free,   150228k buffers
>>>> 
>>>> Swap: 10223608k total,        0k used, 10223608k free,   772980k cached
>>>> 
>>>> 
>>>> 
>>>>   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
>>>> COMMAND
>>>> 
>>>>  5265 root      18   0  541m  79m 5796 S 469.5  1.0   1077:32
>>>> freeswitch
>>>> 
>>>>     1 root      15   0 10348  684  572 S  0.0  0.0   0:01.24
>>>> init
>>>> 
>>>>     2 root      RT  -5     0    0    0 S  0.0  0.0   0:00.00
>>>> migration/0
>>>> 
>>>>     3 root      34  19     0    0    0 S  0.0  0.0   0:00.00
>>>> ksoftirqd/0
>>>> 
>>>>     4 root      RT  -5     0    0    0 S  0.0  0.0   0:00.00
>>>> watchdog/0
>>>> 
>>>>     5 root      RT  -5     0    0    0 S  0.0  0.0   0:00.00
>>>> migration/1
>>>> 
>>>>     6 root      34  19     0    0    0 S  0.0  0.0   0:00
>>>> 
>>>> 
>>>> 
>>>> Is normal the load average?
>>>> 
>>>> 
>>>> 
>>>> This is the output with uname ?a
>>>> 
>>>> 
>>>> 
>>>> [root at siptrcrv2 snmp]# uname -a
>>>> 
>>>> Linux siptrcrv2 2.6.18-194.el5 #1 SMP Fri Apr 2 14:58:14 EDT 2010 x86_64
>>>> x86_64 x86_64 GNU/Linux
>>>> 
>>>> 
>>>> 
>>>> Regards,
>>>> 
>>>> Ricardo.-
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> De: freeswitch-users-bounces at lists.freeswitch.org
>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de
>>>> Brian
>>>> West
>>>> Enviado el: mi?rcoles, 19 de enero de 2011 16:22
>>>> Para: FreeSWITCH Users Help
>>>> 
>>>> Asunto: Re: [Freeswitch-users] Question aboutCPU usage.
>>>> 
>>>> 
>>>> 
>>>> Chances are if you would install a 64bit OS on that there NICE 64bit CPU
>>>> it
>>>> would work much better.
>>>> 
>>>> 
>>>> 
>>>> /b
>>>> 
>>>> 
>>>> 
>>>> On Jan 19, 2011, at 12:43 PM, Ricardo Martinez wrote:
>>>> 
>>>> 
>>>> 
>>>> Hello.
>>>> 
>>>> 
>>>> 
>>>> [root at ser-ng bin]# uname -a
>>>> 
>>>> Linux 2.6.33.5-124.fc13.i686.PAE #1 SMP Fri Jun 11 09:42:24 UTC 2010
>>>> i686
>>>> i686 i386 GNU/Linux
>>>> 
>>>> 
>>>> 
>>>> Ricardo.-
>>>> 
>>>> 
>>>> 
>>>> 
>>>> De: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] En
>>>> nombre de Brian West
>>>> Enviado el: mi?rcoles, 19 de enero de 2011 15:41
>>>> Para: FreeSWITCH Users Help
>>>> Asunto: Re: [Freeswitch-users] Question aboutCPU usage.
>>>> 
>>>> 
>>>> 
>>>> Depend what does uname -a say?
>>>> 
>>>> 
>>>> 
>>>> /b
>>>> 
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>> 
>>> 
>>> 
>>> --
>>> Anthony Minessale II
>>> 
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>> 
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>> 
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From lloyd.aloysius at gmail.com  Sat Jan 22 04:17:56 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 21 Jan 2011 20:17:56 -0500
Subject: [Freeswitch-users] praise of freeSWITCH developers
In-Reply-To: <AANLkTi=13Gv-DPUZ5t=Vz9OFHJ24R0+db7PRaz+_8oJK@mail.gmail.com>
References: <9FE96265969E4BAEB801002D975CC50F@e1705>
	<1295627306.30863.188.camel@luna.tc.commsmundi.com>
	<AANLkTin-cNGJEBBjGDmmFSfPdd-B6kcMV_sZyt9H2-Xm@mail.gmail.com>
	<AANLkTi=nHRE1TqGWtDR=iSnjGcWPwK8x2FeA1nkTGiCo@mail.gmail.com>
	<AANLkTi=13Gv-DPUZ5t=Vz9OFHJ24R0+db7PRaz+_8oJK@mail.gmail.com>
Message-ID: <AANLkTimfXaDdJPMW_zPuo2g5GCDKq66-9mZWvEkVGk=q@mail.gmail.com>

I just donated $100 USD.

Thank you for all the hard work. FreeSWITH Rocks.

Thanks
Lloyd


On Fri, Jan 21, 2011 at 6:36 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> That is very cool, thank you all very much for the support and the
> continued use of FS.
>
>
> 2011/1/21 Jo?o Mesquita <jmesquita at freeswitch.org>:
> > I am not sure where I stand, a developer or a donor but I donated as
> well.
> > My 20 went in a few seconds ago.
> > Regards,
> > Jo?o Mesquita
> >
> >
> > On Fri, Jan 21, 2011 at 6:11 PM, EdPimentl <edpimentl at gmail.com> wrote:
> >>
> >> Will do $20.00 USD
> >>
> >> -E
> >>
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jmesquita at freeswitch.org  Sat Jan 22 04:23:13 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 21 Jan 2011 22:23:13 -0300
Subject: [Freeswitch-users] praise of freeSWITCH developers
In-Reply-To: <AANLkTimfXaDdJPMW_zPuo2g5GCDKq66-9mZWvEkVGk=q@mail.gmail.com>
References: <9FE96265969E4BAEB801002D975CC50F@e1705>
	<1295627306.30863.188.camel@luna.tc.commsmundi.com>
	<AANLkTin-cNGJEBBjGDmmFSfPdd-B6kcMV_sZyt9H2-Xm@mail.gmail.com>
	<AANLkTi=nHRE1TqGWtDR=iSnjGcWPwK8x2FeA1nkTGiCo@mail.gmail.com>
	<AANLkTi=13Gv-DPUZ5t=Vz9OFHJ24R0+db7PRaz+_8oJK@mail.gmail.com>
	<AANLkTimfXaDdJPMW_zPuo2g5GCDKq66-9mZWvEkVGk=q@mail.gmail.com>
Message-ID: <AANLkTi=LoV-Z8O=cj29AtsxV-__K=4mBa5h+1F9cXmZT@mail.gmail.com>

Wow! You make us look bad! Thank you (even tho I don't get the money for
myself)! :-)

Jo?o Mesquita


On Fri, Jan 21, 2011 at 10:17 PM, Aloysius Lloyd
<lloyd.aloysius at gmail.com>wrote:

> I just donated $100 USD.
>
> Thank you for all the hard work. FreeSWITH Rocks.
>
> Thanks
> Lloyd
>
>
> On Fri, Jan 21, 2011 at 6:36 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> That is very cool, thank you all very much for the support and the
>> continued use of FS.
>>
>>
>> 2011/1/21 Jo?o Mesquita <jmesquita at freeswitch.org>:
>> > I am not sure where I stand, a developer or a donor but I donated as
>> well.
>> > My 20 went in a few seconds ago.
>> > Regards,
>> > Jo?o Mesquita
>> >
>> >
>> > On Fri, Jan 21, 2011 at 6:11 PM, EdPimentl <edpimentl at gmail.com> wrote:
>> >>
>> >> Will do $20.00 USD
>> >>
>> >> -E
>> >>
>> >>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Sat Jan 22 04:43:32 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 21 Jan 2011 22:43:32 -0300
Subject: [Freeswitch-users] Run dialplan tools from event socket
In-Reply-To: <AANLkTikDBWtCTp=h0t_Qxtm1S+ZTX_UUtvj5RgbFdt+g@mail.gmail.com>
References: <AANLkTikg5+1aMoWxYr3UYO13iO2qA6FsG9FFi-Zg3_bD@mail.gmail.com>
	<AANLkTin5fn9qMre-8xFAZD5_SUzbLbp_YueTRPPyh-=K@mail.gmail.com>
	<AANLkTin84SkDdDsBE0wfU2QnG025bNK8BhcW=JDvz9gv@mail.gmail.com>
	<AANLkTi=TX6BURrxR38X9bwyUai_52xLK80kMvm9CmZ8T@mail.gmail.com>
	<AANLkTikDBWtCTp=h0t_Qxtm1S+ZTX_UUtvj5RgbFdt+g@mail.gmail.com>
Message-ID: <AANLkTinh7cQp5H9OOv1JsAvYj=PUteS0MtgTXXA3ZvNa@mail.gmail.com>

You hit the right spot... That's what you need to do, we don't need an API,
it already exists. I've tested and it works like a charm.

Regards,
Jo?o Mesquita


On Fri, Jan 21, 2011 at 10:16 AM, Steven Ayre <steveayre at gmail.com> wrote:

> uuid_broadcast appears to have support for executing a dialplan app from
> the api (it uses say as an example):
>
> uuid_broadcast <uuid> app!::args [aleg|bleg|both]
>
> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast
>
> I haven't tried it myself though.
>
> -Steve
>
>
> 2011/1/21 Jo?o Mesquita <jmesquita at freeswitch.org>
>
> Actually, Fraser, I think this won't work...
>>
>> att_xfer uses the signal_bond variable to get the leg connected to the
>> party being transferred. This variable is unset when you park the legs or
>> break the bridge in any way. Maybe I can make a API command that does
>> att_xfer taking the trasferred leg UUID as the transferred party? Do me a
>> favor, make some tests and I will take a deeper look at the att_xfer
>> application code.
>>
>> Regards,
>> Jo?o Mesquita
>>
>>
>>
>> On Fri, Jan 21, 2011 at 1:37 AM, Fraser Redmond <fraserredmond at gmail.com>wrote:
>>
>>> Thanks Jo?o.
>>>
>>> My transfer_call extension runs a couple of js scripts to get and
>>> validate the number to transfer to, then does
>>>                 <action application="att_xfer" data="${dial_string}"/>
>>> (and it has a couple of steps after that to handle failed transfers.)
>>>
>>> So could I use ESL's execute command to run the execute_extension? Not
>>> sure how I missed that option in the wiki. I"ll give it a try, see what
>>> happens.
>>>
>>>
>>> I forgot to say in the original post, but execute_extension seems to be
>>> particularly nice for this use-case, as it falls back through the dialplan
>>> gracefully if there's a problem.
>>>
>>> Cheers,
>>> Fraser
>>>
>>>
>>>
>>>
>>> 2011/1/20 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>
>>> Lucky for you I have been working on this lately and the bad news is ...
>>>> there's no easy way to do it....
>>>>
>>>> You can execute an extension like you said, but you have to park the
>>>> legs first... It would help to know how's the transfer_call extension so
>>>> that I can try to help you out, but maybe it is easier if you think of it
>>>> this way:
>>>>
>>>> When you use an app like att_xfer, the core already knows what to do
>>>> next with a call and parks the legs for you. If you do it on ESL, you've
>>>> done it half way and you didn't really park anything before you transfered
>>>> the call. When the bridge is undone, the leg that was not transfered doesn't
>>>> know what to do, has no applications to be run at this moment and so all
>>>> it's left for it is to let go.
>>>>
>>>> A bit clearer? Att_xfer is a bit of a pain in the butt and it kinda
>>>> requires you to know a bit more of the inner workings of the state machine.
>>>>
>>>> You can always execute att_xfer using ESL's execute (
>>>> http://wiki.freeswitch.org/wiki/Event_Socket_Library#execute) if you
>>>> don't care what happens to the legs afterwards but if you want to have
>>>> control over all 3 legs, no luck for you...
>>>>
>>>> Regards,
>>>> Jo?o Mesquita
>>>>
>>>>
>>>> On Fri, Jan 21, 2011 at 12:13 AM, Fraser Redmond <
>>>> fraserredmond at gmail.com> wrote:
>>>>
>>>>> Is there any way to run a dialplan tool from the event socket?
>>>>>
>>>>> I have a dialplan that uses a dtmf to set up and perform an attended
>>>>> transfer, like so:
>>>>> <action application="bind_meta_app" data="8 a s
>>>>> execute_extension::TransferCall XML transfer_call"/>
>>>>>
>>>>> But I can't see any way to run the same thing from the event socket. I
>>>>> thought doing an  "api uuid_transfer"  might do it, but that hangs up one of
>>>>> the legs (no good for attended transfer.)
>>>>>
>>>>> api uuid_transfer Uuid -bleg TransferCall XML transfer_call
>>>>>
>>>>> As far as I can see, the closest thing is "sendmsg execute", but it
>>>>> looks like you have to park a call/channel first to use that, so I'm not
>>>>> sure that that is much use for attended transfer either.
>>>>>
>>>>> Or should I be lame and do "api uuid_send_dtmf" to send * 8.
>>>>>
>>>>> Cheers,
>>>>> Fraser
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Sat Jan 22 04:44:22 2011
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 22 Jan 2011 09:44:22 +0800
Subject: [Freeswitch-users] Portaudio Improvements
In-Reply-To: <AANLkTikk==DKhXft67HWcj0ytya030bS=4vtt2o7+tD8@mail.gmail.com>
References: <AANLkTim9AmtEGZoK0NyG1yKVsmJ_MSwXAppi-6AcCqnz@mail.gmail.com>
	<AANLkTimm69f6adgkgDz03GUOq8xJw_oaVvrkAGxd1yjG@mail.gmail.com>
	<AANLkTikk==DKhXft67HWcj0ytya030bS=4vtt2o7+tD8@mail.gmail.com>
Message-ID: <AANLkTi=T6a9nU54toi5+fMA7kq_xkmZrRnRT3Fa8Pao5@mail.gmail.com>

I also would like to contribute more on FSComm when I have more free
time. I actually "stoled" some code from FSComm to a client I was
working on. We use a headset so AEC is not a big deal for now. But I
still would like to see AEC work for this. Actually someone reported
ECHO even with a headset I haven't confirm.  Is it possible to get
OSLEC work on Windows/Mac ?

In addition to AEC. I also found PA lacks the ability to change the
sound device volume. I tried to control sound devices in QT but
haven't find a way. Ideas on this?

Thanks.


On Sat, Jan 22, 2011 at 2:22 AM, Mitch Capper <mitch.capper at gmail.com> wrote:
> I did look into AEC a bit, although one of the best options for me looked
> like the DirectX AEC support, but obviously this would be windows only.
> There is oslec for linux which tries to do AEC at a driver level and is in
> the kernel.?? The speex AEC processing does not look very complex
> (http://www.speex.org/docs/manual/speex-manual/node7.html using
> speex_echo_playback/speex_echo_capture).?? Of course easier said then done,
> and does require passing it the audio in the structure it expects.? One of
> my main concerns is I do not also know about how good the speex AEC
> processing is for it to be worthwhile and would be curious if it has been
> found to give good results in voip settings (as I only saw to the contrary).
>
> Finally, it probably hasn't been too high on my list due to the fact that I
> have been using my client with a headset so the need is not really there.
>
> ~Mitch
>
> 2011/1/21 Jo?o Mesquita <jmesquita at freeswitch.org>
>>
>> Mitch, I will work on continuing FSComm for a thick client multi-platform
>> solution but one of the big show stoppers for me was the lack of AEC on
>> mod_pa.
>> I tried the preprocessors embedded on the core by Tony that are using the
>> speexdsp but I got no luck. The main tests I made were on a mac using the
>> speakerphone. Do you have any experience with this type of technology? It
>> seems to me that the only AEC available is the one implemented on speexdsp
>> but I am completely new to this.
>> Regards,
>> Jo?o Mesquita
>>
>>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From infos at madovsky.org  Sat Jan 22 04:52:19 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 21 Jan 2011 20:52:19 -0500
Subject: [Freeswitch-users] praise of freeSWITCH developers
References: <9FE96265969E4BAEB801002D975CC50F@e1705><1295627306.30863.188.camel@luna.tc.commsmundi.com><AANLkTin-cNGJEBBjGDmmFSfPdd-B6kcMV_sZyt9H2-Xm@mail.gmail.com>
	<AANLkTi=nHRE1TqGWtDR=iSnjGcWPwK8x2FeA1nkTGiCo@mail.gmail.com>
Message-ID: <017F6A9CD5984DF28324A73B531978C7@e1705>

> developer or a donor
I think ti's the same when you contribute
your work to the net... ;o)
  ----- Original Message ----- 
  From: Jo?o Mesquita 
  To: FreeSWITCH Users Help 
  Sent: Friday, January 21, 2011 5:52 PM
  Subject: Re: [Freeswitch-users] praise of freeSWITCH developers


  I am not sure where I stand, a developer or a donor but I donated as well. My 20 went in a few seconds ago.


  Regards,
  Jo?o Mesquita



  On Fri, Jan 21, 2011 at 6:11 PM, EdPimentl <edpimentl at gmail.com> wrote:

    Will do $20.00 USD

    -E




    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org






------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From telteclistas at gmail.com  Sat Jan 22 05:23:55 2011
From: telteclistas at gmail.com (leonardo alves)
Date: Fri, 21 Jan 2011 21:23:55 -0500
Subject: [Freeswitch-users] praise of freeSWITCH developers
In-Reply-To: <017F6A9CD5984DF28324A73B531978C7@e1705>
References: <9FE96265969E4BAEB801002D975CC50F@e1705>
	<1295627306.30863.188.camel@luna.tc.commsmundi.com>
	<AANLkTin-cNGJEBBjGDmmFSfPdd-B6kcMV_sZyt9H2-Xm@mail.gmail.com>
	<AANLkTi=nHRE1TqGWtDR=iSnjGcWPwK8x2FeA1nkTGiCo@mail.gmail.com>
	<017F6A9CD5984DF28324A73B531978C7@e1705>
Message-ID: <AANLkTim0Y=GegEXEehQsUTHyWOZzz5PkvCGabUaLa6t+@mail.gmail.com>

Done.Thanks for the incredible software


>  I am not sure where I stand, a developer or a donor but I donated as
> well. My 20 went in a few seconds ago.
>
>
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From steveu at coppice.org  Sat Jan 22 07:17:06 2011
From: steveu at coppice.org (Steve Underwood)
Date: Sat, 22 Jan 2011 12:17:06 +0800
Subject: [Freeswitch-users] Portaudio Improvements
In-Reply-To: <AANLkTikk==DKhXft67HWcj0ytya030bS=4vtt2o7+tD8@mail.gmail.com>
References: <AANLkTim9AmtEGZoK0NyG1yKVsmJ_MSwXAppi-6AcCqnz@mail.gmail.com>	<AANLkTimm69f6adgkgDz03GUOq8xJw_oaVvrkAGxd1yjG@mail.gmail.com>
	<AANLkTikk==DKhXft67HWcj0ytya030bS=4vtt2o7+tD8@mail.gmail.com>
Message-ID: <4D3A5A42.10209@coppice.org>

On 01/22/2011 02:22 AM, Mitch Capper wrote:
> I did look into AEC a bit, although one of the best options for me 
> looked like the DirectX AEC support, but obviously this would be 
> windows only.   There is oslec for linux which tries to do AEC at a 
> driver level and is in the kernel.   The speex AEC processing does not 
> look very complex 
> (http://www.speex.org/docs/manual/speex-manual/node7.html using 
> speex_echo_playback/speex_echo_capture).   Of course easier said then 
> done, and does require passing it the audio in the structure it 
> expects.  One of my main concerns is I do not also know about how good 
> the speex AEC processing is for it to be worthwhile and would be 
> curious if it has been found to give good results in voip settings (as 
> I only saw to the contrary).
>
> Finally, it probably hasn't been too high on my list due to the fact 
> that I have been using my client with a headset so the need is not 
> really there.
OSLEC is not an AEC. It is a line echo canceller, and very much 
optimised for that role. It would be useless for acoustic echoes.

The AEC built into Windows and the speex echo canceller have a problem 
all echo cancellation is now suffering on modern PCs. Its a problem that 
keeps coming up on the speex mailing list. Like most echo cancellers 
they assume the mic and speaker sampling rates match. It turns out that 
with an increasing number of sound cards this is not the case. The 
sample rates are very similar, but they are not locked, and drift in 
relation to each other. This wrecks the performance of the echo 
canceller, and is not an easy problem to work around.

That said, there are products, like skype, which *appear* to do a good 
job of echo cancellation. If you play around with skype, though, you'll 
notice it isn't actually echo cancelling. It seems to be adaptively 
juggling gains to give the appearance of clean duplex communication, and 
the result is fairly pleasing. I think it may be using a similar 
approach to the DSP Group duplex speakerphone chips from the early 90s. 
They were never quite as good as the kind of full AEC seakerphones I was 
developing around the same time. They did, however, give results far 
better than traditional simplex speakerphones using a DSP too 
underpowered to perform full echo cancellation.

Steve




From jmesquita at freeswitch.org  Sat Jan 22 07:56:59 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 22 Jan 2011 01:56:59 -0300
Subject: [Freeswitch-users] Portaudio Improvements
In-Reply-To: <4D3A5A42.10209@coppice.org>
References: <AANLkTim9AmtEGZoK0NyG1yKVsmJ_MSwXAppi-6AcCqnz@mail.gmail.com>
	<AANLkTimm69f6adgkgDz03GUOq8xJw_oaVvrkAGxd1yjG@mail.gmail.com>
	<AANLkTikk==DKhXft67HWcj0ytya030bS=4vtt2o7+tD8@mail.gmail.com>
	<4D3A5A42.10209@coppice.org>
Message-ID: <AANLkTi=uz6k9cffYZ2kOoaXWitXBDrVJHVkrxgk6jV_K@mail.gmail.com>

Steve, I was actually hoping you would tackle this thread, thank you! :-)

I am not a DSP guy and to be honest, I don't think I am much of anything
guy, but I did some shallow study on AEC algos... It seems that the delay
between the recording and the playback can also degrade the AEC performance
considerably. I personally thought that the use of pablio and the lack o
lower latency audio path on mod_pa would help make the speex AEC be
completely useless...

I am not talking about bad performance, I am talking about doing nothing
with the filters and making it even worse if you tweak too much... Now, I
think that before we pursue a solution, we have to find the problem and
that's the question I was going to ask you.

Can you point us towards the right direction? Skype is _very_ useable on a
Mac and "most" windows machines I have used. On the mac they almost never
fail... Are we able to achieve that type of quality? If we are able to, in
which direction should we start going towards?

I honestly thought that replacing mod_pa completely was the solution at
first (and creating some abstraction layer to plug in os dependent drivers
such as coreaudio for mac or dx for windows directly into FS, thus reducing
this delay of buffers such as pablio) but now I am almost convinced that it
was never the problem. If we can decide that before anything else, I think
we could make some progress.

Thank you,
Jo?o Mesquita


On Sat, Jan 22, 2011 at 1:17 AM, Steve Underwood <steveu at coppice.org> wrote:

> On 01/22/2011 02:22 AM, Mitch Capper wrote:
> > I did look into AEC a bit, although one of the best options for me
> > looked like the DirectX AEC support, but obviously this would be
> > windows only.   There is oslec for linux which tries to do AEC at a
> > driver level and is in the kernel.   The speex AEC processing does not
> > look very complex
> > (http://www.speex.org/docs/manual/speex-manual/node7.html using
> > speex_echo_playback/speex_echo_capture).   Of course easier said then
> > done, and does require passing it the audio in the structure it
> > expects.  One of my main concerns is I do not also know about how good
> > the speex AEC processing is for it to be worthwhile and would be
> > curious if it has been found to give good results in voip settings (as
> > I only saw to the contrary).
> >
> > Finally, it probably hasn't been too high on my list due to the fact
> > that I have been using my client with a headset so the need is not
> > really there.
> OSLEC is not an AEC. It is a line echo canceller, and very much
> optimised for that role. It would be useless for acoustic echoes.
>
> The AEC built into Windows and the speex echo canceller have a problem
> all echo cancellation is now suffering on modern PCs. Its a problem that
> keeps coming up on the speex mailing list. Like most echo cancellers
> they assume the mic and speaker sampling rates match. It turns out that
> with an increasing number of sound cards this is not the case. The
> sample rates are very similar, but they are not locked, and drift in
> relation to each other. This wrecks the performance of the echo
> canceller, and is not an easy problem to work around.
>
> That said, there are products, like skype, which *appear* to do a good
> job of echo cancellation. If you play around with skype, though, you'll
> notice it isn't actually echo cancelling. It seems to be adaptively
> juggling gains to give the appearance of clean duplex communication, and
> the result is fairly pleasing. I think it may be using a similar
> approach to the DSP Group duplex speakerphone chips from the early 90s.
> They were never quite as good as the kind of full AEC seakerphones I was
> developing around the same time. They did, however, give results far
> better than traditional simplex speakerphones using a DSP too
> underpowered to perform full echo cancellation.
>
> Steve
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jmesquita at freeswitch.org  Sat Jan 22 07:59:49 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 22 Jan 2011 01:59:49 -0300
Subject: [Freeswitch-users] Portaudio Improvements
In-Reply-To: <AANLkTi=uz6k9cffYZ2kOoaXWitXBDrVJHVkrxgk6jV_K@mail.gmail.com>
References: <AANLkTim9AmtEGZoK0NyG1yKVsmJ_MSwXAppi-6AcCqnz@mail.gmail.com>
	<AANLkTimm69f6adgkgDz03GUOq8xJw_oaVvrkAGxd1yjG@mail.gmail.com>
	<AANLkTikk==DKhXft67HWcj0ytya030bS=4vtt2o7+tD8@mail.gmail.com>
	<4D3A5A42.10209@coppice.org>
	<AANLkTi=uz6k9cffYZ2kOoaXWitXBDrVJHVkrxgk6jV_K@mail.gmail.com>
Message-ID: <AANLkTik3_vt=vncyusZ6Wc=xEWTNTQzq505UALF8_nWq@mail.gmail.com>

And one side not for the ppl who have been following this thread and might
be interested on FSComm. This subject was precisely the reason why I decided
to put FSComm a bit on hold. The idea is not to just make a multi-platform
softphone but rather to make something that behaves better then what we
currently have available. I really DON'T want to have another XLite
experience (no offense) where we all use it because there's really no good
alternative. Not having a decent AEC really caused me the impression that
the main objective will fail dramatically... Comments on this could be
interesting from anyone willing to provide their point of view.

Regards,
Jo?o Mesquita



2011/1/22 Jo?o Mesquita <jmesquita at freeswitch.org>

> Steve, I was actually hoping you would tackle this thread, thank you! :-)
>
> I am not a DSP guy and to be honest, I don't think I am much of anything
> guy, but I did some shallow study on AEC algos... It seems that the delay
> between the recording and the playback can also degrade the AEC performance
> considerably. I personally thought that the use of pablio and the lack o
> lower latency audio path on mod_pa would help make the speex AEC be
> completely useless...
>
> I am not talking about bad performance, I am talking about doing nothing
> with the filters and making it even worse if you tweak too much... Now, I
> think that before we pursue a solution, we have to find the problem and
> that's the question I was going to ask you.
>
> Can you point us towards the right direction? Skype is _very_ useable on a
> Mac and "most" windows machines I have used. On the mac they almost never
> fail... Are we able to achieve that type of quality? If we are able to, in
> which direction should we start going towards?
>
> I honestly thought that replacing mod_pa completely was the solution at
> first (and creating some abstraction layer to plug in os dependent drivers
> such as coreaudio for mac or dx for windows directly into FS, thus reducing
> this delay of buffers such as pablio) but now I am almost convinced that it
> was never the problem. If we can decide that before anything else, I think
> we could make some progress.
>
> Thank you,
> Jo?o Mesquita
>
>
>
> On Sat, Jan 22, 2011 at 1:17 AM, Steve Underwood <steveu at coppice.org>wrote:
>
>> On 01/22/2011 02:22 AM, Mitch Capper wrote:
>> > I did look into AEC a bit, although one of the best options for me
>> > looked like the DirectX AEC support, but obviously this would be
>> > windows only.   There is oslec for linux which tries to do AEC at a
>> > driver level and is in the kernel.   The speex AEC processing does not
>> > look very complex
>> > (http://www.speex.org/docs/manual/speex-manual/node7.html using
>> > speex_echo_playback/speex_echo_capture).   Of course easier said then
>> > done, and does require passing it the audio in the structure it
>> > expects.  One of my main concerns is I do not also know about how good
>> > the speex AEC processing is for it to be worthwhile and would be
>> > curious if it has been found to give good results in voip settings (as
>> > I only saw to the contrary).
>> >
>> > Finally, it probably hasn't been too high on my list due to the fact
>> > that I have been using my client with a headset so the need is not
>> > really there.
>> OSLEC is not an AEC. It is a line echo canceller, and very much
>> optimised for that role. It would be useless for acoustic echoes.
>>
>> The AEC built into Windows and the speex echo canceller have a problem
>> all echo cancellation is now suffering on modern PCs. Its a problem that
>> keeps coming up on the speex mailing list. Like most echo cancellers
>> they assume the mic and speaker sampling rates match. It turns out that
>> with an increasing number of sound cards this is not the case. The
>> sample rates are very similar, but they are not locked, and drift in
>> relation to each other. This wrecks the performance of the echo
>> canceller, and is not an easy problem to work around.
>>
>> That said, there are products, like skype, which *appear* to do a good
>> job of echo cancellation. If you play around with skype, though, you'll
>> notice it isn't actually echo cancelling. It seems to be adaptively
>> juggling gains to give the appearance of clean duplex communication, and
>> the result is fairly pleasing. I think it may be using a similar
>> approach to the DSP Group duplex speakerphone chips from the early 90s.
>> They were never quite as good as the kind of full AEC seakerphones I was
>> developing around the same time. They did, however, give results far
>> better than traditional simplex speakerphones using a DSP too
>> underpowered to perform full echo cancellation.
>>
>> Steve
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From steveu at coppice.org  Sat Jan 22 08:06:41 2011
From: steveu at coppice.org (Steve Underwood)
Date: Sat, 22 Jan 2011 13:06:41 +0800
Subject: [Freeswitch-users] Portaudio Improvements
In-Reply-To: <AANLkTi=T6a9nU54toi5+fMA7kq_xkmZrRnRT3Fa8Pao5@mail.gmail.com>
References: <AANLkTim9AmtEGZoK0NyG1yKVsmJ_MSwXAppi-6AcCqnz@mail.gmail.com>	<AANLkTimm69f6adgkgDz03GUOq8xJw_oaVvrkAGxd1yjG@mail.gmail.com>	<AANLkTikk==DKhXft67HWcj0ytya030bS=4vtt2o7+tD8@mail.gmail.com>
	<AANLkTi=T6a9nU54toi5+fMA7kq_xkmZrRnRT3Fa8Pao5@mail.gmail.com>
Message-ID: <4D3A65E1.1030701@coppice.org>

On 01/22/2011 09:44 AM, Seven Du wrote:
> I also would like to contribute more on FSComm when I have more free
> time. I actually "stoled" some code from FSComm to a client I was
> working on. We use a headset so AEC is not a big deal for now. But I
> still would like to see AEC work for this. Actually someone reported
> ECHO even with a headset I haven't confirm.  Is it possible to get
> OSLEC work on Windows/Mac ?
>
> In addition to AEC. I also found PA lacks the ability to change the
> sound device volume. I tried to control sound devices in QT but
> haven't find a way. Ideas on this?
Echo is a big problem with headsets, especially those with the mic up 
near the earpiece, and a tube to the mouth. Its also a problem in 
traditional telephone handsets. A good IP phone handset should have an 
acoustic echo canceller cleaning up its mic signal. A lot (even many 
expensive ones) don't, and produce considerable echo if you turn up the 
handset volume for the hearing impaired.

Steve



From chat2jesse at gmail.com  Sat Jan 22 08:32:03 2011
From: chat2jesse at gmail.com (jesse)
Date: Fri, 21 Jan 2011 21:32:03 -0800
Subject: [Freeswitch-users] is it possible to create a session between two
	extensions?
Message-ID: <AANLkTimjzAS1WZ86GjwZHtyqVFV6NMZ7Ggd9uD-NjxZd@mail.gmail.com>

In FS, a session is connection between one extension and sip
internal/external  user.

suppose I defined two extensions in diaplan/default.xml one is 2333,
the other is 23334. (both are NOT locally registered endpoints)

I am wondering whether it is possible to create a session with two extensions?
like : original 2333-XML-default 2334-XML-default

-jesse


From hwnorman at hotmail.com  Sat Jan 22 10:04:27 2011
From: hwnorman at hotmail.com (Norman Lam)
Date: Sat, 22 Jan 2011 15:04:27 +0800
Subject: [Freeswitch-users] Iksemel msvs compiling
In-Reply-To: <1295320482333-5934373.post@n2.nabble.com>
References: <1294375438447-5898181.post@n2.nabble.com>	<SNT109-DS11CB0ECE5BD0F9C0384E03C10C0@phx.gbl>	<1294535404901-5903534.post@n2.nabble.com>	<SNT109-DS14B90C48B33126784F294AC10E0@phx.gbl>	<1294673236806-5907287.post@n2.nabble.com>	<SNT109-DS227CE87422187AA91545CFC1F30@phx.gbl>	<1295113990176-5925152.post@n2.nabble.com>	<SNT109-DS2124FB8719BEF6F213D36DC1F40@phx.gbl>	<1295276746914-5932097.post@n2.nabble.com>	<SNT109-DS6447C073B028EC7D60228C1F70@phx.gbl>
	<1295320482333-5934373.post@n2.nabble.com>
Message-ID: <SNT109-DS1063277B95F0AC68C084EDC1FB0@phx.gbl>

Hi Jeff 

I got it to compile, following your guidance below and also edit the
gnutls/gnutls.h file, to insert the definition typedef long ssize_t; after
the line #include <gnutls/compat.h>

I have it compile and working. 

I try update the wiki, but couldn't get in to edit, so maybe you can update
it 

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff
Lenk
Sent: Tuesday, January 18, 2011 11:15 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Iksemel msvs compiling


use an absolute path to find the required includes and libs ex.
c:\gnutls-2.9.9\include

make sure you take into account whether the code is looking for
"path\file.h" or whatever.
-- 
View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/Iksemel-msvs-compiling-tp58912
63p5934373.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From dujinfang at gmail.com  Sat Jan 22 12:31:34 2011
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 22 Jan 2011 17:31:34 +0800
Subject: [Freeswitch-users] is it possible to create a session between
	two extensions?
In-Reply-To: <AANLkTimjzAS1WZ86GjwZHtyqVFV6NMZ7Ggd9uD-NjxZd@mail.gmail.com>
References: <AANLkTimjzAS1WZ86GjwZHtyqVFV6NMZ7Ggd9uD-NjxZd@mail.gmail.com>
Message-ID: <AANLkTinJjW7pt86ySZbJBYk44kQb141-f1pkGdPggFK=@mail.gmail.com>

originate loopback/2333 23334

On Sat, Jan 22, 2011 at 1:32 PM, jesse <chat2jesse at gmail.com> wrote:
> In FS, a session is connection between one extension and sip
> internal/external ?user.
>
> suppose I defined two extensions in diaplan/default.xml one is 2333,
> the other is 23334. (both are NOT locally registered endpoints)
>
> I am wondering whether it is possible to create a session with two extensions?
> like : original 2333-XML-default 2334-XML-default
>
> -jesse
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From christian.loeschenkohl at xpirio.com  Sat Jan 22 12:58:55 2011
From: christian.loeschenkohl at xpirio.com (=?windows-1252?Q?Christian_L=F6schenkohl?=)
Date: Sat, 22 Jan 2011 10:58:55 +0100
Subject: [Freeswitch-users] praise of freeSWITCH developers
In-Reply-To: <9FE96265969E4BAEB801002D975CC50F@e1705>
References: <9FE96265969E4BAEB801002D975CC50F@e1705>
Message-ID: <4D3AAA5F.4020303@xpirio.com>

hello

these words are more than true.
100$ from my side are on the way.

br

On 2011-01-21 17:13, Madovsky wrote:

> I would like to remember to all people on this
> emailist that all freeSWITCH guys work very hard
> to update/upgrade and make FS more powerful, and
> also to answer to hundreds of user emails every day.
> Donations are welcome and appreciated.
> It helps the humans behind the machine
> who give to you the opportunity to create and use amazing projects
> and to continue to make this open source project alive and active.
> Regards
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 5 77 11 - 1000
F  +43 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From john at 247-talk.co.uk  Sat Jan 22 04:57:12 2011
From: john at 247-talk.co.uk (John Carpenter)
Date: Sat, 22 Jan 2011 01:57:12 +0000
Subject: [Freeswitch-users] IVR Bridged Call Dropping after 2 rings
Message-ID: <1295661432.3014.15.camel@John-Home>

Hi, I am trying to setup a very simple IVR using LUA. Call arrives from
a DID SIP trunk and is answered and message is played ok, after a
particular digit is pressed it bridges the call to an extension which is
remotely connected. It works but after 2 rings the call to the extension
is dropped with a SIP message "BYE" from DID provider. If I just route
the call directly to the extension (no IVR) it works fine. It seems like
the DID hangs up when the call is bridged to the extension. Have tried
same thing using the XML IVR Engine and get exactly the same result. The
IVR script is below

pathsep = '/'
session:setAutoHangup(false);
session:answer()
prompt = "ivr" .. pathsep .. "247talk.wav"
invalid = "ivr" .. pathsep .. "ivr-that_was_an_invalid_entry.wav"
freeswitch.consoleLog("INFO", "Prompt file is '" .. prompt .. "'\n")
continue = true

while( session:ready() == true and continue == true) do
        digits = session:playAndGetDigits(1,1,3,7000,"#", prompt,
invalid, "\\d+")
        if (digits == "1") then
                continue = false
                session:execute("bridge","sofia/external/2476%
91.xxx.xx.xx")
        end
        if (digits == "2") then
                session:execute("bridge","sofia/external/2475%
91.xxx.xx.xx")
        end
        if (digits == "3") then
                continue = false
                session:execute("bridge","sofia/external/2475%
91.xxx.xx.xx")
        end
end

session:hangup()

Any help with this greatly appreciated it is driving me nuts.

regards, John Carpenter
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From peter.olsson at visionutveckling.se  Sat Jan 22 15:52:34 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sat, 22 Jan 2011 13:52:34 +0100
Subject: [Freeswitch-users] RFC 2833 DTMF detection - has
	something	changed?
In-Reply-To: <AANLkTi=-d0PnWRdjh7RHJAAaCHqj9DRGGww0+N6_1whB@mail.gmail.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C57ED00ADDA@cooper>
	<9CCA8E46-03D7-4F13-B6BD-8AE8D2564006@freeswitch.org>
	<AANLkTi=yhXDV7yE-BRpGuYU8rQwwMZU85VYBd87Cbo-+@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57EC81C52A@cooper>
	<AANLkTimXXzVCFweC6x18yrNCf8_st9a_eRD8mxAmyega@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57ED00B44B@cooper>,
	<AANLkTi=-d0PnWRdjh7RHJAAaCHqj9DRGGww0+N6_1whB@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57EC81C54B@cooper>

Ok, I've updated FS-2971 to update esl-lib files as well.

Thanks,

Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Anthony Minessale [anthony.minessale at gmail.com]
Skickat: den 21 januari 2011 21:39
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] RFC 2833 DTMF detection - has something    changed?

the new one will go in today.

as for the other 2 you mentioned:

The first two are applied, the last one needs one bit more of work.



On Fri, Jan 21, 2011 at 4:06 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> Thanks again, I tried the change below but it didn't help - however, it gave me a clue for what's going on :)
>
> The problem seems to be that do_flush() will flush valid DTMF packets, and since do_flush() is called every time a file start/stops playing (because of rtp break is called) it is quite possible to be unlucky and miss DTMF's during this short period of time.
>
> I've created a patch for this in jira FS-3002, it refactors the RFC2833 detection code, and after trying this out for the last 24 hours, the problems seems to be solved. Please review and apply this patch if it seems ok to you (or get back to me if you have any thoughts about the implemenation). More detailed information can be found in the jira case.
>
> While you're at it - I have a few other patches laying around in jira - if you could have a look at those also I would really appreciate it :)
>
> FS-2917, FS-2973 and FS-2971
>
> Thanks,
>
> Peter Olsson
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale
> Skickat: den 19 januari 2011 19:22
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] RFC 2833 DTMF detection - has something changed?
>
> comment line 2993 and see if its better.
> If it works better, what is on the other end of the call?, I hate it already.
>
>
> On Wed, Jan 19, 2011 at 12:12 PM, Peter Olsson
> <peter.olsson at visionutveckling.se> wrote:
>> Yep, I'm on today's git, so I do have the last change/fix that was made for this in switch.rtp.c.
>>
>> /Peter
>> ________________________________________
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Steven Ayre [steveayre at gmail.com]
>> Skickat: den 19 januari 2011 18:33
>> Till: FreeSWITCH Users Help
>> ?mne: Re: [Freeswitch-users] RFC 2833 DTMF detection - has something    changed?
>>
>> Brian, he says he's on Git from today...
>>
>>
>> On 19 January 2011 17:10, Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
>> Update we did have one day where it was messed up.
>>
>> /b
>>
>> On Jan 19, 2011, at 11:02 AM, Peter Olsson wrote:
>>
>> Hi,
>>
>> I have some problems with DTMF detection, which I've never seen earlier versions. I know that there was a bug in the rtp code a few days (or week(s)?) back, but I'm running on latest git (as of today). I'm also running on the same machine as before, and more or less no config changes (except for a few dialplan changes, but nothing that changes DTMF detection).
>>
>> Anyway, it seems to me that the 2833-detection is not as accurate as it was before. When using a couple of months old FS version I rarely missed any DTMF's (I can't remember I ever did..:)), but now it seems to happen once in a while. Also, I decided today to get wireshark up and running, and after 5 DTMF's, FS missed the last one. I looked inside my wireshark dump, and I could clearly see all DTMF packets in there, but FS somehow missed this.
>>
>> Is there some kind of debugging I could enable, for instance DEBUG_2833 directive?
>>
>> What's the best way to move this forward? I can send my wireshark dump if that helps, but I guess you will need some more debugging info from FS as well?
>>
>> Thanks,
>>
>> Peter Olsson
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4d39f06632762119121577!



From azatek0 at gmail.com  Sat Jan 22 17:01:41 2011
From: azatek0 at gmail.com (Aza Tek)
Date: Sat, 22 Jan 2011 16:01:41 +0200
Subject: [Freeswitch-users] GSM/PSTN Gateways
In-Reply-To: <AANLkTi=t8_ddWNcabFsJRiZh1M9_TUPDjARG3CwVXJ=b@mail.gmail.com>
References: <0F8036C9-D129-419D-AAA0-C8EA3856D7DD@gmail.com>
	<4D384C4C.60309@puzzled.xs4all.nl>
	<AANLkTi=t8_ddWNcabFsJRiZh1M9_TUPDjARG3CwVXJ=b@mail.gmail.com>
Message-ID: <AANLkTi=dGnshYzXEux19DmdWqoqj3P2ode+NJ8V8u6fK@mail.gmail.com>

Are any of these GSM Gateways capable of passing the original caller's ID?

Thanks


 A more premium solution would be 2N at http://www.2n.cz/en/
>> These work very well and in my experience are reliable.
>>
>> Portech also have external boxes:
>> http://www.portech.com.tw/p3-product1.asp?Cid=6
>>
>> Haven't used them myself but someone in #freenode once mentioned
>> Mobigator, a GSM gateway compatible with FreeSWITCH. Uses a simple USB
>> link to hook up to your freeswitch server. On the software side I think
>> you need to install gsmopen on your FreeSWITCH box (howto on wiki).
>>
>> http://www.mobigater.com/index.php?p=2&s=4
>> http://www.mobigater.com/index.php?p=2&s=6
>>
>> At about $120 or ?90 they are quite affordable but I don't know how well
>> they work.
>>
>> Regards,
>> Patrick
>>
>>
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From steveayre at gmail.com  Sat Jan 22 18:29:54 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 22 Jan 2011 15:29:54 +0000
Subject: [Freeswitch-users] GSM/PSTN Gateways
In-Reply-To: <AANLkTi=dGnshYzXEux19DmdWqoqj3P2ode+NJ8V8u6fK@mail.gmail.com>
References: <0F8036C9-D129-419D-AAA0-C8EA3856D7DD@gmail.com>
	<4D384C4C.60309@puzzled.xs4all.nl>
	<AANLkTi=t8_ddWNcabFsJRiZh1M9_TUPDjARG3CwVXJ=b@mail.gmail.com>
	<AANLkTi=dGnshYzXEux19DmdWqoqj3P2ode+NJ8V8u6fK@mail.gmail.com>
Message-ID: <100216A3-30F7-4723-91F5-5C3EDA1DB3AE@gmail.com>

No, when you make a call from a SIM the phone network only sees the SIM's phone number. You can either send that or withhold it, but not send the other leg's caller id.

Steve on iPhone

On 22 Jan 2011, at 14:01, Aza Tek <azatek0 at gmail.com> wrote:

> Are any of these GSM Gateways capable of passing the original caller's ID?
> 
> Thanks
>  
> 
> A more premium solution would be 2N at http://www.2n.cz/en/
> These work very well and in my experience are reliable.
> 
> Portech also have external boxes:
> http://www.portech.com.tw/p3-product1.asp?Cid=6
> 
> Haven't used them myself but someone in #freenode once mentioned
> Mobigator, a GSM gateway compatible with FreeSWITCH. Uses a simple USB
> link to hook up to your freeswitch server. On the software side I think
> you need to install gsmopen on your FreeSWITCH box (howto on wiki).
> 
> http://www.mobigater.com/index.php?p=2&s=4
> http://www.mobigater.com/index.php?p=2&s=6
> 
> At about $120 or ?90 they are quite affordable but I don't know how well
> they work.
> 
> Regards,
> Patrick
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From infos at madovsky.org  Sat Jan 22 19:44:35 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 22 Jan 2011 11:44:35 -0500
Subject: [Freeswitch-users] praise of freeSWITCH developers
References: <9FE96265969E4BAEB801002D975CC50F@e1705>
	<4D3AAA5F.4020303@xpirio.com>
Message-ID: <59F933FF4FAE4FB49AC06C7C5379A6C1@e1705>

Thanks Chris
and all others for your contribution.
I have always a sentence in my mind that says :
"help people you love, love will help you in return"....
or "give wood to the fire to help it stay alive"....
Anthony, maybe a barbecue in august would be a good idea ? :D


----- Original Message ----- 
From: "Christian L?schenkohl" <christian.loeschenkohl at xpirio.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, January 22, 2011 4:58 AM
Subject: Re: [Freeswitch-users] praise of freeSWITCH developers


hello

these words are more than true.
100$ from my side are on the way.

br

On 2011-01-21 17:13, Madovsky wrote:

> I would like to remember to all people on this
> emailist that all freeSWITCH guys work very hard
> to update/upgrade and make FS more powerful, and
> also to answer to hundreds of user emails every day.
> Donations are welcome and appreciated.
> It helps the humans behind the machine
> who give to you the opportunity to create and use amazing projects
> and to continue to make this open source project alive and active.
> Regards
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 5 77 11 - 1000
F  +43 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 



From rupa at rupa.com  Sat Jan 22 19:54:20 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 22 Jan 2011 10:54:20 -0600
Subject: [Freeswitch-users] conference list
In-Reply-To: <A4B3B1FD7AFE4809A32A69A49F37943E@e1705>
References: <DBCC446AC5A943F1AEB555BF1707ED6C@e1705>
	<AANLkTi=zp_5gMY51Puc-4bpmJ0duZtAZnYM9Vw0t_xF1@mail.gmail.com>
	<021F1447C64046159BEE423C61E947A0@e1705>
	<AANLkTinYCtPT8wzgGXaV8GUScGwCdRv9O4OWJNc7yi+=@mail.gmail.com>
	<4FE385B18ADC4C97B67EC94E8CEA278B@e1705>
	<AANLkTi=qmqiTUrkiqzhFqz2gh568Dq=YscEmXguZgaQh@mail.gmail.com>
	<A4B3B1FD7AFE4809A32A69A49F37943E@e1705>
Message-ID: <AANLkTi=E5d3AU2Q1AzxaU2dfnUq+2RVhbnpKbpdN1drP@mail.gmail.com>

well, yeah. but parsing xml is pretty straightfoward.  Most of the
apis give more detailed info via xml...

On Fri, Jan 21, 2011 at 12:59 PM, Madovsky <infos at madovsky.org> wrote:
> Ah ok right,
> but if I call conference conf_name list xml_list
> from dialplan I haave to parse all the xml code to get the var is_moderator
> isn't it ?
>
> Thanks
>
> ----- Original Message -----
> From: "Rupa Schomaker" <rupa at rupa.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Friday, January 21, 2011 1:38 PM
> Subject: Re: [Freeswitch-users] conference list
>
>
>> conference, ? ? ? ? ? ? <confname> list [delim <string>]
>> ----> ? ? ? ? ? ? ? ?<confname> xml_list
>>
>> use xml_list instead of list
>>
>> On Fri, Jan 21, 2011 at 12:27 PM, Madovsky <infos at madovsky.org> wrote:
>>> Rupa,
>>>
>>> sorry I don't understand how to use xml_list.
>>> it isn't exist as api and as dialplan application
>>>
>>> Thanks
>>>
>>>
>>>
>>> ----- Original Message -----
>>> From: "Rupa Schomaker" <rupa at rupa.com>
>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>> Sent: Friday, January 21, 2011 1:07 PM
>>> Subject: Re: [Freeswitch-users] conference list
>>>
>>>
>>>> does the xml list have what you want?
>>>>
>>>> On Fri, Jan 21, 2011 at 12:13 AM, Madovsky <infos at madovsky.org> wrote:
>>>>> sorry when I say csv it's the csv like list when
>>>>> you type api conference list
>>>>> but there is only flags as speak|hear|floor,
>>>>> the flags-member is not listed
>>>>>
>>>>> ----- Original Message -----
>>>>> From: "Rupa Schomaker" <rupa at rupa.com>
>>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>> Sent: Friday, January 21, 2011 1:08 AM
>>>>> Subject: Re: [Freeswitch-users] conference list
>>>>>
>>>>>
>>>>>> Dunno about csv, but xml_list has a flags node for flags for each
>>>>>> member.
>>>>>>
>>>>>> On Thu, Jan 20, 2011 at 8:50 PM, Madovsky <infos at madovsky.org> wrote:
>>>>>>> Sorry Brian, it's my last question before next week ;)
>>>>>>>
>>>>>>> it's more a suggestion that a question,
>>>>>>> when use api conference myconf list command
>>>>>>> maybe it would be useful to know in the csv list
>>>>>>> which channel is moderator etc...
>>>>>>>
>>>>>>> Thanks
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> -Rupa
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa


From david.ponzone at ipeva.fr  Sat Jan 22 20:01:53 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sat, 22 Jan 2011 18:01:53 +0100
Subject: [Freeswitch-users] GSM/PSTN Gateways
In-Reply-To: <AANLkTi=dGnshYzXEux19DmdWqoqj3P2ode+NJ8V8u6fK@mail.gmail.com>
References: <0F8036C9-D129-419D-AAA0-C8EA3856D7DD@gmail.com>
	<4D384C4C.60309@puzzled.xs4all.nl>
	<AANLkTi=t8_ddWNcabFsJRiZh1M9_TUPDjARG3CwVXJ=b@mail.gmail.com>
	<AANLkTi=dGnshYzXEux19DmdWqoqj3P2ode+NJ8V8u6fK@mail.gmail.com>
Message-ID: <0CA26810-57FC-4FBD-94D1-43365C1E66FD@ipeva.fr>

The issue is not at the gateway level, but at the carrier level.
No GSM carrier, to my knowledge, will let you send the CLID you want.
I am not even sure if the GSM protocol allows the terminal to send a CLID.
I suspect it only sends its IMEI, and that the network adds the CLID to the call.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 22/01/2011 ? 15:01, Aza Tek a ?crit :

> Are any of these GSM Gateways capable of passing the original caller's ID?
> 
> Thanks
>  
> 
> A more premium solution would be 2N at http://www.2n.cz/en/
> These work very well and in my experience are reliable.
> 
> Portech also have external boxes:
> http://www.portech.com.tw/p3-product1.asp?Cid=6
> 
> Haven't used them myself but someone in #freenode once mentioned
> Mobigator, a GSM gateway compatible with FreeSWITCH. Uses a simple USB
> link to hook up to your freeswitch server. On the software side I think
> you need to install gsmopen on your FreeSWITCH box (howto on wiki).
> 
> http://www.mobigater.com/index.php?p=2&s=4
> http://www.mobigater.com/index.php?p=2&s=6
> 
> At about $120 or ?90 they are quite affordable but I don't know how well
> they work.
> 
> Regards,
> Patrick
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From infos at madovsky.org  Sat Jan 22 20:03:49 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 22 Jan 2011 12:03:49 -0500
Subject: [Freeswitch-users] conference list
References: <DBCC446AC5A943F1AEB555BF1707ED6C@e1705><AANLkTi=zp_5gMY51Puc-4bpmJ0duZtAZnYM9Vw0t_xF1@mail.gmail.com><021F1447C64046159BEE423C61E947A0@e1705><AANLkTinYCtPT8wzgGXaV8GUScGwCdRv9O4OWJNc7yi+=@mail.gmail.com><4FE385B18ADC4C97B67EC94E8CEA278B@e1705><AANLkTi=qmqiTUrkiqzhFqz2gh568Dq=YscEmXguZgaQh@mail.gmail.com><A4B3B1FD7AFE4809A32A69A49F37943E@e1705>
	<AANLkTi=E5d3AU2Q1AzxaU2dfnUq+2RVhbnpKbpdN1drP@mail.gmail.com>
Message-ID: <1C16569015964371B37B93C570FC4D4F@e1705>

true, I can also easily use any PHP or Perl xml parser if I call the api
from shell.

----- Original Message ----- 
From: "Rupa Schomaker" <rupa at rupa.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, January 22, 2011 11:54 AM
Subject: Re: [Freeswitch-users] conference list


well, yeah. but parsing xml is pretty straightfoward.  Most of the
apis give more detailed info via xml...

On Fri, Jan 21, 2011 at 12:59 PM, Madovsky <infos at madovsky.org> wrote:
> Ah ok right,
> but if I call conference conf_name list xml_list
> from dialplan I haave to parse all the xml code to get the var 
> is_moderator
> isn't it ?
>
> Thanks
>
> ----- Original Message -----
> From: "Rupa Schomaker" <rupa at rupa.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Friday, January 21, 2011 1:38 PM
> Subject: Re: [Freeswitch-users] conference list
>
>
>> conference, <confname> list [delim <string>]
>> ----> <confname> xml_list
>>
>> use xml_list instead of list
>>
>> On Fri, Jan 21, 2011 at 12:27 PM, Madovsky <infos at madovsky.org> wrote:
>>> Rupa,
>>>
>>> sorry I don't understand how to use xml_list.
>>> it isn't exist as api and as dialplan application
>>>
>>> Thanks
>>>
>>>
>>>
>>> ----- Original Message -----
>>> From: "Rupa Schomaker" <rupa at rupa.com>
>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>> Sent: Friday, January 21, 2011 1:07 PM
>>> Subject: Re: [Freeswitch-users] conference list
>>>
>>>
>>>> does the xml list have what you want?
>>>>
>>>> On Fri, Jan 21, 2011 at 12:13 AM, Madovsky <infos at madovsky.org> wrote:
>>>>> sorry when I say csv it's the csv like list when
>>>>> you type api conference list
>>>>> but there is only flags as speak|hear|floor,
>>>>> the flags-member is not listed
>>>>>
>>>>> ----- Original Message -----
>>>>> From: "Rupa Schomaker" <rupa at rupa.com>
>>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>> Sent: Friday, January 21, 2011 1:08 AM
>>>>> Subject: Re: [Freeswitch-users] conference list
>>>>>
>>>>>
>>>>>> Dunno about csv, but xml_list has a flags node for flags for each
>>>>>> member.
>>>>>>
>>>>>> On Thu, Jan 20, 2011 at 8:50 PM, Madovsky <infos at madovsky.org> wrote:
>>>>>>> Sorry Brian, it's my last question before next week ;)
>>>>>>>
>>>>>>> it's more a suggestion that a question,
>>>>>>> when use api conference myconf list command
>>>>>>> maybe it would be useful to know in the csv list
>>>>>>> which channel is moderator etc...
>>>>>>>
>>>>>>> Thanks
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> -Rupa
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 



From mitch.capper at gmail.com  Sat Jan 22 21:52:13 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Sat, 22 Jan 2011 10:52:13 -0800
Subject: [Freeswitch-users] Portaudio Improvements
In-Reply-To: <AANLkTi=T6a9nU54toi5+fMA7kq_xkmZrRnRT3Fa8Pao5@mail.gmail.com>
References: <AANLkTim9AmtEGZoK0NyG1yKVsmJ_MSwXAppi-6AcCqnz@mail.gmail.com>
	<AANLkTimm69f6adgkgDz03GUOq8xJw_oaVvrkAGxd1yjG@mail.gmail.com>
	<AANLkTikk==DKhXft67HWcj0ytya030bS=4vtt2o7+tD8@mail.gmail.com>
	<AANLkTi=T6a9nU54toi5+fMA7kq_xkmZrRnRT3Fa8Pao5@mail.gmail.com>
Message-ID: <AANLkTimbULX-_ne4ZyirMFxcOcRBaEZwxTfude6=L8F_@mail.gmail.com>

It is true PA lacks volume control, it had been something I was debating.
Adding gain is pretty simple as you are for the most part just multiplying
up the samples I believe so would not be a major item to add to portaudio, I
however decided not to go this route for two reasons.  1) In Windows each
application has its own volume setting (Vista and higher),  and apps are
starting to tie into this than internal volume controls.  The downside is it
doesn't distort audio/have a gain option.   This is when I thought about
allowing for a fixed gain amount added to portaudio.  2) Freeswitch actually
has its own built in volume control settings with set_audio_level.
set_audio_level allows for 9 settings (4 lower 4 higher) and seems to do a
good job at digital volume control.
Avoiding modifying the audio stream itself could also result in less
distortion when using actual volume controls (headsets, etc).

So if you want volume control I would say take a look at set_audio_level as
it may be the simplest method.   Adding it to portaudio would work but one
of the nice advantages of set_audio_level is the fact you can also do it per
channel rather than globally:)

~Mitch

On Fri, Jan 21, 2011 at 5:44 PM, Seven Du <dujinfang at gmail.com> wrote:

> In addition to AEC. I also found PA lacks the ability to change the
> sound device volume. I tried to control sound devices in QT but
> haven't find a way. Ideas on this?
>
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From jmesquita at freeswitch.org  Sat Jan 22 22:27:00 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 22 Jan 2011 16:27:00 -0300
Subject: [Freeswitch-users] Run dialplan tools from event socket
In-Reply-To: <AANLkTimDtnM10Vu5=gFMa_RUg_somgWH6r4TkcWb6h5O@mail.gmail.com>
References: <AANLkTikg5+1aMoWxYr3UYO13iO2qA6FsG9FFi-Zg3_bD@mail.gmail.com>
	<AANLkTin5fn9qMre-8xFAZD5_SUzbLbp_YueTRPPyh-=K@mail.gmail.com>
	<AANLkTin84SkDdDsBE0wfU2QnG025bNK8BhcW=JDvz9gv@mail.gmail.com>
	<AANLkTi=TX6BURrxR38X9bwyUai_52xLK80kMvm9CmZ8T@mail.gmail.com>
	<AANLkTikDBWtCTp=h0t_Qxtm1S+ZTX_UUtvj5RgbFdt+g@mail.gmail.com>
	<AANLkTinh7cQp5H9OOv1JsAvYj=PUteS0MtgTXXA3ZvNa@mail.gmail.com>
	<AANLkTimDtnM10Vu5=gFMa_RUg_somgWH6r4TkcWb6h5O@mail.gmail.com>
Message-ID: <AANLkTim3itY9aADKhz0E+mZ41mFD8s+8Kg9RQDDu2y0=@mail.gmail.com>

Ok, it is oficial now... It "kinda" works with uuid_broadcast ...

When you hangup the channel that initiated the transfer, the peer_channel
gets hungup as well, therefore, no luck with the transfer because all 3
channels get hungup. The rest works fine..

I compared the channel flags from the working and the non-working examples
and I found some interesting things. The failed call lacks the following
flags:

CF_BRIDGED
CF_CONTROLLED
CF_PARK

I am guessing that the important ones that make everything go to hell at the
end is the CF_BRIDGED because apparently, the lack of it means that the
system can let the session go. I don't know what CF_CONTROLLED is and the
lack of CF_PARK really messes up the MOH. You get the channel muted when you
initiate the transfer.

I am investigating more and will let you guys know as I move on.

Regards,
Jo?o Mesquita


2011/1/21 Fraser Redmond <fraserredmond at gmail.com>

> Awesome, thanks Steve and Jo?o!
>
> (Got called away to work on something else today, but will try it
> tomorrow.)
>
> Cheers,
> Fraser
>
>
>
>
>
> 2011/1/21 Jo?o Mesquita <jmesquita at freeswitch.org>
>
>> You hit the right spot... That's what you need to do, we don't need an
>> API, it already exists. I've tested and it works like a charm.
>>
>> Regards,
>> Jo?o Mesquita
>>
>>
>>
>> On Fri, Jan 21, 2011 at 10:16 AM, Steven Ayre <steveayre at gmail.com>wrote:
>>
>>> uuid_broadcast appears to have support for executing a dialplan app from
>>> the api (it uses say as an example):
>>>
>>> uuid_broadcast <uuid> app!::args [aleg|bleg|both]
>>>
>>> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast
>>>
>>> I haven't tried it myself though.
>>>
>>> -Steve
>>>
>>>
>>> 2011/1/21 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>
>>> Actually, Fraser, I think this won't work...
>>>>
>>>> att_xfer uses the signal_bond variable to get the leg connected to the
>>>> party being transferred. This variable is unset when you park the legs or
>>>> break the bridge in any way. Maybe I can make a API command that does
>>>> att_xfer taking the trasferred leg UUID as the transferred party? Do me a
>>>> favor, make some tests and I will take a deeper look at the att_xfer
>>>> application code.
>>>>
>>>> Regards,
>>>> Jo?o Mesquita
>>>>
>>>>
>>>>
>>>> On Fri, Jan 21, 2011 at 1:37 AM, Fraser Redmond <
>>>> fraserredmond at gmail.com> wrote:
>>>>
>>>>> Thanks Jo?o.
>>>>>
>>>>> My transfer_call extension runs a couple of js scripts to get and
>>>>> validate the number to transfer to, then does
>>>>>                 <action application="att_xfer" data="${dial_string}"/>
>>>>> (and it has a couple of steps after that to handle failed transfers.)
>>>>>
>>>>> So could I use ESL's execute command to run the execute_extension? Not
>>>>> sure how I missed that option in the wiki. I"ll give it a try, see what
>>>>> happens.
>>>>>
>>>>>
>>>>> I forgot to say in the original post, but execute_extension seems to be
>>>>> particularly nice for this use-case, as it falls back through the dialplan
>>>>> gracefully if there's a problem.
>>>>>
>>>>> Cheers,
>>>>> Fraser
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> 2011/1/20 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>>>
>>>>> Lucky for you I have been working on this lately and the bad news is
>>>>>> ... there's no easy way to do it....
>>>>>>
>>>>>> You can execute an extension like you said, but you have to park the
>>>>>> legs first... It would help to know how's the transfer_call extension so
>>>>>> that I can try to help you out, but maybe it is easier if you think of it
>>>>>> this way:
>>>>>>
>>>>>> When you use an app like att_xfer, the core already knows what to do
>>>>>> next with a call and parks the legs for you. If you do it on ESL, you've
>>>>>> done it half way and you didn't really park anything before you transfered
>>>>>> the call. When the bridge is undone, the leg that was not transfered doesn't
>>>>>> know what to do, has no applications to be run at this moment and so all
>>>>>> it's left for it is to let go.
>>>>>>
>>>>>> A bit clearer? Att_xfer is a bit of a pain in the butt and it kinda
>>>>>> requires you to know a bit more of the inner workings of the state machine.
>>>>>>
>>>>>> You can always execute att_xfer using ESL's execute (
>>>>>> http://wiki.freeswitch.org/wiki/Event_Socket_Library#execute) if you
>>>>>> don't care what happens to the legs afterwards but if you want to have
>>>>>> control over all 3 legs, no luck for you...
>>>>>>
>>>>>> Regards,
>>>>>> Jo?o Mesquita
>>>>>>
>>>>>>
>>>>>> On Fri, Jan 21, 2011 at 12:13 AM, Fraser Redmond <
>>>>>> fraserredmond at gmail.com> wrote:
>>>>>>
>>>>>>> Is there any way to run a dialplan tool from the event socket?
>>>>>>>
>>>>>>> I have a dialplan that uses a dtmf to set up and perform an attended
>>>>>>> transfer, like so:
>>>>>>> <action application="bind_meta_app" data="8 a s
>>>>>>> execute_extension::TransferCall XML transfer_call"/>
>>>>>>>
>>>>>>> But I can't see any way to run the same thing from the event socket.
>>>>>>> I thought doing an  "api uuid_transfer"  might do it, but that hangs up one
>>>>>>> of the legs (no good for attended transfer.)
>>>>>>>
>>>>>>> api uuid_transfer Uuid -bleg TransferCall XML transfer_call
>>>>>>>
>>>>>>> As far as I can see, the closest thing is "sendmsg execute", but it
>>>>>>> looks like you have to park a call/channel first to use that, so I'm not
>>>>>>> sure that that is much use for attended transfer either.
>>>>>>>
>>>>>>> Or should I be lame and do "api uuid_send_dtmf" to send * 8.
>>>>>>>
>>>>>>> Cheers,
>>>>>>> Fraser
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
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>> FreeSWITCH-users at lists.freeswitch.org
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>
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From chat2jesse at gmail.com  Sun Jan 23 02:01:41 2011
From: chat2jesse at gmail.com (jesse)
Date: Sat, 22 Jan 2011 15:01:41 -0800
Subject: [Freeswitch-users] is it possible to create a session between
	two extensions?
In-Reply-To: <AANLkTinJjW7pt86ySZbJBYk44kQb141-f1pkGdPggFK=@mail.gmail.com>
References: <AANLkTimjzAS1WZ86GjwZHtyqVFV6NMZ7Ggd9uD-NjxZd@mail.gmail.com>
	<AANLkTinJjW7pt86ySZbJBYk44kQb141-f1pkGdPggFK=@mail.gmail.com>
Message-ID: <AANLkTinkVynJAPc443NX_KTJUn3vLUADb0-8GD1HKjnB@mail.gmail.com>

thanks for the tip!

On Sat, Jan 22, 2011 at 1:31 AM, Seven Du <dujinfang at gmail.com> wrote:
> originate loopback/2333 23334
>
> On Sat, Jan 22, 2011 at 1:32 PM, jesse <chat2jesse at gmail.com> wrote:
>> In FS, a session is connection between one extension and sip
>> internal/external ?user.
>>
>> suppose I defined two extensions in diaplan/default.xml one is 2333,
>> the other is 23334. (both are NOT locally registered endpoints)
>>
>> I am wondering whether it is possible to create a session with two extensions?
>> like : original 2333-XML-default 2334-XML-default
>>
>> -jesse
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:? http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From jeff at jefflenk.com  Sun Jan 23 02:18:52 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sat, 22 Jan 2011 15:18:52 -0800 (PST)
Subject: [Freeswitch-users] Iksemel msvs compiling
In-Reply-To: <SNT109-DS1063277B95F0AC68C084EDC1FB0@phx.gbl>
References: <1294535404901-5903534.post@n2.nabble.com>
	<SNT109-DS14B90C48B33126784F294AC10E0@phx.gbl>
	<1294673236806-5907287.post@n2.nabble.com>
	<SNT109-DS227CE87422187AA91545CFC1F30@phx.gbl>
	<1295113990176-5925152.post@n2.nabble.com>
	<SNT109-DS2124FB8719BEF6F213D36DC1F40@phx.gbl>
	<1295276746914-5932097.post@n2.nabble.com>
	<SNT109-DS6447C073B028EC7D60228C1F70@phx.gbl>
	<1295320482333-5934373.post@n2.nabble.com>
	<SNT109-DS1063277B95F0AC68C084EDC1FB0@phx.gbl>
Message-ID: <1295738332589-5951821.post@n2.nabble.com>


Hi Norman,

I'm glad you figured it out!

It seems a little strange that the library does not properly define ssize_t
but maybe the library is built with other tools/compilers under windows(not
sure doesnt really matter to me). 

You should be able to add your notes to the wiki - just create an account
for the wiki.

-Jeff
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Iksemel-msvs-compiling-tp5891263p5951821.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From infos at madovsky.org  Sun Jan 23 02:22:21 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 22 Jan 2011 18:22:21 -0500
Subject: [Freeswitch-users] mod_conference with cluster ODBC
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705><AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com><AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com><AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com><AANLkTimutwOyeubcS=tax5EbDEniCD+Nc4iHMydQxpFb@mail.gmail.com><37761574FEE44D13BEBE4A7DE0089083@e1705><AANLkTi=VZ8JZ+PuyXoz_EYNVovY-AOuPyz9+yfKoUc3k@mail.gmail.com><337274A8CA2343259DF18D8135AF2DCE@e1705><AANLkTimVaxdbwVHCTqpYC5w3q+Vi+t7-oMekrkxu4Kfd@mail.gmail.com><5B0F4D23C27A47A08D36C36B78AB981B@e1705><A7B3B85184604E1DB648F9AD0F99CD7C@stor1>
	<AANLkTi=4EM2gs+viY6gFY+qa=CGQ4dv7vMio6gj6Hi6L@mail.gmail.com>
Message-ID: <9FA0F047DF024E66B6CFCB23FA8E5CEA@e1705>

If I create an ESL deamon with for example Perl,
it means that every event from FS will be sent to this daemon
so I only need to catch the right ones and do the job ?

Thanks
  ----- Original Message ----- 
  From: Jo?o Mesquita 
  To: FreeSWITCH Users Help 
  Sent: Friday, January 07, 2011 5:29 PM
  Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


  Even tho both of your approaches are good for a "small" scale system, I think we are missing the point here.


  If you don't need to have several conferences bridged because you don't lack machine power to hold the conference onto one server, you can use ESL to make an INVITE and then a REPLACES. If I am not mistaken, you are able to use the uuid_simplify command to make the replaces after the bridge is done. Although, most people looking to have multiple conferences on multiple servers are looking for scalability where you can have one single (or multiple) conferences spread over several boxes that can even be geographically spread out look like a single conference to the user and/or systems involved.


  This is the real challenge and that might be worth thinking about hacking C, the rest is just dialplan and a bit of ESL to make the control. The first one is way out of my league.

  Jo?o Mesquita



  On Fri, Jan 7, 2011 at 6:43 PM, Kris <kris at livecall.com> wrote:

    Just an idea..soon I will have to put people that are answered on multiple
    servers into the same conference. I am thinking about having a table on the
    central SQLServer like this: ConferenceName, ServerName. . I would lookup
    the server a particular conference is on and then transfer the caller to
    that server and extension that will put the caller into the appropriate
    conference (dial something.. at SERVER)- I guess. I've seen the export word
    that maybe the way to pass on variables to the other server such as the
    ConferenceName, UserName

    Then the server hosting the conference will have an extension that has the
    forums profile and controls
         <action application="conference" data="${ConferenceName}@forums"/>
    That way all the users are in the same conference and can be controlled
    there instead of having only one link to a bunch of callers on another
    server.

    If you get it going, could you email the dial strings, extensions you
    used.etc.I am curious..
    Kris


    ----- Original Message -----
    From: "Madovsky" <infos at madovsky.org>
    To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

    Sent: Friday, January 07, 2011 10:28 AM
    Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


    I got it thanks,
    but do you think it would be more interesting to reduce
    bandwidth and latency between nodes and centralize the conference on one
    node only
    by transferring the incoming user to the right node ?
     ----- Original Message -----
     From: Jo?o Mesquita
     To: FreeSWITCH Users Help
     Sent: Thursday, January 06, 2011 4:33 PM
     Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


     Ok, let me see if I can get this into your head. (giggles)


     A conference means that the audio needs to mixed in together so that all
    participants can talk/hear each other, right? If you implement something in
    C on mod_conference, you are going to essentially do the same as what an ESL
    app does. You _need_ to call in from one server to the other so that you can
    mix the audio of all the participants. The real advantage would be the
    management API being only one for everything and the challenge is exactly
    that. How to mute certain users on a conference that is spanning over 10
    servers or deaf them, etc...


     A SIP "user" is easier because you don't have to bridge audio from another
    server necessarily. Got it?


     Regards,
     Jo?o Mesquita



     On Thu, Jan 6, 2011 at 6:25 PM, Madovsky <infos at madovsky.org> wrote:

       Rupa,

       I don't want bother anyone with this thread but why not
       to manage conference as SIP user ?
       if someone from server A call an other who is registered on server B, so
       FS do it automatically, why not with conference ? Or maybe create a
    param
       in mod_conference that let the choice of the admin to manage unique name
    in
       all cluster or not.
       like <param name="cluster_unique_name" value="true"/>
       I will try to understand the C code to hack something like this...


       ----- Original Message -----
       From: "Rupa Schomaker" <rupa at rupa.com>
       To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

       Sent: Thursday, January 06, 2011 3:01 PM
       Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC


       Yes

       On Thu, Jan 6, 2011 at 1:40 PM, Madovsky <infos at madovsky.org> wrote:
       > in case of you have 8 servers you have to do it for each ?
       >
       > Thanks
       >
       > ----- Original Message -----
       > From: joy this
       > To: FreeSWITCH Users Help
       > Sent: Thursday, January 06, 2011 2:51 AM
       > Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
       > It works. Thank you everyone.
       >
       > 2011/1/5 Rupa Schomaker <rupa at rupa.com>
       >>
       >> Use the api: conference <confname> dial [{dial string
       >> options}]<endpoint_module_name>/<destination> [<callerid_number>
       >> [<callerid_name>]]
       >> To initiate the call from within conference A on server 1. Have a
       >> corresponding dialplan entry on server 2 to accept the call and add
    it
       >> into
       >> the conference A on server 2. You've now bridged the two conferences
    in
       >> the
       >> two servers.
       >
       > ________________________________
       >
       > _______________________________________________
       > FreeSWITCH-users mailing list
       > FreeSWITCH-users at lists.freeswitch.org
       > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
       >
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
       > http://www.freeswitch.org
       >
       > --
       > This message has been scanned for viruses and
       > dangerous content by MailScanner, and is
       > believed to be clean.
       > _______________________________________________
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       --
       -Rupa

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From jmesquita at freeswitch.org  Sun Jan 23 04:21:17 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 22 Jan 2011 22:21:17 -0300
Subject: [Freeswitch-users] mod_conference with cluster ODBC
In-Reply-To: <9FA0F047DF024E66B6CFCB23FA8E5CEA@e1705>
References: <46549E68636443D3BD6AA2E90AE5A86A@e1705>
	<AANLkTimHxMhJz94DHc0=obV+wZG5byqUHnMh90uwMxhH@mail.gmail.com>
	<AANLkTinEPb01tyVheoAyfdEYmp6L+vMj7nwSqi8tnRxR@mail.gmail.com>
	<AANLkTi=FA7AZvOd6REepNZkZ4+U3jgMnHDxPE0ocDc17@mail.gmail.com>
	<AANLkTimutwOyeubcS=tax5EbDEniCD+Nc4iHMydQxpFb@mail.gmail.com>
	<37761574FEE44D13BEBE4A7DE0089083@e1705>
	<AANLkTi=VZ8JZ+PuyXoz_EYNVovY-AOuPyz9+yfKoUc3k@mail.gmail.com>
	<337274A8CA2343259DF18D8135AF2DCE@e1705>
	<AANLkTimVaxdbwVHCTqpYC5w3q+Vi+t7-oMekrkxu4Kfd@mail.gmail.com>
	<5B0F4D23C27A47A08D36C36B78AB981B@e1705>
	<A7B3B85184604E1DB648F9AD0F99CD7C@stor1>
	<AANLkTi=4EM2gs+viY6gFY+qa=CGQ4dv7vMio6gj6Hi6L@mail.gmail.com>
	<9FA0F047DF024E66B6CFCB23FA8E5CEA@e1705>
Message-ID: <AANLkTinPUJj5ZWybS3cFLtzNWXtKfw8=mg3mNE821qpO@mail.gmail.com>

Yes, you got the hang of it.

Regards,
Jo?o Mesquita


On Sat, Jan 22, 2011 at 8:22 PM, Madovsky <infos at madovsky.org> wrote:

>  If I create an ESL deamon with for example Perl,
> it means that every event from FS will be sent to this daemon
> so I only need to catch the right ones and do the job ?
>
> Thanks
>
> ----- Original Message -----
> *From:* Jo?o Mesquita <jmesquita at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Friday, January 07, 2011 5:29 PM
> *Subject:* Re: [Freeswitch-users] mod_conference with cluster ODBC
>
> Even tho both of your approaches are good for a "small" scale system, I
> think we are missing the point here.
>
> If you don't need to have several conferences bridged because you don't
> lack machine power to hold the conference onto one server, you can use ESL
> to make an INVITE and then a REPLACES. If I am not mistaken, you are able to
> use the uuid_simplify command to make the replaces after the bridge is done.
> Although, most people looking to have multiple conferences on multiple
> servers are looking for scalability where you can have one single (or
> multiple) conferences spread over several boxes that can even be
> geographically spread out look like a single conference to the user and/or
> systems involved.
>
> This is the real challenge and that might be worth thinking about hacking
> C, the rest is just dialplan and a bit of ESL to make the control. The first
> one is way out of my league.
>
> Jo?o Mesquita
>
>
> On Fri, Jan 7, 2011 at 6:43 PM, Kris <kris at livecall.com> wrote:
>
>> Just an idea..soon I will have to put people that are answered on multiple
>> servers into the same conference. I am thinking about having a table on
>> the
>> central SQLServer like this: ConferenceName, ServerName. . I would lookup
>> the server a particular conference is on and then transfer the caller to
>> that server and extension that will put the caller into the appropriate
>> conference (dial something.. at SERVER)- I guess. I've seen the export word
>> that maybe the way to pass on variables to the other server such as the
>> ConferenceName, UserName
>>
>> Then the server hosting the conference will have an extension that has the
>> forums profile and controls
>>      <action application="conference" data="${ConferenceName}@forums"/>
>> That way all the users are in the same conference and can be controlled
>> there instead of having only one link to a bunch of callers on another
>> server.
>>
>> If you get it going, could you email the dial strings, extensions you
>> used.etc.I am curious..
>> Kris
>>
>> ----- Original Message -----
>> From: "Madovsky" <infos at madovsky.org>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>  Sent: Friday, January 07, 2011 10:28 AM
>> Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
>>
>>
>> I got it thanks,
>> but do you think it would be more interesting to reduce
>> bandwidth and latency between nodes and centralize the conference on one
>> node only
>> by transferring the incoming user to the right node ?
>>  ----- Original Message -----
>>  From: Jo?o Mesquita
>>  To: FreeSWITCH Users Help
>>  Sent: Thursday, January 06, 2011 4:33 PM
>>  Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
>>
>>
>>  Ok, let me see if I can get this into your head. (giggles)
>>
>>
>>  A conference means that the audio needs to mixed in together so that all
>> participants can talk/hear each other, right? If you implement something
>> in
>> C on mod_conference, you are going to essentially do the same as what an
>> ESL
>> app does. You _need_ to call in from one server to the other so that you
>> can
>> mix the audio of all the participants. The real advantage would be the
>> management API being only one for everything and the challenge is exactly
>> that. How to mute certain users on a conference that is spanning over 10
>> servers or deaf them, etc...
>>
>>
>>  A SIP "user" is easier because you don't have to bridge audio from
>> another
>> server necessarily. Got it?
>>
>>
>>  Regards,
>>  Jo?o Mesquita
>>
>>
>>
>>  On Thu, Jan 6, 2011 at 6:25 PM, Madovsky <infos at madovsky.org> wrote:
>>
>>    Rupa,
>>
>>    I don't want bother anyone with this thread but why not
>>    to manage conference as SIP user ?
>>    if someone from server A call an other who is registered on server B,
>> so
>>    FS do it automatically, why not with conference ? Or maybe create a
>> param
>>    in mod_conference that let the choice of the admin to manage unique
>> name
>> in
>>    all cluster or not.
>>    like <param name="cluster_unique_name" value="true"/>
>>    I will try to understand the C code to hack something like this...
>>
>>
>>    ----- Original Message -----
>>    From: "Rupa Schomaker" <rupa at rupa.com>
>>    To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>
>>    Sent: Thursday, January 06, 2011 3:01 PM
>>    Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
>>
>>
>>    Yes
>>
>>    On Thu, Jan 6, 2011 at 1:40 PM, Madovsky <infos at madovsky.org> wrote:
>>    > in case of you have 8 servers you have to do it for each ?
>>    >
>>    > Thanks
>>    >
>>    > ----- Original Message -----
>>    > From: joy this
>>    > To: FreeSWITCH Users Help
>>    > Sent: Thursday, January 06, 2011 2:51 AM
>>    > Subject: Re: [Freeswitch-users] mod_conference with cluster ODBC
>>    > It works. Thank you everyone.
>>    >
>>    > 2011/1/5 Rupa Schomaker <rupa at rupa.com>
>>    >>
>>    >> Use the api: conference <confname> dial [{dial string
>>    >> options}]<endpoint_module_name>/<destination> [<callerid_number>
>>    >> [<callerid_name>]]
>>    >> To initiate the call from within conference A on server 1. Have a
>>    >> corresponding dialplan entry on server 2 to accept the call and add
>> it
>>    >> into
>>    >> the conference A on server 2. You've now bridged the two conferences
>> in
>>    >> the
>>    >> two servers.
>>    >
>>    > ________________________________
>>    >
>>    > _______________________________________________
>>    > FreeSWITCH-users mailing list
>>    > FreeSWITCH-users at lists.freeswitch.org
>>    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>    >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>    > http://www.freeswitch.org
>>    >
>>    > --
>>    > This message has been scanned for viruses and
>>    > dangerous content by MailScanner, and is
>>    > believed to be clean.
>>    > _______________________________________________
>>    > FreeSWITCH-users mailing list
>>    > FreeSWITCH-users at lists.freeswitch.org
>>    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>    >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>    > http://www.freeswitch.org
>>    >
>>    >
>>
>>
>>
>>    --
>>    -Rupa
>>
>>    _______________________________________________
>>    FreeSWITCH-users mailing list
>>    FreeSWITCH-users at lists.freeswitch.org
>>    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>    UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>    http://www.freeswitch.org
>>
>>
>>    _______________________________________________
>>    FreeSWITCH-users mailing list
>>    FreeSWITCH-users at lists.freeswitch.org
>>    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>    UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>    http://www.freeswitch.org
>>
>>
>>
>>
>>
>>
>> ------------------------------------------------------------------------------
>>
>>
>>  _______________________________________________
>>  FreeSWITCH-users mailing list
>>  FreeSWITCH-users at lists.freeswitch.org
>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Sun Jan 23 04:26:10 2011
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 23 Jan 2011 09:26:10 +0800
Subject: [Freeswitch-users] Portaudio Improvements
In-Reply-To: <AANLkTimbULX-_ne4ZyirMFxcOcRBaEZwxTfude6=L8F_@mail.gmail.com>
References: <AANLkTim9AmtEGZoK0NyG1yKVsmJ_MSwXAppi-6AcCqnz@mail.gmail.com>
	<AANLkTimm69f6adgkgDz03GUOq8xJw_oaVvrkAGxd1yjG@mail.gmail.com>
	<AANLkTikk==DKhXft67HWcj0ytya030bS=4vtt2o7+tD8@mail.gmail.com>
	<AANLkTi=T6a9nU54toi5+fMA7kq_xkmZrRnRT3Fa8Pao5@mail.gmail.com>
	<AANLkTimbULX-_ne4ZyirMFxcOcRBaEZwxTfude6=L8F_@mail.gmail.com>
Message-ID: <AANLkTin_ue-i-Mw5CTQiZo3CTWi3Kcvt_4pdGYTS1jk7@mail.gmail.com>

Thanks Mitch, I actually had tried the equivalent uuid_audio and looks
like it's the only way to do it for now. We had wanted to get energy
levels dynamically to show a live indication but we given up that
feature.


On Sun, Jan 23, 2011 at 2:52 AM, Mitch Capper <mitch.capper at gmail.com> wrote:
> It is true PA lacks volume control, it had been something I was debating.
> Adding gain is pretty simple as you are for the most part just multiplying
> up the samples I believe so would not be a major item to add to portaudio, I
> however decided not to go this route for two reasons.? 1) In Windows each
> application has its own volume setting (Vista and higher),? and apps are
> starting to tie into this than internal volume controls.? The downside is it
> doesn't distort audio/have a gain option.?? This is when I thought about
> allowing for a fixed gain amount added to portaudio.? 2) Freeswitch actually
> has its own built in volume control settings with set_audio_level.
> set_audio_level allows for 9 settings (4 lower 4 higher) and seems to do a
> good job at digital volume control.
> Avoiding modifying the audio stream itself could also result in less
> distortion when using actual volume controls (headsets, etc).
>
> So if you want volume control I would say take a look at set_audio_level as
> it may be the simplest method.?? Adding it to portaudio would work but one
> of the nice advantages of set_audio_level is the fact you can also do it per
> channel rather than globally:)
>
> ~Mitch
>
> On Fri, Jan 21, 2011 at 5:44 PM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> In addition to AEC. I also found PA lacks the ability to change the
>> sound device volume. I tried to control sound devices in QT but
>> haven't find a way. Ideas on this?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From Nabble at slickdeals.endjunk.com  Sun Jan 23 05:44:21 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sat, 22 Jan 2011 18:44:21 -0800 (PST)
Subject: [Freeswitch-users] mod_opal vs mod_h323
Message-ID: <1295750661945-5952066.post@n2.nabble.com>


As I understood it correctly, both mod_h323 and mod_opal can handle H323
calls to/from a HotMail/MSN account. If so, what are the differences between
the two modules?

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-opal-vs-mod-h323-tp5952066p5952066.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mitch.capper at gmail.com  Sun Jan 23 08:05:22 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Sat, 22 Jan 2011 21:05:22 -0800
Subject: [Freeswitch-users] Portaudio Improvements
In-Reply-To: <AANLkTin_ue-i-Mw5CTQiZo3CTWi3Kcvt_4pdGYTS1jk7@mail.gmail.com>
References: <AANLkTim9AmtEGZoK0NyG1yKVsmJ_MSwXAppi-6AcCqnz@mail.gmail.com>
	<AANLkTimm69f6adgkgDz03GUOq8xJw_oaVvrkAGxd1yjG@mail.gmail.com>
	<AANLkTikk==DKhXft67HWcj0ytya030bS=4vtt2o7+tD8@mail.gmail.com>
	<AANLkTi=T6a9nU54toi5+fMA7kq_xkmZrRnRT3Fa8Pao5@mail.gmail.com>
	<AANLkTimbULX-_ne4ZyirMFxcOcRBaEZwxTfude6=L8F_@mail.gmail.com>
	<AANLkTin_ue-i-Mw5CTQiZo3CTWi3Kcvt_4pdGYTS1jk7@mail.gmail.com>
Message-ID: <AANLkTi=GRGJbQwE8yF=y_yUWJobUMnhdYfYpnGge5qf-@mail.gmail.com>

Ah well viewing volume levels had not been something I had looked into.  It
wouldn't be too hard but the question is how do you constantly report that
(FS event every second? every half second? port audio commands to return
that value frequently?).  If your client is windows however again the native
windows volume handler for apps does also keep track of their current
output.

~Mitch

On Sat, Jan 22, 2011 at 5:26 PM, Seven Du <dujinfang at gmail.com> wrote:

> Thanks Mitch, I actually had tried the equivalent uuid_audio and looks
> like it's the only way to do it for now. We had wanted to get energy
> levels dynamically to show a live indication but we given up that
> feature.
>
>
> On Sun, Jan 23, 2011 at 2:52 AM, Mitch Capper <mitch.capper at gmail.com>
> wrote:
> > It is true PA lacks volume control, it had been something I was debating.
> > Adding gain is pretty simple as you are for the most part just
> multiplying
> > up the samples I believe so would not be a major item to add to
> portaudio, I
> > however decided not to go this route for two reasons.  1) In Windows each
> > application has its own volume setting (Vista and higher),  and apps are
> > starting to tie into this than internal volume controls.  The downside is
> it
> > doesn't distort audio/have a gain option.   This is when I thought about
> > allowing for a fixed gain amount added to portaudio.  2) Freeswitch
> actually
> > has its own built in volume control settings with set_audio_level.
> > set_audio_level allows for 9 settings (4 lower 4 higher) and seems to do
> a
> > good job at digital volume control.
> > Avoiding modifying the audio stream itself could also result in less
> > distortion when using actual volume controls (headsets, etc).
> >
> > So if you want volume control I would say take a look at set_audio_level
> as
> > it may be the simplest method.   Adding it to portaudio would work but
> one
> > of the nice advantages of set_audio_level is the fact you can also do it
> per
> > channel rather than globally:)
> >
> > ~Mitch
> >
> > On Fri, Jan 21, 2011 at 5:44 PM, Seven Du <dujinfang at gmail.com> wrote:
> >>
> >> In addition to AEC. I also found PA lacks the ability to change the
> >> sound device volume. I tried to control sound devices in QT but
> >> haven't find a way. Ideas on this?
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bernhard.suttner at winet.ch  Sun Jan 23 13:13:22 2011
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Sun, 23 Jan 2011 11:13:22 +0100
Subject: [Freeswitch-users] praise of freeSWITCH developers
Message-ID: <20110123111322.5f78cd1d@mail.winet.ch>

Just donated 35$ for all the great work of the FreeSWITCH developers 

----- Original Message -----
From: Madovsky [mailto:infos at madovsky.org]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Sat, 22 Jan 2011 17:44:35 +0100
Subject: Re: [Freeswitch-users] praise of freeSWITCH developers


> Thanks Chris
> and all others for your contribution.
> I have always a sentence in my mind that says :
> "help people you love, love will help you in return"....
> or "give wood to the fire to help it stay alive"....
> Anthony, maybe a barbecue in august would be a good idea ? :D
> 
> 
> ----- Original Message ----- 
> From: "Christian L?schenkohl" <christian.loeschenkohl at xpirio.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Saturday, January 22, 2011 4:58 AM
> Subject: Re: [Freeswitch-users] praise of freeSWITCH developers
> 
> 
> hello
> 
> these words are more than true.
> 100$ from my side are on the way.
> 
> br
> 
> On 2011-01-21 17:13, Madovsky wrote:
> 
> > I would like to remember to all people on this
> > emailist that all freeSWITCH guys work very hard
> > to update/upgrade and make FS more powerful, and
> > also to answer to hundreds of user emails every day.
> > Donations are welcome and appreciated.
> > It helps the humans behind the machine
> > who give to you the opportunity to create and use amazing projects
> > and to continue to make this open source project alive and active.
> > Regards
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> -- 
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
> 
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
> 
> T  +43 5 77 11 - 1000
> F  +43 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From rafonline at hotmail.com  Sun Jan 23 17:00:18 2011
From: rafonline at hotmail.com (Rafqat .)
Date: Sun, 23 Jan 2011 14:00:18 +0000
Subject: [Freeswitch-users] Bounty Offered for A2Billing equivalent for
	FreeSWITCH
Message-ID: <COL122-W5649D559590F7C32C5F5F2B5FA0@phx.gbl>



Hi,

I was wondering if anyone is interested in developing a billing application that is on-par with the current functionality offered by A2Billing but for the freeswitch platform. I have attempted to make a calling card app myself using modules like mod_nibblebill and mod_lcr but I am struggling with failover working the way I thought it would work (maybe my ignorance) and also issues with ## to make follow on calls.? I know I could simply try and resolve these issues but after looking at A2Billing it seems to offer a lot:

http://www.star2billing.com/solutions/call-through/
http://www.star2billing.com/solutions/call-back/


I would like the application to be event socket based (preferably written in Java) and would most likely sit on some other server, leaving freeswitch to do what it does best.

I am not sure the kind of bounty I should be offering for this.? Please bear in mind this is not a private venture as such but something for all of us to be able to use free of charge.

Please let me know if anyone is interested (particularly the core freeswitch team) and how much you would like for it.


Cheers

Raf




 		 	   		  

From cjbujold at accra.ca  Sun Jan 23 18:07:27 2011
From: cjbujold at accra.ca (Charles Bujold)
Date: Sun, 23 Jan 2011 11:07:27 -0400
Subject: [Freeswitch-users] Looking for lost article
Message-ID: <00ea01cbbb0f$401fa3b0$c05eeb10$@accra.ca>

Recently I saw in a newsgroup or some other location a reference that
somebody had found a new GUI for Freeswitch.  If I am not mistaking I
believe it was based from somewhere in Europe and it was switching from
Asterix based to Freeswitch.  If you have come across this post  can you
please send me the web site.  We are trying to keep a list of all frontends
for freeswitch.

 

Thanks

cjb

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From u2nsam at gmail.com  Sun Jan 23 18:35:56 2011
From: u2nsam at gmail.com (Sam)
Date: Sun, 23 Jan 2011 21:05:56 +0530
Subject: [Freeswitch-users] modules
Message-ID: <AANLkTim9EJFGCk+4VfGMhfzv9EeX192_E+t-hD6U3=9L@mail.gmail.com>

Hello,

Is there single doc explaining each of the modules briefly about their
functionality.
It would be helpful for others to go on a index like page depending upon the
modules and find along.

Regards
Sam
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From edpimentl at gmail.com  Sun Jan 23 18:49:05 2011
From: edpimentl at gmail.com (EdPimentl)
Date: Sun, 23 Jan 2011 10:49:05 -0500
Subject: [Freeswitch-users] Bounty Offered for A2Billing equivalent for
	FreeSWITCH
In-Reply-To: <COL122-W5649D559590F7C32C5F5F2B5FA0@phx.gbl>
References: <COL122-W5649D559590F7C32C5F5F2B5FA0@phx.gbl>
Message-ID: <AANLkTimhJEt+KQ23p3b0F6Q04GdkK7rtkC+MPPHOZLPF@mail.gmail.com>

Why not use ASTPP?

-E
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From edpimentl at gmail.com  Sun Jan 23 18:59:36 2011
From: edpimentl at gmail.com (EdPimentl)
Date: Sun, 23 Jan 2011 10:59:36 -0500
Subject: [Freeswitch-users] Bounty Offered for A2Billing equivalent for
	FreeSWITCH
In-Reply-To: <COL122-W5649D559590F7C32C5F5F2B5FA0@phx.gbl>
References: <COL122-W5649D559590F7C32C5F5F2B5FA0@phx.gbl>
Message-ID: <AANLkTik1+kJdgOxFa1wgvh0hgGOK62xEG2ncnOsRF4hC@mail.gmail.com>

ASTPP already support FreeSwitch.

Review the features missing you want to implement.

If you decide ASTPP is a viable solution, I would contribute to the bounty.

http://www.astpp.org/?q=node/165

-E
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From chris at cloudtel.com  Sun Jan 23 19:01:07 2011
From: chris at cloudtel.com (Chris Burns)
Date: Sun, 23 Jan 2011 11:01:07 -0500
Subject: [Freeswitch-users] modules
In-Reply-To: <AANLkTim9EJFGCk+4VfGMhfzv9EeX192_E+t-hD6U3=9L@mail.gmail.com>
References: <AANLkTim9EJFGCk+4VfGMhfzv9EeX192_E+t-hD6U3=9L@mail.gmail.com>
Message-ID: <AANLkTinuezyPgUueKEWgMrxMgikX-=eRu0ckUHUJO0Gw@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Modules

On Sun, Jan 23, 2011 at 10:35 AM, Sam <u2nsam at gmail.com> wrote:

> Hello,
>
> Is there single doc explaining each of the modules briefly about their
> functionality.
> It would be helpful for others to go on a index like page depending upon
> the modules and find along.
>
> Regards
> Sam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From freeswitch-list at puzzled.xs4all.nl  Sun Jan 23 19:42:56 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Sun, 23 Jan 2011 17:42:56 +0100
Subject: [Freeswitch-users] Looking for lost article
In-Reply-To: <00ea01cbbb0f$401fa3b0$c05eeb10$@accra.ca>
References: <00ea01cbbb0f$401fa3b0$c05eeb10$@accra.ca>
Message-ID: <4D3C5A90.8010300@puzzled.xs4all.nl>

On 01/23/2011 04:07 PM, Charles Bujold wrote:
> Recently I saw in a newsgroup or some other location a reference that
> somebody had found a new GUI for Freeswitch. If I am not mistaking I
> believe it was based from somewhere in Europe and it was switching from
> Asterix based to Freeswitch. If you have come across this post can you
> please send me the web site. We are trying to keep a list of all
> frontends for freeswitch.

Do you mean http://www.amooma.de/gemeinschaft ? The site is in German 
only. The switch to FreeSWITCH in Gemeinschaft 4.0 is mentioned here:
http://www.golem.de/1101/80818.html
http://www.linux-magazin.de/NEWS/Freeswitch-statt-Asterisk-BSI-laesst-sichere-Telefonanlage-entwickeln

Regards,
Patrick


From Nabble at slickdeals.endjunk.com  Sun Jan 23 19:47:51 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 23 Jan 2011 08:47:51 -0800 (PST)
Subject: [Freeswitch-users] Looking for lost article
In-Reply-To: <00ea01cbbb0f$401fa3b0$c05eeb10$@accra.ca>
References: <00ea01cbbb0f$401fa3b0$c05eeb10$@accra.ca>
Message-ID: <1295801271754-5953047.post@n2.nabble.com>


R U talking about a German software package called  http://amooma.de/
Gemeinschaft ?

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Looking-for-lost-article-tp5952877p5953047.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From rafonline at hotmail.com  Sun Jan 23 20:14:14 2011
From: rafonline at hotmail.com (Rafqat .)
Date: Sun, 23 Jan 2011 17:14:14 +0000
Subject: [Freeswitch-users] Bounty Offered for A2Billing equivalent for
 FreeSWITCH
In-Reply-To: <AANLkTik1+kJdgOxFa1wgvh0hgGOK62xEG2ncnOsRF4hC@mail.gmail.com>
References: <COL122-W5649D559590F7C32C5F5F2B5FA0@phx.gbl>,
	<AANLkTik1+kJdgOxFa1wgvh0hgGOK62xEG2ncnOsRF4hC@mail.gmail.com>
Message-ID: <COL122-W29A9437CE5C8CDF605425B5FA0@phx.gbl>



Hi,

I have taken a look at ASTPP but unfortunately their website seems a little stale, so i'm not sure if it's still worked on/supported.

I also had trouble trying to install it.

Cheers

Raf

________________________________
> From: edpimentl at gmail.com
> Date: Sun, 23 Jan 2011 10:59:36 -0500
> To: freeswitch-users at lists.freeswitch.org
> CC: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Bounty Offered for A2Billing equivalent
> for FreeSWITCH
>
>
> ASTPP already support FreeSwitch.
>
> Review the features missing you want to implement.
>
> If you decide ASTPP is a viable solution, I would contribute to the bounty.
>
> http://www.astpp.org/?q=node/165
>
> -E
>
>
>
> _______________________________________________ FreeSWITCH-users
> mailing list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  

From infos at madovsky.org  Sun Jan 23 20:19:01 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 23 Jan 2011 12:19:01 -0500
Subject: [Freeswitch-users] praise of freeSWITCH developers
References: <20110123111322.5f78cd1d@mail.winet.ch>
Message-ID: <F09BD788825D4370B9E542E0AF3CDFB3@e1705>

Great spirit on FS emailist.
I think it's one of the best emailist I ever been
since my developer life (13 years ago....)
This is a real social network :D

----- Original Message ----- 
From: "Bernhard Suttner" <bernhard.suttner at winet.ch>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Sunday, January 23, 2011 5:13 AM
Subject: Re: [Freeswitch-users] praise of freeSWITCH developers


> Just donated 35$ for all the great work of the FreeSWITCH developers
>
> ----- Original Message -----
> From: Madovsky [mailto:infos at madovsky.org]
> To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
> Sent: Sat, 22 Jan 2011 17:44:35 +0100
> Subject: Re: [Freeswitch-users] praise of freeSWITCH developers
>
>
>> Thanks Chris
>> and all others for your contribution.
>> I have always a sentence in my mind that says :
>> "help people you love, love will help you in return"....
>> or "give wood to the fire to help it stay alive"....
>> Anthony, maybe a barbecue in august would be a good idea ? :D
>>
>>
>> ----- Original Message ----- 
>> From: "Christian L?schenkohl" <christian.loeschenkohl at xpirio.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Saturday, January 22, 2011 4:58 AM
>> Subject: Re: [Freeswitch-users] praise of freeSWITCH developers
>>
>>
>> hello
>>
>> these words are more than true.
>> 100$ from my side are on the way.
>>
>> br
>>
>> On 2011-01-21 17:13, Madovsky wrote:
>>
>> > I would like to remember to all people on this
>> > emailist that all freeSWITCH guys work very hard
>> > to update/upgrade and make FS more powerful, and
>> > also to answer to hundreds of user emails every day.
>> > Donations are welcome and appreciated.
>> > It helps the humans behind the machine
>> > who give to you the opportunity to create and use amazing projects
>> > and to continue to make this open source project alive and active.
>> > Regards
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> -- 
>> Ing. Christian L?schenkohl
>> Technische Leitung, Forschung & Entwicklung VoIP
>>
>> xpirio
>> Telekommunikation & Service GmbH
>> Lakeside B04
>> 9020 Klagenfurt
>> Austria
>>
>> T  +43 5 77 11 - 1000
>> F  +43 5 77 11 - 1002
>> E  christian.loeschenkohl at xpirio.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From infos at madovsky.org  Sun Jan 23 20:37:06 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 23 Jan 2011 12:37:06 -0500
Subject: [Freeswitch-users] nibblebill pause
Message-ID: <EBC1B7CEDEA64E2EBB8C114DEBFBBC55@e1705>

Is it a normal behavior
that the dialplan stalls when there is nibblebill pause ?

Thanks
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From darren at aleph-com.net  Sun Jan 23 20:44:47 2011
From: darren at aleph-com.net (Darren Wiebe)
Date: Sun, 23 Jan 2011 10:44:47 -0700
Subject: [Freeswitch-users] Bounty Offered for A2Billing equivalent for
 FreeSWITCH
In-Reply-To: <COL122-W29A9437CE5C8CDF605425B5FA0@phx.gbl>
References: <COL122-W5649D559590F7C32C5F5F2B5FA0@phx.gbl>,
	<AANLkTik1+kJdgOxFa1wgvh0hgGOK62xEG2ncnOsRF4hC@mail.gmail.com>
	<COL122-W29A9437CE5C8CDF605425B5FA0@phx.gbl>
Message-ID: <4D3C690F.5070507@aleph-com.net>

Hello,

The website certainly is a little (maybe more than that) stale.  I've 
been doing a lot of work on the FreeSWITCH calling card support in the 
last few weeks.

Darren Wiebe

On 11-01-23 10:14 AM, Rafqat . wrote:
>
> Hi,
>
> I have taken a look at ASTPP but unfortunately their website seems a little stale, so i'm not sure if it's still worked on/supported.
>
> I also had trouble trying to install it.
>
> Cheers
>
> Raf
>
> ________________________________
>> From: edpimentl at gmail.com
>> Date: Sun, 23 Jan 2011 10:59:36 -0500
>> To: freeswitch-users at lists.freeswitch.org
>> CC: freeswitch-dev at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Bounty Offered for A2Billing equivalent
>> for FreeSWITCH
>>
>>
>> ASTPP already support FreeSwitch.
>>
>> Review the features missing you want to implement.
>>
>> If you decide ASTPP is a viable solution, I would contribute to the bounty.
>>
>> http://www.astpp.org/?q=node/165
>>
>> -E
>>
>>
>>
>> _______________________________________________ FreeSWITCH-users
>> mailing list FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>   		 	   		
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From rupa at rupa.com  Sun Jan 23 20:48:00 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Sun, 23 Jan 2011 11:48:00 -0600
Subject: [Freeswitch-users] nibblebill pause
In-Reply-To: <EBC1B7CEDEA64E2EBB8C114DEBFBBC55@e1705>
References: <EBC1B7CEDEA64E2EBB8C114DEBFBBC55@e1705>
Message-ID: <AANLkTi=2VE9TPTg_BaaeKjk17NoTna+f08oajwv-3-Ga@mail.gmail.com>

nope, can you put together a simple test-case demonstrating this problem?

On Sun, Jan 23, 2011 at 11:37 AM, Madovsky <infos at madovsky.org> wrote:
> Is it a normal behavior
> that the dialplan stalls when there is nibblebill pause ?
>
> Thanks
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From lloyd.aloysius at gmail.com  Sun Jan 23 21:21:29 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 23 Jan 2011 13:21:29 -0500
Subject: [Freeswitch-users] Bounty Offered for A2Billing equivalent for
	FreeSWITCH
In-Reply-To: <4D3C690F.5070507@aleph-com.net>
References: <COL122-W5649D559590F7C32C5F5F2B5FA0@phx.gbl>
	<AANLkTik1+kJdgOxFa1wgvh0hgGOK62xEG2ncnOsRF4hC@mail.gmail.com>
	<COL122-W29A9437CE5C8CDF605425B5FA0@phx.gbl>
	<4D3C690F.5070507@aleph-com.net>
Message-ID: <AANLkTikzt0p4djwDJmf1Lg3gbLqjVmJD5ua5oeitVHee@mail.gmail.com>

Rafquat,

As you know already A2Billing have three high level components.

1. Web Interface Manage the Database ( Customer, Reseller and
Administration)

2. Asterisk PHP-AGI for the Call through Service(Calling Card , ANI
Authentication etc ..)

3. Python used for the Callback service.

I mention one of my earlier posts. You can still use the A2Billing Web
Interface  ( Customer, Reseller and Administration) for the FreeSWITCH
Calling card application...

Then developing the Telephony Application

1. Call through service using Lua + A2Billing Database

2. Call Back in FreeSWITCH is very easy to implement, compare to Asterisk
Callback implementation.


FYI ... http://www.callwithus.com/  initially used Asterisk + A2Billing. Now
the back end is FreeSWITCH and Management using A2Billing web interface and
database.

Hope this will help you.

Thanks
Lloyd


On Sun, Jan 23, 2011 at 12:44 PM, Darren Wiebe <darren at aleph-com.net> wrote:

> Hello,
>
> The website certainly is a little (maybe more than that) stale.  I've
> been doing a lot of work on the FreeSWITCH calling card support in the
> last few weeks.
>
> Darren Wiebe
>
> On 11-01-23 10:14 AM, Rafqat . wrote:
> >
> > Hi,
> >
> > I have taken a look at ASTPP but unfortunately their website seems a
> little stale, so i'm not sure if it's still worked on/supported.
> >
> > I also had trouble trying to install it.
> >
> > Cheers
> >
> > Raf
> >
> > ________________________________
> >> From: edpimentl at gmail.com
> >> Date: Sun, 23 Jan 2011 10:59:36 -0500
> >> To: freeswitch-users at lists.freeswitch.org
> >> CC: freeswitch-dev at lists.freeswitch.org
> >> Subject: Re: [Freeswitch-users] Bounty Offered for A2Billing equivalent
> >> for FreeSWITCH
> >>
> >>
> >> ASTPP already support FreeSwitch.
> >>
> >> Review the features missing you want to implement.
> >>
> >> If you decide ASTPP is a viable solution, I would contribute to the
> bounty.
> >>
> >> http://www.astpp.org/?q=node/165
> >>
> >> -E
> >>
> >>
> >>
> >> _______________________________________________ FreeSWITCH-users
> >> mailing list FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Nabble at slickdeals.endjunk.com  Sun Jan 23 21:27:17 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 23 Jan 2011 10:27:17 -0800 (PST)
Subject: [Freeswitch-users] mod_jabber crashes if caller hangs up first
Message-ID: <1295807237099-5953247.post@n2.nabble.com>


I wonder if anyone out here who uses the FS with mod_jabber to handle GV
calls has run into some problems. My FS v1.0.6 with mod_jabber patches from
FS git, hosted on a Seagate DockStar, will crash if a caller who calls my GV
line hangs up first before I get a chance to hang up. I wonder if this is
only a local issue on cross-compiled FS v1.0.6.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-jabber-crashes-if-caller-hangs-up-first-tp5953247p5953247.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From sos at sokhapkin.dyndns.org  Sun Jan 23 21:46:21 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sun, 23 Jan 2011 13:46:21 -0500
Subject: [Freeswitch-users] Bounty Offered for A2Billing equivalent for
	FreeSWITCH
In-Reply-To: <AANLkTikzt0p4djwDJmf1Lg3gbLqjVmJD5ua5oeitVHee@mail.gmail.com>
References: <COL122-W5649D559590F7C32C5F5F2B5FA0@phx.gbl>
	<4D3C690F.5070507@aleph-com.net>
	<AANLkTikzt0p4djwDJmf1Lg3gbLqjVmJD5ua5oeitVHee@mail.gmail.com>
Message-ID: <201101231346.21531.sos@sokhapkin.dyndns.org>

To process calls on callwithus.com, I use mod_xml_curl to get FS dial plan 
from web server. PHP running on web server builds dialplan (sequence of 
"bridge" applications) using a2billing database. CDRs are posted using 
mod_xml_cdr. There are some enhancements comparing to a2billing, for example, 
mod_nibblebill is employed to disconnect customer's calls when account balance 
falls below credit limit.

On Sunday 23 January 2011, Aloysius Lloyd wrote:
> Rafquat,
> 
> As you know already A2Billing have three high level components.
> 
> 1. Web Interface Manage the Database ( Customer, Reseller and
> Administration)
> 
> 2. Asterisk PHP-AGI for the Call through Service(Calling Card , ANI
> Authentication etc ..)
> 
> 3. Python used for the Callback service.
> 
> I mention one of my earlier posts. You can still use the A2Billing Web
> Interface  ( Customer, Reseller and Administration) for the FreeSWITCH
> Calling card application...
> 
> Then developing the Telephony Application
> 
> 1. Call through service using Lua + A2Billing Database
> 
> 2. Call Back in FreeSWITCH is very easy to implement, compare to Asterisk
> Callback implementation.
> 
> 
> FYI ... http://www.callwithus.com/  initially used Asterisk + A2Billing.
> Now the back end is FreeSWITCH and Management using A2Billing web
> interface and database.
> 
> Hope this will help you.
> 
> Thanks
> Lloyd
> 
> On Sun, Jan 23, 2011 at 12:44 PM, Darren Wiebe <darren at aleph-com.net> wrote:
> > Hello,
> > 
> > The website certainly is a little (maybe more than that) stale.  I've
> > been doing a lot of work on the FreeSWITCH calling card support in the
> > last few weeks.
> > 
> > Darren Wiebe
> > 
> > On 11-01-23 10:14 AM, Rafqat . wrote:
> > > Hi,
> > > 
> > > I have taken a look at ASTPP but unfortunately their website seems a
> > 
> > little stale, so i'm not sure if it's still worked on/supported.
> > 
> > > I also had trouble trying to install it.
> > > 
> > > Cheers
> > > 
> > > Raf
> > > 
> > > ________________________________
> > > 
> > >> From: edpimentl at gmail.com
> > >> Date: Sun, 23 Jan 2011 10:59:36 -0500
> > >> To: freeswitch-users at lists.freeswitch.org
> > >> CC: freeswitch-dev at lists.freeswitch.org
> > >> Subject: Re: [Freeswitch-users] Bounty Offered for A2Billing
> > >> equivalent for FreeSWITCH
> > >> 
> > >> 
> > >> ASTPP already support FreeSwitch.
> > >> 
> > >> Review the features missing you want to implement.
> > >> 
> > >> If you decide ASTPP is a viable solution, I would contribute to the
> > 
> > bounty.
> > 
> > >> http://www.astpp.org/?q=node/165
> > >> 
> > >> -E
> > >> 
> > >> 
> > >> 
> > >> _______________________________________________ FreeSWITCH-users
> > >> mailing list FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > 
> > >> UNSUBSCRIBE:
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 
> > >> http://www.freeswitch.org
> > > 
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> > > s http://www.freeswitch.org
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org



From lloyd.aloysius at gmail.com  Sun Jan 23 21:59:11 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 23 Jan 2011 13:59:11 -0500
Subject: [Freeswitch-users] Bounty Offered for A2Billing equivalent for
	FreeSWITCH
In-Reply-To: <201101231346.21531.sos@sokhapkin.dyndns.org>
References: <COL122-W5649D559590F7C32C5F5F2B5FA0@phx.gbl>
	<4D3C690F.5070507@aleph-com.net>
	<AANLkTikzt0p4djwDJmf1Lg3gbLqjVmJD5ua5oeitVHee@mail.gmail.com>
	<201101231346.21531.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTimyczRWYaovaLg5i+Cm2cbUzQ2+pgERLOWroA4x@mail.gmail.com>

Sergey,

What do you use for the IVR ? LUA?

I know there is limitation in Asterisk + A2billing maximum of 200 calls on a
box.

How may calls can be achieved in the FreeSWITCH solution?

Thanks  again for the information.

Thanks
Lloyd


On Sun, Jan 23, 2011 at 1:46 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> To process calls on callwithus.com, I use mod_xml_curl to get FS dial plan
> from web server. PHP running on web server builds dialplan (sequence of
> "bridge" applications) using a2billing database. CDRs are posted using
> mod_xml_cdr. There are some enhancements comparing to a2billing, for
> example,
> mod_nibblebill is employed to disconnect customer's calls when account
> balance
> falls below credit limit.
>
> On Sunday 23 January 2011, Aloysius Lloyd wrote:
> > Rafquat,
> >
> > As you know already A2Billing have three high level components.
> >
> > 1. Web Interface Manage the Database ( Customer, Reseller and
> > Administration)
> >
> > 2. Asterisk PHP-AGI for the Call through Service(Calling Card , ANI
> > Authentication etc ..)
> >
> > 3. Python used for the Callback service.
> >
> > I mention one of my earlier posts. You can still use the A2Billing Web
> > Interface  ( Customer, Reseller and Administration) for the FreeSWITCH
> > Calling card application...
> >
> > Then developing the Telephony Application
> >
> > 1. Call through service using Lua + A2Billing Database
> >
> > 2. Call Back in FreeSWITCH is very easy to implement, compare to Asterisk
> > Callback implementation.
> >
> >
> > FYI ... http://www.callwithus.com/  initially used Asterisk + A2Billing.
> > Now the back end is FreeSWITCH and Management using A2Billing web
> > interface and database.
> >
> > Hope this will help you.
> >
> > Thanks
> > Lloyd
> >
> > On Sun, Jan 23, 2011 at 12:44 PM, Darren Wiebe <darren at aleph-com.net>
> wrote:
> > > Hello,
> > >
> > > The website certainly is a little (maybe more than that) stale.  I've
> > > been doing a lot of work on the FreeSWITCH calling card support in the
> > > last few weeks.
> > >
> > > Darren Wiebe
> > >
> > > On 11-01-23 10:14 AM, Rafqat . wrote:
> > > > Hi,
> > > >
> > > > I have taken a look at ASTPP but unfortunately their website seems a
> > >
> > > little stale, so i'm not sure if it's still worked on/supported.
> > >
> > > > I also had trouble trying to install it.
> > > >
> > > > Cheers
> > > >
> > > > Raf
> > > >
> > > > ________________________________
> > > >
> > > >> From: edpimentl at gmail.com
> > > >> Date: Sun, 23 Jan 2011 10:59:36 -0500
> > > >> To: freeswitch-users at lists.freeswitch.org
> > > >> CC: freeswitch-dev at lists.freeswitch.org
> > > >> Subject: Re: [Freeswitch-users] Bounty Offered for A2Billing
> > > >> equivalent for FreeSWITCH
> > > >>
> > > >>
> > > >> ASTPP already support FreeSwitch.
> > > >>
> > > >> Review the features missing you want to implement.
> > > >>
> > > >> If you decide ASTPP is a viable solution, I would contribute to the
> > >
> > > bounty.
> > >
> > > >> http://www.astpp.org/?q=node/165
> > > >>
> > > >> -E
> > > >>
> > > >>
> > > >>
> > > >> _______________________________________________ FreeSWITCH-users
> > > >> mailing list FreeSWITCH-users at lists.freeswitch.org
> > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > >> UNSUBSCRIBE:
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > >> http://www.freeswitch.org
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-user
> > > > s http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
>
>
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From sos at sokhapkin.dyndns.org  Sun Jan 23 22:10:25 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sun, 23 Jan 2011 14:10:25 -0500
Subject: [Freeswitch-users] Bounty Offered for A2Billing equivalent for
	FreeSWITCH
In-Reply-To: <AANLkTimyczRWYaovaLg5i+Cm2cbUzQ2+pgERLOWroA4x@mail.gmail.com>
References: <COL122-W5649D559590F7C32C5F5F2B5FA0@phx.gbl>
	<201101231346.21531.sos@sokhapkin.dyndns.org>
	<AANLkTimyczRWYaovaLg5i+Cm2cbUzQ2+pgERLOWroA4x@mail.gmail.com>
Message-ID: <201101231410.25725.sos@sokhapkin.dyndns.org>

Which IVR? Do you mean calling card prompts? I almost do not use calling card, 
I provide service to SIP clients. Calling card access numbers are still using 
asterisk. There is almost no traffic, no worth to me to port them to FS.

On Sunday 23 January 2011, Aloysius Lloyd wrote:
> Sergey,
> 
> What do you use for the IVR ? LUA?
> 
> I know there is limitation in Asterisk + A2billing maximum of 200 calls on
> a box.
> 
> How may calls can be achieved in the FreeSWITCH solution?
> 
> Thanks  again for the information.
> 
> Thanks
> Lloyd
> 
> 
> On Sun, Jan 23, 2011 at 1:46 PM, Sergey Okhapkin
> 
> <sos at sokhapkin.dyndns.org>wrote:
> > To process calls on callwithus.com, I use mod_xml_curl to get FS dial
> > plan from web server. PHP running on web server builds dialplan
> > (sequence of "bridge" applications) using a2billing database. CDRs are
> > posted using mod_xml_cdr. There are some enhancements comparing to
> > a2billing, for example,
> > mod_nibblebill is employed to disconnect customer's calls when account
> > balance
> > falls below credit limit.
> > 
> > On Sunday 23 January 2011, Aloysius Lloyd wrote:
> > > Rafquat,
> > > 
> > > As you know already A2Billing have three high level components.
> > > 
> > > 1. Web Interface Manage the Database ( Customer, Reseller and
> > > Administration)
> > > 
> > > 2. Asterisk PHP-AGI for the Call through Service(Calling Card , ANI
> > > Authentication etc ..)
> > > 
> > > 3. Python used for the Callback service.
> > > 
> > > I mention one of my earlier posts. You can still use the A2Billing Web
> > > Interface  ( Customer, Reseller and Administration) for the FreeSWITCH
> > > Calling card application...
> > > 
> > > Then developing the Telephony Application
> > > 
> > > 1. Call through service using Lua + A2Billing Database
> > > 
> > > 2. Call Back in FreeSWITCH is very easy to implement, compare to
> > > Asterisk Callback implementation.
> > > 
> > > 
> > > FYI ... http://www.callwithus.com/  initially used Asterisk +
> > > A2Billing. Now the back end is FreeSWITCH and Management using
> > > A2Billing web interface and database.
> > > 
> > > Hope this will help you.
> > > 
> > > Thanks
> > > Lloyd
> > > 
> > > On Sun, Jan 23, 2011 at 12:44 PM, Darren Wiebe <darren at aleph-com.net>
> > 
> > wrote:
> > > > Hello,
> > > > 
> > > > The website certainly is a little (maybe more than that) stale.  I've
> > > > been doing a lot of work on the FreeSWITCH calling card support in
> > > > the last few weeks.
> > > > 
> > > > Darren Wiebe
> > > > 
> > > > On 11-01-23 10:14 AM, Rafqat . wrote:
> > > > > Hi,
> > > > > 
> > > > > I have taken a look at ASTPP but unfortunately their website seems
> > > > > a
> > > > 
> > > > little stale, so i'm not sure if it's still worked on/supported.
> > > > 
> > > > > I also had trouble trying to install it.
> > > > > 
> > > > > Cheers
> > > > > 
> > > > > Raf
> > > > > 
> > > > > ________________________________
> > > > > 
> > > > >> From: edpimentl at gmail.com
> > > > >> Date: Sun, 23 Jan 2011 10:59:36 -0500
> > > > >> To: freeswitch-users at lists.freeswitch.org
> > > > >> CC: freeswitch-dev at lists.freeswitch.org
> > > > >> Subject: Re: [Freeswitch-users] Bounty Offered for A2Billing
> > > > >> equivalent for FreeSWITCH
> > > > >> 
> > > > >> 
> > > > >> ASTPP already support FreeSwitch.
> > > > >> 
> > > > >> Review the features missing you want to implement.
> > > > >> 
> > > > >> If you decide ASTPP is a viable solution, I would contribute to
> > > > >> the
> > > > 
> > > > bounty.
> > > > 
> > > > >> http://www.astpp.org/?q=node/165
> > > > >> 
> > > > >> -E
> > > > >> 
> > > > >> 
> > > > >> 
> > > > >> _______________________________________________ FreeSWITCH-users
> > > > >> mailing list FreeSWITCH-users at lists.freeswitch.org
> > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > 
> > > > >> UNSUBSCRIBE:
> > > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > 
> > > > >> http://www.freeswitch.org
> > > > > 
> > > > > _______________________________________________
> > > > > FreeSWITCH-users mailing list
> > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > 
> > > > > UNSUBSCRIBE:
> > http://lists.freeswitch.org/mailman/options/freeswitch-user
> > 
> > > > > s http://www.freeswitch.org
> > > > 
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > 
> > > > UNSUBSCRIBE:
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 
> > > > http://www.freeswitch.org



From lloyd.aloysius at gmail.com  Sun Jan 23 22:12:14 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 23 Jan 2011 14:12:14 -0500
Subject: [Freeswitch-users] Bounty Offered for A2Billing equivalent for
	FreeSWITCH
In-Reply-To: <201101231410.25725.sos@sokhapkin.dyndns.org>
References: <COL122-W5649D559590F7C32C5F5F2B5FA0@phx.gbl>
	<201101231346.21531.sos@sokhapkin.dyndns.org>
	<AANLkTimyczRWYaovaLg5i+Cm2cbUzQ2+pgERLOWroA4x@mail.gmail.com>
	<201101231410.25725.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTinbgbPT9XUeAujNOLF-Bqxcr7v=NdK4P4KFpzhf@mail.gmail.com>

That is true.

Thanks for the information.

Thanks
Lloyd

On Sun, Jan 23, 2011 at 2:10 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> Which IVR? Do you mean calling card prompts? I almost do not use calling
> card,
> I provide service to SIP clients. Calling card access numbers are still
> using
> asterisk. There is almost no traffic, no worth to me to port them to FS.
>
> On Sunday 23 January 2011, Aloysius Lloyd wrote:
> > Sergey,
> >
> > What do you use for the IVR ? LUA?
> >
> > I know there is limitation in Asterisk + A2billing maximum of 200 calls
> on
> > a box.
> >
> > How may calls can be achieved in the FreeSWITCH solution?
> >
> > Thanks  again for the information.
> >
> > Thanks
> > Lloyd
> >
> >
> > On Sun, Jan 23, 2011 at 1:46 PM, Sergey Okhapkin
> >
> > <sos at sokhapkin.dyndns.org>wrote:
> > > To process calls on callwithus.com, I use mod_xml_curl to get FS dial
> > > plan from web server. PHP running on web server builds dialplan
> > > (sequence of "bridge" applications) using a2billing database. CDRs are
> > > posted using mod_xml_cdr. There are some enhancements comparing to
> > > a2billing, for example,
> > > mod_nibblebill is employed to disconnect customer's calls when account
> > > balance
> > > falls below credit limit.
> > >
> > > On Sunday 23 January 2011, Aloysius Lloyd wrote:
> > > > Rafquat,
> > > >
> > > > As you know already A2Billing have three high level components.
> > > >
> > > > 1. Web Interface Manage the Database ( Customer, Reseller and
> > > > Administration)
> > > >
> > > > 2. Asterisk PHP-AGI for the Call through Service(Calling Card , ANI
> > > > Authentication etc ..)
> > > >
> > > > 3. Python used for the Callback service.
> > > >
> > > > I mention one of my earlier posts. You can still use the A2Billing
> Web
> > > > Interface  ( Customer, Reseller and Administration) for the
> FreeSWITCH
> > > > Calling card application...
> > > >
> > > > Then developing the Telephony Application
> > > >
> > > > 1. Call through service using Lua + A2Billing Database
> > > >
> > > > 2. Call Back in FreeSWITCH is very easy to implement, compare to
> > > > Asterisk Callback implementation.
> > > >
> > > >
> > > > FYI ... http://www.callwithus.com/  initially used Asterisk +
> > > > A2Billing. Now the back end is FreeSWITCH and Management using
> > > > A2Billing web interface and database.
> > > >
> > > > Hope this will help you.
> > > >
> > > > Thanks
> > > > Lloyd
> > > >
> > > > On Sun, Jan 23, 2011 at 12:44 PM, Darren Wiebe <darren at aleph-com.net
> >
> > >
> > > wrote:
> > > > > Hello,
> > > > >
> > > > > The website certainly is a little (maybe more than that) stale.
>  I've
> > > > > been doing a lot of work on the FreeSWITCH calling card support in
> > > > > the last few weeks.
> > > > >
> > > > > Darren Wiebe
> > > > >
> > > > > On 11-01-23 10:14 AM, Rafqat . wrote:
> > > > > > Hi,
> > > > > >
> > > > > > I have taken a look at ASTPP but unfortunately their website
> seems
> > > > > > a
> > > > >
> > > > > little stale, so i'm not sure if it's still worked on/supported.
> > > > >
> > > > > > I also had trouble trying to install it.
> > > > > >
> > > > > > Cheers
> > > > > >
> > > > > > Raf
> > > > > >
> > > > > > ________________________________
> > > > > >
> > > > > >> From: edpimentl at gmail.com
> > > > > >> Date: Sun, 23 Jan 2011 10:59:36 -0500
> > > > > >> To: freeswitch-users at lists.freeswitch.org
> > > > > >> CC: freeswitch-dev at lists.freeswitch.org
> > > > > >> Subject: Re: [Freeswitch-users] Bounty Offered for A2Billing
> > > > > >> equivalent for FreeSWITCH
> > > > > >>
> > > > > >>
> > > > > >> ASTPP already support FreeSwitch.
> > > > > >>
> > > > > >> Review the features missing you want to implement.
> > > > > >>
> > > > > >> If you decide ASTPP is a viable solution, I would contribute to
> > > > > >> the
> > > > >
> > > > > bounty.
> > > > >
> > > > > >> http://www.astpp.org/?q=node/165
> > > > > >>
> > > > > >> -E
> > > > > >>
> > > > > >>
> > > > > >>
> > > > > >> _______________________________________________ FreeSWITCH-users
> > > > > >> mailing list FreeSWITCH-users at lists.freeswitch.org
> > > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > >
> > > > > >> UNSUBSCRIBE:
> > > > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > >
> > > > > >> http://www.freeswitch.org
> > > > > >
> > > > > > _______________________________________________
> > > > > > FreeSWITCH-users mailing list
> > > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > > > > UNSUBSCRIBE:
> > > http://lists.freeswitch.org/mailman/options/freeswitch-user
> > >
> > > > > > s http://www.freeswitch.org
> > > > >
> > > > > _______________________________________________
> > > > > FreeSWITCH-users mailing list
> > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > > > UNSUBSCRIBE:
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > > > http://www.freeswitch.org
>
>
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From erik.dekkers at wvds.nl  Sun Jan 23 23:48:27 2011
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Sun, 23 Jan 2011 21:48:27 +0100
Subject: [Freeswitch-users] Javascript IVR session question
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE07872CFA48@FS1.wvds.local>

Hey ppl,

At the moment im building a Javascript based IVR but im kind of stuck on a part.

The IVR should do this:

- Answer session (working)
- Play some wav files (working)
- Record a message to file (working)
- Hang up the first session (working)
- Call the second session (not working)
- Play the previous recorded file (not working)

After I dial the second session, the console says "channel is hungup already". How should i do this?

Kind regards,

Erik Dekkers (wvds-nl on IRC)



my script:

var allDigits = "";
function on_dtmf(session, type, digits, arg)
{
    if (digits.digit == "#") {
        return allDigits;
    }
    if (digits.digit == "*") {
        return false; //stop the recording.
    }
    console_log("digit: " + digits.digit + "\n");
    allDigits += digits.digit;
    return(allDigits);
}
session.answer();
if (session.ready()) {
    allDigits = "";
    var rtn;
    rtn = session.streamFile("/home/edekkers/sounds/10_spreek_in.wav", on_dtmf, "");
    if (session.ready()) {
        var tmp_Filename = "/tmp/test.wav";
        if (session.ready()) {
            rtn = session.recordFile(tmp_Filename, on_dtmf, "", 120);
            }
        rtn = session.streamFile("/home/edekkers/sounds/11_bericht_is_ontvangen.wav", on_dtmf, "");
        if (session.ready()) {
            session.hangup();
        }
    }
}
session.execute("bridge","user/202")
if (session.ready()) {
    session.streamFile("/tmp/test.wav");
}


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From brian at freeswitch.org  Mon Jan 24 01:16:11 2011
From: brian at freeswitch.org (Brian West)
Date: Sun, 23 Jan 2011 16:16:11 -0600
Subject: [Freeswitch-users] praise of freeSWITCH developers
In-Reply-To: <20110123111322.5f78cd1d@mail.winet.ch>
References: <20110123111322.5f78cd1d@mail.winet.ch>
Message-ID: <7FAC1ED2-E969-42C1-9E05-3D741DFFC1DD@freeswitch.org>

Thank you is all I can say.  We are going to all be together on the week of Feb 7th so we could do another Developer dinner that week... 

/b

On Jan 23, 2011, at 4:13 AM, Bernhard Suttner wrote:

> Just donated 35$ for all the great work of the FreeSWITCH developers 



From kees at mroffice.org  Mon Jan 24 03:14:42 2011
From: kees at mroffice.org (Kees Varekamp)
Date: Mon, 24 Jan 2011 13:14:42 +1300
Subject: [Freeswitch-users] praise of freeSWITCH developers
In-Reply-To: <7FAC1ED2-E969-42C1-9E05-3D741DFFC1DD@freeswitch.org>
References: <20110123111322.5f78cd1d@mail.winet.ch>
	<7FAC1ED2-E969-42C1-9E05-3D741DFFC1DD@freeswitch.org>
Message-ID: <AANLkTi=BuqzsYY0eu=sRioUB3eyBf-nzJWGspDrrxxx=@mail.gmail.com>

And another 50 usd from NZ :-)

On Mon, Jan 24, 2011 at 11:16, Brian West <brian at freeswitch.org> wrote:

> Thank you is all I can say.  We are going to all be together on the week of
> Feb 7th so we could do another Developer dinner that week...
>
> /b
>
> On Jan 23, 2011, at 4:13 AM, Bernhard Suttner wrote:
>
> > Just donated 35$ for all the great work of the FreeSWITCH developers
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Mon Jan 24 03:59:37 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 23 Jan 2011 19:59:37 -0500
Subject: [Freeswitch-users] nibblebill pause
References: <EBC1B7CEDEA64E2EBB8C114DEBFBBC55@e1705>
	<AANLkTi=2VE9TPTg_BaaeKjk17NoTna+f08oajwv-3-Ga@mail.gmail.com>
Message-ID: <90A521C1096F48218EA1D3296A8FEB65@e1705>

I have a very simple extension

        <extension name="callnow">
                <condition field="${sip_to_uri}" 
expression="^\d{10,15})@$${domain}$">
                        <action application="set" 
data="dialed_extension=$1"/>
                        <action application="nibblebill" data="pause"/>
                        <action application="sleep" data="2000"/>
                        <action application="answer"/>
                        <action application="transfer" data="ivr XML 
features"/>
                </condition>
        </extension>

the dialplan logs lock at nibblebill pause,
so for now I set nibblebill rate to 0 but there are SQL queries for 
nothing.... :(


----- Original Message ----- 
From: "Rupa Schomaker" <rupa at rupa.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Sunday, January 23, 2011 12:48 PM
Subject: Re: [Freeswitch-users] nibblebill pause


> nope, can you put together a simple test-case demonstrating this problem?
>
> On Sun, Jan 23, 2011 at 11:37 AM, Madovsky <infos at madovsky.org> wrote:
>> Is it a normal behavior
>> that the dialplan stalls when there is nibblebill pause ?
>>
>> Thanks
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> -- 
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From infos at madovsky.org  Mon Jan 24 09:13:39 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 24 Jan 2011 01:13:39 -0500
Subject: [Freeswitch-users] IVRD
Message-ID: <0040622336AF46389A17B7F04DE28BF7@e1705>

I tried to use IVRD from  wiki example

http://wiki.freeswitch.org/wiki/Ivrd

and server2.pl in ESL directory

copy and paste in my dialplan ans settings
so the daemon is running well, but if I attempt
to call nothing happens unless hangup.
on the log I can see only

EXECUTE sofia/internal/9999999999999 at default socket(127.0.0.1:8084 full)

I tried the tests of troubleshooting without error
I don't understand why the events are not received in the perl script
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From u2nsam at gmail.com  Mon Jan 24 09:18:48 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 24 Jan 2011 11:48:48 +0530
Subject: [Freeswitch-users] info
In-Reply-To: <AANLkTimM0fXNSF8e2HWbmJ_ZmHxsCfr0d=-_D9RZ8u-F@mail.gmail.com>
References: <AANLkTimM0fXNSF8e2HWbmJ_ZmHxsCfr0d=-_D9RZ8u-F@mail.gmail.com>
Message-ID: <AANLkTinRr1K50ebNwHr_t2Jx_r8HqRmZXnmTqdQTizg7@mail.gmail.com>

Pls ignore this email, my bad i didn't reloaded ... sorry to bother.

Regds
Sam

On Mon, Jan 24, 2011 at 11:27 AM, Sam <u2nsam at gmail.com> wrote:

> Have any one tried using <action application="info"/> in git-1c95ad9
> 2011-01-20 22-43-50 -0300 .
> because its not working for me in this version.
>
> Regds
> Sam
>
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From lakindia89 at gmail.com  Mon Jan 24 09:21:35 2011
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Mon, 24 Jan 2011 11:51:35 +0530
Subject: [Freeswitch-users] Help regard to park_timeout
In-Reply-To: <AANLkTi=edjFv90+q3G2kkzoJ76kWgx+NBe=Qx3CYA2Jd@mail.gmail.com>
References: <AANLkTi=edjFv90+q3G2kkzoJ76kWgx+NBe=Qx3CYA2Jd@mail.gmail.com>
Message-ID: <AANLkTinHSCfH6Ydh5ipGfhMQbpcHc+uJd6oxGhJNNFPy@mail.gmail.com>

Hi all,
I've done a further experimentation and I need a clarification.

>From CLI>
originate {ignore_early_media=true,park_timeout=50,api_hangup_hook='perl
/root/a.pl',exec_after_bridge_app=park}freetdm/grp1/a/9952248266 &park()

Then I executed the following commands from CLI.
uuid_getvar 9b59d172-2781-11e0-8586-8390cbcc860f park_timeout
_undef_
uuid_getvar 9b59d172-2781-11e0-8586-8390cbcc860f api_hangup_hook
perl /root/a.pl

I don't know why uuid_getvar, returns undef for park_timeout variable. But
the call is hangup once 50 seconds is reached. Can some one pls explain what
it is printing as __undef__


On Fri, Jan 21, 2011 at 6:37 PM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Dear all,
> I was using park_timeout and I come across the following scenario which I
> felt something is missing.
> I've originated a call as follows.
>
> originate
> {ignore_early_media=true,exec_after_bridge_app=park,park_timeout=60,api_hangup_hook='perl
> /root/a.pl'}freetdm/grp1/a/9952248266 &park()
>
> Once the call is answered I originated another call.
> originate {ignore_early_media=true,park_timeout=60,api_hangup_hook='perl
> /root/a.pl'}freetdm/grp1/a/9843171457 &park()
>
> Once this call is also answered, I said "uuid_bridge <uuid1> <uuid2>". Both
> call gets bridged. After some time, I hangup the second call (9843171457).
> Now the first call goes into park().
>
> I expect that the first call will hangup after 60 seconds, but it didn't.
>
> The freeswitch log is here
> http://pastebin.freeswitch.org/15099
>
> When I start to use the park_timeout, I thought once a leg is in park, then
> the timer will start, and once it is unparked for various reason the timer
> will be reseted. After sometime, when the leg again comes in park, the timer
> will start. Is this correct?
>
>
>
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From milan.m.masek at gmail.com  Mon Jan 24 06:43:24 2011
From: milan.m.masek at gmail.com (Milan)
Date: Mon, 24 Jan 2011 03:43:24 +0000 (UTC)
Subject: [Freeswitch-users] Bria 3 and Freeswitch TLS configuration
References: <AANLkTinntjwHEckZeT3N3=LuFwguMgK0bS=vUD_CWQUv@mail.gmail.com>
Message-ID: <loom.20110124T043717-856@post.gmane.org>

Milan Masek <milan.m.masek at ...> writes:

> 
> Hi there,What do you think I am doing wrong when my connection to Freeswitch
over TLS from Bria softphone ends with:"could not be enabled. Problem at server
(SIP error 503)"Configuration:Server: CentOS 5, Freeswitch 1.0.6 (fresh
installation from a tarball)
> Client: (K)Ubuntu 10.04, Bria 3.1-firewall is OK, telnet on port 5061 OK,
iptables OK, tcpdump shows traffic-server CA certificate imported on client
computer to system-wide keystore database (dpkg-reconfigure ca-certificates)
> -TLS enabled in vars.xmlAlso (with no TLS connection) - I can register phone
(Bria, Empathy), I can dial to another user, hear ring tone, pickup the phone
call - but I can not hear person on other side. I can just see that my
microphone works.Thank you for help. Any advice appreciated.Cheers,Milan


Freeswitch works excelent. Firewall and codecs is important to setup well.
But what about SIPS signaling traffic configuration on default port 5061? I am
getting still error 503.
Can you help me, please?
Cheers,
M.



From spencer at 5ninesolutions.com  Mon Jan 24 10:10:05 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 23 Jan 2011 23:10:05 -0800
Subject: [Freeswitch-users] BLF with Valet Parking
In-Reply-To: <AANLkTi=7z_BtTOjZVCat7T1C70V+H=Roba+pdXs+MuXW@mail.gmail.com>
References: <5E3329D2-286C-4F3E-9F04-F11F5CAACF6B@5ninesolutions.com>
	<AANLkTimkN-bdsRxxR8RhaKams=apiHLeRt9CPf42wvXJ@mail.gmail.com>
	<31AC9159-35FD-450F-9E9A-E648AA99AF8B@freeswitch.org>
	<AANLkTim6JVF6Db8PYMH0dVtOAa4Lg=XAUQ8py-vUP9Uq@mail.gmail.com>
	<AANLkTi=BXXPovnFGfETomdwc3gwQwuNZ==ieVdycPme4@mail.gmail.com>
	<45DC17EC-C795-42B0-B31C-CE70434297C9@5ninesolutions.com>
	<AANLkTi=7z_BtTOjZVCat7T1C70V+H=Roba+pdXs+MuXW@mail.gmail.com>
Message-ID: <929F0481-CB5F-4DEC-A909-641696B33E37@5ninesolutions.com>

Thanks for the info!  Using fifo has been working great.  What I was  
really hoping to do was use the built in park key on these phones.  I  
was looking at the default dialplan and there seems to be a starting  
point already there using this fuction.  When you press the park key,  
the phone prompts for a lot number.  If for instance I press 1, the  
phone dispatches an INFO message like the following:

recv 344 bytes from udp/[x.x.x.x]:43491 at 06:56:10.683660:
     
------------------------------------------------------------------------
    INFO sip:callpark at x.x.x.x;orbit=1 SIP/2.0
    Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK-9884b5e3
    From: "Test" <sip:1000 at x.x.x.x>;tag=12850188b984f129o0
    To: <sip:callpark at x.x.x.x;orbit=1>
    Call-ID: 3d4e50eb-c490b481 at 192.168.2.2
    CSeq: 5617 INFO
    Max-Forwards: 70
    User-Agent: Cisco/SPA509G-7.4.7
    Content-Length: 0

     
------------------------------------------------------------------------
2011-01-24 06:56:10.683749 [DEBUG] sofia.c:6131 dispatched freeswitch  
event for INFO

To which I get a 200 OK from Freeswitch.  The phone then says "Invalid  
Park Number".

It seems the phone would send a REFER instead of INFO??  I have the  
following in my dialplan:

     <extension name="park">
       <condition field="source" expression="mod_sofia"/>
       <condition field="destination_number" expression="callpark"/>
       <condition field="${sip_refer_to}">
         <expression><![CDATA[<sip:callpark at .*;orbit=(\d+)>]]></ 
expression>
         <action application="fifo" data="ParkingSlot$1@${domain_name}  
in undef $${hold_music}"/>
       </condition>
     </extension>

Any pointers would be greatly appreciated.

Thanks,
Spencer



On Jan 17, 2011, at 11:30 AM, S W wrote:

> FYI, on the wiki there is a little further documentation on the config
> I wrote up there in the email:
>
> http://wiki.freeswitch.org/wiki/Park_%26_Retrieve
>
> I've been using this code for quite a while.  The fact that fifo just
> works solid in freeswitch is wonderful, to say the least.
>
> On Mon, Jan 17, 2011 at 2:02 PM, Spencer Thomason
> <spencer at 5ninesolutions.com> wrote:
>> Thanks for the info.  Initially I was trying to mimic the Asterisk
>> parking behavior and show the status of a call in a slot.  I'll  
>> rework
>> this using fifo.
>>
>> Spencer
>>
>> On Jan 17, 2011, at 9:07 AM, Anthony Minessale wrote:
>>
>>> yes using fifo for parking is the more elegant solution.
>>> the valet_parking is more for people who miss the asterisk flavored
>>> parking.
>>>
>>>
>>> On Mon, Jan 17, 2011 at 11:00 AM, S W <steve.d.ward at gmail.com>  
>>> wrote:
>>>> Travis:
>>>>
>>>> Here is something that might interest you.  I use 509gs, and I have
>>>> some of them set up with BLF/speeddials set up on "parking slots".
>>>> If
>>>> no call is in the slot, you just press the button to park the call.
>>>> If you want to retrieve a call from the slot, press that same  
>>>> button.
>>>> Following is the config I use to be able to park calls and have
>>>> BLF/speed dials.
>>>>
>>>> In autoconf/fifo.conf.xml :
>>>> -------------------------------------
>>>> <fifo name="ParkingSlot1@$${domain}" importance="0"/>
>>>> -------------------------------------
>>>>
>>>> In dialplan, e.g. 100_parking_slots.xml :
>>>>
>>>>   <extension name="parking_slots">
>>>>    <condition field="destination_number"
>>>> expression="^(ParkingSlot\d+)$" break="on-false">
>>>>      <!-- Get count of callers in the specified parking slot (fifo)
>>>> -->
>>>>      <action inline="true" application="set"
>>>> data="slot_count=${fifo(count $1@$${domain})}"/>
>>>>      <action inline="true" application="set"
>>>> data="slot_count=${slot_count:-3:2}"/>
>>>>    </condition>
>>>>    <condition field="${slot_count}" expression="^\:0$"
>>>> break="always">
>>>>      <!-- For FIFO count of 0: place the caller into the FIFO
>>>> (actions
>>>> are    -->
>>>>      <!-- executed).
>>>>                             -->
>>>>      <!-- fifo settings for parking fifos -->
>>>>      <action application="unset" data="fifo_chime_list"/>
>>>>      <action application="set" data="fifo_chime_freq=0"/>
>>>>     <!-- Parked callers listen to MOH -->
>>>>      <action application="fifo" data="${destination_number}@$$
>>>> {domain}
>>>> in undef local_stream://moh"/>
>>>>     <!-- This hangup should never happen. -->
>>>>      <action application="hangup"/>
>>>>      <!-- For non-zero FIFO count: connect the caller to the parked
>>>> channel   -->
>>>>      <!-- (remove the channel from the FIFO and bridge it to the
>>>> caller using -->
>>>>      <!-- fifo ... out)
>>>>                                  -->
>>>>      <anti-action application="fifo"
>>>> data="${destination_number}@$${domain} out nowait"/>
>>>>     <!-- This hangup should never happen -->
>>>>      <anti-action application="hangup"/>
>>>>    </condition>
>>>> </extension>
>>>>
>>>> And here is an example SPA509g config (I actually use this):
>>>>
>>>> fnc=blf+sd;sub=ParkingSlot1@$PROXY;ext=ParkingSlot1@$PROXY
>>>>
>>>> Note that, above, my FreeSWITCH box that runs the fifo code above  
>>>> is
>>>> the $PROXY to which my 509g is also registered, etc.
>>>>
>>>> Let me know if you have any questions on that; I hope it helps.
>>>>
>>>>
>>>> On Mon, Jan 17, 2011 at 10:29 AM, Brian West <brian at freeswitch.org>
>>>> wrote:
>>>>>
>>>>> I think he wants BLF status of a lot on a key.  That I don't think
>>>>> is there.
>>>>> /b
>>>>> On Jan 17, 2011, at 9:01 AM, Travis Kemen wrote:
>>>>>
>>>>> BLF works fine here with valet parking using polycom/snom phones.
>>>>>
>>>>> Travis
>>>>>
>>>>> On Sun, Jan 16, 2011 at 6:15 PM, Spencer Thomason <spencer at 5ninesolutions.com
>>>>>> wrote:
>>>>>>
>>>>>> Hello all,
>>>>>> Is it possible to use BLFs with valet parking?  Basically I have
>>>>>> several Linksys SPA 509G phones and after a call is parked, I'd
>>>>>> like
>>>>>> use use a speed dial/BLF key on the phone to pickup the call.   
>>>>>> I do
>>>>>> have presence configured and working with the extensions.
>>>>>>
>>>>>> Thanks,
>>>>>> Spencer
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From u2nsam at gmail.com  Mon Jan 24 08:57:20 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 24 Jan 2011 11:27:20 +0530
Subject: [Freeswitch-users] info
Message-ID: <AANLkTimM0fXNSF8e2HWbmJ_ZmHxsCfr0d=-_D9RZ8u-F@mail.gmail.com>

Have any one tried using <action application="info"/> in git-1c95ad9
2011-01-20 22-43-50 -0300 .
because its not working for me in this version.

Regds
Sam
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From abubacker at bksystems.co.in  Mon Jan 24 08:32:25 2011
From: abubacker at bksystems.co.in (abubacker)
Date: Mon, 24 Jan 2011 11:02:25 +0530
Subject: [Freeswitch-users] FreeSWITCH 1.0.7
In-Reply-To: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org>
References: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org>
Message-ID: <4D3D0EE9.4070200@bksys.co.in>

On Friday 14 January 2011 09:55 PM, Brian West wrote:
> _http://latest.freeswitch.org/_
>
> Enjoy!
>
> /b
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    
Great job , Thanks to every one !

-- 
Best regards,
N.Abubacker ,
Associate system engineer ,
bk systems pvt ltd ,
Ph : 9144-43902701

Disclaimer: http://www.bksystems.co.in/email-policy

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From u2nsam at gmail.com  Mon Jan 24 12:05:18 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 24 Jan 2011 14:35:18 +0530
Subject: [Freeswitch-users] deflect
In-Reply-To: <AANLkTiksYLOeki2_XDJC5YwMW6hjUxvdpT7-Mvg7vmKZ@mail.gmail.com>
References: <AANLkTiksYLOeki2_XDJC5YwMW6hjUxvdpT7-Mvg7vmKZ@mail.gmail.com>
Message-ID: <AANLkTi=EMn55D1xGBrWuGgVE_dW_5itoJ4WpzByyagTE@mail.gmail.com>

Hi,

Is it possible by having b2bua in between , would the leg A be deflected to
the another FS server from first server ?

Regds
Sam

On Wed, Jan 12, 2011 at 11:42 AM, Sam <u2nsam at gmail.com> wrote:

> Hi,
>
> When call comes on 1 server and plays an application and after execution of
> the
> application the call is bridge to the other server ,but here after bridging
> the call
> should refer/deflect to other server, how this can be done ?
>
> Here just using the deflect variable is not recommended as there is proxy
> in between,
> so once the call is bridge the next step would be deflect the leg totally
> to another server via proxy.
>
> Regards
> Sam
>
>
>
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From fdelawarde at wirelessmundi.com  Mon Jan 24 12:18:15 2011
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Mon, 24 Jan 2011 10:18:15 +0100
Subject: [Freeswitch-users] praise of freeSWITCH developers
In-Reply-To: <AANLkTi=BuqzsYY0eu=sRioUB3eyBf-nzJWGspDrrxxx=@mail.gmail.com>
References: <20110123111322.5f78cd1d@mail.winet.ch>
	<7FAC1ED2-E969-42C1-9E05-3D741DFFC1DD@freeswitch.org>
	<AANLkTi=BuqzsYY0eu=sRioUB3eyBf-nzJWGspDrrxxx=@mail.gmail.com>
Message-ID: <1295860695.32615.23.camel@luna.tc.commsmundi.com>

On Mon, 2011-01-24 at 13:14 +1300, Kees Varekamp wrote:
> And another 50 usd from NZ :-)

Nice! We already have raised $385 total so far on this thread. Remember
that if they can afford fancy cocktails and silk underwear, our favorite
devs will code faster.

Carry on, just a little more to go for the $500 step!

Fran?ois.



From marcdecorny at gmail.com  Mon Jan 24 12:20:12 2011
From: marcdecorny at gmail.com (Marc de Corny)
Date: Mon, 24 Jan 2011 09:20:12 +0000
Subject: [Freeswitch-users] Caller ID using Fifo
In-Reply-To: <AANLkTi=m_hW9vQV5nbC2jwcos0bKy2rCf7YAr6aA23oz@mail.gmail.com>
References: <AANLkTik6pdgXgND6kmzZA4qe8vDZQ0_7e47+ciVWPfub@mail.gmail.com>
	<AANLkTikKDO3qru-9=nuPEQcW+iCN7ccizyJAh9sqUkCN@mail.gmail.com>
	<AANLkTi=p7_CsUDWS62+weZv50+PBZ8RLop0Q04y-c4hB@mail.gmail.com>
	<AANLkTi=m_hW9vQV5nbC2jwcos0bKy2rCf7YAr6aA23oz@mail.gmail.com>
Message-ID: <AANLkTimE-0g-Gc92tseC-GWOuwV-icvJU7ndLnhPOCo=@mail.gmail.com>

Hi All

I have since had a play with mod_callcenter and have not been able to send
the call to an agent with the caller_id_name as the name of the queue.

I keep on getting the CLI on both.

As this cannot be done in FIFO either, do any of you have any ideas? or
managed to get it working ?

thanks
Marc

On Fri, Jan 14, 2011 at 4:35 PM, Marc de Corny <marcdecorny at gmail.com>wrote:

> Just to follow up on this subject.
>
> I have done a lot of testing on the fifo trying to get the caller_id_name
> changed on the outbound call to the agent and to be honest I cannot
> understand the explanation.
>
> If mod_fifo does not know which call it will connect until the agent
> answers, how come it displays the CLI correctly, jsut won;t let me change
> it.
>
> Still seems strange. I am looking into the Mod_callcentre to check if it
> sends caller_id information. but the same logic if valid could apply
>
> Also maybe someone should change the Wiki ( I would but do not have enough
> expertise on the subject) because the following is a bit misleading
>
>  "Note: If you wish to specify the caller ID presented when a fifo calls an
> agent, set the origination_caller_id_name and origination_caller_id_num
> variables to the values desired. These could be set within the {} of the
> dialstring, or they could be set using the set application in the dialplan
> which places the caller into the fifo (before the 'fifo in' executed on the
> caller). "
> thanks
> Marc
>   On Thu, Jan 13, 2011 at 10:47 PM, Joao Leme <joaocarlosleme at gmail.com>wrote:
>
>> What about showing the Caller ID after it is answered? Any way to do that?
>>
>> 2011/1/12 Jo?o Mesquita <jmesquita at freeswitch.org>
>>
>> Jo?o Leme,
>>>
>>> The caller id is not passed when the phone is ringing because mod_fifo
>>> does not know which call is going to be sent to that channel once it is
>>> answered until it is really answered. I don't know if mod_callcenter does
>>> show anything but you should consider looking at the documentation if you
>>> really need this feature.
>>>
>>> Regards,
>>> Jo?o Mesquita
>>>
>>>
>>>   On Wed, Jan 12, 2011 at 9:15 PM, Joao Leme <joaocarlosleme at gmail.com>wrote:
>>>
>>>>  Hi there,
>>>> I would like to know if there is a way to see the caller ID on my Sip
>>>> Client (X-Lite for example) of the caller that I answear from a Fifo queue?
>>>> Thanks,
>>>> John
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From steveayre at gmail.com  Mon Jan 24 12:42:24 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 24 Jan 2011 09:42:24 +0000
Subject: [Freeswitch-users] deflect
In-Reply-To: <AANLkTi=EMn55D1xGBrWuGgVE_dW_5itoJ4WpzByyagTE@mail.gmail.com>
References: <AANLkTiksYLOeki2_XDJC5YwMW6hjUxvdpT7-Mvg7vmKZ@mail.gmail.com>
	<AANLkTi=EMn55D1xGBrWuGgVE_dW_5itoJ4WpzByyagTE@mail.gmail.com>
Message-ID: <AANLkTi=F5Lhsk1tjcG5z5LTUjZb4e-cQdNXQjK3OsVD7@mail.gmail.com>

You could try uuid_simplify with the api_on_answer hook

http://wiki.freeswitch.org/wiki/Mod_commands#uuid_simplify
http://wiki.freeswitch.org/wiki/Variable_api_on_answer

-Steve



On 24 January 2011 09:05, Sam <u2nsam at gmail.com> wrote:

> Hi,
>
> Is it possible by having b2bua in between , would the leg A be deflected to
> the another FS server from first server ?
>
> Regds
> Sam
>
>
> On Wed, Jan 12, 2011 at 11:42 AM, Sam <u2nsam at gmail.com> wrote:
>
>> Hi,
>>
>> When call comes on 1 server and plays an application and after execution
>> of the
>> application the call is bridge to the other server ,but here after
>> bridging the call
>> should refer/deflect to other server, how this can be done ?
>>
>> Here just using the deflect variable is not recommended as there is proxy
>> in between,
>> so once the call is bridge the next step would be deflect the leg totally
>> to another server via proxy.
>>
>> Regards
>> Sam
>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dome at tel.co.th  Mon Jan 24 12:58:08 2011
From: dome at tel.co.th (dome at tel.co.th)
Date: Mon, 24 Jan 2011 16:58:08 +0700
Subject: [Freeswitch-users] praise of freeSWITCH developers
In-Reply-To: <1295860695.32615.23.camel@luna.tc.commsmundi.com>
References: <20110123111322.5f78cd1d@mail.winet.ch>
	<7FAC1ED2-E969-42C1-9E05-3D741DFFC1DD@freeswitch.org>
	<AANLkTi=BuqzsYY0eu=sRioUB3eyBf-nzJWGspDrrxxx=@mail.gmail.com>
	<1295860695.32615.23.camel@luna.tc.commsmundi.com>
Message-ID: <AANLkTimn-qLpqhoso0cWEgZwSFK8szKTOrd03B+uoR2B@mail.gmail.com>

50 usd from Thailand. :)
I use my wife paypal account. she don't know FS. but after i tell her
she said why not ?...
And then she agree :)

BG

Dome C.


From u2nsam at gmail.com  Mon Jan 24 13:01:29 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 24 Jan 2011 15:31:29 +0530
Subject: [Freeswitch-users] deflect
In-Reply-To: <AANLkTi=F5Lhsk1tjcG5z5LTUjZb4e-cQdNXQjK3OsVD7@mail.gmail.com>
References: <AANLkTiksYLOeki2_XDJC5YwMW6hjUxvdpT7-Mvg7vmKZ@mail.gmail.com>
	<AANLkTi=EMn55D1xGBrWuGgVE_dW_5itoJ4WpzByyagTE@mail.gmail.com>
	<AANLkTi=F5Lhsk1tjcG5z5LTUjZb4e-cQdNXQjK3OsVD7@mail.gmail.com>
Message-ID: <AANLkTik9j3-o7GGr0=uudVgxLwnmcdTwC8ue8bVrv264@mail.gmail.com>

So by this ,

<action application="deflect" data="{api_on_answer='uuid_transfer  ${uuid}
both'}sofia/external/9999 at 192.168.2.130"/>

Will transfer both legs of call to 192.168.2.130  ?

Regards
Sam




On Mon, Jan 24, 2011 at 3:12 PM, Steven Ayre <steveayre at gmail.com> wrote:

> You could try uuid_simplify with the api_on_answer hook
>
> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_simplify
> http://wiki.freeswitch.org/wiki/Variable_api_on_answer
>
> -Steve
>
>
>
> On 24 January 2011 09:05, Sam <u2nsam at gmail.com> wrote:
>
>> Hi,
>>
>> Is it possible by having b2bua in between , would the leg A be deflected
>> to the another FS server from first server ?
>>
>> Regds
>> Sam
>>
>>
>> On Wed, Jan 12, 2011 at 11:42 AM, Sam <u2nsam at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> When call comes on 1 server and plays an application and after execution
>>> of the
>>> application the call is bridge to the other server ,but here after
>>> bridging the call
>>> should refer/deflect to other server, how this can be done ?
>>>
>>> Here just using the deflect variable is not recommended as there is proxy
>>> in between,
>>> so once the call is bridge the next step would be deflect the leg totally
>>> to another server via proxy.
>>>
>>> Regards
>>> Sam
>>>
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Mon Jan 24 13:29:13 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 24 Jan 2011 15:59:13 +0530
Subject: [Freeswitch-users] deflect
In-Reply-To: <AANLkTi=F5Lhsk1tjcG5z5LTUjZb4e-cQdNXQjK3OsVD7@mail.gmail.com>
References: <AANLkTiksYLOeki2_XDJC5YwMW6hjUxvdpT7-Mvg7vmKZ@mail.gmail.com>
	<AANLkTi=EMn55D1xGBrWuGgVE_dW_5itoJ4WpzByyagTE@mail.gmail.com>
	<AANLkTi=F5Lhsk1tjcG5z5LTUjZb4e-cQdNXQjK3OsVD7@mail.gmail.com>
Message-ID: <AANLkTikn_mbo5bmQ5NxE=qmHS4rNyu_EaGgTN6gWrjQQ@mail.gmail.com>

The whole media & signaling would be deflected or just signaling ?

Regds
Sam


On Mon, Jan 24, 2011 at 3:12 PM, Steven Ayre <steveayre at gmail.com> wrote:

> You could try uuid_simplify with the api_on_answer hook
>
> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_simplify
> http://wiki.freeswitch.org/wiki/Variable_api_on_answer
>
> -Steve
>
>
>
> On 24 January 2011 09:05, Sam <u2nsam at gmail.com> wrote:
>
>> Hi,
>>
>> Is it possible by having b2bua in between , would the leg A be deflected
>> to the another FS server from first server ?
>>
>> Regds
>> Sam
>>
>>
>> On Wed, Jan 12, 2011 at 11:42 AM, Sam <u2nsam at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> When call comes on 1 server and plays an application and after execution
>>> of the
>>> application the call is bridge to the other server ,but here after
>>> bridging the call
>>> should refer/deflect to other server, how this can be done ?
>>>
>>> Here just using the deflect variable is not recommended as there is proxy
>>> in between,
>>> so once the call is bridge the next step would be deflect the leg totally
>>> to another server via proxy.
>>>
>>> Regards
>>> Sam
>>>
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Mon Jan 24 14:08:33 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 24 Jan 2011 11:08:33 +0000
Subject: [Freeswitch-users] deflect
In-Reply-To: <AANLkTik9j3-o7GGr0=uudVgxLwnmcdTwC8ue8bVrv264@mail.gmail.com>
References: <AANLkTiksYLOeki2_XDJC5YwMW6hjUxvdpT7-Mvg7vmKZ@mail.gmail.com>
	<AANLkTi=EMn55D1xGBrWuGgVE_dW_5itoJ4WpzByyagTE@mail.gmail.com>
	<AANLkTi=F5Lhsk1tjcG5z5LTUjZb4e-cQdNXQjK3OsVD7@mail.gmail.com>
	<AANLkTik9j3-o7GGr0=uudVgxLwnmcdTwC8ue8bVrv264@mail.gmail.com>
Message-ID: <AANLkTim6h42+8ZDmFL00dYMNOmAz-3JtG43q+2oMMZ36@mail.gmail.com>

No, bridge application.
<action application="bridge" data="{api_on_answer='uuid_transfer  ${uuid}
both'}sofia/external/9999 at 192.168.2.130"/>
The call will be setup going through FS, then when it's answered FS will try
to do a reinvite to remove itself from the call path at which point both
signalling and media will not go through FS. It'll only work if both
endpoints can see each other though.

Perhaps I misunderstood what you're trying to do though?

The deflect app sends a REFER request on the a-leg for an answered call that
tells the caller to redirect to another server. FS won't be in the call path
for signalling or media after the redirect. There's no bleg in this
scenario.
<action application="deflect"
data="sip:9999 at 192.168.2.130<sip%3A9999 at 192.168.2.130>
"/>

So for example you can answer the call, do IVR, then redirect the caller to
an extension on another server without the call going through FS.

-Steve



On 24 January 2011 10:01, Sam <u2nsam at gmail.com> wrote:

> So by this ,
>
> <action application="deflect" data="{api_on_answer='uuid_transfer  ${uuid}
> both'}sofia/external/9999 at 192.168.2.130"/>
>
> Will transfer both legs of call to 192.168.2.130  ?
>
> Regards
> Sam
>
>
>
>
>
> On Mon, Jan 24, 2011 at 3:12 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> You could try uuid_simplify with the api_on_answer hook
>>
>> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_simplify
>> http://wiki.freeswitch.org/wiki/Variable_api_on_answer
>>
>> -Steve
>>
>>
>>
>> On 24 January 2011 09:05, Sam <u2nsam at gmail.com> wrote:
>>
>>>  Hi,
>>>
>>> Is it possible by having b2bua in between , would the leg A be deflected
>>> to the another FS server from first server ?
>>>
>>> Regds
>>> Sam
>>>
>>>
>>> On Wed, Jan 12, 2011 at 11:42 AM, Sam <u2nsam at gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> When call comes on 1 server and plays an application and after execution
>>>> of the
>>>> application the call is bridge to the other server ,but here after
>>>> bridging the call
>>>> should refer/deflect to other server, how this can be done ?
>>>>
>>>> Here just using the deflect variable is not recommended as there is
>>>> proxy in between,
>>>> so once the call is bridge the next step would be deflect the leg
>>>> totally to another server via proxy.
>>>>
>>>> Regards
>>>> Sam
>>>>
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fabio.bigliardi at gmail.com  Mon Jan 24 17:26:35 2011
From: fabio.bigliardi at gmail.com (Fabio Bigliardi)
Date: Mon, 24 Jan 2011 15:26:35 +0100
Subject: [Freeswitch-users] How to change global variable at runtime
Message-ID: <AANLkTikmTO3i=gT7qtZGZCEkYtzA=HsX-N7OnrK_vjSA@mail.gmail.com>

Hi all,
I would like to define a global variable in vars.xml and then change
its value at runtime. This value has to be read in dialplan and in
fifo.conf.xml.
I tried *global_setvar* from the CLI but from the log I can see that
the $${var} is expanded to the old value set in vars.xml, not to the
new one.

What is the real effect of global_setvar command?

Have I got to somehow write vars.xml and then issue reloadxml in order
the change to take effect?

Thanks a lot for your support.

Best regards,
F. Bigliardi
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From lists at telefaks.de  Mon Jan 24 18:45:07 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Mon, 24 Jan 2011 16:45:07 +0100
Subject: [Freeswitch-users] Playing Google translation tts
In-Reply-To: <4D2FB01C.8010504@telefaks.de>
References: <4D2E19AB.9020908@telefaks.de>	<AANLkTikzqem2YpaDN77FWpBmRavHrd8qADGZP0FrRrWW@mail.gmail.com>
	<1294969633389-5920422.post@n2.nabble.com>
	<4D2FB01C.8010504@telefaks.de>
Message-ID: <4D3D9E83.4040909@telefaks.de>

Yesterday I updated the discussed Freeswitch machine to latest GIT and the problem is gone.

Thanks to all
Peter



Peter Steinbach schrieb:
> Yesterday I updated a test FS (make current on Ubuntu) and that one
> played the file successfully.
> I have not tried it with our main FS (on Debian where the problem
> occured) yet. So I don't know yet, whether it's from FS or from any
> library used.
>
> Best regards
> Peter
> mazilo schrieb:
>   
>> Norman Tomlins wrote:
>>   
>>     
>>> Peter,
>>>
>>> I have used this in the past and it has been working.
>>>
>>> <extension name="Free_Google_Text_To_Speech">
>>>   <condition field="destination_number" expression="^2115$">
>>>     <action application="answer" data=""/>
>>>     <action application="playback"
>>> data="shout://translate.google.com/translate_tts?tl=en&q=Buy+Cheap+dids+at+www+dot+voice+network+dot+see+eh"/>
>>>   </condition>
>>> </extension>
>>>     
>>>       
>> Thanks Norman. I tried your dialplan above and got the following even though
>> I have mod_shout loaded.
>> EXECUTE sofia/internal/1003 at 192.168.1.15
>> playback(shout://translate.google.com/translate_tts?tl=en&q=Hello+,+It+is+me)
>> 2011-01-13 20:41:57.388295 [ERR] mod_shout.c:800 Error: MPG123 Error at
>> __FILE__:__LINE__.
>> 2011-01-13 20:41:57.389383 [ERR] mod_shout.c:803 Error from mpg123: Invalid
>> mpg123 handle. (code 10)
>> 2011-01-13 20:41:57.397162 [DEBUG] sofia.c:4626 Channel
>> sofia/internal/1003 at 192.168.1.15 entering state [ready][200]
>> 2011-01-13 20:41:57.402589 [NOTICE] switch_core_state_machine.c:189
>> sofia/internal/1003 at 192.168.1.15 has executed the last dialplan instruction,
>> hanging up.
>> 2011-01-13 20:41:57.402589 [DEBUG] switch_channel.c:2535
>> (sofia/internal/1003 at 192.168.1.15) Callstate Change ACTIVE -> HANGUP
>>
>> -----
>> don't and stop are the ONLY two 4-letter words considered offensive to men,
>> but not when used together.
>>   
>>     
>
>
>   


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de




From kbdfck at gmail.com  Mon Jan 24 19:01:45 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Mon, 24 Jan 2011 19:01:45 +0300
Subject: [Freeswitch-users] Is it possible to use att_xfer on channels
	already bridged via loopback?
Message-ID: <AANLkTimWAPV52349=Yge2zjnJyP40EdgDJXgyH-QcOCd@mail.gmail.com>

Is att_xfer or mod_loopback is broken in FS-current?
I use FreeSWITCH Version 1.0.head (git-7eceff4 2011-01-16 22-33-50 +0000)
Seems there were no updates of att_xfer or mod_loopback since that.

I use loopback channel as destination when doing att_xfer to re-enter dialplan.
With loopback_bowout=false and loopback_bowout_on_execute=false this
works. But when any of connected parties tries to do att_xfer again,
all channels get hangup on transferer hangup.

Scenario:

A calls B, B answers
A launches att_xfer via *7, B listens to MOH
A dials C and we do att_xfer to loopback/C
C answers, A hangs up to complete transfer
C and B are now bridged via loopback, `show channels` shows 4 channels
include 2 loopback legs.

Now, C also tries to do in-call transfer with *7.
C launches att_xfer via *7, B listens to MOH
C dials D and do att_xfer to loopback/D
D answers, C hangs up to complete transfer
B and D are hung up instead of be bridged together.

There are also issues with MOH wile running att_xfer, but they are not
so important as att_xfer behavior itself.



-- 
Best regards,

Dmitry Sytchev,
IT Engineer


From anthony.minessale at gmail.com  Mon Jan 24 19:17:53 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Jan 2011 10:17:53 -0600
Subject: [Freeswitch-users] How to change global variable at runtime
In-Reply-To: <AANLkTikmTO3i=gT7qtZGZCEkYtzA=HsX-N7OnrK_vjSA@mail.gmail.com>
References: <AANLkTikmTO3i=gT7qtZGZCEkYtzA=HsX-N7OnrK_vjSA@mail.gmail.com>
Message-ID: <AANLkTikRXU1Ftdbr8tRDsY9o-2RbuysPK09eKMQZeBkO@mail.gmail.com>

use only one $ not 2.

$$ does not mean global, it means eval only once when you load xml.
use ${var} and global_setvar together

On Mon, Jan 24, 2011 at 8:26 AM, Fabio Bigliardi
<fabio.bigliardi at gmail.com> wrote:
> Hi all,
> I would like to define a global variable in vars.xml and then change its
> value at runtime. This value has to be read in dialplan and in
> fifo.conf.xml.
> I tried global_setvar from the CLI but from the log I can see that the
> $${var} is expanded to the old value set in vars.xml, not to the new one.
>
>
> What is the real effect of global_setvar command?
>
> Have I got to somehow write vars.xml and then issue reloadxml in order the
> change to take effect?
>
> Thanks a lot for your support.
>
> Best regards,
>
> F. Bigliardi
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Mon Jan 24 19:20:27 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Jan 2011 10:20:27 -0600
Subject: [Freeswitch-users] IVRD
In-Reply-To: <0040622336AF46389A17B7F04DE28BF7@e1705>
References: <0040622336AF46389A17B7F04DE28BF7@e1705>
Message-ID: <AANLkTik5OZgPOda5JSGSjxnJ+mu_N16xkb+JaPNMsBwF@mail.gmail.com>

check for proper path and execute permissions on the file and perl -c
to make sure it compiles.


On Mon, Jan 24, 2011 at 12:13 AM, Madovsky <infos at madovsky.org> wrote:
> I tried to use IVRD from? wiki example
>
> http://wiki.freeswitch.org/wiki/Ivrd
>
> and server2.pl in ESL directory
> copy and paste in my dialplan ans settings
> so the daemon is running well, but if I attempt
> to call nothing happens unless hangup.
> on the log I can see only
>
> EXECUTE sofia/internal/9999999999999 at default socket(127.0.0.1:8084 full)
>
> I tried the tests of troubleshooting without error
> I don't understand why the events are not received in the perl script
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From u2nsam at gmail.com  Mon Jan 24 19:48:01 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 24 Jan 2011 22:18:01 +0530
Subject: [Freeswitch-users] # in prefix
Message-ID: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>

Hello,

As i send call with prefix 999# the prefix is not passed to the provider
from FS.

Customer sends 999#12127773456 ---> FS ---> 12127773456 ,
it goes without prefix to the provider because of '#', is there any method
to send to the provider 999#12127773456 .

Regds
Sam
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From david.ponzone at ipeva.fr  Mon Jan 24 19:57:24 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 24 Jan 2011 17:57:24 +0100
Subject: [Freeswitch-users] # in prefix
In-Reply-To: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
References: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
Message-ID: <495E0F29-72CB-4383-8FCC-C704CB72A3C8@ipeva.fr>

How are we supposed to provide help without seeing your config ?
You assume that FS does not pass the #.
FS does what you ask it to, so it will send # to leg B if you configure your dialplan accordingly.
The only char I had issue to send to leg B so far is @ (must be replaced by %40).

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 24/01/2011 ? 17:48, Sam a ?crit :

> Hello,
> 
> As i send call with prefix 999# the prefix is not passed to the provider from FS.
> 
> Customer sends 999#12127773456 ---> FS ---> 12127773456 ,
> it goes without prefix to the provider because of '#', is there any method to send to the provider 999#12127773456 .
> 
> Regds
> Sam
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From infos at madovsky.org  Mon Jan 24 20:04:35 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 24 Jan 2011 12:04:35 -0500
Subject: [Freeswitch-users] praise of freeSWITCH developers
References: <20110123111322.5f78cd1d@mail.winet.ch><7FAC1ED2-E969-42C1-9E05-3D741DFFC1DD@freeswitch.org><AANLkTi=BuqzsYY0eu=sRioUB3eyBf-nzJWGspDrrxxx=@mail.gmail.com>
	<1295860695.32615.23.camel@luna.tc.commsmundi.com>
Message-ID: <0A4D008BA10A4EAF8588CC826AB4250A@e1705>

I think there are also people who gave money
without to say anything, so your total is wrong ! :D
nice solidarity, thanks all

----- Original Message ----- 
From: "Fran?ois Delawarde" <fdelawarde at wirelessmundi.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, January 24, 2011 4:18 AM
Subject: Re: [Freeswitch-users] praise of freeSWITCH developers


> On Mon, 2011-01-24 at 13:14 +1300, Kees Varekamp wrote:
>> And another 50 usd from NZ :-)
>
> Nice! We already have raised $385 total so far on this thread. Remember
> that if they can afford fancy cocktails and silk underwear, our favorite
> devs will code faster.
>
> Carry on, just a little more to go for the $500 step!
>
> Fran?ois.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From brian at freeswitch.org  Mon Jan 24 20:04:05 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 Jan 2011 11:04:05 -0600
Subject: [Freeswitch-users] # in prefix
In-Reply-To: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
References: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
Message-ID: <2BAF59AA-B828-4AF6-9BD8-F4E2EAB390A8@freeswitch.org>

You MUST url encode the # key as %23 since its illegal in the URI.  I think we will LET you send it in the URI but some stuff will throw a fit because its invalid.

/b

On Jan 24, 2011, at 10:48 AM, Sam wrote:

> Hello,
> 
> As i send call with prefix 999# the prefix is not passed to the provider from FS.
> 
> Customer sends 999#12127773456 ---> FS ---> 12127773456 ,
> it goes without prefix to the provider because of '#', is there any method to send to the provider 999#12127773456 .
> 
> Regds
> Sam



From david.ponzone at ipeva.fr  Mon Jan 24 20:14:40 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 24 Jan 2011 18:14:40 +0100
Subject: [Freeswitch-users] # in prefix
In-Reply-To: <2BAF59AA-B828-4AF6-9BD8-F4E2EAB390A8@freeswitch.org>
References: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
	<2BAF59AA-B828-4AF6-9BD8-F4E2EAB390A8@freeswitch.org>
Message-ID: <5E44E0E4-D315-4FC9-8CCD-D7A5510BE1FC@ipeva.fr>

Brian,

I can confirm that on my side, I can send it litterally as #.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 24/01/2011 ? 18:04, Brian West a ?crit :

> You MUST url encode the # key as %23 since its illegal in the URI.  I think we will LET you send it in the URI but some stuff will throw a fit because its invalid.
> 
> /b
> 
> On Jan 24, 2011, at 10:48 AM, Sam wrote:
> 
>> Hello,
>> 
>> As i send call with prefix 999# the prefix is not passed to the provider from FS.
>> 
>> Customer sends 999#12127773456 ---> FS ---> 12127773456 ,
>> it goes without prefix to the provider because of '#', is there any method to send to the provider 999#12127773456 .
>> 
>> Regds
>> Sam
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From infos at madovsky.org  Mon Jan 24 20:16:32 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 24 Jan 2011 12:16:32 -0500
Subject: [Freeswitch-users] info
References: <AANLkTimM0fXNSF8e2HWbmJ_ZmHxsCfr0d=-_D9RZ8u-F@mail.gmail.com>
	<AANLkTinRr1K50ebNwHr_t2Jx_r8HqRmZXnmTqdQTizg7@mail.gmail.com>
Message-ID: <59A136AAB8334CCB9E941B6D2870EF87@e1705>

Please next time try to make a max test before to ask in this emailist ;)
  ----- Original Message ----- 
  From: Sam 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 24, 2011 1:18 AM
  Subject: Re: [Freeswitch-users] info


  Pls ignore this email, my bad i didn't reloaded ... sorry to bother.

  Regds
  Sam


  On Mon, Jan 24, 2011 at 11:27 AM, Sam <u2nsam at gmail.com> wrote:

    Have any one tried using <action application="info"/> in git-1c95ad9 2011-01-20 22-43-50 -0300 .
    because its not working for me in this version.

    Regds
    Sam 





------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From jmesquita at freeswitch.org  Mon Jan 24 20:23:09 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 24 Jan 2011 14:23:09 -0300
Subject: [Freeswitch-users] Is it possible to use att_xfer on channels
 already bridged via loopback?
In-Reply-To: <AANLkTimWAPV52349=Yge2zjnJyP40EdgDJXgyH-QcOCd@mail.gmail.com>
References: <AANLkTimWAPV52349=Yge2zjnJyP40EdgDJXgyH-QcOCd@mail.gmail.com>
Message-ID: <AANLkTin6SoYj-_DggPttAPZcwJqfc+HGHhUJvLzWGs-Z@mail.gmail.com>

I am just now discussing this with another developer and the question that
is never answered is:

Why are you trying to use att_xfer if it is your endpoint's duty to make the
transfer? Are you using SIP?

Regards,
Jo?o Mesquita


On Mon, Jan 24, 2011 at 1:01 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:

> Is att_xfer or mod_loopback is broken in FS-current?
> I use FreeSWITCH Version 1.0.head (git-7eceff4 2011-01-16 22-33-50 +0000)
> Seems there were no updates of att_xfer or mod_loopback since that.
>
> I use loopback channel as destination when doing att_xfer to re-enter
> dialplan.
> With loopback_bowout=false and loopback_bowout_on_execute=false this
> works. But when any of connected parties tries to do att_xfer again,
> all channels get hangup on transferer hangup.
>
> Scenario:
>
> A calls B, B answers
> A launches att_xfer via *7, B listens to MOH
> A dials C and we do att_xfer to loopback/C
> C answers, A hangs up to complete transfer
> C and B are now bridged via loopback, `show channels` shows 4 channels
> include 2 loopback legs.
>
> Now, C also tries to do in-call transfer with *7.
> C launches att_xfer via *7, B listens to MOH
> C dials D and do att_xfer to loopback/D
> D answers, C hangs up to complete transfer
> B and D are hung up instead of be bridged together.
>
> There are also issues with MOH wile running att_xfer, but they are not
> so important as att_xfer behavior itself.
>
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Jan 24 20:30:57 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Jan 2011 09:30:57 -0800
Subject: [Freeswitch-users] praise of freeSWITCH developers
In-Reply-To: <0A4D008BA10A4EAF8588CC826AB4250A@e1705>
References: <20110123111322.5f78cd1d@mail.winet.ch>
	<7FAC1ED2-E969-42C1-9E05-3D741DFFC1DD@freeswitch.org>
	<AANLkTi=BuqzsYY0eu=sRioUB3eyBf-nzJWGspDrrxxx=@mail.gmail.com>
	<1295860695.32615.23.camel@luna.tc.commsmundi.com>
	<0A4D008BA10A4EAF8588CC826AB4250A@e1705>
Message-ID: <AANLkTinVdL8SACEhSB_DnQAiXCwVTYy54J28jmqEJ+22@mail.gmail.com>

Thank you all for the unsolicited donations! The FreeSWITCH devs are most
appreciative of our awesome community. Keep spreading the word about how
awesome FreeSWITCH - and its community - really is.

-MC

On Mon, Jan 24, 2011 at 9:04 AM, Madovsky <infos at madovsky.org> wrote:

> I think there are also people who gave money
> without to say anything, so your total is wrong ! :D
> nice solidarity, thanks all
>
> ----- Original Message -----
> From: "Fran?ois Delawarde" <fdelawarde at wirelessmundi.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, January 24, 2011 4:18 AM
> Subject: Re: [Freeswitch-users] praise of freeSWITCH developers
>
>
> > On Mon, 2011-01-24 at 13:14 +1300, Kees Varekamp wrote:
> >> And another 50 usd from NZ :-)
> >
> > Nice! We already have raised $385 total so far on this thread. Remember
> > that if they can afford fancy cocktails and silk underwear, our favorite
> > devs will code faster.
> >
> > Carry on, just a little more to go for the $500 step!
> >
> > Fran?ois.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From milan.m.masek at gmail.com  Mon Jan 24 13:34:09 2011
From: milan.m.masek at gmail.com (Milan Masek)
Date: Mon, 24 Jan 2011 02:34:09 -0800
Subject: [Freeswitch-users] Bria 3 and Freeswitch TLS configuration
	(Milan)
Message-ID: <AANLkTikv5Y478Rpy5pueSOMSUteyGtA18UVHc_=tGUU0@mail.gmail.com>

Help yourself.
With ssldump I am getting
1 5  0.2448 (0.1591)  C>S  Alert
    level           fatal
    value           unknown_ca

Anybody know what can make Bria happy on client computer? How to install
self signed CA certificate for Bria in Ubuntu and MacOSX appropriately?
Thx
M.
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From john at 247-talk.co.uk  Mon Jan 24 14:33:07 2011
From: john at 247-talk.co.uk (John Carpenter)
Date: Mon, 24 Jan 2011 11:33:07 +0000
Subject: [Freeswitch-users] IVR Bridged Call Dropping after 2 rings
In-Reply-To: <1295661432.3014.15.camel@John-Home>
References: <1295661432.3014.15.camel@John-Home>
Message-ID: <1295868787.3088.5.camel@John-Home>

I have spoken to my SIP trunk supplier and they say they have a timeout
of 10 seconds on their system, so if no signalling or RTP media it
disconnects. So question is now how can I get FS to send some signalling
or RTP messages back to SIP trunk while my IVR is dialling an extension.

regards, John Carpenter


On Sat, 2011-01-22 at 01:57 +0000, John Carpenter wrote:

> Hi, I am trying to setup a very simple IVR using LUA. Call arrives
> from a DID SIP trunk and is answered and message is played ok, after a
> particular digit is pressed it bridges the call to an extension which
> is remotely connected. It works but after 2 rings the call to the
> extension is dropped with a SIP message "BYE" from DID provider. If I
> just route the call directly to the extension (no IVR) it works fine.
> It seems like the DID hangs up when the call is bridged to the
> extension. Have tried same thing using the XML IVR Engine and get
> exactly the same result. The IVR script is below
> 
> pathsep = '/'
> session:setAutoHangup(false);
> session:answer()
> prompt = "ivr" .. pathsep .. "247talk.wav"
> invalid = "ivr" .. pathsep .. "ivr-that_was_an_invalid_entry.wav"
> freeswitch.consoleLog("INFO", "Prompt file is '" .. prompt .. "'\n")
> continue = true
> 
> while( session:ready() == true and continue == true) do
>         digits = session:playAndGetDigits(1,1,3,7000,"#", prompt,
> invalid, "\\d+")
>         if (digits == "1") then
>                 continue = false
>                 session:execute("bridge","sofia/external/2476%
> 91.xxx.xx.xx")
>         end
>         if (digits == "2") then
>                 session:execute("bridge","sofia/external/2475%
> 91.xxx.xx.xx")
>         end
>         if (digits == "3") then
>                 continue = false
>                 session:execute("bridge","sofia/external/2475%
> 91.xxx.xx.xx")
>         end
> end
> 
> session:hangup()
> 
> Any help with this greatly appreciated it is driving me nuts.
> 
> regards, John Carpenter 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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From milan.m.masek at gmail.com  Mon Jan 24 14:39:35 2011
From: milan.m.masek at gmail.com (Milan Masek)
Date: Mon, 24 Jan 2011 03:39:35 -0800
Subject: [Freeswitch-users] Howto setup FS SSL certificates with Thawte.
Message-ID: <AANLkTin1qkn=BySX3Erez36TeidvrmgCzxp72HOYQ9WJ@mail.gmail.com>

Hello folks,
Could anybody please improve wiki and write a chapter how to setup FS with
certificates signed by real CA?
Automatic script for self signed certificate gentls_cert is a clue, I
suppose.
Thx
Milan
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From singhujjwal at gmail.com  Mon Jan 24 12:23:53 2011
From: singhujjwal at gmail.com (Ujjwal SIngh)
Date: Mon, 24 Jan 2011 14:53:53 +0530
Subject: [Freeswitch-users] Best effort SRTP offer from FreeSWITCH
Message-ID: <AANLkTimpNnTFH3VhBLLiaV2rnRdroYeoJzFf+d0ryJ6D@mail.gmail.com>

Hi,

Can we initiate Best Effort SRTP offer from FreeSWITCH in the same way as
Polycom phones do,
i.e sending two "m=" lines for the same media, the first line having a SAVP
profile, while the second one having the normal AVP profile.

Regards,
Ujjwal
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From infos at madovsky.org  Mon Jan 24 20:48:13 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 24 Jan 2011 12:48:13 -0500
Subject: [Freeswitch-users] IVRD
References: <0040622336AF46389A17B7F04DE28BF7@e1705>
	<AANLkTik5OZgPOda5JSGSjxnJ+mu_N16xkb+JaPNMsBwF@mail.gmail.com>
Message-ID: <31141E9C13A547D2855217E1B113FBE4@e1705>

Concerning the use of fs_ivrd:

path is ok
permission is root 755
perl -c gives 
../scripts/perl/ivrd-hello_world.pl syntax OK

in dialplan I have :





----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, January 24, 2011 11:20 AM
Subject: Re: [Freeswitch-users] IVRD


check for proper path and execute permissions on the file and perl -c
to make sure it compiles.


On Mon, Jan 24, 2011 at 12:13 AM, Madovsky <infos at madovsky.org> wrote:
> I tried to use IVRD from wiki example
>
> http://wiki.freeswitch.org/wiki/Ivrd
>
> and server2.pl in ESL directory
> copy and paste in my dialplan ans settings
> so the daemon is running well, but if I attempt
> to call nothing happens unless hangup.
> on the log I can see only
>
> EXECUTE sofia/internal/9999999999999 at default socket(127.0.0.1:8084 full)
>
> I tried the tests of troubleshooting without error
> I don't understand why the events are not received in the perl script
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From curriegrad2004 at gmail.com  Mon Jan 24 20:51:23 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 24 Jan 2011 09:51:23 -0800
Subject: [Freeswitch-users] Howto setup FS SSL certificates with Thawte.
In-Reply-To: <AANLkTin1qkn=BySX3Erez36TeidvrmgCzxp72HOYQ9WJ@mail.gmail.com>
References: <AANLkTin1qkn=BySX3Erez36TeidvrmgCzxp72HOYQ9WJ@mail.gmail.com>
Message-ID: <AANLkTimTmUGcoTW6RFoG--icuQC7mS_rQ0L8VfrD3vH_@mail.gmail.com>

You just simply rename the key/cert to agent.pem. Shouldn't be too
hard to do it in the first place.

On Mon, Jan 24, 2011 at 3:39 AM, Milan Masek <milan.m.masek at gmail.com> wrote:
> Hello folks,
> Could anybody please improve wiki and write a chapter how to setup FS with
> certificates signed by real CA?
> Automatic script for self signed certificate gentls_cert is a clue, I
> suppose.
> Thx
> Milan
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From infos at madovsky.org  Mon Jan 24 20:57:06 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 24 Jan 2011 12:57:06 -0500
Subject: [Freeswitch-users] IVRD
Message-ID: <B890EB1A467549D48DB330F5795906E9@e1705>

Sorry I sent this email by mistake without to finish it.
so in my dialplan I have :
                <extension name="gluglu">
                    <condition field="destination_number" 
expression="^999$">
                        <action application="set" 
data="ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl"/>
                        <action application="socket" data="127.0.0.1:9090 
full"/>
                        <action application="answer"/>
                        <action application="voicemail" data="default 
${domain_name} ${dialed_extension}"/>
                    <condition>
                </extension>

and when I call this extension in log level 7 I can see :

Dialplan: sofia/internal/9999999999999 at default Action 
set(ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl)
Dialplan: sofia/internal/9999999999999 at default Action socket(127.0.0.1:9090 
full)
Starting ivrd-hello_world.pl...

and no ivrd-hello_world.pl code is executed in the dialplan unless the print 
"Starting ivrd-hello_world.pl";

the same if I replace fs_ivrd with server2.pl for example

Thanks


----- Original Message ----- 
From: "Madovsky" <infos at madovsky.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, January 24, 2011 12:48 PM
Subject: Re: [Freeswitch-users] IVRD


> Concerning the use of fs_ivrd:
>
> path is ok
> permission is root 755
> perl -c gives ../scripts/perl/ivrd-hello_world.pl syntax OK
>
> in dialplan I have :
>
>
>
>
>
> ----- Original Message ----- 
> From: "Anthony Minessale" <anthony.minessale at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, January 24, 2011 11:20 AM
> Subject: Re: [Freeswitch-users] IVRD
>
>
> check for proper path and execute permissions on the file and perl -c
> to make sure it compiles.
>
>
> On Mon, Jan 24, 2011 at 12:13 AM, Madovsky <infos at madovsky.org> wrote:
>> I tried to use IVRD from wiki example
>>
>> http://wiki.freeswitch.org/wiki/Ivrd
>>
>> and server2.pl in ESL directory
>> copy and paste in my dialplan ans settings
>> so the daemon is running well, but if I attempt
>> to call nothing happens unless hangup.
>> on the log I can see only
>>
>> EXECUTE sofia/internal/9999999999999 at default socket(127.0.0.1:8084 full)
>>
>> I tried the tests of troubleshooting without error
>> I don't understand why the events are not received in the perl script
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From robert.hadley at teotech.com  Mon Jan 24 20:58:56 2011
From: robert.hadley at teotech.com (Robert Hadley)
Date: Mon, 24 Jan 2011 09:58:56 -0800
Subject: [Freeswitch-users] praise of freeSWITCH developers
In-Reply-To: <AANLkTinVdL8SACEhSB_DnQAiXCwVTYy54J28jmqEJ+22@mail.gmail.com>
References: <20110123111322.5f78cd1d@mail.winet.ch>
	<7FAC1ED2-E969-42C1-9E05-3D741DFFC1DD@freeswitch.org>
	<AANLkTi=BuqzsYY0eu=sRioUB3eyBf-nzJWGspDrrxxx=@mail.gmail.com>
	<1295860695.32615.23.camel@luna.tc.commsmundi.com>
	<0A4D008BA10A4EAF8588CC826AB4250A@e1705>
	<AANLkTinVdL8SACEhSB_DnQAiXCwVTYy54J28jmqEJ+22@mail.gmail.com>
Message-ID: <E826C94A3C6BE44198FAF4D66B5AB4D80A51086A5A@VA3DIAXVS351.RED001.local>


Thanks,
I just caught up reading UL emails, and just added my $50 Thank You to the FS developers
-Robert

From: Michael Collins [mailto:msc at freeswitch.org]
Sent: Monday, January 24, 2011 9:31 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] praise of freeSWITCH developers

Thank you all for the unsolicited donations! The FreeSWITCH devs are most appreciative of our awesome community. Keep spreading the word about how awesome FreeSWITCH - and its community - really is.

-MC
On Mon, Jan 24, 2011 at 9:04 AM, Madovsky <infos at madovsky.org<mailto:infos at madovsky.org>> wrote:
I think there are also people who gave money
without to say anything, so your total is wrong ! :D
nice solidarity, thanks all

----- Original Message -----
From: "Fran?ois Delawarde" <fdelawarde at wirelessmundi.com<mailto:fdelawarde at wirelessmundi.com>>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Sent: Monday, January 24, 2011 4:18 AM
Subject: Re: [Freeswitch-users] praise of freeSWITCH developers

> On Mon, 2011-01-24 at 13:14 +1300, Kees Varekamp wrote:
>> And another 50 usd from NZ :-)
>
> Nice! We already have raised $385 total so far on this thread. Remember
> that if they can afford fancy cocktails and silk underwear, our favorite
> devs will code faster.
>
> Carry on, just a little more to go for the $500 step!
>
> Fran?ois.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From chris at cloudtel.com  Mon Jan 24 21:22:45 2011
From: chris at cloudtel.com (Chris Burns)
Date: Mon, 24 Jan 2011 13:22:45 -0500
Subject: [Freeswitch-users] Bria 3 and Freeswitch TLS configuration
	(Milan)
In-Reply-To: <AANLkTikv5Y478Rpy5pueSOMSUteyGtA18UVHc_=tGUU0@mail.gmail.com>
References: <AANLkTikv5Y478Rpy5pueSOMSUteyGtA18UVHc_=tGUU0@mail.gmail.com>
Message-ID: <AANLkTi=m6_e3oC=R+DgV_-xyvR5CRMe2tyfEeAYLbNVS@mail.gmail.com>

Put the the ca cert in your ca store if it needs to be there :D

In Deb they go in /usr/share/ca-certificates belonging to the package
bearing the same name. Ubuntu prob the same as Deb. Windows I dont know,
theres prob some cutesy wizard for it.

On Mon, Jan 24, 2011 at 5:34 AM, Milan Masek <milan.m.masek at gmail.com>wrote:

> Help yourself.
> With ssldump I am getting
> 1 5  0.2448 (0.1591)  C>S  Alert
>     level           fatal
>     value           unknown_ca
>
> Anybody know what can make Bria happy on client computer? How to install
> self signed CA certificate for Bria in Ubuntu and MacOSX appropriately?
> Thx
> M.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kbdfck at gmail.com  Mon Jan 24 21:51:45 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Mon, 24 Jan 2011 21:51:45 +0300
Subject: [Freeswitch-users] Is it possible to use att_xfer on channels
 already bridged via loopback?
In-Reply-To: <AANLkTin6SoYj-_DggPttAPZcwJqfc+HGHhUJvLzWGs-Z@mail.gmail.com>
References: <AANLkTimWAPV52349=Yge2zjnJyP40EdgDJXgyH-QcOCd@mail.gmail.com>
	<AANLkTin6SoYj-_DggPttAPZcwJqfc+HGHhUJvLzWGs-Z@mail.gmail.com>
Message-ID: <AANLkTi=VGmzii5_GkU06orX-BZxtZkj-a12qtPE+fkNG@mail.gmail.com>

Yes, I'm using SIP as my endpoints.

There are number of reasons why I want to use in-call triggered transfers.

1. We are migrating several large ISP PBX to Freeswitch from Asterisk.
In-call transfers in Asterisk via res_features are mature and stable,
and many users use it - if we remove this feature, we will loose our
users.

2. Not all SIP endpoints support call transfer, and despite majority
of devices support this, we want fine control over who is allowed to
do transfer in which situation - there are many paths of call in our
PBXs.

3. We also need to provide call transfer functionality to external
PSTN users, which are dialed via SIP proxy and then SIP-E1 gateway
like audiocodes or cisco. There is no simple universal solution to
trigger such transfers from PSTN side besides in-call transfers with
bind_meta_app.

I agree, in ideal world we would live without server-side attended
transfers, but for now att_xfer is an ultimate feature of real PBX,
even old Avaya and Samsung boxes support this :)


2011/1/24 Jo?o Mesquita <jmesquita at freeswitch.org>:
> I am just now discussing this with another developer and the question that
> is never answered is:
> Why are you trying to use att_xfer if it is your endpoint's duty to make the
> transfer? Are you using SIP?
> Regards,
> Jo?o Mesquita
>
>
> On Mon, Jan 24, 2011 at 1:01 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>
>> Is att_xfer or mod_loopback is broken in FS-current?
>> I use FreeSWITCH Version 1.0.head (git-7eceff4 2011-01-16 22-33-50 +0000)
>> Seems there were no updates of att_xfer or mod_loopback since that.
>>
>> I use loopback channel as destination when doing att_xfer to re-enter
>> dialplan.
>> With loopback_bowout=false and loopback_bowout_on_execute=false this
>> works. But when any of connected parties tries to do att_xfer again,
>> all channels get hangup on transferer hangup.
>>
>> Scenario:
>>
>> A calls B, B answers
>> A launches att_xfer via *7, B listens to MOH
>> A dials C and we do att_xfer to loopback/C
>> C answers, A hangs up to complete transfer
>> C and B are now bridged via loopback, `show channels` shows 4 channels
>> include 2 loopback legs.
>>
>> Now, C also tries to do in-call transfer with *7.
>> C launches att_xfer via *7, B listens to MOH
>> C dials D and do att_xfer to loopback/D
>> D answers, C hangs up to complete transfer
>> B and D are hung up instead of be bridged together.
>>
>> There are also issues with MOH wile running att_xfer, but they are not
>> so important as att_xfer behavior itself.
>>
>>
>>
>> --
>> Best regards,
>>
>> Dmitry Sytchev,
>> IT Engineer
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Best regards,

Dmitry Sytchev,
IT Engineer


From msc at freeswitch.org  Mon Jan 24 22:46:55 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Jan 2011 11:46:55 -0800
Subject: [Freeswitch-users] Freeswitch Sends RTP To Private IP Address
In-Reply-To: <AANLkTinVK90fOOpCq3Nj7ihb6FZom_3k33cNowcCo8T-@mail.gmail.com>
References: <AANLkTimru1i31uSYW+bGK_3z=GiW=QNPzjvTZRRwP5sq@mail.gmail.com>
	<AANLkTinVK90fOOpCq3Nj7ihb6FZom_3k33cNowcCo8T-@mail.gmail.com>
Message-ID: <AANLkTimZB7b68LmF-RmYXe91_Y1K+k5xkt+=hARgzaj2@mail.gmail.com>

Your candor is refreshing... and thanks for posting the "solution" to your
"problem"! :)

-MC

On Fri, Jan 21, 2011 at 3:19 PM, Mike <motosota at gmail.com> wrote:

> And the Cretin-Of-The-Day award goes to .... 'me'.
>
> Having re-looked at the Wireshark trace I realised I hadn't opened up
> enough ports in the FreeSWITCH firewall to let the RTP from the Polycom
> phone through - so it was unable to learn what the real public IP address
> and port number.
>
> This is what I get for breaking my own rules about working too late!
>
> Forget I asked....move along....nothing to see here.
>
>
> On Fri, Jan 21, 2011 at 10:44 PM, Mike <motosota at gmail.com> wrote:
>
>> Scenario. Polycom phone, private IP address, behind Cisco router doing NAT
>> overload to a public IP address (SIP 'fixup' disabled, so the Cisco isn't
>> mangling anything).
>>
>> RTP stream from FreeSWITCH is sent to the private, not public address of
>> the phone.
>>
>> I've got this working on 1.0.6  but I've tried this with FreeSWITCH
>> Version 1.0.head (git-7070061 2011-01-20 13-52-00 -0600) and it doesn't seem
>> to work for me.
>>
>> Here http://pastebin.freeswitch.org/15109 is my sip profile.
>>
>> Here http://pastebin.freeswitch.org/15110 is a FreeSWITCH log for a call
>> made from one of the Polycoms to voicemail with SIP tracing enabled.
>>
>> And here http://pastebin.freeswitch.org/15111 is the wireshark trace
>> showing the media stream being sent by FreeSWITCH to the wrong port.
>>
>> I may be (hopefully) missing something very obvious here.
>>
>> Mike
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Mon Jan 24 22:50:44 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 24 Jan 2011 20:50:44 +0100
Subject: [Freeswitch-users] FS HA
Message-ID: <AANLkTi=_BPuE5KX4D48JDoOW9U3netsr9uDU9Lo5CgYO@mail.gmail.com>

hello guys,


i configured FS HA and looks like its trying to recover the call .. but the
re-INVITE fails due to "wrong/missed" codec capability.




freeswitch at internal>
freeswitch at internal> sofia profile external siptrace on
Enabled sip debugging on external
freeswitch at internal> sofia profile internal siptrace on
Enabled sip debugging on internal
freeswitch at internal>
freeswitch at internal>
freeswitch at internal>
freeswitch at internal> sofia recover
Recovered 1 call(s)

2011-01-24 21:32:31.667404 [CRIT] switch_odbc.c:205 The sql server is not
responding for DSN COREFSdrv [STATE: HY000 CODE 7 ERROR: [unixODBC]Unknown
error;
FATAL:  terminating connection due to administrator command
][176]
freeswitch at internal> 2011-01-24 21:32:31.667404 [INFO] switch_odbc.c:210 The
connection has been re-established
2011-01-24 21:32:32.726402 [NOTICE] switch_channel.c:669 New Channel
sofia/external/385914392122 at 195.88.212.39[808b0a98-8928-4ca2-a320-8e026ba94018]
*2011-01-24 21:32:32.726402 [DEBUG] sofia_glue.c:2354 Set Codec
sofia/external/385914392122 at 195.88.212.39 PCMA/8000 20 ms 160 samples*
2011-01-24 21:32:32.726402 [DEBUG] sofia_glue.c:2594 AUDIO RTP
[sofia/external/385914392122 at 195.88.212.39] 195.88.212.30 port 20092 ->
195.88.212.39 port 17750 codec: 8 ms: 20
2011-01-24 21:32:32.726402 [DEBUG] switch_rtp.c:1182 Starting timer [soft]
160 bytes per 20ms
2011-01-24 21:32:32.728288 [DEBUG] sofia_glue.c:2774 Set 2833 dtmf send
payload to 101
2011-01-24 21:32:32.728288 [DEBUG] sofia_glue.c:2779 Set 2833 dtmf receive
payload to 101
2011-01-24 21:32:32.728288 [DEBUG] sofia_glue.c:2790 Set comfort noise
payload to 13
2011-01-24 21:32:32.779414 [DEBUG] sofia_glue.c:4246 (sofia/external/
385914392122 at 195.88.212.39) State Change CS_NEW -> CS_INIT
2011-01-24 21:32:32.779414 [DEBUG] switch_core_session.c:1021 Send signal
sofia/external/385914392122 at 195.88.212.39 [BREAK]
2011-01-24 21:32:32.779414 [NOTICE] sofia_glue.c:4249 Resurrecting fallen
channel sofia/external/385914392122 at 195.88.212.39
2011-01-24 21:32:32.780381 [DEBUG] switch_core_state_machine.c:314
(sofia/external/385914392122 at 195.88.212.39) Running State Change CS_INIT
2011-01-24 21:32:32.780381 [DEBUG] switch_core_state_machine.c:338
(sofia/external/385914392122 at 195.88.212.39) State INIT
2011-01-24 21:32:32.780381 [DEBUG] mod_sofia.c:83 sofia/external/
385914392122 at 195.88.212.39 SOFIA INIT
2011-01-24 21:32:32.780381 [DEBUG] mod_sofia.c:114 (sofia/external/
385914392122 at 195.88.212.39) State Change CS_INIT -> CS_EXECUTE
2011-01-24 21:32:32.780381 [DEBUG] switch_core_session.c:1021 Send signal
sofia/external/385914392122 at 195.88.212.39 [BREAK]
2011-01-24 21:32:32.780381 [DEBUG] switch_core_state_machine.c:338
(sofia/external/385914392122 at 195.88.212.39) State INIT going to sleep
2011-01-24 21:32:32.780381 [DEBUG] switch_core_state_machine.c:314
(sofia/external/385914392122 at 195.88.212.39) Running State Change CS_EXECUTE
2011-01-24 21:32:32.780381 [DEBUG] switch_core_state_machine.c:348
(sofia/external/385914392122 at 195.88.212.39) State EXECUTE
2011-01-24 21:32:32.780381 [DEBUG] mod_sofia.c:226 sofia/external/
385914392122 at 195.88.212.39 SOFIA EXECUTE
2011-01-24 21:32:32.780381 [DEBUG] switch_core_state_machine.c:157
sofia/external/385914392122 at 195.88.212.39 Standard EXECUTE
EXECUTE sofia/external/385914392122 at 195.88.212.39 set(playback_delimiter=!)
send 1029 bytes to udp/[195.88.212.39]:62342 at 20:32:32.781241:
   ------------------------------------------------------------------------
   INVITE sip:385914392122 at 195.88.212.39
<sip%3A385914392122 at 195.88.212.39>SIP/2.0
   Via: SIP/2.0/UDP 195.88.212.30:5080;rport;branch=z9hG4bKKDX31Z2Q71B3j
   Route: <sip:385914392122 at 195.88.212.39:62342;lr>
   Max-Forwards: 69
   From: <sip:38518880050 at 195.88.212.30 <sip%3A38518880050 at 195.88.212.30>
>;tag=DFejDg1B59y9r
   To: <sip:385914392122 at 195.88.212.39 <sip%3A385914392122 at 195.88.212.39>
>;tag=C4B0BE6C-1B8
   Call-ID: 1785B58B-272611E0-8909D88B-B59C2BD1 at 195.88.212.39
   CSeq: 7614512 INVITE
   Contact: <sip:385914392122 at 195.88.212.30:5080>
   User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 168
   X-FS-Support: update_display
   Remote-Party-ID: "38518880050"
<sip:38518880050 at 195.88.212.30<sip%3A38518880050 at 195.88.212.30>
>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1295881060 1295881062 IN IP4 195.88.212.30
   s=FreeSWITCH
   c=IN IP4 195.88.212.30
   t=0 0*
   m=audio 20092 RTP/AVP 0 13*
*   a=rtpmap:13 CN/8000*
   a=ptime:20
   ------------------------------------------------------------------------
2011-01-24 21:32:32.781430 [DEBUG] sofia.c:4153 Channel sofia/external/
385914392122 at 195.88.212.39 entering state [calling][0]
recv 496 bytes from udp/[195.88.212.39]:5060 at 20:32:32.784077:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 195.88.212.30:5080;rport;branch=z9hG4bKKDX31Z2Q71B3j
   From: <sip:38518880050 at 195.88.212.30 <sip%3A38518880050 at 195.88.212.30>
>;tag=DFejDg1B59y9r
   To: <sip:385914392122 at 195.88.212.39 <sip%3A385914392122 at 195.88.212.39>
>;tag=C4B0BE6C-1B8
   Date: Mon, 24 Jan 2011 19:22:34 GMT
   Call-ID: 1785B58B-272611E0-8909D88B-B59C2BD1 at 195.88.212.39
   Server: Cisco-SIPGateway/IOS-12.x
   CSeq: 7614512 INVITE
   Allow-Events: telephone-event
   Remote-Party-ID:
<sip:385914392122 at 195.88.212.39<sip%3A385914392122 at 195.88.212.39>
>;party=called;screen=yes;privacy=off
   Content-Length: 0

   ------------------------------------------------------------------------
recv 376 bytes from udp/[195.88.212.39]:5060 at 20:32:32.784195:
   ------------------------------------------------------------------------
   SIP/2.0 488 Not Acceptable Media
   Via: SIP/2.0/UDP 195.88.212.30:5080;rport;branch=z9hG4bKKDX31Z2Q71B3j
   From: <sip:38518880050 at 195.88.212.30 <sip%3A38518880050 at 195.88.212.30>
>;tag=DFejDg1B59y9r
   To: <sip:385914392122 at 195.88.212.39 <sip%3A385914392122 at 195.88.212.39>
>;tag=C4B0BE6C-1B8
   Call-ID: 1785B58B-272611E0-8909D88B-B59C2BD1 at 195.88.212.39
   Warning: 304 195.88.212.39 "Media Type(s) Unavailable"
   CSeq: 7614512 INVITE
   Content-Length: 0

   ------------------------------------------------------------------------
send 395 bytes to udp/[195.88.212.39]:62342 at 20:32:32.784303:
   ------------------------------------------------------------------------
   ACK sip:385914392122 at 195.88.212.39 <sip%3A385914392122 at 195.88.212.39>SIP/2.0
   Via: SIP/2.0/UDP 195.88.212.30:5080;rport;branch=z9hG4bKKDX31Z2Q71B3j
   Route: <sip:385914392122 at 195.88.212.39:62342;lr>
   Max-Forwards: 69
   From: <sip:38518880050 at 195.88.212.30 <sip%3A38518880050 at 195.88.212.30>
>;tag=DFejDg1B59y9r
   To: <sip:385914392122 at 195.88.212.39 <sip%3A385914392122 at 195.88.212.39>
>;tag=C4B0BE6C-1B8
   Call-ID: 1785B58B-272611E0-8909D88B-B59C2BD1 at 195.88.212.39
   CSeq: 7614512 ACK
   Content-Length: 0

   ------------------------------------------------------------------------
2011-01-24 21:32:32.784397 [DEBUG] sofia.c:4153 Channel sofia/external/
385914392122 at 195.88.212.39 entering state [terminated][488]
2011-01-24 21:32:32.784397 [NOTICE] sofia.c:4789 Hangup sofia/external/
385914392122 at 195.88.212.39 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
2011-01-24 21:32:32.784397 [DEBUG] switch_channel.c:2102 Send signal
sofia/external/385914392122 at 195.88.212.39 [KILL]
2011-01-24 21:32:32.784397 [DEBUG] switch_core_session.c:1021 Send signal
sofia/external/385914392122 at 195.88.212.39 [BREAK]
2011-01-24 21:32:32.852411 [DEBUG] mod_dptools.c:816 sofia/external/
385914392122 at 195.88.212.39 SET [playback_delimiter]=[!]
2011-01-24 21:32:32.853505 [DEBUG] switch_core_state_machine.c:348
(sofia/external/385914392122 at 195.88.212.39) State EXECUTE going to sleep
2011-01-24 21:32:32.853505 [DEBUG] switch_core_state_machine.c:314
(sofia/external/385914392122 at 195.88.212.39) Running State Change CS_HANGUP
2011-01-24 21:32:32.853505 [DEBUG] switch_core_state_machine.c:499
(sofia/external/385914392122 at 195.88.212.39) State HANGUP
2011-01-24 21:32:32.861414 [DEBUG] mod_sofia.c:408 sofia/external/
385914392122 at 195.88.212.39 Overriding SIP cause 488 with 488 from the other
leg
2011-01-24 21:32:32.861414 [DEBUG] mod_sofia.c:414 Channel sofia/external/
385914392122 at 195.88.212.39 hanging up, cause: INCOMPATIBLE_DESTINATION
2011-01-24 21:32:32.863454 [DEBUG] switch_core_state_machine.c:46
sofia/external/385914392122 at 195.88.212.39 Standard HANGUP, cause:
INCOMPATIBLE_DESTINATION
2011-01-24 21:32:32.863454 [DEBUG] switch_core_state_machine.c:499
(sofia/external/385914392122 at 195.88.212.39) State HANGUP going to sleep
2011-01-24 21:32:32.863454 [DEBUG] switch_core_state_machine.c:333
(sofia/external/385914392122 at 195.88.212.39) State Change CS_HANGUP ->
CS_REPORTING
2011-01-24 21:32:32.863454 [DEBUG] switch_core_session.c:1021 Send signal
sofia/external/385914392122 at 195.88.212.39 [BREAK]
2011-01-24 21:32:32.863454 [DEBUG] switch_core_state_machine.c:314
(sofia/external/385914392122 at 195.88.212.39) Running State Change
CS_REPORTING
2011-01-24 21:32:32.863454 [DEBUG] switch_core_state_machine.c:590
(sofia/external/385914392122 at 195.88.212.39) State REPORTING
2011-01-24 21:32:32.864328 [DEBUG] switch_core_state_machine.c:53
sofia/external/385914392122 at 195.88.212.39 Standard REPORTING, cause:
INCOMPATIBLE_DESTINATION
2011-01-24 21:32:32.864328 [DEBUG] switch_core_state_machine.c:590
(sofia/external/385914392122 at 195.88.212.39) State REPORTING going to sleep
2011-01-24 21:32:32.864328 [DEBUG] switch_core_state_machine.c:327
(sofia/external/385914392122 at 195.88.212.39) State Change CS_REPORTING ->
CS_DESTROY
2011-01-24 21:32:32.864328 [DEBUG] switch_core_session.c:1021 Send signal
sofia/external/385914392122 at 195.88.212.39 [BREAK]
2011-01-24 21:32:32.864328 [DEBUG] switch_core_session.c:1164 Session 1
(sofia/external/385914392122 at 195.88.212.39) Locked, Waiting on external
entities
2011-01-24 21:32:32.864328 [NOTICE] switch_core_session.c:1182 Session 1
(sofia/external/385914392122 at 195.88.212.39) Ended
2011-01-24 21:32:32.864328 [NOTICE] switch_core_session.c:1184 Close Channel
sofia/external/385914392122 at 195.88.212.39 [CS_DESTROY]
2011-01-24 21:32:32.864328 [DEBUG] switch_core_state_machine.c:428
(sofia/external/385914392122 at 195.88.212.39) Running State Change CS_DESTROY
2011-01-24 21:32:32.864328 [DEBUG] switch_core_state_machine.c:439
(sofia/external/385914392122 at 195.88.212.39) State DESTROY
2011-01-24 21:32:32.864328 [DEBUG] mod_sofia.c:341 sofia/external/
385914392122 at 195.88.212.39 SOFIA DESTROY
2011-01-24 21:32:32.864328 [DEBUG] switch_core_state_machine.c:60
sofia/external/385914392122 at 195.88.212.39 Standard DESTROY
2011-01-24 21:32:32.864328 [DEBUG] switch_core_state_machine.c:439
(sofia/external/385914392122 at 195.88.212.39) State DESTROY going to sleep

freeswitch at internal>
freeswitch at internal>
freeswitch at internal> 2011-01-24 21:32:51.813359 [CRIT] switch_odbc.c:205 The
sql server is not responding for DSN COREFSdrv [STATE: HY000 CODE 7 ERROR:
[unixODBC]Unknown error;
FATAL:  terminating connection due to administrator command
][176]
2011-01-24 21:32:51.813359 [INFO] switch_odbc.c:210 The connection has been
re-established
2011-01-24 21:32:52.810408 [CRIT] switch_odbc.c:205 The sql server is not
responding for DSN COREFSdrv [STATE: HY000 CODE 7 ERROR: [unixODBC]Unknown
error;
FATAL:  terminating connection due to administrator command
][176]
2011-01-24 21:32:52.810408 [INFO] switch_odbc.c:210 The connection has been
re-established
2011-01-24 21:32:53.817405 [CRIT] switch_odbc.c:205 The sql server is not
responding for DSN COREFSdrv [STATE: HY000 CODE 7 ERROR: [unixODBC]Unknown
error;
FATAL:  terminating connection due to administrator command
][176]
2011-01-24 21:32:53.817405 [INFO] switch_odbc.c:210 The connection has been
re-established
recv 432 bytes from udp/[195.88.212.39]:62342 at 20:33:06.045642:
   ------------------------------------------------------------------------
   BYE sip:385914392122 at 195.88.212.30:5080 SIP/2.0
   Via: SIP/2.0/UDP  195.88.212.39:5060;branch=z9hG4bK240FA
   From: <sip:385914392122 at 195.88.212.39 <sip%3A385914392122 at 195.88.212.39>
>;tag=C4B0BE6C-1B8
   To: <sip:38518880050 at 195.88.212.30 <sip%3A38518880050 at 195.88.212.30>
>;tag=DFejDg1B59y9r
   Date: Mon, 24 Jan 2011 19:22:34 GMT
   Call-ID: 1785B58B-272611E0-8909D88B-B59C2BD1 at 195.88.212.39
   User-Agent: Cisco-SIPGateway/IOS-12.x
   Max-Forwards: 70
   Timestamp: 1295896987
   CSeq: 102 BYE
   Content-Length: 0

   ------------------------------------------------------------------------
send 522 bytes to udp/[195.88.212.39]:5060 at 20:33:06.045858:
   ------------------------------------------------------------------------
   SIP/2.0 481 Call Does Not Exist
   Via: SIP/2.0/UDP  195.88.212.39:5060;branch=z9hG4bK240FA
   From: <sip:385914392122 at 195.88.212.39 <sip%3A385914392122 at 195.88.212.39>
>;tag=C4B0BE6C-1B8
   To: <sip:38518880050 at 195.88.212.30 <sip%3A38518880050 at 195.88.212.30>
>;tag=DFejDg1B59y9r
   Call-ID: 1785B58B-272611E0-8909D88B-B59C2BD1 at 195.88.212.39
   CSeq: 102 BYE
   Timestamp: 1295896987 0.000191
   User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Content-Length: 0

   ------------------------------------------------------------------------

freeswitch at internal>
freeswitch at internal>






my codec settings in vars.conf
  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,GSM"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA,GSM"/>


my codec settings in sip_profiles/*
    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>



now, where the is FS picking up the codec list for the re-INVITE after
"sofia recover"




from the debug:

2011-01-24 21:32:32.726402 [DEBUG] sofia_glue.c:2354 Set Codec
sofia/external/385914392122 at 195.88.212.39 PCMA/8000 20 ms 160 samples

i can see sofia knows what codec is being used, but somehow its not sending
it in re-INVITE.



BTW: yea, i know, i know .. the version is a bit outdated but we can still
make it working .. :)
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From msc at freeswitch.org  Mon Jan 24 23:13:33 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Jan 2011 12:13:33 -0800
Subject: [Freeswitch-users] Javascript IVR session question
In-Reply-To: <CDF5E00590C9BA4AB96BE7CDD28C86AE07872CFA48@FS1.wvds.local>
References: <CDF5E00590C9BA4AB96BE7CDD28C86AE07872CFA48@FS1.wvds.local>
Message-ID: <AANLkTimaJDm8Fc1EKk1hKrnF60cWBXyg4bzdQ+77NMG=@mail.gmail.com>

If I read this correctly you are hanging up the channel that you later wish
to bridge to user/202. Why do you need to hangup? Perhaps you could describe
a little more about the application? I'm sure we can help you iron out the
details.

-MC

On Sun, Jan 23, 2011 at 12:48 PM, Erik Dekkers <erik.dekkers at wvds.nl> wrote:

>   Hey ppl,
>
> At the moment im building a Javascript based IVR but im kind of stuck on a
> part.
>
> The IVR should do this:
>
> - Answer session (working)
> - Play some wav files (working)
> - Record a message to file (working)
> - Hang up the first session (working)
> - Call the second session (not working)
> - Play the previous recorded file (not working)
>
> After I dial the second session, the console says "channel is hungup
> already". How should i do this?
>
> Kind regards,
>
> Erik Dekkers (wvds-nl on IRC)
>
>
>
> my script:
>
> var allDigits = "";
> function on_dtmf(session, type, digits, arg)
> {
>     if (digits.digit == "#") {
>         return allDigits;
>     }
>     if (digits.digit == "*") {
>         return false; //stop the recording.
>     }
>     console_log("digit: " + digits.digit + "\n");
>     allDigits += digits.digit;
>     return(allDigits);
> }
> session.answer();
> if (session.ready()) {
>     allDigits = "";
>     var rtn;
>     rtn = session.streamFile("/home/edekkers/sounds/10_spreek_in.wav",
> on_dtmf, "");
>     if (session.ready()) {
>         var tmp_Filename = "/tmp/test.wav";
>         if (session.ready()) {
>             rtn = session.recordFile(tmp_Filename, on_dtmf, "", 120);
>             }
>         rtn =
> session.streamFile("/home/edekkers/sounds/11_bericht_is_ontvangen.wav",
> on_dtmf, "");
>         if (session.ready()) {
>             session.hangup();
>         }
>     }
> }
> session.execute("bridge","user/202")
> if (session.ready()) {
>     session.streamFile("/tmp/test.wav");
> }
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Jan 24 23:16:20 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Jan 2011 12:16:20 -0800
Subject: [Freeswitch-users] Help regard to park_timeout
In-Reply-To: <AANLkTinHSCfH6Ydh5ipGfhMQbpcHc+uJd6oxGhJNNFPy@mail.gmail.com>
References: <AANLkTi=edjFv90+q3G2kkzoJ76kWgx+NBe=Qx3CYA2Jd@mail.gmail.com>
	<AANLkTinHSCfH6Ydh5ipGfhMQbpcHc+uJd6oxGhJNNFPy@mail.gmail.com>
Message-ID: <AANLkTi=GvaDh7EsjJmsV-UzDw6LH=99KVgYjg8HxJ-Ek@mail.gmail.com>

Do a uuid_dump on the channel and see what all is there. Also, try doing
this:
eval uuid:9b59d172-2781-11e0-8586-8390cbcc860f ${park_timeout}

See what happens.

-MC

On Sun, Jan 23, 2011 at 10:21 PM, lakshmanan ganapathy <lakindia89 at gmail.com
> wrote:

> Hi all,
> I've done a further experimentation and I need a clarification.
>
> From CLI>
> originate {ignore_early_media=true,park_timeout=50,api_hangup_hook='perl
> /root/a.pl',exec_after_bridge_app=park}freetdm/grp1/a/9952248266 &park()
>
> Then I executed the following commands from CLI.
> uuid_getvar 9b59d172-2781-11e0-8586-8390cbcc860f park_timeout
> _undef_
> uuid_getvar 9b59d172-2781-11e0-8586-8390cbcc860f api_hangup_hook
> perl /root/a.pl
>
> I don't know why uuid_getvar, returns undef for park_timeout variable. But
> the call is hangup once 50 seconds is reached. Can some one pls explain what
> it is printing as __undef__
>
>
>
> On Fri, Jan 21, 2011 at 6:37 PM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> Dear all,
>> I was using park_timeout and I come across the following scenario which I
>> felt something is missing.
>> I've originated a call as follows.
>>
>> originate
>> {ignore_early_media=true,exec_after_bridge_app=park,park_timeout=60,api_hangup_hook='perl
>> /root/a.pl'}freetdm/grp1/a/9952248266 &park()
>>
>> Once the call is answered I originated another call.
>> originate {ignore_early_media=true,park_timeout=60,api_hangup_hook='perl
>> /root/a.pl'}freetdm/grp1/a/9843171457 &park()
>>
>> Once this call is also answered, I said "uuid_bridge <uuid1> <uuid2>".
>> Both call gets bridged. After some time, I hangup the second call
>> (9843171457). Now the first call goes into park().
>>
>> I expect that the first call will hangup after 60 seconds, but it didn't.
>>
>> The freeswitch log is here
>> http://pastebin.freeswitch.org/15099
>>
>> When I start to use the park_timeout, I thought once a leg is in park,
>> then the timer will start, and once it is unparked for various reason the
>> timer will be reseted. After sometime, when the leg again comes in park, the
>> timer will start. Is this correct?
>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Jan 24 23:17:20 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 Jan 2011 14:17:20 -0600
Subject: [Freeswitch-users] FS HA
In-Reply-To: <AANLkTi=_BPuE5KX4D48JDoOW9U3netsr9uDU9Lo5CgYO@mail.gmail.com>
References: <AANLkTi=_BPuE5KX4D48JDoOW9U3netsr9uDU9Lo5CgYO@mail.gmail.com>
Message-ID: <FACE40F5-BB80-479A-BFFC-F07F6BA681D2@freeswitch.org>

What makes you think that fails?  It has ULAW and CN in the codec list!  Sounds like you need the verbose sdp... set the global variable "verbose_sdp=true"

/b

On Jan 24, 2011, at 1:50 PM, Tihomir Culjaga wrote:

> 
> 
> i configured FS HA and looks like its trying to recover the call .. but the re-INVITE fails due to "wrong/missed" codec capability.



From brian at freeswitch.org  Mon Jan 24 23:18:51 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 Jan 2011 14:18:51 -0600
Subject: [Freeswitch-users] Freeswitch Sends RTP To Private IP Address
In-Reply-To: <AANLkTimZB7b68LmF-RmYXe91_Y1K+k5xkt+=hARgzaj2@mail.gmail.com>
References: <AANLkTimru1i31uSYW+bGK_3z=GiW=QNPzjvTZRRwP5sq@mail.gmail.com>
	<AANLkTinVK90fOOpCq3Nj7ihb6FZom_3k33cNowcCo8T-@mail.gmail.com>
	<AANLkTimZB7b68LmF-RmYXe91_Y1K+k5xkt+=hARgzaj2@mail.gmail.com>
Message-ID: <44F79970-08DE-41B1-878D-D19E04833590@freeswitch.org>

This is a self help forum isn't it?  :P

/b

On Jan 24, 2011, at 1:46 PM, Michael Collins wrote:

> Your candor is refreshing... and thanks for posting the "solution" to your "problem"! :)
> 
> -MC



From msc at freeswitch.org  Mon Jan 24 23:23:22 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Jan 2011 12:23:22 -0800
Subject: [Freeswitch-users] Freeswitch Sends RTP To Private IP Address
In-Reply-To: <44F79970-08DE-41B1-878D-D19E04833590@freeswitch.org>
References: <AANLkTimru1i31uSYW+bGK_3z=GiW=QNPzjvTZRRwP5sq@mail.gmail.com>
	<AANLkTinVK90fOOpCq3Nj7ihb6FZom_3k33cNowcCo8T-@mail.gmail.com>
	<AANLkTimZB7b68LmF-RmYXe91_Y1K+k5xkt+=hARgzaj2@mail.gmail.com>
	<44F79970-08DE-41B1-878D-D19E04833590@freeswitch.org>
Message-ID: <AANLkTinhnAaUE3iUsszvLDUSBayt1jL-ZK5_7XZvZwfT@mail.gmail.com>

FIYDS (Fix It Yo Damm Seff!)

On Mon, Jan 24, 2011 at 12:18 PM, Brian West <brian at freeswitch.org> wrote:

> This is a self help forum isn't it?  :P
>
> /b
>
> On Jan 24, 2011, at 1:46 PM, Michael Collins wrote:
>
> > Your candor is refreshing... and thanks for posting the "solution" to
> your "problem"! :)
> >
> > -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Jan 24 23:25:50 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Jan 2011 12:25:50 -0800
Subject: [Freeswitch-users] info
In-Reply-To: <59A136AAB8334CCB9E941B6D2870EF87@e1705>
References: <AANLkTimM0fXNSF8e2HWbmJ_ZmHxsCfr0d=-_D9RZ8u-F@mail.gmail.com>
	<AANLkTinRr1K50ebNwHr_t2Jx_r8HqRmZXnmTqdQTizg7@mail.gmail.com>
	<59A136AAB8334CCB9E941B6D2870EF87@e1705>
Message-ID: <AANLkTi=qY-ASihDQ723r8ttZDf8+73OT+YXWL15RMKag@mail.gmail.com>

Like Brian said, this is the self-help forum! We'd be reach if we could
figure out how to charge everyone who fixed their own problems. :)

-MC

On Mon, Jan 24, 2011 at 9:16 AM, Madovsky <infos at madovsky.org> wrote:

>  Please next time try to make a max test before to ask in this emailist ;)
>
> ----- Original Message -----
> *From:* Sam <u2nsam at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, January 24, 2011 1:18 AM
> *Subject:* Re: [Freeswitch-users] info
>
> Pls ignore this email, my bad i didn't reloaded ... sorry to bother.
>
> Regds
> Sam
>
> On Mon, Jan 24, 2011 at 11:27 AM, Sam <u2nsam at gmail.com> wrote:
>
>> Have any one tried using <action application="info"/> in git-1c95ad9
>> 2011-01-20 22-43-50 -0300 .
>> because its not working for me in this version.
>>
>> Regds
>> Sam
>>
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Jan 24 23:28:50 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Jan 2011 12:28:50 -0800
Subject: [Freeswitch-users] IVR Bridged Call Dropping after 2 rings
In-Reply-To: <1295661432.3014.15.camel@John-Home>
References: <1295661432.3014.15.camel@John-Home>
Message-ID: <AANLkTikMTAKiakvQFu20xdMh7yrrynJxXd8X9a9RG911@mail.gmail.com>

Can you pastebin a debug log with a siptrace? Also, pastebin your dialplan.
I think we can help with this but I want to see what you're doing before I
suggest anything.

-MC

On Fri, Jan 21, 2011 at 5:57 PM, John Carpenter <john at 247-talk.co.uk> wrote:

>  Hi, I am trying to setup a very simple IVR using LUA. Call arrives from a
> DID SIP trunk and is answered and message is played ok, after a particular
> digit is pressed it bridges the call to an extension which is remotely
> connected. It works but after 2 rings the call to the extension is dropped
> with a SIP message "BYE" from DID provider. If I just route the call
> directly to the extension (no IVR) it works fine. It seems like the DID
> hangs up when the call is bridged to the extension. Have tried same thing
> using the XML IVR Engine and get exactly the same result. The IVR script is
> below
>
> pathsep = '/'
> session:setAutoHangup(false);
> session:answer()
> prompt = "ivr" .. pathsep .. "247talk.wav"
> invalid = "ivr" .. pathsep .. "ivr-that_was_an_invalid_entry.wav"
> freeswitch.consoleLog("INFO", "Prompt file is '" .. prompt .. "'\n")
> continue = true
>
> while( session:ready() == true and continue == true) do
>         digits = session:playAndGetDigits(1,1,3,7000,"#", prompt, invalid,
> "\\d+")
>         if (digits == "1") then
>                 continue = false
>
> session:execute("bridge","sofia/external/2476%91.xxx.xx.xx")
>         end
>         if (digits == "2") then
>
> session:execute("bridge","sofia/external/2475%91.xxx.xx.xx")
>         end
>         if (digits == "3") then
>                 continue = false
>
> session:execute("bridge","sofia/external/2475%91.xxx.xx.xx")
>         end
> end
>
> session:hangup()
>
> Any help with this greatly appreciated it is driving me nuts.
>
> regards, John Carpenter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Mon Jan 24 23:30:41 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Jan 2011 14:30:41 -0600
Subject: [Freeswitch-users] IVRD
In-Reply-To: <B890EB1A467549D48DB330F5795906E9@e1705>
References: <B890EB1A467549D48DB330F5795906E9@e1705>
Message-ID: <AANLkTin54y6CJHhSc8DbEZBnx7oBW+4ppD8E3-AQnSk4@mail.gmail.com>

did you run the fs_ivrd from a shell and look for output in stderr?
it has to be related to the script executing failing etc.


On Mon, Jan 24, 2011 at 11:57 AM, Madovsky <infos at madovsky.org> wrote:
> Sorry I sent this email by mistake without to finish it.
> so in my dialplan I have :
> ? ? ? ? ? ? ? ?<extension name="gluglu">
> ? ? ? ? ? ? ? ? ? ?<condition field="destination_number"
> expression="^999$">
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set"
> data="ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl"/>
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="socket" data="127.0.0.1:9090
> full"/>
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="answer"/>
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="voicemail" data="default
> ${domain_name} ${dialed_extension}"/>
> ? ? ? ? ? ? ? ? ? ?<condition>
> ? ? ? ? ? ? ? ?</extension>
>
> and when I call this extension in log level 7 I can see :
>
> Dialplan: sofia/internal/9999999999999 at default Action
> set(ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl)
> Dialplan: sofia/internal/9999999999999 at default Action socket(127.0.0.1:9090
> full)
> Starting ivrd-hello_world.pl...
>
> and no ivrd-hello_world.pl code is executed in the dialplan unless the print
> "Starting ivrd-hello_world.pl";
>
> the same if I replace fs_ivrd with server2.pl for example
>
> Thanks
>
>
> ----- Original Message -----
> From: "Madovsky" <infos at madovsky.org>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, January 24, 2011 12:48 PM
> Subject: Re: [Freeswitch-users] IVRD
>
>
>> Concerning the use of fs_ivrd:
>>
>> path is ok
>> permission is root 755
>> perl -c gives ../scripts/perl/ivrd-hello_world.pl syntax OK
>>
>> in dialplan I have :
>>
>>
>>
>>
>>
>> ----- Original Message -----
>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, January 24, 2011 11:20 AM
>> Subject: Re: [Freeswitch-users] IVRD
>>
>>
>> check for proper path and execute permissions on the file and perl -c
>> to make sure it compiles.
>>
>>
>> On Mon, Jan 24, 2011 at 12:13 AM, Madovsky <infos at madovsky.org> wrote:
>>> I tried to use IVRD from wiki example
>>>
>>> http://wiki.freeswitch.org/wiki/Ivrd
>>>
>>> and server2.pl in ESL directory
>>> copy and paste in my dialplan ans settings
>>> so the daemon is running well, but if I attempt
>>> to call nothing happens unless hangup.
>>> on the log I can see only
>>>
>>> EXECUTE sofia/internal/9999999999999 at default socket(127.0.0.1:8084 full)
>>>
>>> I tried the tests of troubleshooting without error
>>> I don't understand why the events are not received in the perl script
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Mon Jan 24 23:44:03 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Jan 2011 14:44:03 -0600
Subject: [Freeswitch-users] Is it possible to use att_xfer on channels
 already bridged via loopback?
In-Reply-To: <AANLkTi=VGmzii5_GkU06orX-BZxtZkj-a12qtPE+fkNG@mail.gmail.com>
References: <AANLkTimWAPV52349=Yge2zjnJyP40EdgDJXgyH-QcOCd@mail.gmail.com>
	<AANLkTin6SoYj-_DggPttAPZcwJqfc+HGHhUJvLzWGs-Z@mail.gmail.com>
	<AANLkTi=VGmzii5_GkU06orX-BZxtZkj-a12qtPE+fkNG@mail.gmail.com>
Message-ID: <AANLkTi=trSayCkbTgsOyUxWYiSMnxey6T4pzxaojXugx@mail.gmail.com>

Guess who wrote that code in res_features ;)


On Mon, Jan 24, 2011 at 12:51 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> Yes, I'm using SIP as my endpoints.
>
> There are number of reasons why I want to use in-call triggered transfers.
>
> 1. We are migrating several large ISP PBX to Freeswitch from Asterisk.
> In-call transfers in Asterisk via res_features are mature and stable,
> and many users use it - if we remove this feature, we will loose our
> users.
>
> 2. Not all SIP endpoints support call transfer, and despite majority
> of devices support this, we want fine control over who is allowed to
> do transfer in which situation - there are many paths of call in our
> PBXs.
>
> 3. We also need to provide call transfer functionality to external
> PSTN users, which are dialed via SIP proxy and then SIP-E1 gateway
> like audiocodes or cisco. There is no simple universal solution to
> trigger such transfers from PSTN side besides in-call transfers with
> bind_meta_app.
>
> I agree, in ideal world we would live without server-side attended
> transfers, but for now att_xfer is an ultimate feature of real PBX,
> even old Avaya and Samsung boxes support this :)
>
>
> 2011/1/24 Jo?o Mesquita <jmesquita at freeswitch.org>:
>> I am just now discussing this with another developer and the question that
>> is never answered is:
>> Why are you trying to use att_xfer if it is your endpoint's duty to make the
>> transfer? Are you using SIP?
>> Regards,
>> Jo?o Mesquita
>>
>>
>> On Mon, Jan 24, 2011 at 1:01 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>>
>>> Is att_xfer or mod_loopback is broken in FS-current?
>>> I use FreeSWITCH Version 1.0.head (git-7eceff4 2011-01-16 22-33-50 +0000)
>>> Seems there were no updates of att_xfer or mod_loopback since that.
>>>
>>> I use loopback channel as destination when doing att_xfer to re-enter
>>> dialplan.
>>> With loopback_bowout=false and loopback_bowout_on_execute=false this
>>> works. But when any of connected parties tries to do att_xfer again,
>>> all channels get hangup on transferer hangup.
>>>
>>> Scenario:
>>>
>>> A calls B, B answers
>>> A launches att_xfer via *7, B listens to MOH
>>> A dials C and we do att_xfer to loopback/C
>>> C answers, A hangs up to complete transfer
>>> C and B are now bridged via loopback, `show channels` shows 4 channels
>>> include 2 loopback legs.
>>>
>>> Now, C also tries to do in-call transfer with *7.
>>> C launches att_xfer via *7, B listens to MOH
>>> C dials D and do att_xfer to loopback/D
>>> D answers, C hangs up to complete transfer
>>> B and D are hung up instead of be bridged together.
>>>
>>> There are also issues with MOH wile running att_xfer, but they are not
>>> so important as att_xfer behavior itself.
>>>
>>>
>>>
>>> --
>>> Best regards,
>>>
>>> Dmitry Sytchev,
>>> IT Engineer
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From msc at freeswitch.org  Mon Jan 24 23:45:35 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Jan 2011 12:45:35 -0800
Subject: [Freeswitch-users] Is it possible to use att_xfer on channels
 already bridged via loopback?
In-Reply-To: <AANLkTi=trSayCkbTgsOyUxWYiSMnxey6T4pzxaojXugx@mail.gmail.com>
References: <AANLkTimWAPV52349=Yge2zjnJyP40EdgDJXgyH-QcOCd@mail.gmail.com>
	<AANLkTin6SoYj-_DggPttAPZcwJqfc+HGHhUJvLzWGs-Z@mail.gmail.com>
	<AANLkTi=VGmzii5_GkU06orX-BZxtZkj-a12qtPE+fkNG@mail.gmail.com>
	<AANLkTi=trSayCkbTgsOyUxWYiSMnxey6T4pzxaojXugx@mail.gmail.com>
Message-ID: <AANLkTimNTpfadZooOzihZ+TJrxazFvUaPhGeezSyKThV@mail.gmail.com>

On Mon, Jan 24, 2011 at 12:44 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Guess who wrote that code in res_features ;)
>
> Was that another hack done on a whim like mod_yaml? :P
-MC
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From msc at freeswitch.org  Mon Jan 24 23:46:27 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Jan 2011 12:46:27 -0800
Subject: [Freeswitch-users] Is it possible to use att_xfer on channels
 already bridged via loopback?
In-Reply-To: <AANLkTimWAPV52349=Yge2zjnJyP40EdgDJXgyH-QcOCd@mail.gmail.com>
References: <AANLkTimWAPV52349=Yge2zjnJyP40EdgDJXgyH-QcOCd@mail.gmail.com>
Message-ID: <AANLkTi=WKADqDk-qqevfAh3vR1mZgZdwEEey_cuFUEz8@mail.gmail.com>

Dmitry,

I'd like to try this myself on one of my boxes. Would you pastebin the
dialplan you are using?

Thanks,
MC

On Mon, Jan 24, 2011 at 8:01 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:

> Is att_xfer or mod_loopback is broken in FS-current?
> I use FreeSWITCH Version 1.0.head (git-7eceff4 2011-01-16 22-33-50 +0000)
> Seems there were no updates of att_xfer or mod_loopback since that.
>
> I use loopback channel as destination when doing att_xfer to re-enter
> dialplan.
> With loopback_bowout=false and loopback_bowout_on_execute=false this
> works. But when any of connected parties tries to do att_xfer again,
> all channels get hangup on transferer hangup.
>
> Scenario:
>
> A calls B, B answers
> A launches att_xfer via *7, B listens to MOH
> A dials C and we do att_xfer to loopback/C
> C answers, A hangs up to complete transfer
> C and B are now bridged via loopback, `show channels` shows 4 channels
> include 2 loopback legs.
>
> Now, C also tries to do in-call transfer with *7.
> C launches att_xfer via *7, B listens to MOH
> C dials D and do att_xfer to loopback/D
> D answers, C hangs up to complete transfer
> B and D are hung up instead of be bridged together.
>
> There are also issues with MOH wile running att_xfer, but they are not
> so important as att_xfer behavior itself.
>
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bernhard.suttner at winet.ch  Mon Jan 24 23:58:30 2011
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Mon, 24 Jan 2011 21:58:30 +0100
Subject: [Freeswitch-users] Outgoing registrations within a fs pool
Message-ID: <20110124215830.b392645e@mail.winet.ch>

Hi,

does someone has an idea what happens if I would add the gateway (with registration) to all the freeswitch servers of the pool with shared database?

Best regards,
Bernhard

----- Original Message -----
From: Bernhard Suttner [mailto:bernhard.suttner at winet.ch]
To: 'FreeSWITCH Users Help' [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Fri, 14 Jan 2011 10:42:10 +0100
Subject: Re: [Freeswitch-users] Outgoing registrations within a fs pool


> Hi,
> 
> Kamailio is able to do a registration with the uacreg module, but it is not
> as stable as on FreeSWITCH (its not the main task of Kamailio to implement
> B2BUA features). Therefore I asked if FreeSWITCH is able to register "from a
> pool of FS servers".
> 
> Best regards,
> Bernhard
> 
> -----Urspr?ngliche Nachricht-----
> Von: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Paul
> Cupis
> Gesendet: Donnerstag, 13. Januar 2011 21:59
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] Outgoing registrations within a fs pool
> 
> Perhaps look at getting the kamailio server to register to the provider
> rather than one of the FS from the pool?
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From bernhard.suttner at winet.ch  Mon Jan 24 23:58:31 2011
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Mon, 24 Jan 2011 21:58:31 +0100
Subject: [Freeswitch-users] Outgoing registrations within a fs pool
Message-ID: <20110124215831.3ba1d88d@mail.winet.ch>

Hi,

does someone has an idea what happens if I would add the gateway (with registration) to all the freeswitch servers of the pool with shared database?

Best regards,
Bernhard

----- Original Message -----
From: Bernhard Suttner [mailto:bernhard.suttner at winet.ch]
To: 'FreeSWITCH Users Help' [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Fri, 14 Jan 2011 10:42:10 +0100
Subject: Re: [Freeswitch-users] Outgoing registrations within a fs pool


> Hi,
> 
> Kamailio is able to do a registration with the uacreg module, but it is not
> as stable as on FreeSWITCH (its not the main task of Kamailio to implement
> B2BUA features). Therefore I asked if FreeSWITCH is able to register "from a
> pool of FS servers".
> 
> Best regards,
> Bernhard
> 
> -----Urspr?ngliche Nachricht-----
> Von: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Paul
> Cupis
> Gesendet: Donnerstag, 13. Januar 2011 21:59
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] Outgoing registrations within a fs pool
> 
> Perhaps look at getting the kamailio server to register to the provider
> rather than one of the FS from the pool?
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From kbdfck at gmail.com  Tue Jan 25 00:04:01 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Tue, 25 Jan 2011 00:04:01 +0300
Subject: [Freeswitch-users] Is it possible to use att_xfer on channels
 already bridged via loopback?
In-Reply-To: <AANLkTi=WKADqDk-qqevfAh3vR1mZgZdwEEey_cuFUEz8@mail.gmail.com>
References: <AANLkTimWAPV52349=Yge2zjnJyP40EdgDJXgyH-QcOCd@mail.gmail.com>
	<AANLkTi=WKADqDk-qqevfAh3vR1mZgZdwEEey_cuFUEz8@mail.gmail.com>
Message-ID: <AANLkTikDTV_nveYRL8ysh_24+RCk=bx4p9L9ozvMGaXi@mail.gmail.com>

=))  res_features is cool :))

I'll clean out my dialplan and provide it to you tomorrow. How do I
authenticate to pastebin? With my Jira account?
There is another problem, I'm using
mod_spidermonkey/mod_spidermonkey_odbc to get virtual extension from
my database, but I'll try to convert it to simple dialplan.
BTW, maybe my scripts can be a reason of my troubles?

2011/1/24 Michael Collins <msc at freeswitch.org>:
> Dmitry,
> I'd like to try this myself on one of my boxes. Would you pastebin the
> dialplan you are using?
> Thanks,
> MC
>
> On Mon, Jan 24, 2011 at 8:01 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>
>> Is att_xfer or mod_loopback is broken in FS-current?
>> I use FreeSWITCH Version 1.0.head (git-7eceff4 2011-01-16 22-33-50 +0000)
>> Seems there were no updates of att_xfer or mod_loopback since that.
>>
>> I use loopback channel as destination when doing att_xfer to re-enter
>> dialplan.
>> With loopback_bowout=false and loopback_bowout_on_execute=false this
>> works. But when any of connected parties tries to do att_xfer again,
>> all channels get hangup on transferer hangup.
>>
>> Scenario:
>>
>> A calls B, B answers
>> A launches att_xfer via *7, B listens to MOH
>> A dials C and we do att_xfer to loopback/C
>> C answers, A hangs up to complete transfer
>> C and B are now bridged via loopback, `show channels` shows 4 channels
>> include 2 loopback legs.
>>
>> Now, C also tries to do in-call transfer with *7.
>> C launches att_xfer via *7, B listens to MOH
>> C dials D and do att_xfer to loopback/D
>> D answers, C hangs up to complete transfer
>> B and D are hung up instead of be bridged together.
>>
>> There are also issues with MOH wile running att_xfer, but they are not
>> so important as att_xfer behavior itself.
>>
>>
>>
>> --
>> Best regards,
>>
>> Dmitry Sytchev,
>> IT Engineer
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Best regards,

Dmitry Sytchev,
IT Engineer


From milan.m.masek at gmail.com  Mon Jan 24 23:51:02 2011
From: milan.m.masek at gmail.com (Milan Masek)
Date: Mon, 24 Jan 2011 12:51:02 -0800
Subject: [Freeswitch-users] Bria 3 on Ubuntu and Freeswitch self signed
	CA cert
Message-ID: <AANLkTin=vJ9mNz3YHFVEmrxbpaB8i7MCz9F_OLdCj-xj@mail.gmail.com>

Thank you Chris for your help. I did this allready.
I put the ../CA/cacert.pem from FS server to my keystore and enable it with
sudo dpkg-reconfigure ca-certificates.
Bria still complains.

---------- Forwarded message ----------
From: Chris Burns <chris at cloudtel.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: Mon, 24 Jan 2011 13:22:45 -0500
Subject: Re: [Freeswitch-users] Bria 3 and Freeswitch TLS configuration
(Milan)
Put the the ca cert in your ca store if it needs to be there :D

In Deb they go in /usr/share/ca-certificates belonging to the package
bearing the same name. Ubuntu prob the same as Deb. Windows I dont know,
theres prob some cutesy wizard for it.

On Mon, Jan 24, 2011 at 5:34 AM, Milan Masek <milan.m.masek at gmail.com>wrote:
Help yourself.
With ssldump I am getting
1 5  0.2448 (0.1591)  C>S  Alert
    level           fatal
    value           unknown_ca

Anybody know what can make Bria happy on client computer? How to install
self signed CA certificate for Bria in Ubuntu and MacOSX appropriately?
Thx
M.
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From william.suffill at gmail.com  Tue Jan 25 00:15:25 2011
From: william.suffill at gmail.com (William Suffill)
Date: Mon, 24 Jan 2011 16:15:25 -0500
Subject: [Freeswitch-users] Is it possible to use att_xfer on channels
 already bridged via loopback?
In-Reply-To: <AANLkTikDTV_nveYRL8ysh_24+RCk=bx4p9L9ozvMGaXi@mail.gmail.com>
References: <AANLkTimWAPV52349=Yge2zjnJyP40EdgDJXgyH-QcOCd@mail.gmail.com>
	<AANLkTi=WKADqDk-qqevfAh3vR1mZgZdwEEey_cuFUEz8@mail.gmail.com>
	<AANLkTikDTV_nveYRL8ysh_24+RCk=bx4p9L9ozvMGaXi@mail.gmail.com>
Message-ID: <AANLkTi=W4ATyZdymo_5JcjCsjYJ=4HfnkfmE+T49rpCD@mail.gmail.com>

Read the text in the password prompt for the pastebin. Magic what you shall
find. =)

-- W

PS: Feel free to e-mail off list if you still don't get what the
username/password is.
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From freeswitch-list at puzzled.xs4all.nl  Tue Jan 25 01:07:55 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Mon, 24 Jan 2011 23:07:55 +0100
Subject: [Freeswitch-users] Bria 3 on Ubuntu and Freeswitch self signed
 CA cert
In-Reply-To: <AANLkTin=vJ9mNz3YHFVEmrxbpaB8i7MCz9F_OLdCj-xj@mail.gmail.com>
References: <AANLkTin=vJ9mNz3YHFVEmrxbpaB8i7MCz9F_OLdCj-xj@mail.gmail.com>
Message-ID: <4D3DF83B.1090504@puzzled.xs4all.nl>

On 01/24/2011 09:51 PM, Milan Masek wrote:
> Thank you Chris for your help. I did this allready.
> I put the ../CA/cacert.pem from FS server to my keystore and enable it
> with sudo dpkg-reconfigure ca-certificates.
> Bria still complains.

Don't know if it's related but I have seen some apps give weird errors 
because they could not handle 4096bit CA certs. Creating an intermediate 
CA with 2048bit would solve it.

Regards,
Patrick


From steveayre at gmail.com  Tue Jan 25 01:29:45 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 24 Jan 2011 22:29:45 +0000
Subject: [Freeswitch-users] FS HA
In-Reply-To: <FACE40F5-BB80-479A-BFFC-F07F6BA681D2@freeswitch.org>
References: <AANLkTi=_BPuE5KX4D48JDoOW9U3netsr9uDU9Lo5CgYO@mail.gmail.com>
	<FACE40F5-BB80-479A-BFFC-F07F6BA681D2@freeswitch.org>
Message-ID: <350D44B3-A176-47A9-84F4-72EB62F08299@gmail.com>

Brian, it has PCMU in the sdp, but Sofia thinks it's setting PCMA?

Steve on iPhone

On 24 Jan 2011, at 20:17, Brian West <brian at freeswitch.org> wrote:

> What makes you think that fails?  It has ULAW and CN in the codec list!  Sounds like you need the verbose sdp... set the global variable "verbose_sdp=true"
> 
> /b
> 
> On Jan 24, 2011, at 1:50 PM, Tihomir Culjaga wrote:
> 
>> 
>> 
>> i configured FS HA and looks like its trying to recover the call .. but the re-INVITE fails due to "wrong/missed" codec capability.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From anthony.minessale at gmail.com  Tue Jan 25 01:31:48 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Jan 2011 16:31:48 -0600
Subject: [Freeswitch-users] Outgoing registrations within a fs pool
In-Reply-To: <20110124215830.b392645e@mail.winet.ch>
References: <20110124215830.b392645e@mail.winet.ch>
Message-ID: <AANLkTimm8n-mK-w2XXHns_XL-wPZeMKLqmm_H9vuuWvd@mail.gmail.com>

I think gateways are all in-memory so it should be ok

On Mon, Jan 24, 2011 at 2:58 PM, Bernhard Suttner
<bernhard.suttner at winet.ch> wrote:
> Hi,
>
> does someone has an idea what happens if I would add the gateway (with registration) to all the freeswitch servers of the pool with shared database?
>
> Best regards,
> Bernhard
>
> ----- Original Message -----
> From: Bernhard Suttner [mailto:bernhard.suttner at winet.ch]
> To: 'FreeSWITCH Users Help' [mailto:freeswitch-users at lists.freeswitch.org]
> Sent: Fri, 14 Jan 2011 10:42:10 +0100
> Subject: Re: [Freeswitch-users] Outgoing registrations within a fs pool
>
>
>> Hi,
>>
>> Kamailio is able to do a registration with the uacreg module, but it is not
>> as stable as on FreeSWITCH (its not the main task of Kamailio to implement
>> B2BUA features). Therefore I asked if FreeSWITCH is able to register "from a
>> pool of FS servers".
>>
>> Best regards,
>> Bernhard
>>
>> -----Urspr?ngliche Nachricht-----
>> Von: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Paul
>> Cupis
>> Gesendet: Donnerstag, 13. Januar 2011 21:59
>> An: FreeSWITCH Users Help
>> Betreff: Re: [Freeswitch-users] Outgoing registrations within a fs pool
>>
>> Perhaps look at getting the kamailio server to register to the provider
>> rather than one of the FS from the pool?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From joaocarlosleme at gmail.com  Tue Jan 25 01:55:45 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Mon, 24 Jan 2011 14:55:45 -0800
Subject: [Freeswitch-users] Caller ID using Fifo
In-Reply-To: <AANLkTimE-0g-Gc92tseC-GWOuwV-icvJU7ndLnhPOCo=@mail.gmail.com>
References: <AANLkTik6pdgXgND6kmzZA4qe8vDZQ0_7e47+ciVWPfub@mail.gmail.com>
	<AANLkTikKDO3qru-9=nuPEQcW+iCN7ccizyJAh9sqUkCN@mail.gmail.com>
	<AANLkTi=p7_CsUDWS62+weZv50+PBZ8RLop0Q04y-c4hB@mail.gmail.com>
	<AANLkTi=m_hW9vQV5nbC2jwcos0bKy2rCf7YAr6aA23oz@mail.gmail.com>
	<AANLkTimE-0g-Gc92tseC-GWOuwV-icvJU7ndLnhPOCo=@mail.gmail.com>
Message-ID: <AANLkTim8QUcxH0rbcZymMr_EAnqpdE8OUnjpJ1tWN7sn@mail.gmail.com>

I haven't been able to find a solution either (havent' had much time), but
have you already tried the following to make sure it does not work?

"Note: If you wish to specify the caller ID presented when a fifo calls an
agent, set the origination_caller_id_name and origination_caller_id_num
variables to the values desired. These could be set within the {} of the
dialstring, or they could be set using the set application in the dialplan
which places the caller into the fifo (before the 'fifo in' executed on the
caller)."

I haven't had time to look into but that's where I would start.
It was added to the Fifo wiki by User "Sward" (no futher info on user) on:

(cur<http://wiki.freeswitch.org/index.php?title=Mod_fifo&diff=24330&oldid=20224>)
(prev<http://wiki.freeswitch.org/index.php?title=Mod_fifo&diff=20224&oldid=20175>)
15:21, 9 June 2010<http://wiki.freeswitch.org/index.php?title=Mod_fifo&oldid=20224>
Sward<http://wiki.freeswitch.org/index.php?title=User:Sward&action=edit&redlink=1>
(Talk<http://wiki.freeswitch.org/index.php?title=User_talk:Sward&action=edit&redlink=1>|
contribs <http://wiki.freeswitch.org/wiki/Special:Contributions/Sward>) (26,518
bytes) (?<http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback>Configure
for Agent Callback)

If anyone know this "Sward" guy, he may have some answers.
Thanks,
John

On Mon, Jan 24, 2011 at 1:20 AM, Marc de Corny <marcdecorny at gmail.com>wrote:

> Hi All
>
> I have since had a play with mod_callcenter and have not been able to send
> the call to an agent with the caller_id_name as the name of the queue.
>
> I keep on getting the CLI on both.
>
> As this cannot be done in FIFO either, do any of you have any ideas? or
> managed to get it working ?
>
> thanks
> Marc
>
> On Fri, Jan 14, 2011 at 4:35 PM, Marc de Corny <marcdecorny at gmail.com>wrote:
>
>> Just to follow up on this subject.
>>
>> I have done a lot of testing on the fifo trying to get the caller_id_name
>> changed on the outbound call to the agent and to be honest I cannot
>> understand the explanation.
>>
>> If mod_fifo does not know which call it will connect until the agent
>> answers, how come it displays the CLI correctly, jsut won;t let me change
>> it.
>>
>> Still seems strange. I am looking into the Mod_callcentre to check if it
>> sends caller_id information. but the same logic if valid could apply
>>
>> Also maybe someone should change the Wiki ( I would but do not have enough
>> expertise on the subject) because the following is a bit misleading
>>
>>  "Note: If you wish to specify the caller ID presented when a fifo calls
>> an agent, set the origination_caller_id_name and origination_caller_id_num
>> variables to the values desired. These could be set within the {} of the
>> dialstring, or they could be set using the set application in the dialplan
>> which places the caller into the fifo (before the 'fifo in' executed on the
>> caller). "
>>  thanks
>> Marc
>>   On Thu, Jan 13, 2011 at 10:47 PM, Joao Leme <joaocarlosleme at gmail.com>wrote:
>>
>>> What about showing the Caller ID after it is answered? Any way to do
>>> that?
>>>
>>> 2011/1/12 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>
>>> Jo?o Leme,
>>>>
>>>> The caller id is not passed when the phone is ringing because mod_fifo
>>>> does not know which call is going to be sent to that channel once it is
>>>> answered until it is really answered. I don't know if mod_callcenter does
>>>> show anything but you should consider looking at the documentation if you
>>>> really need this feature.
>>>>
>>>> Regards,
>>>> Jo?o Mesquita
>>>>
>>>>
>>>>   On Wed, Jan 12, 2011 at 9:15 PM, Joao Leme <joaocarlosleme at gmail.com>wrote:
>>>>
>>>>>  Hi there,
>>>>> I would like to know if there is a way to see the caller ID on my Sip
>>>>> Client (X-Lite for example) of the caller that I answear from a Fifo queue?
>>>>> Thanks,
>>>>> John
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Tue Jan 25 02:01:12 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Jan 2011 00:01:12 +0100
Subject: [Freeswitch-users] FS HA
In-Reply-To: <350D44B3-A176-47A9-84F4-72EB62F08299@gmail.com>
References: <AANLkTi=_BPuE5KX4D48JDoOW9U3netsr9uDU9Lo5CgYO@mail.gmail.com>
	<FACE40F5-BB80-479A-BFFC-F07F6BA681D2@freeswitch.org>
	<350D44B3-A176-47A9-84F4-72EB62F08299@gmail.com>
Message-ID: <AANLkTimJLMjPJwthN8AhDq=1_JnDbdmocbgd=-=usyTT@mail.gmail.com>

yap, i do have PCMA ... and the debug shows it correctly :=)

i will try to see what it does with verbose. Post new debug tomorrow.

ty.


On Mon, Jan 24, 2011 at 11:29 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Brian, it has PCMU in the sdp, but Sofia thinks it's setting PCMA?
>
> Steve on iPhone
>
> On 24 Jan 2011, at 20:17, Brian West <brian at freeswitch.org> wrote:
>
> > What makes you think that fails?  It has ULAW and CN in the codec list!
>  Sounds like you need the verbose sdp... set the global variable
> "verbose_sdp=true"
> >
> > /b
> >
> > On Jan 24, 2011, at 1:50 PM, Tihomir Culjaga wrote:
> >
> >>
> >>
> >> i configured FS HA and looks like its trying to recover the call .. but
> the re-INVITE fails due to "wrong/missed" codec capability.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bernhard.suttner at winet.ch  Tue Jan 25 02:11:30 2011
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Tue, 25 Jan 2011 00:11:30 +0100
Subject: [Freeswitch-users] Outgoing registrations within a fs pool
Message-ID: <20110125001130.74166c5a@mail.winet.ch>

Hi,

thanks for your answer. If there are 5 freeswitch servers within the pool (shared db) I would add the gateway to all of the freeswitch server. Will all of them try to register or would the first freeswitch server register and the other servers would check if the (re)-registration was already done? 

The point is, that all the freeswitch servers do have the same configuration and it does not matter which freeswitch server registers because a incoming call would be handled the same way on all of the servers with shared database.

Best regards,
Bernhard

----- Original Message -----
From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Mon, 24 Jan 2011 23:31:48 +0100
Subject: Re: [Freeswitch-users] Outgoing registrations within a fs pool


> I think gateways are all in-memory so it should be ok
> 
> On Mon, Jan 24, 2011 at 2:58 PM, Bernhard Suttner
> <bernhard.suttner at winet.ch> wrote:
> > Hi,
> >
> > does someone has an idea what happens if I would add the gateway (with
> registration) to all the freeswitch servers of the pool with shared
> database?
> >
> > Best regards,
> > Bernhard
> >
> > ----- Original Message -----
> > From: Bernhard Suttner [mailto:bernhard.suttner at winet.ch]
> > To: 'FreeSWITCH Users Help' [mailto:freeswitch-users at lists.freeswitch.org]
> > Sent: Fri, 14 Jan 2011 10:42:10 +0100
> > Subject: Re: [Freeswitch-users] Outgoing registrations within a fs pool
> >
> >
> >> Hi,
> >>
> >> Kamailio is able to do a registration with the uacreg module, but it is
> not
> >> as stable as on FreeSWITCH (its not the main task of Kamailio to
> implement
> >> B2BUA features). Therefore I asked if FreeSWITCH is able to register
> "from a
> >> pool of FS servers".
> >>
> >> Best regards,
> >> Bernhard
> >>
> >> -----Urspr?ngliche Nachricht-----
> >> Von: freeswitch-users-bounces at lists.freeswitch.org
> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
> Paul
> >> Cupis
> >> Gesendet: Donnerstag, 13. Januar 2011 21:59
> >> An: FreeSWITCH Users Help
> >> Betreff: Re: [Freeswitch-users] Outgoing registrations within a fs pool
> >>
> >> Perhaps look at getting the kamailio server to register to the provider
> >> rather than one of the FS from the pool?
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From infos at madovsky.org  Tue Jan 25 02:19:57 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 24 Jan 2011 18:19:57 -0500
Subject: [Freeswitch-users] IVRD
References: <B890EB1A467549D48DB330F5795906E9@e1705>
	<AANLkTin54y6CJHhSc8DbEZBnx7oBW+4ppD8E3-AQnSk4@mail.gmail.com>
Message-ID: <2E1D414AFF8443239EF805F64F22AAA1@e1705>

yes,

and there's only this whe the dialplan call the socket

/usr/local/freeswitch/bin/fs_ivrd -h 127.0.0.1 -p 9090
Starting ivrd-hello_world.pl...

thx

----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, January 24, 2011 3:30 PM
Subject: Re: [Freeswitch-users] IVRD


did you run the fs_ivrd from a shell and look for output in stderr?
it has to be related to the script executing failing etc.


On Mon, Jan 24, 2011 at 11:57 AM, Madovsky <infos at madovsky.org> wrote:
> Sorry I sent this email by mistake without to finish it.
> so in my dialplan I have :
> <extension name="gluglu">
> <condition field="destination_number"
> expression="^999$">
> <action application="set"
> data="ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl"/>
> <action application="socket" data="127.0.0.1:9090
> full"/>
> <action application="answer"/>
> <action application="voicemail" data="default
> ${domain_name} ${dialed_extension}"/>
> <condition>
> </extension>
>
> and when I call this extension in log level 7 I can see :
>
> Dialplan: sofia/internal/9999999999999 at default Action
> set(ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl)
> Dialplan: sofia/internal/9999999999999 at default Action 
> socket(127.0.0.1:9090
> full)
> Starting ivrd-hello_world.pl...
>
> and no ivrd-hello_world.pl code is executed in the dialplan unless the 
> print
> "Starting ivrd-hello_world.pl";
>
> the same if I replace fs_ivrd with server2.pl for example
>
> Thanks
>
>
> ----- Original Message -----
> From: "Madovsky" <infos at madovsky.org>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, January 24, 2011 12:48 PM
> Subject: Re: [Freeswitch-users] IVRD
>
>
>> Concerning the use of fs_ivrd:
>>
>> path is ok
>> permission is root 755
>> perl -c gives ../scripts/perl/ivrd-hello_world.pl syntax OK
>>
>> in dialplan I have :
>>
>>
>>
>>
>>
>> ----- Original Message -----
>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, January 24, 2011 11:20 AM
>> Subject: Re: [Freeswitch-users] IVRD
>>
>>
>> check for proper path and execute permissions on the file and perl -c
>> to make sure it compiles.
>>
>>
>> On Mon, Jan 24, 2011 at 12:13 AM, Madovsky <infos at madovsky.org> wrote:
>>> I tried to use IVRD from wiki example
>>>
>>> http://wiki.freeswitch.org/wiki/Ivrd
>>>
>>> and server2.pl in ESL directory
>>> copy and paste in my dialplan ans settings
>>> so the daemon is running well, but if I attempt
>>> to call nothing happens unless hangup.
>>> on the log I can see only
>>>
>>> EXECUTE sofia/internal/9999999999999 at default socket(127.0.0.1:8084 full)
>>>
>>> I tried the tests of troubleshooting without error
>>> I don't understand why the events are not received in the perl script
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 



From infos at madovsky.org  Tue Jan 25 02:26:58 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 24 Jan 2011 18:26:58 -0500
Subject: [Freeswitch-users] IVRD
References: <B890EB1A467549D48DB330F5795906E9@e1705>
	<AANLkTin54y6CJHhSc8DbEZBnx7oBW+4ppD8E3-AQnSk4@mail.gmail.com>
Message-ID: <A842C399BF77474D86093E7D4FDF2F24@e1705>

Apparently the script stalls at

## Create the connection object which is basically an IVR
my $con = new ESL::IVR;

I'm looking into IVR.pm to know wha'ts happening

----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, January 24, 2011 3:30 PM
Subject: Re: [Freeswitch-users] IVRD


did you run the fs_ivrd from a shell and look for output in stderr?
it has to be related to the script executing failing etc.


On Mon, Jan 24, 2011 at 11:57 AM, Madovsky <infos at madovsky.org> wrote:
> Sorry I sent this email by mistake without to finish it.
> so in my dialplan I have :
> <extension name="gluglu">
> <condition field="destination_number"
> expression="^999$">
> <action application="set"
> data="ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl"/>
> <action application="socket" data="127.0.0.1:9090
> full"/>
> <action application="answer"/>
> <action application="voicemail" data="default
> ${domain_name} ${dialed_extension}"/>
> <condition>
> </extension>
>
> and when I call this extension in log level 7 I can see :
>
> Dialplan: sofia/internal/9999999999999 at default Action
> set(ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl)
> Dialplan: sofia/internal/9999999999999 at default Action 
> socket(127.0.0.1:9090
> full)
> Starting ivrd-hello_world.pl...
>
> and no ivrd-hello_world.pl code is executed in the dialplan unless the 
> print
> "Starting ivrd-hello_world.pl";
>
> the same if I replace fs_ivrd with server2.pl for example
>
> Thanks
>
>
> ----- Original Message -----
> From: "Madovsky" <infos at madovsky.org>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, January 24, 2011 12:48 PM
> Subject: Re: [Freeswitch-users] IVRD
>
>
>> Concerning the use of fs_ivrd:
>>
>> path is ok
>> permission is root 755
>> perl -c gives ../scripts/perl/ivrd-hello_world.pl syntax OK
>>
>> in dialplan I have :
>>
>>
>>
>>
>>
>> ----- Original Message -----
>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, January 24, 2011 11:20 AM
>> Subject: Re: [Freeswitch-users] IVRD
>>
>>
>> check for proper path and execute permissions on the file and perl -c
>> to make sure it compiles.
>>
>>
>> On Mon, Jan 24, 2011 at 12:13 AM, Madovsky <infos at madovsky.org> wrote:
>>> I tried to use IVRD from wiki example
>>>
>>> http://wiki.freeswitch.org/wiki/Ivrd
>>>
>>> and server2.pl in ESL directory
>>> copy and paste in my dialplan ans settings
>>> so the daemon is running well, but if I attempt
>>> to call nothing happens unless hangup.
>>> on the log I can see only
>>>
>>> EXECUTE sofia/internal/9999999999999 at default socket(127.0.0.1:8084 full)
>>>
>>> I tried the tests of troubleshooting without error
>>> I don't understand why the events are not received in the perl script
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 



From anthony.minessale at gmail.com  Tue Jan 25 02:32:08 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Jan 2011 17:32:08 -0600
Subject: [Freeswitch-users] IVRD
In-Reply-To: <A842C399BF77474D86093E7D4FDF2F24@e1705>
References: <B890EB1A467549D48DB330F5795906E9@e1705>
	<AANLkTin54y6CJHhSc8DbEZBnx7oBW+4ppD8E3-AQnSk4@mail.gmail.com>
	<A842C399BF77474D86093E7D4FDF2F24@e1705>
Message-ID: <AANLkTi=CwgFtn3Xv0-9mM5nzsk4hWUpuEDYjeW7dw97=@mail.gmail.com>

did you maybe update FS and not update all the ESL stuff?
Do you have it in a nostandard location?


On Mon, Jan 24, 2011 at 5:26 PM, Madovsky <infos at madovsky.org> wrote:
> Apparently the script stalls at
>
> ## Create the connection object which is basically an IVR
> my $con = new ESL::IVR;
>
> I'm looking into IVR.pm to know wha'ts happening
>
> ----- Original Message -----
> From: "Anthony Minessale" <anthony.minessale at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, January 24, 2011 3:30 PM
> Subject: Re: [Freeswitch-users] IVRD
>
>
> did you run the fs_ivrd from a shell and look for output in stderr?
> it has to be related to the script executing failing etc.
>
>
> On Mon, Jan 24, 2011 at 11:57 AM, Madovsky <infos at madovsky.org> wrote:
>> Sorry I sent this email by mistake without to finish it.
>> so in my dialplan I have :
>> <extension name="gluglu">
>> <condition field="destination_number"
>> expression="^999$">
>> <action application="set"
>> data="ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl"/>
>> <action application="socket" data="127.0.0.1:9090
>> full"/>
>> <action application="answer"/>
>> <action application="voicemail" data="default
>> ${domain_name} ${dialed_extension}"/>
>> <condition>
>> </extension>
>>
>> and when I call this extension in log level 7 I can see :
>>
>> Dialplan: sofia/internal/9999999999999 at default Action
>> set(ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl)
>> Dialplan: sofia/internal/9999999999999 at default Action
>> socket(127.0.0.1:9090
>> full)
>> Starting ivrd-hello_world.pl...
>>
>> and no ivrd-hello_world.pl code is executed in the dialplan unless the
>> print
>> "Starting ivrd-hello_world.pl";
>>
>> the same if I replace fs_ivrd with server2.pl for example
>>
>> Thanks
>>
>>
>> ----- Original Message -----
>> From: "Madovsky" <infos at madovsky.org>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, January 24, 2011 12:48 PM
>> Subject: Re: [Freeswitch-users] IVRD
>>
>>
>>> Concerning the use of fs_ivrd:
>>>
>>> path is ok
>>> permission is root 755
>>> perl -c gives ../scripts/perl/ivrd-hello_world.pl syntax OK
>>>
>>> in dialplan I have :
>>>
>>>
>>>
>>>
>>>
>>> ----- Original Message -----
>>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>> Sent: Monday, January 24, 2011 11:20 AM
>>> Subject: Re: [Freeswitch-users] IVRD
>>>
>>>
>>> check for proper path and execute permissions on the file and perl -c
>>> to make sure it compiles.
>>>
>>>
>>> On Mon, Jan 24, 2011 at 12:13 AM, Madovsky <infos at madovsky.org> wrote:
>>>> I tried to use IVRD from wiki example
>>>>
>>>> http://wiki.freeswitch.org/wiki/Ivrd
>>>>
>>>> and server2.pl in ESL directory
>>>> copy and paste in my dialplan ans settings
>>>> so the daemon is running well, but if I attempt
>>>> to call nothing happens unless hangup.
>>>> on the log I can see only
>>>>
>>>> EXECUTE sofia/internal/9999999999999 at default socket(127.0.0.1:8084 full)
>>>>
>>>> I tried the tests of troubleshooting without error
>>>> I don't understand why the events are not received in the perl script
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Tue Jan 25 02:38:26 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Jan 2011 17:38:26 -0600
Subject: [Freeswitch-users] Caller ID using Fifo
In-Reply-To: <AANLkTim8QUcxH0rbcZymMr_EAnqpdE8OUnjpJ1tWN7sn@mail.gmail.com>
References: <AANLkTik6pdgXgND6kmzZA4qe8vDZQ0_7e47+ciVWPfub@mail.gmail.com>
	<AANLkTikKDO3qru-9=nuPEQcW+iCN7ccizyJAh9sqUkCN@mail.gmail.com>
	<AANLkTi=p7_CsUDWS62+weZv50+PBZ8RLop0Q04y-c4hB@mail.gmail.com>
	<AANLkTi=m_hW9vQV5nbC2jwcos0bKy2rCf7YAr6aA23oz@mail.gmail.com>
	<AANLkTimE-0g-Gc92tseC-GWOuwV-icvJU7ndLnhPOCo=@mail.gmail.com>
	<AANLkTim8QUcxH0rbcZymMr_EAnqpdE8OUnjpJ1tWN7sn@mail.gmail.com>
Message-ID: <AANLkTi=7rFemnLG1jtDTH49Y0Ty15OysSzwjH9SFuqBB@mail.gmail.com>

there is a param in fifo "outbound_name" if its set, it will put it in
parens along with the caller's caller_id_name

(sales) Fred Smith
2121231234



On Mon, Jan 24, 2011 at 4:55 PM, Joao Leme <joaocarlosleme at gmail.com> wrote:
> I haven't been able to find a solution either (havent' had much time), but
> have you already tried the following to make sure it does not work?
> "Note: If you wish to specify the caller ID presented when a fifo calls an
> agent, set the origination_caller_id_name and origination_caller_id_num
> variables to the values desired. These could be set within the {} of the
> dialstring, or they could be set using the set application in the dialplan
> which places the caller into the fifo (before the 'fifo in' executed on the
> caller)."
> I haven't had time to look into but that's where I would start.
> It was added to the Fifo wiki by User "Sward" (no futher info on user) on:
> (cur) (prev) 15:21, 9 June 2010 Sward (Talk | contribs) (26,518 bytes)
> (?Configure for Agent Callback)
> If anyone know this "Sward" guy, he may have some?answers.
> Thanks,
> John
>
> On Mon, Jan 24, 2011 at 1:20 AM, Marc de Corny <marcdecorny at gmail.com>
> wrote:
>>
>> Hi All
>>
>> I have since had a play with mod_callcenter and have not been able to send
>> the call to an agent with the caller_id_name as the name of the queue.
>>
>> I keep on getting the CLI on both.
>>
>> As this cannot be done in FIFO either, do any of you have any ideas? or
>> managed to get it working ?
>>
>> thanks
>> Marc
>>
>> On Fri, Jan 14, 2011 at 4:35 PM, Marc de Corny <marcdecorny at gmail.com>
>> wrote:
>>>
>>> Just to follow up on this subject.
>>>
>>> I have done a lot of testing on the fifo trying to get the caller_id_name
>>> changed on the outbound call to the agent and to be honest I cannot
>>> understand the explanation.
>>>
>>> If mod_fifo does not know which call it will connect until the agent
>>> answers, how come it displays the CLI correctly, jsut won;t let me change
>>> it.
>>>
>>> Still seems strange. I am looking into the Mod_callcentre to check if it
>>> sends caller_id information. but the same logic if valid could apply
>>>
>>> Also maybe someone should change the Wiki ( I would but do not have
>>> enough expertise on the subject) because the following?is a bit misleading
>>>
>>> ?"Note: If you wish to specify the caller ID presented when a fifo calls
>>> an agent, set the origination_caller_id_name and origination_caller_id_num
>>> variables to the values desired. These could be set within the {} of the
>>> dialstring, or they could be set using the set application in the dialplan
>>> which places the caller into the fifo (before the 'fifo in' executed on the
>>> caller). "
>>> thanks
>>> Marc
>>> On Thu, Jan 13, 2011 at 10:47 PM, Joao Leme <joaocarlosleme at gmail.com>
>>> wrote:
>>>>
>>>> What about showing the Caller ID after it is?answered? Any way to do
>>>> that?
>>>>
>>>> 2011/1/12 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>>>
>>>>> Jo?o Leme,
>>>>> The caller id is not passed when the phone is ringing because mod_fifo
>>>>> does not know which call is going to be sent to that channel once it is
>>>>> answered until it is really answered. I don't know if mod_callcenter does
>>>>> show anything but you should consider looking at the documentation if you
>>>>> really need this feature.
>>>>> Regards,
>>>>> Jo?o Mesquita
>>>>>
>>>>>
>>>>> On Wed, Jan 12, 2011 at 9:15 PM, Joao Leme <joaocarlosleme at gmail.com>
>>>>> wrote:
>>>>>>
>>>>>> Hi there,
>>>>>> I would like to know if there is a way to see the caller ID on my Sip
>>>>>> Client (X-Lite for example) of the caller that I answear from a Fifo queue?
>>>>>> Thanks,
>>>>>> John
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From steveayre at gmail.com  Tue Jan 25 02:50:21 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 24 Jan 2011 23:50:21 +0000
Subject: [Freeswitch-users] Outgoing registrations within a fs pool
In-Reply-To: <20110125001130.74166c5a@mail.winet.ch>
References: <20110125001130.74166c5a@mail.winet.ch>
Message-ID: <AANLkTikQCgc=g3ThyKFsnCK280N+unJjEQPZmOzg43oh@mail.gmail.com>

They will all register. They have to because they're all different
endpoints.

-Steve


On 24 January 2011 23:11, Bernhard Suttner <bernhard.suttner at winet.ch>wrote:

> Hi,
>
> thanks for your answer. If there are 5 freeswitch servers within the pool
> (shared db) I would add the gateway to all of the freeswitch server. Will
> all of them try to register or would the first freeswitch server register
> and the other servers would check if the (re)-registration was already done?
>
> The point is, that all the freeswitch servers do have the same
> configuration and it does not matter which freeswitch server registers
> because a incoming call would be handled the same way on all of the servers
> with shared database.
>
> Best regards,
> Bernhard
>
> ----- Original Message -----
> From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
> To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
> Sent: Mon, 24 Jan 2011 23:31:48 +0100
> Subject: Re: [Freeswitch-users] Outgoing registrations within a fs pool
>
>
> > I think gateways are all in-memory so it should be ok
> >
> > On Mon, Jan 24, 2011 at 2:58 PM, Bernhard Suttner
> > <bernhard.suttner at winet.ch> wrote:
> > > Hi,
> > >
> > > does someone has an idea what happens if I would add the gateway (with
> > registration) to all the freeswitch servers of the pool with shared
> > database?
> > >
> > > Best regards,
> > > Bernhard
> > >
> > > ----- Original Message -----
> > > From: Bernhard Suttner [mailto:bernhard.suttner at winet.ch]
> > > To: 'FreeSWITCH Users Help' [mailto:
> freeswitch-users at lists.freeswitch.org]
> > > Sent: Fri, 14 Jan 2011 10:42:10 +0100
> > > Subject: Re: [Freeswitch-users] Outgoing registrations within a fs pool
> > >
> > >
> > >> Hi,
> > >>
> > >> Kamailio is able to do a registration with the uacreg module, but it
> is
> > not
> > >> as stable as on FreeSWITCH (its not the main task of Kamailio to
> > implement
> > >> B2BUA features). Therefore I asked if FreeSWITCH is able to register
> > "from a
> > >> pool of FS servers".
> > >>
> > >> Best regards,
> > >> Bernhard
> > >>
> > >> -----Urspr?ngliche Nachricht-----
> > >> Von: freeswitch-users-bounces at lists.freeswitch.org
> > >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
> > Paul
> > >> Cupis
> > >> Gesendet: Donnerstag, 13. Januar 2011 21:59
> > >> An: FreeSWITCH Users Help
> > >> Betreff: Re: [Freeswitch-users] Outgoing registrations within a fs
> pool
> > >>
> > >> Perhaps look at getting the kamailio server to register to the
> provider
> > >> rather than one of the FS from the pool?
> > >>
> > >>
> > >> _______________________________________________
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> http://www.freeswitch.org
> > >>
> > >>
> > >>
> > >> _______________________________________________
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> http://www.freeswitch.org
> > >>
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Tue Jan 25 02:55:25 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 24 Jan 2011 18:55:25 -0500
Subject: [Freeswitch-users] IVRD
References: <B890EB1A467549D48DB330F5795906E9@e1705><AANLkTin54y6CJHhSc8DbEZBnx7oBW+4ppD8E3-AQnSk4@mail.gmail.com><A842C399BF77474D86093E7D4FDF2F24@e1705>
	<AANLkTi=CwgFtn3Xv0-9mM5nzsk4hWUpuEDYjeW7dw97=@mail.gmail.com>
Message-ID: <4CEE1B8B58EE4E6FB13FC4B626A47AE3@e1705>

I compiled ESL from the last source tree of FS I have (git from about 5 days 
ago)
and followed the instructions on wiki ESL Perl
ESL stuff are in

/usr/local/lib64/perl5/site_perl/5.10.0/x86_64-linux-thread-multi/ESL



----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, January 24, 2011 6:32 PM
Subject: Re: [Freeswitch-users] IVRD


> did you maybe update FS and not update all the ESL stuff?
> Do you have it in a nostandard location?
>
>
> On Mon, Jan 24, 2011 at 5:26 PM, Madovsky <infos at madovsky.org> wrote:
>> Apparently the script stalls at
>>
>> ## Create the connection object which is basically an IVR
>> my $con = new ESL::IVR;
>>
>> I'm looking into IVR.pm to know wha'ts happening
>>
>> ----- Original Message -----
>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, January 24, 2011 3:30 PM
>> Subject: Re: [Freeswitch-users] IVRD
>>
>>
>> did you run the fs_ivrd from a shell and look for output in stderr?
>> it has to be related to the script executing failing etc.
>>
>>
>> On Mon, Jan 24, 2011 at 11:57 AM, Madovsky <infos at madovsky.org> wrote:
>>> Sorry I sent this email by mistake without to finish it.
>>> so in my dialplan I have :
>>> <extension name="gluglu">
>>> <condition field="destination_number"
>>> expression="^999$">
>>> <action application="set"
>>> data="ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl"/>
>>> <action application="socket" data="127.0.0.1:9090
>>> full"/>
>>> <action application="answer"/>
>>> <action application="voicemail" data="default
>>> ${domain_name} ${dialed_extension}"/>
>>> <condition>
>>> </extension>
>>>
>>> and when I call this extension in log level 7 I can see :
>>>
>>> Dialplan: sofia/internal/9999999999999 at default Action
>>> set(ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl)
>>> Dialplan: sofia/internal/9999999999999 at default Action
>>> socket(127.0.0.1:9090
>>> full)
>>> Starting ivrd-hello_world.pl...
>>>
>>> and no ivrd-hello_world.pl code is executed in the dialplan unless the
>>> print
>>> "Starting ivrd-hello_world.pl";
>>>
>>> the same if I replace fs_ivrd with server2.pl for example
>>>
>>> Thanks
>>>
>>>
>>> ----- Original Message -----
>>> From: "Madovsky" <infos at madovsky.org>
>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>> Sent: Monday, January 24, 2011 12:48 PM
>>> Subject: Re: [Freeswitch-users] IVRD
>>>
>>>
>>>> Concerning the use of fs_ivrd:
>>>>
>>>> path is ok
>>>> permission is root 755
>>>> perl -c gives ../scripts/perl/ivrd-hello_world.pl syntax OK
>>>>
>>>> in dialplan I have :
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> ----- Original Message -----
>>>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Monday, January 24, 2011 11:20 AM
>>>> Subject: Re: [Freeswitch-users] IVRD
>>>>
>>>>
>>>> check for proper path and execute permissions on the file and perl -c
>>>> to make sure it compiles.
>>>>
>>>>
>>>> On Mon, Jan 24, 2011 at 12:13 AM, Madovsky <infos at madovsky.org> wrote:
>>>>> I tried to use IVRD from wiki example
>>>>>
>>>>> http://wiki.freeswitch.org/wiki/Ivrd
>>>>>
>>>>> and server2.pl in ESL directory
>>>>> copy and paste in my dialplan ans settings
>>>>> so the daemon is running well, but if I attempt
>>>>> to call nothing happens unless hangup.
>>>>> on the log I can see only
>>>>>
>>>>> EXECUTE sofia/internal/9999999999999 at default socket(127.0.0.1:8084 
>>>>> full)
>>>>>
>>>>> I tried the tests of troubleshooting without error
>>>>> I don't understand why the events are not received in the perl script
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From anthony.minessale at gmail.com  Tue Jan 25 03:02:28 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Jan 2011 18:02:28 -0600
Subject: [Freeswitch-users] IVRD
In-Reply-To: <4CEE1B8B58EE4E6FB13FC4B626A47AE3@e1705>
References: <B890EB1A467549D48DB330F5795906E9@e1705>
	<AANLkTin54y6CJHhSc8DbEZBnx7oBW+4ppD8E3-AQnSk4@mail.gmail.com>
	<A842C399BF77474D86093E7D4FDF2F24@e1705>
	<AANLkTi=CwgFtn3Xv0-9mM5nzsk4hWUpuEDYjeW7dw97=@mail.gmail.com>
	<4CEE1B8B58EE4E6FB13FC4B626A47AE3@e1705>
Message-ID: <AANLkTikr5_3U6E1WHpsgNExN9a0M2_+papGWgDcoDZ8O@mail.gmail.com>

you may want to go over everything again:
We use that extensively so it's unlikely there is a problem.

search your dir for any stale ESL.so or .pm files and recopy them all
from your source tree so they match the version of FS you are on.


Did you try the exact test example?
I can test it on my end tomorrow if you are still stuck.



On Mon, Jan 24, 2011 at 5:55 PM, Madovsky <infos at madovsky.org> wrote:
> I compiled ESL from the last source tree of FS I have (git from about 5 days
> ago)
> and followed the instructions on wiki ESL Perl
> ESL stuff are in
>
> /usr/local/lib64/perl5/site_perl/5.10.0/x86_64-linux-thread-multi/ESL
>
>
>
> ----- Original Message -----
> From: "Anthony Minessale" <anthony.minessale at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, January 24, 2011 6:32 PM
> Subject: Re: [Freeswitch-users] IVRD
>
>
>> did you maybe update FS and not update all the ESL stuff?
>> Do you have it in a nostandard location?
>>
>>
>> On Mon, Jan 24, 2011 at 5:26 PM, Madovsky <infos at madovsky.org> wrote:
>>> Apparently the script stalls at
>>>
>>> ## Create the connection object which is basically an IVR
>>> my $con = new ESL::IVR;
>>>
>>> I'm looking into IVR.pm to know wha'ts happening
>>>
>>> ----- Original Message -----
>>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>> Sent: Monday, January 24, 2011 3:30 PM
>>> Subject: Re: [Freeswitch-users] IVRD
>>>
>>>
>>> did you run the fs_ivrd from a shell and look for output in stderr?
>>> it has to be related to the script executing failing etc.
>>>
>>>
>>> On Mon, Jan 24, 2011 at 11:57 AM, Madovsky <infos at madovsky.org> wrote:
>>>> Sorry I sent this email by mistake without to finish it.
>>>> so in my dialplan I have :
>>>> <extension name="gluglu">
>>>> <condition field="destination_number"
>>>> expression="^999$">
>>>> <action application="set"
>>>> data="ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl"/>
>>>> <action application="socket" data="127.0.0.1:9090
>>>> full"/>
>>>> <action application="answer"/>
>>>> <action application="voicemail" data="default
>>>> ${domain_name} ${dialed_extension}"/>
>>>> <condition>
>>>> </extension>
>>>>
>>>> and when I call this extension in log level 7 I can see :
>>>>
>>>> Dialplan: sofia/internal/9999999999999 at default Action
>>>> set(ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl)
>>>> Dialplan: sofia/internal/9999999999999 at default Action
>>>> socket(127.0.0.1:9090
>>>> full)
>>>> Starting ivrd-hello_world.pl...
>>>>
>>>> and no ivrd-hello_world.pl code is executed in the dialplan unless the
>>>> print
>>>> "Starting ivrd-hello_world.pl";
>>>>
>>>> the same if I replace fs_ivrd with server2.pl for example
>>>>
>>>> Thanks
>>>>
>>>>
>>>> ----- Original Message -----
>>>> From: "Madovsky" <infos at madovsky.org>
>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Monday, January 24, 2011 12:48 PM
>>>> Subject: Re: [Freeswitch-users] IVRD
>>>>
>>>>
>>>>> Concerning the use of fs_ivrd:
>>>>>
>>>>> path is ok
>>>>> permission is root 755
>>>>> perl -c gives ../scripts/perl/ivrd-hello_world.pl syntax OK
>>>>>
>>>>> in dialplan I have :
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> ----- Original Message -----
>>>>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>> Sent: Monday, January 24, 2011 11:20 AM
>>>>> Subject: Re: [Freeswitch-users] IVRD
>>>>>
>>>>>
>>>>> check for proper path and execute permissions on the file and perl -c
>>>>> to make sure it compiles.
>>>>>
>>>>>
>>>>> On Mon, Jan 24, 2011 at 12:13 AM, Madovsky <infos at madovsky.org> wrote:
>>>>>> I tried to use IVRD from wiki example
>>>>>>
>>>>>> http://wiki.freeswitch.org/wiki/Ivrd
>>>>>>
>>>>>> and server2.pl in ESL directory
>>>>>> copy and paste in my dialplan ans settings
>>>>>> so the daemon is running well, but if I attempt
>>>>>> to call nothing happens unless hangup.
>>>>>> on the log I can see only
>>>>>>
>>>>>> EXECUTE sofia/internal/9999999999999 at default socket(127.0.0.1:8084
>>>>>> full)
>>>>>>
>>>>>> I tried the tests of troubleshooting without error
>>>>>> I don't understand why the events are not received in the perl script
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org
>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From infos at madovsky.org  Tue Jan 25 03:12:07 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 24 Jan 2011 19:12:07 -0500
Subject: [Freeswitch-users] IVRD
References: <B890EB1A467549D48DB330F5795906E9@e1705><AANLkTin54y6CJHhSc8DbEZBnx7oBW+4ppD8E3-AQnSk4@mail.gmail.com><A842C399BF77474D86093E7D4FDF2F24@e1705><AANLkTi=CwgFtn3Xv0-9mM5nzsk4hWUpuEDYjeW7dw97=@mail.gmail.com><4CEE1B8B58EE4E6FB13FC4B626A47AE3@e1705>
	<AANLkTikr5_3U6E1WHpsgNExN9a0M2_+papGWgDcoDZ8O@mail.gmail.com>
Message-ID: <6DE4198B7D834FE28C758DDB2AD01ACE@e1705>

> Did you try the exact test example?
yes I did.

I'm going to update git source and reinstall all now.
write you once done.

thanks

----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, January 24, 2011 7:02 PM
Subject: Re: [Freeswitch-users] IVRD


> you may want to go over everything again:
> We use that extensively so it's unlikely there is a problem.
>
> search your dir for any stale ESL.so or .pm files and recopy them all
> from your source tree so they match the version of FS you are on.
>
>
> Did you try the exact test example?
> I can test it on my end tomorrow if you are still stuck.
>
>
>
> On Mon, Jan 24, 2011 at 5:55 PM, Madovsky <infos at madovsky.org> wrote:
>> I compiled ESL from the last source tree of FS I have (git from about 5 
>> days
>> ago)
>> and followed the instructions on wiki ESL Perl
>> ESL stuff are in
>>
>> /usr/local/lib64/perl5/site_perl/5.10.0/x86_64-linux-thread-multi/ESL
>>
>>
>>
>> ----- Original Message -----
>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, January 24, 2011 6:32 PM
>> Subject: Re: [Freeswitch-users] IVRD
>>
>>
>>> did you maybe update FS and not update all the ESL stuff?
>>> Do you have it in a nostandard location?
>>>
>>>
>>> On Mon, Jan 24, 2011 at 5:26 PM, Madovsky <infos at madovsky.org> wrote:
>>>> Apparently the script stalls at
>>>>
>>>> ## Create the connection object which is basically an IVR
>>>> my $con = new ESL::IVR;
>>>>
>>>> I'm looking into IVR.pm to know wha'ts happening
>>>>
>>>> ----- Original Message -----
>>>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Monday, January 24, 2011 3:30 PM
>>>> Subject: Re: [Freeswitch-users] IVRD
>>>>
>>>>
>>>> did you run the fs_ivrd from a shell and look for output in stderr?
>>>> it has to be related to the script executing failing etc.
>>>>
>>>>
>>>> On Mon, Jan 24, 2011 at 11:57 AM, Madovsky <infos at madovsky.org> wrote:
>>>>> Sorry I sent this email by mistake without to finish it.
>>>>> so in my dialplan I have :
>>>>> <extension name="gluglu">
>>>>> <condition field="destination_number"
>>>>> expression="^999$">
>>>>> <action application="set"
>>>>> data="ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl"/>
>>>>> <action application="socket" data="127.0.0.1:9090
>>>>> full"/>
>>>>> <action application="answer"/>
>>>>> <action application="voicemail" data="default
>>>>> ${domain_name} ${dialed_extension}"/>
>>>>> <condition>
>>>>> </extension>
>>>>>
>>>>> and when I call this extension in log level 7 I can see :
>>>>>
>>>>> Dialplan: sofia/internal/9999999999999 at default Action
>>>>> set(ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl)
>>>>> Dialplan: sofia/internal/9999999999999 at default Action
>>>>> socket(127.0.0.1:9090
>>>>> full)
>>>>> Starting ivrd-hello_world.pl...
>>>>>
>>>>> and no ivrd-hello_world.pl code is executed in the dialplan unless the
>>>>> print
>>>>> "Starting ivrd-hello_world.pl";
>>>>>
>>>>> the same if I replace fs_ivrd with server2.pl for example
>>>>>
>>>>> Thanks
>>>>>
>>>>>
>>>>> ----- Original Message -----
>>>>> From: "Madovsky" <infos at madovsky.org>
>>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>> Sent: Monday, January 24, 2011 12:48 PM
>>>>> Subject: Re: [Freeswitch-users] IVRD
>>>>>
>>>>>
>>>>>> Concerning the use of fs_ivrd:
>>>>>>
>>>>>> path is ok
>>>>>> permission is root 755
>>>>>> perl -c gives ../scripts/perl/ivrd-hello_world.pl syntax OK
>>>>>>
>>>>>> in dialplan I have :
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> ----- Original Message -----
>>>>>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>>>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>>> Sent: Monday, January 24, 2011 11:20 AM
>>>>>> Subject: Re: [Freeswitch-users] IVRD
>>>>>>
>>>>>>
>>>>>> check for proper path and execute permissions on the file and perl -c
>>>>>> to make sure it compiles.
>>>>>>
>>>>>>
>>>>>> On Mon, Jan 24, 2011 at 12:13 AM, Madovsky <infos at madovsky.org> 
>>>>>> wrote:
>>>>>>> I tried to use IVRD from wiki example
>>>>>>>
>>>>>>> http://wiki.freeswitch.org/wiki/Ivrd
>>>>>>>
>>>>>>> and server2.pl in ESL directory
>>>>>>> copy and paste in my dialplan ans settings
>>>>>>> so the daemon is running well, but if I attempt
>>>>>>> to call nothing happens unless hangup.
>>>>>>> on the log I can see only
>>>>>>>
>>>>>>> EXECUTE sofia/internal/9999999999999 at default socket(127.0.0.1:8084
>>>>>>> full)
>>>>>>>
>>>>>>> I tried the tests of troubleshooting without error
>>>>>>> I don't understand why the events are not received in the perl 
>>>>>>> script
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II
>>>>>>
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>
>>>>>> AIM: anthm
>>>>>> MSN:anthony_minessale at hotmail.com
>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>
>>>>>> FreeSWITCH Developer Conference
>>>>>> sip:888 at conference.freeswitch.org
>>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>>> pstn:+19193869900
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From joaocarlosleme at gmail.com  Tue Jan 25 03:58:02 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Mon, 24 Jan 2011 16:58:02 -0800
Subject: [Freeswitch-users] Caller ID using Fifo
In-Reply-To: <AANLkTi=7rFemnLG1jtDTH49Y0Ty15OysSzwjH9SFuqBB@mail.gmail.com>
References: <AANLkTik6pdgXgND6kmzZA4qe8vDZQ0_7e47+ciVWPfub@mail.gmail.com>
	<AANLkTikKDO3qru-9=nuPEQcW+iCN7ccizyJAh9sqUkCN@mail.gmail.com>
	<AANLkTi=p7_CsUDWS62+weZv50+PBZ8RLop0Q04y-c4hB@mail.gmail.com>
	<AANLkTi=m_hW9vQV5nbC2jwcos0bKy2rCf7YAr6aA23oz@mail.gmail.com>
	<AANLkTimE-0g-Gc92tseC-GWOuwV-icvJU7ndLnhPOCo=@mail.gmail.com>
	<AANLkTim8QUcxH0rbcZymMr_EAnqpdE8OUnjpJ1tWN7sn@mail.gmail.com>
	<AANLkTi=7rFemnLG1jtDTH49Y0Ty15OysSzwjH9SFuqBB@mail.gmail.com>
Message-ID: <AANLkTimTa6h+LKKqQ+8pq8SmJHj8NXBXDgcGBFzcOsMR@mail.gmail.com>

Hi Fred,

I just tried:

<action application="set" data="outbound_name=${caller_id_number}"/>

before

<action application="fifo" data="sales_fifo@$${domain} in undef
$${base_dir}/sounds/music/8000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav"/>

and it didn't work.

Did I do anything wrong?

Thanks,
John

On Mon, Jan 24, 2011 at 3:38 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> there is a param in fifo "outbound_name" if its set, it will put it in
> parens along with the caller's caller_id_name
>
> (sales) Fred Smith
> 2121231234
>
>
>
> On Mon, Jan 24, 2011 at 4:55 PM, Joao Leme <joaocarlosleme at gmail.com>
> wrote:
> > I haven't been able to find a solution either (havent' had much time),
> but
> > have you already tried the following to make sure it does not work?
> > "Note: If you wish to specify the caller ID presented when a fifo calls
> an
> > agent, set the origination_caller_id_name and origination_caller_id_num
> > variables to the values desired. These could be set within the {} of the
> > dialstring, or they could be set using the set application in the
> dialplan
> > which places the caller into the fifo (before the 'fifo in' executed on
> the
> > caller)."
> > I haven't had time to look into but that's where I would start.
> > It was added to the Fifo wiki by User "Sward" (no futher info on user)
> on:
> > (cur) (prev) 15:21, 9 June 2010 Sward (Talk | contribs) (26,518 bytes)
> > (?Configure for Agent Callback)
> > If anyone know this "Sward" guy, he may have some answers.
> > Thanks,
> > John
> >
> > On Mon, Jan 24, 2011 at 1:20 AM, Marc de Corny <marcdecorny at gmail.com>
> > wrote:
> >>
> >> Hi All
> >>
> >> I have since had a play with mod_callcenter and have not been able to
> send
> >> the call to an agent with the caller_id_name as the name of the queue.
> >>
> >> I keep on getting the CLI on both.
> >>
> >> As this cannot be done in FIFO either, do any of you have any ideas? or
> >> managed to get it working ?
> >>
> >> thanks
> >> Marc
> >>
> >> On Fri, Jan 14, 2011 at 4:35 PM, Marc de Corny <marcdecorny at gmail.com>
> >> wrote:
> >>>
> >>> Just to follow up on this subject.
> >>>
> >>> I have done a lot of testing on the fifo trying to get the
> caller_id_name
> >>> changed on the outbound call to the agent and to be honest I cannot
> >>> understand the explanation.
> >>>
> >>> If mod_fifo does not know which call it will connect until the agent
> >>> answers, how come it displays the CLI correctly, jsut won;t let me
> change
> >>> it.
> >>>
> >>> Still seems strange. I am looking into the Mod_callcentre to check if
> it
> >>> sends caller_id information. but the same logic if valid could apply
> >>>
> >>> Also maybe someone should change the Wiki ( I would but do not have
> >>> enough expertise on the subject) because the following is a bit
> misleading
> >>>
> >>>  "Note: If you wish to specify the caller ID presented when a fifo
> calls
> >>> an agent, set the origination_caller_id_name and
> origination_caller_id_num
> >>> variables to the values desired. These could be set within the {} of
> the
> >>> dialstring, or they could be set using the set application in the
> dialplan
> >>> which places the caller into the fifo (before the 'fifo in' executed on
> the
> >>> caller). "
> >>> thanks
> >>> Marc
> >>> On Thu, Jan 13, 2011 at 10:47 PM, Joao Leme <joaocarlosleme at gmail.com>
> >>> wrote:
> >>>>
> >>>> What about showing the Caller ID after it is answered? Any way to do
> >>>> that?
> >>>>
> >>>> 2011/1/12 Jo?o Mesquita <jmesquita at freeswitch.org>
> >>>>>
> >>>>> Jo?o Leme,
> >>>>> The caller id is not passed when the phone is ringing because
> mod_fifo
> >>>>> does not know which call is going to be sent to that channel once it
> is
> >>>>> answered until it is really answered. I don't know if mod_callcenter
> does
> >>>>> show anything but you should consider looking at the documentation if
> you
> >>>>> really need this feature.
> >>>>> Regards,
> >>>>> Jo?o Mesquita
> >>>>>
> >>>>>
> >>>>> On Wed, Jan 12, 2011 at 9:15 PM, Joao Leme <joaocarlosleme at gmail.com
> >
> >>>>> wrote:
> >>>>>>
> >>>>>> Hi there,
> >>>>>> I would like to know if there is a way to see the caller ID on my
> Sip
> >>>>>> Client (X-Lite for example) of the caller that I answear from a Fifo
> queue?
> >>>>>> Thanks,
> >>>>>> John
> >>>>>> _______________________________________________
> >>>>>> FreeSWITCH-users mailing list
> >>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>
> >>>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>> http://www.freeswitch.org
> >>>>>>
> >>>>>
> >>>>>
> >>>>> _______________________________________________
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>
> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>>
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From michal.bielicki at seventhsignal.de  Tue Jan 25 04:12:35 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Tue, 25 Jan 2011 02:12:35 +0100
Subject: [Freeswitch-users] Caller ID using Fifo
In-Reply-To: <AANLkTimTa6h+LKKqQ+8pq8SmJHj8NXBXDgcGBFzcOsMR@mail.gmail.com>
References: <AANLkTik6pdgXgND6kmzZA4qe8vDZQ0_7e47+ciVWPfub@mail.gmail.com>
	<AANLkTikKDO3qru-9=nuPEQcW+iCN7ccizyJAh9sqUkCN@mail.gmail.com>
	<AANLkTi=p7_CsUDWS62+weZv50+PBZ8RLop0Q04y-c4hB@mail.gmail.com>
	<AANLkTi=m_hW9vQV5nbC2jwcos0bKy2rCf7YAr6aA23oz@mail.gmail.com>
	<AANLkTimE-0g-Gc92tseC-GWOuwV-icvJU7ndLnhPOCo=@mail.gmail.com>
	<AANLkTim8QUcxH0rbcZymMr_EAnqpdE8OUnjpJ1tWN7sn@mail.gmail.com>
	<AANLkTi=7rFemnLG1jtDTH49Y0Ty15OysSzwjH9SFuqBB@mail.gmail.com>
	<AANLkTimTa6h+LKKqQ+8pq8SmJHj8NXBXDgcGBFzcOsMR@mail.gmail.com>
Message-ID: <7A71F0B3-C463-4021-A52A-30C1816CF6E3@seventhsignal.de>

Try export instead of set.

Am 25.01.2011 um 01:58 schrieb Joao Leme:

> Hi Fred,
> 
> I just tried:
> 
> <action application="set" data="outbound_name=${caller_id_number}"/>
> 
> before
> 
> <action application="fifo" data="sales_fifo@$${domain} in undef $${base_dir}/sounds/music/8000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav"/>
> 
> and it didn't work.
> 
> Did I do anything wrong?
> 
> Thanks,
> John
> 
> On Mon, Jan 24, 2011 at 3:38 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> there is a param in fifo "outbound_name" if its set, it will put it in
> parens along with the caller's caller_id_name
> 
> (sales) Fred Smith
> 2121231234
> 
> 
> 
> On Mon, Jan 24, 2011 at 4:55 PM, Joao Leme <joaocarlosleme at gmail.com> wrote:
> > I haven't been able to find a solution either (havent' had much time), but
> > have you already tried the following to make sure it does not work?
> > "Note: If you wish to specify the caller ID presented when a fifo calls an
> > agent, set the origination_caller_id_name and origination_caller_id_num
> > variables to the values desired. These could be set within the {} of the
> > dialstring, or they could be set using the set application in the dialplan
> > which places the caller into the fifo (before the 'fifo in' executed on the
> > caller)."
> > I haven't had time to look into but that's where I would start.
> > It was added to the Fifo wiki by User "Sward" (no futher info on user) on:
> > (cur) (prev) 15:21, 9 June 2010 Sward (Talk | contribs) (26,518 bytes)
> > (?Configure for Agent Callback)
> > If anyone know this "Sward" guy, he may have some answers.
> > Thanks,
> > John
> >
> > On Mon, Jan 24, 2011 at 1:20 AM, Marc de Corny <marcdecorny at gmail.com>
> > wrote:
> >>
> >> Hi All
> >>
> >> I have since had a play with mod_callcenter and have not been able to send
> >> the call to an agent with the caller_id_name as the name of the queue.
> >>
> >> I keep on getting the CLI on both.
> >>
> >> As this cannot be done in FIFO either, do any of you have any ideas? or
> >> managed to get it working ?
> >>
> >> thanks
> >> Marc
> >>
> >> On Fri, Jan 14, 2011 at 4:35 PM, Marc de Corny <marcdecorny at gmail.com>
> >> wrote:
> >>>
> >>> Just to follow up on this subject.
> >>>
> >>> I have done a lot of testing on the fifo trying to get the caller_id_name
> >>> changed on the outbound call to the agent and to be honest I cannot
> >>> understand the explanation.
> >>>
> >>> If mod_fifo does not know which call it will connect until the agent
> >>> answers, how come it displays the CLI correctly, jsut won;t let me change
> >>> it.
> >>>
> >>> Still seems strange. I am looking into the Mod_callcentre to check if it
> >>> sends caller_id information. but the same logic if valid could apply
> >>>
> >>> Also maybe someone should change the Wiki ( I would but do not have
> >>> enough expertise on the subject) because the following is a bit misleading
> >>>
> >>>  "Note: If you wish to specify the caller ID presented when a fifo calls
> >>> an agent, set the origination_caller_id_name and origination_caller_id_num
> >>> variables to the values desired. These could be set within the {} of the
> >>> dialstring, or they could be set using the set application in the dialplan
> >>> which places the caller into the fifo (before the 'fifo in' executed on the
> >>> caller). "
> >>> thanks
> >>> Marc
> >>> On Thu, Jan 13, 2011 at 10:47 PM, Joao Leme <joaocarlosleme at gmail.com>
> >>> wrote:
> >>>>
> >>>> What about showing the Caller ID after it is answered? Any way to do
> >>>> that?
> >>>>
> >>>> 2011/1/12 Jo?o Mesquita <jmesquita at freeswitch.org>
> >>>>>
> >>>>> Jo?o Leme,
> >>>>> The caller id is not passed when the phone is ringing because mod_fifo
> >>>>> does not know which call is going to be sent to that channel once it is
> >>>>> answered until it is really answered. I don't know if mod_callcenter does
> >>>>> show anything but you should consider looking at the documentation if you
> >>>>> really need this feature.
> >>>>> Regards,
> >>>>> Jo?o Mesquita
> >>>>>
> >>>>>
> >>>>> On Wed, Jan 12, 2011 at 9:15 PM, Joao Leme <joaocarlosleme at gmail.com>
> >>>>> wrote:
> >>>>>>
> >>>>>> Hi there,
> >>>>>> I would like to know if there is a way to see the caller ID on my Sip
> >>>>>> Client (X-Lite for example) of the caller that I answear from a Fifo queue?
> >>>>>> Thanks,
> >>>>>> John
> >>>>>> _______________________________________________
> >>>>>> FreeSWITCH-users mailing list
> >>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>
> >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>> http://www.freeswitch.org
> >>>>>>
> >>>>>
> >>>>>
> >>>>> _______________________________________________
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>
> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>>
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> 
> 
> --
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de

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From anthony.minessale at gmail.com  Tue Jan 25 04:28:25 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Jan 2011 19:28:25 -0600
Subject: [Freeswitch-users] Caller ID using Fifo
In-Reply-To: <AANLkTi=m_hW9vQV5nbC2jwcos0bKy2rCf7YAr6aA23oz@mail.gmail.com>
References: <AANLkTik6pdgXgND6kmzZA4qe8vDZQ0_7e47+ciVWPfub@mail.gmail.com>
	<AANLkTikKDO3qru-9=nuPEQcW+iCN7ccizyJAh9sqUkCN@mail.gmail.com>
	<AANLkTi=p7_CsUDWS62+weZv50+PBZ8RLop0Q04y-c4hB@mail.gmail.com>
	<AANLkTi=m_hW9vQV5nbC2jwcos0bKy2rCf7YAr6aA23oz@mail.gmail.com>
Message-ID: <AANLkTinX7fZOO9edgHs739RxhcBJwuh70E9f07mM0mzL@mail.gmail.com>

You should all confer to make sure you are all using fs latest git because
that is the version I am talking about.  Fifo has some major new features in
latest that do not exist in older versions including showing the customers
cid when it calls agents.  The dilemma jm describes used to be true but is
no longer the case with the default ringall strategy on latest git.

The customers cid is sent to the agent and if the fifo xml defines
outbound_name param that will be included as well.

If you want to override it you must do what you quoted in the wiki in the
dialstring contained in the member tag of the xml for that membership not in
the dialplan.

On Jan 14, 2011 10:36 AM, "Marc de Corny" <marcdecorny at gmail.com> wrote:
>
> Just to follow up on this subject.
>
> I have done a lot of testing on the fifo trying to get the caller_id_name
changed on the outbound call to the agent and to be honest I cannot
understand the explanation.
>
> If mod_fifo does not know which call it will connect until the agent
answers, how come it displays the CLI correctly, jsut won;t let me change
it.
>
> Still seems strange. I am looking into the Mod_callcentre to check if it
sends caller_id information. but the same logic if valid could apply
>
> Also maybe someone should change the Wiki ( I would but do not have enough
expertise on the subject) because the following is a bit misleading
>
>  "Note: If you wish to specify the caller ID presented when a fifo calls
an agent, set the origination_caller_id_name and origination_caller_id_num
variables to the values desired. These could be set within the {} of the
dialstring, or they could be set using the set application in the dialplan
which places the caller into the fifo (before the 'fifo in' executed on the
caller). "
> thanks
> Marc
> On Thu, Jan 13, 2011 at 10:47 PM, Joao Leme <joaocarlosleme at gmail.com>
wrote:
>>
>> What about showing the Caller ID after it is answered? Any way to do
that?
>>
>> 2011/1/12 Jo?o Mesquita <jmesquita at freeswitch.org>
>>
>>> Jo?o Leme,
>>>
>>> The caller id is not passed when the phone is ringing because mod_fifo
does not know which call is going to be sent to that channel once it is
answered until it is really answered. I don't know if mod_callcenter does
show anything but you should consider looking at the documentation if you
really need this feature.
>>>
>>> Regards,
>>> Jo?o Mesquita
>>>
>>>
>>> On Wed, Jan 12, 2011 at 9:15 PM, Joao Leme <joaocarlosleme at gmail.com>
wrote:
>>>>
>>>> Hi there,
>>>> I would like to know if there is a way to see the caller ID on my Sip
Client (X-Lite for example) of the caller that I answear from a Fifo queue?
>>>> Thanks,
>>>> John
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From joaocarlosleme at gmail.com  Tue Jan 25 04:59:03 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Mon, 24 Jan 2011 17:59:03 -0800
Subject: [Freeswitch-users] Caller ID using Fifo
In-Reply-To: <7A71F0B3-C463-4021-A52A-30C1816CF6E3@seventhsignal.de>
References: <AANLkTik6pdgXgND6kmzZA4qe8vDZQ0_7e47+ciVWPfub@mail.gmail.com>
	<AANLkTikKDO3qru-9=nuPEQcW+iCN7ccizyJAh9sqUkCN@mail.gmail.com>
	<AANLkTi=p7_CsUDWS62+weZv50+PBZ8RLop0Q04y-c4hB@mail.gmail.com>
	<AANLkTi=m_hW9vQV5nbC2jwcos0bKy2rCf7YAr6aA23oz@mail.gmail.com>
	<AANLkTimE-0g-Gc92tseC-GWOuwV-icvJU7ndLnhPOCo=@mail.gmail.com>
	<AANLkTim8QUcxH0rbcZymMr_EAnqpdE8OUnjpJ1tWN7sn@mail.gmail.com>
	<AANLkTi=7rFemnLG1jtDTH49Y0Ty15OysSzwjH9SFuqBB@mail.gmail.com>
	<AANLkTimTa6h+LKKqQ+8pq8SmJHj8NXBXDgcGBFzcOsMR@mail.gmail.com>
	<7A71F0B3-C463-4021-A52A-30C1816CF6E3@seventhsignal.de>
Message-ID: <AANLkTi=A8Ukkg9F7ZiN3fCP=v=PKBwcepTfrsU2hG+s8@mail.gmail.com>

Thanks but no luck. I keeps showing "Queue
<fifo+sales_fifo at 192.168.1.58<fifo%2Bsales_fifo at 192.168.1.58>>"
where "sales_fifo@$${domain}" is the fifo name on fifo.conf.xml.

On Mon, Jan 24, 2011 at 5:12 PM, Michal Bielicki <
michal.bielicki at seventhsignal.de> wrote:

> Try export instead of set.
>
> Am 25.01.2011 um 01:58 schrieb Joao Leme:
>
> Hi Fred,
>
> I just tried:
>
> <action application="set" data="outbound_name=${caller_id_number}"/>
>
> before
>
> <action application="fifo" data="sales_fifo@$${domain} in undef
> $${base_dir}/sounds/music/8000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav"/>
>
> and it didn't work.
>
> Did I do anything wrong?
>
> Thanks,
> John
>
> On Mon, Jan 24, 2011 at 3:38 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> there is a param in fifo "outbound_name" if its set, it will put it in
>> parens along with the caller's caller_id_name
>>
>> (sales) Fred Smith
>> 2121231234
>>
>>
>>
>> On Mon, Jan 24, 2011 at 4:55 PM, Joao Leme <joaocarlosleme at gmail.com>
>> wrote:
>> > I haven't been able to find a solution either (havent' had much time),
>> but
>> > have you already tried the following to make sure it does not work?
>> > "Note: If you wish to specify the caller ID presented when a fifo calls
>> an
>> > agent, set the origination_caller_id_name and origination_caller_id_num
>> > variables to the values desired. These could be set within the {} of the
>> > dialstring, or they could be set using the set application in the
>> dialplan
>> > which places the caller into the fifo (before the 'fifo in' executed on
>> the
>> > caller)."
>> > I haven't had time to look into but that's where I would start.
>> > It was added to the Fifo wiki by User "Sward" (no futher info on user)
>> on:
>> > (cur) (prev) 15:21, 9 June 2010 Sward (Talk | contribs) (26,518 bytes)
>> > (?Configure for Agent Callback)
>> > If anyone know this "Sward" guy, he may have some answers.
>> > Thanks,
>> > John
>> >
>> > On Mon, Jan 24, 2011 at 1:20 AM, Marc de Corny <marcdecorny at gmail.com>
>> > wrote:
>> >>
>> >> Hi All
>> >>
>> >> I have since had a play with mod_callcenter and have not been able to
>> send
>> >> the call to an agent with the caller_id_name as the name of the queue.
>> >>
>> >> I keep on getting the CLI on both.
>> >>
>> >> As this cannot be done in FIFO either, do any of you have any ideas? or
>> >> managed to get it working ?
>> >>
>> >> thanks
>> >> Marc
>> >>
>> >> On Fri, Jan 14, 2011 at 4:35 PM, Marc de Corny <marcdecorny at gmail.com>
>> >> wrote:
>> >>>
>> >>> Just to follow up on this subject.
>> >>>
>> >>> I have done a lot of testing on the fifo trying to get the
>> caller_id_name
>> >>> changed on the outbound call to the agent and to be honest I cannot
>> >>> understand the explanation.
>> >>>
>> >>> If mod_fifo does not know which call it will connect until the agent
>> >>> answers, how come it displays the CLI correctly, jsut won;t let me
>> change
>> >>> it.
>> >>>
>> >>> Still seems strange. I am looking into the Mod_callcentre to check if
>> it
>> >>> sends caller_id information. but the same logic if valid could apply
>> >>>
>> >>> Also maybe someone should change the Wiki ( I would but do not have
>> >>> enough expertise on the subject) because the following is a bit
>> misleading
>> >>>
>> >>>  "Note: If you wish to specify the caller ID presented when a fifo
>> calls
>> >>> an agent, set the origination_caller_id_name and
>> origination_caller_id_num
>> >>> variables to the values desired. These could be set within the {} of
>> the
>> >>> dialstring, or they could be set using the set application in the
>> dialplan
>> >>> which places the caller into the fifo (before the 'fifo in' executed
>> on the
>> >>> caller). "
>> >>> thanks
>> >>> Marc
>> >>> On Thu, Jan 13, 2011 at 10:47 PM, Joao Leme <joaocarlosleme at gmail.com
>> >
>> >>> wrote:
>> >>>>
>> >>>> What about showing the Caller ID after it is answered? Any way to do
>> >>>> that?
>> >>>>
>> >>>> 2011/1/12 Jo?o Mesquita <jmesquita at freeswitch.org>
>> >>>>>
>> >>>>> Jo?o Leme,
>> >>>>> The caller id is not passed when the phone is ringing because
>> mod_fifo
>> >>>>> does not know which call is going to be sent to that channel once it
>> is
>> >>>>> answered until it is really answered. I don't know if mod_callcenter
>> does
>> >>>>> show anything but you should consider looking at the documentation
>> if you
>> >>>>> really need this feature.
>> >>>>> Regards,
>> >>>>> Jo?o Mesquita
>> >>>>>
>> >>>>>
>> >>>>> On Wed, Jan 12, 2011 at 9:15 PM, Joao Leme <
>> joaocarlosleme at gmail.com>
>> >>>>> wrote:
>> >>>>>>
>> >>>>>> Hi there,
>> >>>>>> I would like to know if there is a way to see the caller ID on my
>> Sip
>> >>>>>> Client (X-Lite for example) of the caller that I answear from a
>> Fifo queue?
>> >>>>>> Thanks,
>> >>>>>> John
>> >>>>>> _______________________________________________
>> >>>>>> FreeSWITCH-users mailing list
>> >>>>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>>>
>> >>>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>>> http://www.freeswitch.org
>> >>>>>>
>> >>>>>
>> >>>>>
>> >>>>> _______________________________________________
>> >>>>> FreeSWITCH-users mailing list
>> >>>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>>
>> >>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>> http://www.freeswitch.org
>> >>>>>
>> >>>>
>> >>>>
>> >>>> _______________________________________________
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>>
>> >>>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> *Michal Bielicki*
> Gesch?ftsf?hrer / CEO
>
> *Seventh Signal Ltd. & Co. KG*
> Weigandufer 45, B?ro 115, D-12059 Berlin
> Voice: +49 30 60988730
>
> Amtsgericht Charlottenburg HRA 44413 B
> Ust.-ID: DE266981999
> Gesch?ftsf?hrer: Michal Bielicki
> Pers?nlich Haftende Gesellschafterin:
> Seventh Signal Ltd, 69 Great Hampton St. Birmingham,
> B18 6EW, GB, Company Nr.: 06889439
> WWW.: http://www.seventhsignal.de
>
>
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From u2nsam at gmail.com  Tue Jan 25 06:11:29 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 25 Jan 2011 08:41:29 +0530
Subject: [Freeswitch-users] # in prefix
In-Reply-To: <2BAF59AA-B828-4AF6-9BD8-F4E2EAB390A8@freeswitch.org>
References: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
	<2BAF59AA-B828-4AF6-9BD8-F4E2EAB390A8@freeswitch.org>
Message-ID: <AANLkTim2Fd_t0GWqjip7jjKarvkqxqMM_P=gjVbQS0JZ@mail.gmail.com>

The more info on FS is that it is working in proxy mode , so what ever
customer sends it sends to the next hop,
I will try using %23 for # while sending calls.

Regards
Sam

On Mon, Jan 24, 2011 at 10:34 PM, Brian West <brian at freeswitch.org> wrote:

> You MUST url encode the # key as %23 since its illegal in the URI.  I think
> we will LET you send it in the URI but some stuff will throw a fit because
> its invalid.
>
> /b
>
> On Jan 24, 2011, at 10:48 AM, Sam wrote:
>
> > Hello,
> >
> > As i send call with prefix 999# the prefix is not passed to the provider
> from FS.
> >
> > Customer sends 999#12127773456 ---> FS ---> 12127773456 ,
> > it goes without prefix to the provider because of '#', is there any
> method to send to the provider 999#12127773456 .
> >
> > Regds
> > Sam
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Tue Jan 25 06:15:01 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 Jan 2011 21:15:01 -0600
Subject: [Freeswitch-users] # in prefix
In-Reply-To: <AANLkTim2Fd_t0GWqjip7jjKarvkqxqMM_P=gjVbQS0JZ@mail.gmail.com>
References: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
	<2BAF59AA-B828-4AF6-9BD8-F4E2EAB390A8@freeswitch.org>
	<AANLkTim2Fd_t0GWqjip7jjKarvkqxqMM_P=gjVbQS0JZ@mail.gmail.com>
Message-ID: <29DBF085-91BA-4110-AE87-CF8F040B6280@freeswitch.org>

We aren't a proxy.. what exactly are you talking bout?  Can you paste us a dialplan example?

/b

On Jan 24, 2011, at 9:11 PM, Sam wrote:

> The more info on FS is that it is working in proxy mode , so what ever customer sends it sends to the next hop,
> I will try using %23 for # while sending calls.
> 
> Regards
> Sam



From rob4manhere at gmail.com  Tue Jan 25 06:20:14 2011
From: rob4manhere at gmail.com (Rob Forman)
Date: Mon, 24 Jan 2011 21:20:14 -0600
Subject: [Freeswitch-users] praise of freeSWITCH developers
In-Reply-To: <E826C94A3C6BE44198FAF4D66B5AB4D80A51086A5A@VA3DIAXVS351.RED001.local>
References: <20110123111322.5f78cd1d@mail.winet.ch>
	<7FAC1ED2-E969-42C1-9E05-3D741DFFC1DD@freeswitch.org>
	<AANLkTi=BuqzsYY0eu=sRioUB3eyBf-nzJWGspDrrxxx=@mail.gmail.com>
	<1295860695.32615.23.camel@luna.tc.commsmundi.com>
	<0A4D008BA10A4EAF8588CC826AB4250A@e1705>
	<AANLkTinVdL8SACEhSB_DnQAiXCwVTYy54J28jmqEJ+22@mail.gmail.com>
	<E826C94A3C6BE44198FAF4D66B5AB4D80A51086A5A@VA3DIAXVS351.RED001.local>
Message-ID: <AANLkTi=K+-z4nBMzXUR_Owt5UiZM9125MYS6oQLS9QcE@mail.gmail.com>

Late to the party.. but just added my donation and wanted to say thanks to
the team for a product that has always exceeded expectations.  Its both rare
and refreshing!

Rob


On Mon, Jan 24, 2011 at 11:58 AM, Robert Hadley
<robert.hadley at teotech.com>wrote:

>
>
> Thanks,
>
> I just caught up reading UL emails, and just added my $50 Thank You to the
> FS developers
>
> -Robert
>
>
>
> *From:* Michael Collins [mailto:msc at freeswitch.org]
> *Sent:* Monday, January 24, 2011 9:31 AM
> *To:* FreeSWITCH Users Help
>
> *Subject:* Re: [Freeswitch-users] praise of freeSWITCH developers
>
>
>
> Thank you all for the unsolicited donations! The FreeSWITCH devs are most
> appreciative of our awesome community. Keep spreading the word about how
> awesome FreeSWITCH - and its community - really is.
>
>
>
> -MC
>
> On Mon, Jan 24, 2011 at 9:04 AM, Madovsky <infos at madovsky.org> wrote:
>
> I think there are also people who gave money
> without to say anything, so your total is wrong ! :D
> nice solidarity, thanks all
>
>
> ----- Original Message -----
> From: "Fran?ois Delawarde" <fdelawarde at wirelessmundi.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>
> Sent: Monday, January 24, 2011 4:18 AM
> Subject: Re: [Freeswitch-users] praise of freeSWITCH developers
>
> > On Mon, 2011-01-24 at 13:14 +1300, Kees Varekamp wrote:
> >> And another 50 usd from NZ :-)
> >
> > Nice! We already have raised $385 total so far on this thread. Remember
> > that if they can afford fancy cocktails and silk underwear, our favorite
> > devs will code faster.
> >
> > Carry on, just a little more to go for the $500 step!
> >
> > Fran?ois.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Tue Jan 25 06:28:10 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 25 Jan 2011 08:58:10 +0530
Subject: [Freeswitch-users] # in prefix
In-Reply-To: <29DBF085-91BA-4110-AE87-CF8F040B6280@freeswitch.org>
References: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
	<2BAF59AA-B828-4AF6-9BD8-F4E2EAB390A8@freeswitch.org>
	<AANLkTim2Fd_t0GWqjip7jjKarvkqxqMM_P=gjVbQS0JZ@mail.gmail.com>
	<29DBF085-91BA-4110-AE87-CF8F040B6280@freeswitch.org>
Message-ID: <AANLkTim3dcKu_PO_ZXeTKViN66wEL-rVuXaME48QPNmP@mail.gmail.com>

Hi,

This is in the dialplan.

http://pastebin.freeswitch.org/15131

Also Brian , how can i ignore 183 without sdp,
what happens is the provider sends 183 without sdp and by applying "<action
application="set" data="sip_ignore_183nosdp=true"/>"  the FS sends 180 to
the customer.
Here i want to block the 183 with SDP just like router as b2bua and send
nothing to customer, and when actual 183 with sdp comes it should send .

Regds
Sam

On Tue, Jan 25, 2011 at 8:45 AM, Brian West <brian at freeswitch.org> wrote:

> We aren't a proxy.. what exactly are you talking bout?  Can you paste us a
> dialplan example?
>
> /b
>
> On Jan 24, 2011, at 9:11 PM, Sam wrote:
>
> > The more info on FS is that it is working in proxy mode , so what ever
> customer sends it sends to the next hop,
> > I will try using %23 for # while sending calls.
> >
> > Regards
> > Sam
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Tue Jan 25 06:58:03 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 24 Jan 2011 22:58:03 -0500
Subject: [Freeswitch-users] IVRD
References: <B890EB1A467549D48DB330F5795906E9@e1705><AANLkTin54y6CJHhSc8DbEZBnx7oBW+4ppD8E3-AQnSk4@mail.gmail.com><A842C399BF77474D86093E7D4FDF2F24@e1705><AANLkTi=CwgFtn3Xv0-9mM5nzsk4hWUpuEDYjeW7dw97=@mail.gmail.com><4CEE1B8B58EE4E6FB13FC4B626A47AE3@e1705>
	<AANLkTikr5_3U6E1WHpsgNExN9a0M2_+papGWgDcoDZ8O@mail.gmail.com>
Message-ID: <EFB4DC78D7CA42749E181E44EE8B5CBF@e1705>

just updated to the last git,
and retried the server2.pl example with debug level 7

so the last lines are the same as FS log after call the extension that calls 
the socket.

[DEBUG] src/esl.c:975 esl_recv_event() RECV HEADER 
[variable_playback_seconds] = [16]
[DEBUG] src/esl.c:975 esl_recv_event() RECV HEADER [variable_playback_ms] = 
[16384]
[DEBUG] src/esl.c:975 esl_recv_event() RECV HEADER 
[variable_playback_samples] = [262144]
[DEBUG] src/esl.c:975 esl_recv_event() RECV HEADER [variable_nibble_account] 
= [9999999999999]
[DEBUG] src/esl.c:975 esl_recv_event() RECV HEADER [variable_nibble_rate] = 
[0]
[DEBUG] src/esl.c:975 esl_recv_event() RECV HEADER 
[variable_current_application_data] = [127.0.0.1:8084 async full]
[DEBUG] src/esl.c:975 esl_recv_event() RECV HEADER 
[variable_current_application] = [socket]
[DEBUG] src/esl.c:975 esl_recv_event() RECV HEADER [variable_socket_host] = 
[127.0.0.1]
[DEBUG] src/esl.c:975 esl_recv_event() RECV HEADER [Content-Type] = 
[command/reply]
[DEBUG] src/esl.c:975 esl_recv_event() RECV HEADER [Reply-Text] = [+OK
]
[DEBUG] src/esl.c:975 esl_recv_event() RECV HEADER [Socket-Mode] = [async]
[DEBUG] src/esl.c:975 esl_recv_event() RECV HEADER [Control] = [full]

and the rest of the server2.pl script is not executed and printed.

Anyway, I think I will use a php socket client with a while loop and kill it
when the call/conference hangup.

Thank you

Franck



----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, January 24, 2011 7:02 PM
Subject: Re: [Freeswitch-users] IVRD


> you may want to go over everything again:
> We use that extensively so it's unlikely there is a problem.
>
> search your dir for any stale ESL.so or .pm files and recopy them all
> from your source tree so they match the version of FS you are on.
>
>
> Did you try the exact test example?
> I can test it on my end tomorrow if you are still stuck.
>
>
>
> On Mon, Jan 24, 2011 at 5:55 PM, Madovsky <infos at madovsky.org> wrote:
>> I compiled ESL from the last source tree of FS I have (git from about 5 
>> days
>> ago)
>> and followed the instructions on wiki ESL Perl
>> ESL stuff are in
>>
>> /usr/local/lib64/perl5/site_perl/5.10.0/x86_64-linux-thread-multi/ESL
>>
>>
>>
>> ----- Original Message -----
>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, January 24, 2011 6:32 PM
>> Subject: Re: [Freeswitch-users] IVRD
>>
>>
>>> did you maybe update FS and not update all the ESL stuff?
>>> Do you have it in a nostandard location?
>>>
>>>
>>> On Mon, Jan 24, 2011 at 5:26 PM, Madovsky <infos at madovsky.org> wrote:
>>>> Apparently the script stalls at
>>>>
>>>> ## Create the connection object which is basically an IVR
>>>> my $con = new ESL::IVR;
>>>>
>>>> I'm looking into IVR.pm to know wha'ts happening
>>>>
>>>> ----- Original Message -----
>>>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Monday, January 24, 2011 3:30 PM
>>>> Subject: Re: [Freeswitch-users] IVRD
>>>>
>>>>
>>>> did you run the fs_ivrd from a shell and look for output in stderr?
>>>> it has to be related to the script executing failing etc.
>>>>
>>>>
>>>> On Mon, Jan 24, 2011 at 11:57 AM, Madovsky <infos at madovsky.org> wrote:
>>>>> Sorry I sent this email by mistake without to finish it.
>>>>> so in my dialplan I have :
>>>>> <extension name="gluglu">
>>>>> <condition field="destination_number"
>>>>> expression="^999$">
>>>>> <action application="set"
>>>>> data="ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl"/>
>>>>> <action application="socket" data="127.0.0.1:9090
>>>>> full"/>
>>>>> <action application="answer"/>
>>>>> <action application="voicemail" data="default
>>>>> ${domain_name} ${dialed_extension}"/>
>>>>> <condition>
>>>>> </extension>
>>>>>
>>>>> and when I call this extension in log level 7 I can see :
>>>>>
>>>>> Dialplan: sofia/internal/9999999999999 at default Action
>>>>> set(ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl)
>>>>> Dialplan: sofia/internal/9999999999999 at default Action
>>>>> socket(127.0.0.1:9090
>>>>> full)
>>>>> Starting ivrd-hello_world.pl...
>>>>>
>>>>> and no ivrd-hello_world.pl code is executed in the dialplan unless the
>>>>> print
>>>>> "Starting ivrd-hello_world.pl";
>>>>>
>>>>> the same if I replace fs_ivrd with server2.pl for example
>>>>>
>>>>> Thanks
>>>>>
>>>>>
>>>>> ----- Original Message -----
>>>>> From: "Madovsky" <infos at madovsky.org>
>>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>> Sent: Monday, January 24, 2011 12:48 PM
>>>>> Subject: Re: [Freeswitch-users] IVRD
>>>>>
>>>>>
>>>>>> Concerning the use of fs_ivrd:
>>>>>>
>>>>>> path is ok
>>>>>> permission is root 755
>>>>>> perl -c gives ../scripts/perl/ivrd-hello_world.pl syntax OK
>>>>>>
>>>>>> in dialplan I have :
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> ----- Original Message -----
>>>>>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>>>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>>> Sent: Monday, January 24, 2011 11:20 AM
>>>>>> Subject: Re: [Freeswitch-users] IVRD
>>>>>>
>>>>>>
>>>>>> check for proper path and execute permissions on the file and perl -c
>>>>>> to make sure it compiles.
>>>>>>
>>>>>>
>>>>>> On Mon, Jan 24, 2011 at 12:13 AM, Madovsky <infos at madovsky.org> 
>>>>>> wrote:
>>>>>>> I tried to use IVRD from wiki example
>>>>>>>
>>>>>>> http://wiki.freeswitch.org/wiki/Ivrd
>>>>>>>
>>>>>>> and server2.pl in ESL directory
>>>>>>> copy and paste in my dialplan ans settings
>>>>>>> so the daemon is running well, but if I attempt
>>>>>>> to call nothing happens unless hangup.
>>>>>>> on the log I can see only
>>>>>>>
>>>>>>> EXECUTE sofia/internal/9999999999999 at default socket(127.0.0.1:8084
>>>>>>> full)
>>>>>>>
>>>>>>> I tried the tests of troubleshooting without error
>>>>>>> I don't understand why the events are not received in the perl 
>>>>>>> script
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II
>>>>>>
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>
>>>>>> AIM: anthm
>>>>>> MSN:anthony_minessale at hotmail.com
>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>
>>>>>> FreeSWITCH Developer Conference
>>>>>> sip:888 at conference.freeswitch.org
>>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>>> pstn:+19193869900
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From infos at madovsky.org  Tue Jan 25 07:33:47 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 24 Jan 2011 23:33:47 -0500
Subject: [Freeswitch-users] IVRD
References: <B890EB1A467549D48DB330F5795906E9@e1705><AANLkTin54y6CJHhSc8DbEZBnx7oBW+4ppD8E3-AQnSk4@mail.gmail.com><A842C399BF77474D86093E7D4FDF2F24@e1705><AANLkTi=CwgFtn3Xv0-9mM5nzsk4hWUpuEDYjeW7dw97=@mail.gmail.com><4CEE1B8B58EE4E6FB13FC4B626A47AE3@e1705>
	<AANLkTikr5_3U6E1WHpsgNExN9a0M2_+papGWgDcoDZ8O@mail.gmail.com>
Message-ID: <C35D701C93804A8FB9B2F87E424BCD44@e1705>

I tried also the ivrd-demo.php with fs_ivrd from this thread

http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg14816.html

and add a mail functio in the script to know if fs_ivrd call the php script 
but it doesn't.

I really don't know what's happen.

thanks

----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, January 24, 2011 7:02 PM
Subject: Re: [Freeswitch-users] IVRD


> you may want to go over everything again:
> We use that extensively so it's unlikely there is a problem.
>
> search your dir for any stale ESL.so or .pm files and recopy them all
> from your source tree so they match the version of FS you are on.
>
>
> Did you try the exact test example?
> I can test it on my end tomorrow if you are still stuck.
>
>
>
> On Mon, Jan 24, 2011 at 5:55 PM, Madovsky <infos at madovsky.org> wrote:
>> I compiled ESL from the last source tree of FS I have (git from about 5 
>> days
>> ago)
>> and followed the instructions on wiki ESL Perl
>> ESL stuff are in
>>
>> /usr/local/lib64/perl5/site_perl/5.10.0/x86_64-linux-thread-multi/ESL
>>
>>
>>
>> ----- Original Message -----
>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, January 24, 2011 6:32 PM
>> Subject: Re: [Freeswitch-users] IVRD
>>
>>
>>> did you maybe update FS and not update all the ESL stuff?
>>> Do you have it in a nostandard location?
>>>
>>>
>>> On Mon, Jan 24, 2011 at 5:26 PM, Madovsky <infos at madovsky.org> wrote:
>>>> Apparently the script stalls at
>>>>
>>>> ## Create the connection object which is basically an IVR
>>>> my $con = new ESL::IVR;
>>>>
>>>> I'm looking into IVR.pm to know wha'ts happening
>>>>
>>>> ----- Original Message -----
>>>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Monday, January 24, 2011 3:30 PM
>>>> Subject: Re: [Freeswitch-users] IVRD
>>>>
>>>>
>>>> did you run the fs_ivrd from a shell and look for output in stderr?
>>>> it has to be related to the script executing failing etc.
>>>>
>>>>
>>>> On Mon, Jan 24, 2011 at 11:57 AM, Madovsky <infos at madovsky.org> wrote:
>>>>> Sorry I sent this email by mistake without to finish it.
>>>>> so in my dialplan I have :
>>>>> <extension name="gluglu">
>>>>> <condition field="destination_number"
>>>>> expression="^999$">
>>>>> <action application="set"
>>>>> data="ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl"/>
>>>>> <action application="socket" data="127.0.0.1:9090
>>>>> full"/>
>>>>> <action application="answer"/>
>>>>> <action application="voicemail" data="default
>>>>> ${domain_name} ${dialed_extension}"/>
>>>>> <condition>
>>>>> </extension>
>>>>>
>>>>> and when I call this extension in log level 7 I can see :
>>>>>
>>>>> Dialplan: sofia/internal/9999999999999 at default Action
>>>>> set(ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl)
>>>>> Dialplan: sofia/internal/9999999999999 at default Action
>>>>> socket(127.0.0.1:9090
>>>>> full)
>>>>> Starting ivrd-hello_world.pl...
>>>>>
>>>>> and no ivrd-hello_world.pl code is executed in the dialplan unless the
>>>>> print
>>>>> "Starting ivrd-hello_world.pl";
>>>>>
>>>>> the same if I replace fs_ivrd with server2.pl for example
>>>>>
>>>>> Thanks
>>>>>
>>>>>
>>>>> ----- Original Message -----
>>>>> From: "Madovsky" <infos at madovsky.org>
>>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>> Sent: Monday, January 24, 2011 12:48 PM
>>>>> Subject: Re: [Freeswitch-users] IVRD
>>>>>
>>>>>
>>>>>> Concerning the use of fs_ivrd:
>>>>>>
>>>>>> path is ok
>>>>>> permission is root 755
>>>>>> perl -c gives ../scripts/perl/ivrd-hello_world.pl syntax OK
>>>>>>
>>>>>> in dialplan I have :
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> ----- Original Message -----
>>>>>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>>>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>>> Sent: Monday, January 24, 2011 11:20 AM
>>>>>> Subject: Re: [Freeswitch-users] IVRD
>>>>>>
>>>>>>
>>>>>> check for proper path and execute permissions on the file and perl -c
>>>>>> to make sure it compiles.
>>>>>>
>>>>>>
>>>>>> On Mon, Jan 24, 2011 at 12:13 AM, Madovsky <infos at madovsky.org> 
>>>>>> wrote:
>>>>>>> I tried to use IVRD from wiki example
>>>>>>>
>>>>>>> http://wiki.freeswitch.org/wiki/Ivrd
>>>>>>>
>>>>>>> and server2.pl in ESL directory
>>>>>>> copy and paste in my dialplan ans settings
>>>>>>> so the daemon is running well, but if I attempt
>>>>>>> to call nothing happens unless hangup.
>>>>>>> on the log I can see only
>>>>>>>
>>>>>>> EXECUTE sofia/internal/9999999999999 at default socket(127.0.0.1:8084
>>>>>>> full)
>>>>>>>
>>>>>>> I tried the tests of troubleshooting without error
>>>>>>> I don't understand why the events are not received in the perl 
>>>>>>> script
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II
>>>>>>
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>
>>>>>> AIM: anthm
>>>>>> MSN:anthony_minessale at hotmail.com
>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>
>>>>>> FreeSWITCH Developer Conference
>>>>>> sip:888 at conference.freeswitch.org
>>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>>> pstn:+19193869900
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From u2nsam at gmail.com  Tue Jan 25 07:45:00 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 25 Jan 2011 10:15:00 +0530
Subject: [Freeswitch-users] # in prefix
In-Reply-To: <AANLkTim3dcKu_PO_ZXeTKViN66wEL-rVuXaME48QPNmP@mail.gmail.com>
References: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
	<2BAF59AA-B828-4AF6-9BD8-F4E2EAB390A8@freeswitch.org>
	<AANLkTim2Fd_t0GWqjip7jjKarvkqxqMM_P=gjVbQS0JZ@mail.gmail.com>
	<29DBF085-91BA-4110-AE87-CF8F040B6280@freeswitch.org>
	<AANLkTim3dcKu_PO_ZXeTKViN66wEL-rVuXaME48QPNmP@mail.gmail.com>
Message-ID: <AANLkTimkBj87jW789gVdFqSZ+90Fyo-UM=gS31=CgYYA@mail.gmail.com>

Here,
When i do %23 the call is passing with 999%2312127773456 to the provider its
not passing 999#12127773456

Regds
Sam

On Tue, Jan 25, 2011 at 8:58 AM, Sam <u2nsam at gmail.com> wrote:

> Hi,
>
> This is in the dialplan.
>
> http://pastebin.freeswitch.org/15131
>
> Also Brian , how can i ignore 183 without sdp,
> what happens is the provider sends 183 without sdp and by applying "<action
> application="set" data="sip_ignore_183nosdp=true"/>"  the FS sends 180 to
> the customer.
> Here i want to block the 183 with SDP just like router as b2bua and send
> nothing to customer, and when actual 183 with sdp comes it should send .
>
> Regds
> Sam
>
>
> On Tue, Jan 25, 2011 at 8:45 AM, Brian West <brian at freeswitch.org> wrote:
>
>> We aren't a proxy.. what exactly are you talking bout?  Can you paste us a
>> dialplan example?
>>
>> /b
>>
>> On Jan 24, 2011, at 9:11 PM, Sam wrote:
>>
>> > The more info on FS is that it is working in proxy mode , so what ever
>> customer sends it sends to the next hop,
>> > I will try using %23 for # while sending calls.
>> >
>> > Regards
>> > Sam
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From u2nsam at gmail.com  Tue Jan 25 08:17:42 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 25 Jan 2011 10:47:42 +0530
Subject: [Freeswitch-users] # in prefix
In-Reply-To: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
References: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
Message-ID: <AANLkTimRp=oBadOA3n_b+qvNebRgukzMys-XrEuwduy2@mail.gmail.com>

Hi,

What is the significance of '#' on freeswitch ? as it strips all the numbers
before # sign.

Regds


On Mon, Jan 24, 2011 at 10:18 PM, Sam <u2nsam at gmail.com> wrote:

> Hello,
>
> As i send call with prefix 999# the prefix is not passed to the provider
> from FS.
>
> Customer sends 999#12127773456 ---> FS ---> 12127773456 ,
> it goes without prefix to the provider because of '#', is there any method
> to send to the provider 999#12127773456 .
>
> Regds
> Sam
>
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From infos at madovsky.org  Tue Jan 25 08:28:56 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 25 Jan 2011 00:28:56 -0500
Subject: [Freeswitch-users] ESL php client
Message-ID: <0B91AC9D8F024867B984B3DC4FCE9E06@e1705>

no luck from libs/esl
make phpmod

make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/home/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/home/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C php
make[1]: Entering directory `/home/src/freeswitch/libs/esl/php'
g++  -I/home/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable -I/usr/local/include/php -I/usr/local/include/php/main -I/usr/local/include/php/TSRM -I/usr/local/include/php/Zend -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o
cc1plus: warnings being treated as errors
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
make[1]: *** [esl_wrap.o] Error 1
make[1]: Leaving directory `/home/src/freeswitch/libs/esl/php'
make: *** [phpmod] Error 2

PHP 4.4.9 is instlalled on my server

Thanks
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From lakindia89 at gmail.com  Tue Jan 25 10:02:24 2011
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Tue, 25 Jan 2011 12:32:24 +0530
Subject: [Freeswitch-users] Help regard to park_timeout
In-Reply-To: <AANLkTi=GvaDh7EsjJmsV-UzDw6LH=99KVgYjg8HxJ-Ek@mail.gmail.com>
References: <AANLkTi=edjFv90+q3G2kkzoJ76kWgx+NBe=Qx3CYA2Jd@mail.gmail.com>
	<AANLkTinHSCfH6Ydh5ipGfhMQbpcHc+uJd6oxGhJNNFPy@mail.gmail.com>
	<AANLkTi=GvaDh7EsjJmsV-UzDw6LH=99KVgYjg8HxJ-Ek@mail.gmail.com>
Message-ID: <AANLkTimSoL86hyWa94L8WGTynS37QC=yMyeq18MX8hZn@mail.gmail.com>

The output of a uuid_dump is here:

> uuid_dump ff64db72-284d-11e0-abd4-772c8caa0462

Event-Name: CHANNEL_DATA
Core-UUID: 05595dbc-27b0-11e0-ab9f-772c8caa0462
FreeSWITCH-Hostname: FMS-Demo
FreeSWITCH-IPv4: 192.168.1.72
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2011-01-25%2012%3A10%3A49
Event-Date-GMT: Tue,%2025%20Jan%202011%2006%3A40%3A49%20GMT
Event-Date-Timestamp: 1295937649908745
Event-Calling-File: mod_commands.c
Event-Calling-Function: uuid_dump_function
Event-Calling-Line-Number: 4130
Channel-State: CS_EXECUTE
Channel-Call-State: ACTIVE
Channel-State-Number: 4
Channel-Name: FreeTDM/1%3A1/9952248266
Unique-ID: ff64db72-284d-11e0-abd4-772c8caa0462
Call-Direction: outbound
Presence-Call-Direction: outbound
Channel-Call-UUID: ff64cfc4-284d-11e0-abd3-772c8caa0462
Answer-State: answered
Channel-Read-Codec-Name: PCMA
Channel-Read-Codec-Rate: 8000
Channel-Read-Codec-Bit-Rate: 64000
Channel-Write-Codec-Name: PCMA
Channel-Write-Codec-Rate: 8000
Channel-Write-Codec-Bit-Rate: 64000
Caller-Direction: outbound
Caller-Destination-Number: 9952248266
Caller-Unique-ID: ff64db72-284d-11e0-abd4-772c8caa0462
Caller-Source: src/switch_ivr_originate.c
Caller-Context: default
Caller-Channel-Name: FreeTDM/1%3A1/9952248266
Caller-Profile-Index: 1
Caller-Profile-Created-Time: 1295937628350711
Caller-Channel-Created-Time: 1295937628350711
Caller-Channel-Answered-Time: 1295937635463788
Caller-Channel-Progress-Time: 0
Caller-Channel-Progress-Media-Time: 1295937632566721
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0
Caller-Screen-Bit: true
Caller-Privacy-Hide-Name: false
Caller-Privacy-Hide-Number: false
variable_direction: outbound
variable_uuid: ff64db72-284d-11e0-abd4-772c8caa0462
variable_read_codec: PCMA
variable_read_rate: 8000
variable_write_codec: PCMA
variable_write_rate: 8000
variable_channel_name: FreeTDM/1%3A1/9952248266
variable_freetdm_span_name: wp1
variable_freetdm_span_number: 1
variable_freetdm_chan_number: 1
variable_is_outbound: true
variable_call_uuid: ff64cfc4-284d-11e0-abd3-772c8caa0462
variable_ignore_early_media: true
variable_api_hangup_hook: perl%20/root/a.pl
variable_exec_after_bridge_app: park
variable_originate_early_media: false
variable_endpoint_disposition: ANSWER
variable_current_application: park

> eval uuid:ff64db72-284d-11e0-abd4-772c8caa0462
${variable_exec_after_bridge_app}
park

> eval uuid:ff64db72-284d-11e0-abd4-772c8caa0462 ${variable_park_timeout}
-ERR no reply

When I looked into the source I found that, the variable park_timeout is
being set to NULL, once it is read in switch_ivr.c under switch_ivr_park()
function. So I think that's why uuid_getvar returns __undef__.

After seeing this, I experimented as follows.
I made 2 calls.
CLI> originate
{ignore_early_media=true,park_timeout=50,api_hangup_hook='perl
/root/a.pl',exec_after_bridge_app=park}freetdm/grp1/a/9952248266
&park()
CLI > originate
{ignore_early_media=true,park_timeout=50,api_hangup_hook='perl
/root/a.pl'}freetdm/grp1/a/9843171457
&park()

Then I bridged them using uuid_bridge.
CLI> uuid_bridge 6e53bbae-277f-11e0-8580-8390cbcc860f
75126f12-277f-11e0-8582-8390cbcc860f
Now:
uuid_getvar 6e53bbae-277f-11e0-8580-8390cbcc860f park_timeout ( return
__undef__. So I set the park_timeout from the CLI explicitly ).

CLI> uuid_setvar 6e53bbae-277f-11e0-8580-8390cbcc860f park_timeout 20
CLI> uuid_getvar 6e53bbae-277f-11e0-8580-8390cbcc860f park_timeout ( returns
20 ).

Now I hangup the call with 75126f12-277f-11e0-8582-8390cbcc860f UUID.

After that, 6e53bbae-277f-11e0-8580-8390cbcc860f call goes in to park, and
after 20 seconds the call got ended, and it works as I expected.

Now I need to know, whether any specific purpose is there to set the
park_timeout to NULL once it is read?  The variable park_after_bridge is
also set to NULL? I'm unable to find reasons for this. Someone please light
me on this!.

Thanks for your response.




On Tue, Jan 25, 2011 at 1:46 AM, Michael Collins <msc at freeswitch.org> wrote:

> Do a uuid_dump on the channel and see what all is there. Also, try doing
> this:
> eval uuid:9b59d172-2781-11e0-8586-8390cbcc860f ${park_timeout}
>
> See what happens.
>
> -MC
>
> On Sun, Jan 23, 2011 at 10:21 PM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> Hi all,
>> I've done a further experimentation and I need a clarification.
>>
>> From CLI>
>> originate {ignore_early_media=true,park_timeout=50,api_hangup_hook='perl
>> /root/a.pl',exec_after_bridge_app=park}freetdm/grp1/a/9952248266 &park()
>>
>> Then I executed the following commands from CLI.
>> uuid_getvar 9b59d172-2781-11e0-8586-8390cbcc860f park_timeout
>> _undef_
>> uuid_getvar 9b59d172-2781-11e0-8586-8390cbcc860f api_hangup_hook
>> perl /root/a.pl
>>
>> I don't know why uuid_getvar, returns undef for park_timeout variable. But
>> the call is hangup once 50 seconds is reached. Can some one pls explain what
>> it is printing as __undef__
>>
>>
>>
>> On Fri, Jan 21, 2011 at 6:37 PM, lakshmanan ganapathy <
>> lakindia89 at gmail.com> wrote:
>>
>>> Dear all,
>>> I was using park_timeout and I come across the following scenario which I
>>> felt something is missing.
>>> I've originated a call as follows.
>>>
>>> originate
>>> {ignore_early_media=true,exec_after_bridge_app=park,park_timeout=60,api_hangup_hook='perl
>>> /root/a.pl'}freetdm/grp1/a/9952248266 &park()
>>>
>>> Once the call is answered I originated another call.
>>> originate {ignore_early_media=true,park_timeout=60,api_hangup_hook='perl
>>> /root/a.pl'}freetdm/grp1/a/9843171457 &park()
>>>
>>> Once this call is also answered, I said "uuid_bridge <uuid1> <uuid2>".
>>> Both call gets bridged. After some time, I hangup the second call
>>> (9843171457). Now the first call goes into park().
>>>
>>> I expect that the first call will hangup after 60 seconds, but it didn't.
>>>
>>> The freeswitch log is here
>>> http://pastebin.freeswitch.org/15099
>>>
>>> When I start to use the park_timeout, I thought once a leg is in park,
>>> then the timer will start, and once it is unparked for various reason the
>>> timer will be reseted. After sometime, when the leg again comes in park, the
>>> timer will start. Is this correct?
>>>
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Tue Jan 25 10:24:41 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 25 Jan 2011 08:24:41 +0100
Subject: [Freeswitch-users] # in prefix
In-Reply-To: <AANLkTim3dcKu_PO_ZXeTKViN66wEL-rVuXaME48QPNmP@mail.gmail.com>
References: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
	<2BAF59AA-B828-4AF6-9BD8-F4E2EAB390A8@freeswitch.org>
	<AANLkTim2Fd_t0GWqjip7jjKarvkqxqMM_P=gjVbQS0JZ@mail.gmail.com>
	<29DBF085-91BA-4110-AE87-CF8F040B6280@freeswitch.org>
	<AANLkTim3dcKu_PO_ZXeTKViN66wEL-rVuXaME48QPNmP@mail.gmail.com>
Message-ID: <0337AF2E-D3BD-4D77-B7C7-5993A6B9EAF4@ipeva.fr>

Your regexp is (\d+)$.
AFAIK, \d does not match #.
So your regexp does not match anything up to the #.

Your regexp should be:

^(\d+#\d+)$ (or something like that, I never tried to match #, so there could be a special trick for that).

Or if you want to be less specific:

^(.*)$

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 25/01/2011 ? 04:28, Sam a ?crit :

> Hi,
> 
> This is in the dialplan.
> 
> http://pastebin.freeswitch.org/15131
> 
> Also Brian , how can i ignore 183 without sdp,
> what happens is the provider sends 183 without sdp and by applying "<action application="set" data="sip_ignore_183nosdp=true"/>"  the FS sends 180 to the customer.
> Here i want to block the 183 with SDP just like router as b2bua and send nothing to customer, and when actual 183 with sdp comes it should send .
> 
> Regds
> Sam
> 
> On Tue, Jan 25, 2011 at 8:45 AM, Brian West <brian at freeswitch.org> wrote:
> We aren't a proxy.. what exactly are you talking bout?  Can you paste us a dialplan example?
> 
> /b
> 
> On Jan 24, 2011, at 9:11 PM, Sam wrote:
> 
> > The more info on FS is that it is working in proxy mode , so what ever customer sends it sends to the next hop,
> > I will try using %23 for # while sending calls.
> >
> > Regards
> > Sam
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From bernhard.suttner at winet.ch  Tue Jan 25 10:45:07 2011
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Tue, 25 Jan 2011 08:45:07 +0100
Subject: [Freeswitch-users] Outgoing registrations within a fs pool
In-Reply-To: <AANLkTikQCgc=g3ThyKFsnCK280N+unJjEQPZmOzg43oh@mail.gmail.com>
References: <20110125001130.74166c5a@mail.winet.ch>
	<AANLkTikQCgc=g3ThyKFsnCK280N+unJjEQPZmOzg43oh@mail.gmail.com>
Message-ID: <e6dc88a8-2744-4fbe-84dc-e52255e0de61@winet.ch>

Hi,

ok. I understand. It's maybe a feature request to have "shared outgoing registrations".

Best regards,
Bernhard

-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Steven Ayre
Gesendet: Dienstag, 25. Januar 2011 00:50
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Outgoing registrations within a fs pool

They will all register. They have to because they're all different
endpoints.

-Steve


On 24 January 2011 23:11, Bernhard Suttner <bernhard.suttner at winet.ch>wrote:

> Hi,
>
> thanks for your answer. If there are 5 freeswitch servers within the pool
> (shared db) I would add the gateway to all of the freeswitch server. Will
> all of them try to register or would the first freeswitch server register
> and the other servers would check if the (re)-registration was already done?
>
> The point is, that all the freeswitch servers do have the same
> configuration and it does not matter which freeswitch server registers
> because a incoming call would be handled the same way on all of the servers
> with shared database.
>
> Best regards,
> Bernhard
>
> ----- Original Message -----
> From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
> To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
> Sent: Mon, 24 Jan 2011 23:31:48 +0100
> Subject: Re: [Freeswitch-users] Outgoing registrations within a fs pool
>
>
> > I think gateways are all in-memory so it should be ok
> >
> > On Mon, Jan 24, 2011 at 2:58 PM, Bernhard Suttner
> > <bernhard.suttner at winet.ch> wrote:
> > > Hi,
> > >
> > > does someone has an idea what happens if I would add the gateway (with
> > registration) to all the freeswitch servers of the pool with shared
> > database?
> > >
> > > Best regards,
> > > Bernhard
> > >
> > > ----- Original Message -----
> > > From: Bernhard Suttner [mailto:bernhard.suttner at winet.ch]
> > > To: 'FreeSWITCH Users Help' [mailto:
> freeswitch-users at lists.freeswitch.org]
> > > Sent: Fri, 14 Jan 2011 10:42:10 +0100
> > > Subject: Re: [Freeswitch-users] Outgoing registrations within a fs pool
> > >
> > >
> > >> Hi,
> > >>
> > >> Kamailio is able to do a registration with the uacreg module, but it
> is
> > not
> > >> as stable as on FreeSWITCH (its not the main task of Kamailio to
> > implement
> > >> B2BUA features). Therefore I asked if FreeSWITCH is able to register
> > "from a
> > >> pool of FS servers".
> > >>
> > >> Best regards,
> > >> Bernhard
> > >>
> > >> -----Urspr?ngliche Nachricht-----
> > >> Von: freeswitch-users-bounces at lists.freeswitch.org
> > >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
> > Paul
> > >> Cupis
> > >> Gesendet: Donnerstag, 13. Januar 2011 21:59
> > >> An: FreeSWITCH Users Help
> > >> Betreff: Re: [Freeswitch-users] Outgoing registrations within a fs
> pool
> > >>
> > >> Perhaps look at getting the kamailio server to register to the
> provider
> > >> rather than one of the FS from the pool?
> > >>
> > >>
> > >> _______________________________________________
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> http://www.freeswitch.org
> > >>
> > >>
> > >>
> > >> _______________________________________________
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> http://www.freeswitch.org
> > >>
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From steveayre at gmail.com  Tue Jan 25 11:42:08 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 25 Jan 2011 08:42:08 +0000
Subject: [Freeswitch-users] Outgoing registrations within a fs pool
In-Reply-To: <e6dc88a8-2744-4fbe-84dc-e52255e0de61@winet.ch>
References: <20110125001130.74166c5a@mail.winet.ch>
	<AANLkTikQCgc=g3ThyKFsnCK280N+unJjEQPZmOzg43oh@mail.gmail.com>
	<e6dc88a8-2744-4fbe-84dc-e52255e0de61@winet.ch>
Message-ID: <A820A9F6-1776-4B1C-8B17-BAC7A4AD4306@gmail.com>

You misunderstand. Each one is registered from a different ip and port, so a registration from one isn't valid for another.

Steve on iPhone

On 25 Jan 2011, at 07:45, "Bernhard Suttner" <bernhard.suttner at winet.ch> wrote:

> Hi,
> 
> ok. I understand. It's maybe a feature request to have "shared outgoing registrations".
> 
> Best regards,
> Bernhard
> 
> -----Urspr?ngliche Nachricht-----
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Steven Ayre
> Gesendet: Dienstag, 25. Januar 2011 00:50
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] Outgoing registrations within a fs pool
> 
> They will all register. They have to because they're all different
> endpoints.
> 
> -Steve
> 
> 
> On 24 January 2011 23:11, Bernhard Suttner <bernhard.suttner at winet.ch>wrote:
> 
>> Hi,
>> 
>> thanks for your answer. If there are 5 freeswitch servers within the pool
>> (shared db) I would add the gateway to all of the freeswitch server. Will
>> all of them try to register or would the first freeswitch server register
>> and the other servers would check if the (re)-registration was already done?
>> 
>> The point is, that all the freeswitch servers do have the same
>> configuration and it does not matter which freeswitch server registers
>> because a incoming call would be handled the same way on all of the servers
>> with shared database.
>> 
>> Best regards,
>> Bernhard
>> 
>> ----- Original Message -----
>> From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
>> To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
>> Sent: Mon, 24 Jan 2011 23:31:48 +0100
>> Subject: Re: [Freeswitch-users] Outgoing registrations within a fs pool
>> 
>> 
>>> I think gateways are all in-memory so it should be ok
>>> 
>>> On Mon, Jan 24, 2011 at 2:58 PM, Bernhard Suttner
>>> <bernhard.suttner at winet.ch> wrote:
>>>> Hi,
>>>> 
>>>> does someone has an idea what happens if I would add the gateway (with
>>> registration) to all the freeswitch servers of the pool with shared
>>> database?
>>>> 
>>>> Best regards,
>>>> Bernhard
>>>> 
>>>> ----- Original Message -----
>>>> From: Bernhard Suttner [mailto:bernhard.suttner at winet.ch]
>>>> To: 'FreeSWITCH Users Help' [mailto:
>> freeswitch-users at lists.freeswitch.org]
>>>> Sent: Fri, 14 Jan 2011 10:42:10 +0100
>>>> Subject: Re: [Freeswitch-users] Outgoing registrations within a fs pool
>>>> 
>>>> 
>>>>> Hi,
>>>>> 
>>>>> Kamailio is able to do a registration with the uacreg module, but it
>> is
>>> not
>>>>> as stable as on FreeSWITCH (its not the main task of Kamailio to
>>> implement
>>>>> B2BUA features). Therefore I asked if FreeSWITCH is able to register
>>> "from a
>>>>> pool of FS servers".
>>>>> 
>>>>> Best regards,
>>>>> Bernhard
>>>>> 
>>>>> -----Urspr?ngliche Nachricht-----
>>>>> Von: freeswitch-users-bounces at lists.freeswitch.org
>>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
>>> Paul
>>>>> Cupis
>>>>> Gesendet: Donnerstag, 13. Januar 2011 21:59
>>>>> An: FreeSWITCH Users Help
>>>>> Betreff: Re: [Freeswitch-users] Outgoing registrations within a fs
>> pool
>>>>> 
>>>>> Perhaps look at getting the kamailio server to register to the
>> provider
>>>>> rather than one of the FS from the pool?
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>> 
>>> 
>>> 
>>> --
>>> Anthony Minessale II
>>> 
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>> 
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>> 
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From steveayre at gmail.com  Tue Jan 25 11:46:01 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 25 Jan 2011 08:46:01 +0000
Subject: [Freeswitch-users] # in prefix
In-Reply-To: <0337AF2E-D3BD-4D77-B7C7-5993A6B9EAF4@ipeva.fr>
References: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
	<2BAF59AA-B828-4AF6-9BD8-F4E2EAB390A8@freeswitch.org>
	<AANLkTim2Fd_t0GWqjip7jjKarvkqxqMM_P=gjVbQS0JZ@mail.gmail.com>
	<29DBF085-91BA-4110-AE87-CF8F040B6280@freeswitch.org>
	<AANLkTim3dcKu_PO_ZXeTKViN66wEL-rVuXaME48QPNmP@mail.gmail.com>
	<0337AF2E-D3BD-4D77-B7C7-5993A6B9EAF4@ipeva.fr>
Message-ID: <A67787E6-9206-4E46-BAF4-F9DCD56D6582@gmail.com>

David, you're correct. \d means digits and only digits (0-9).

I don't think # requires escaping, try that first and if it doesn't work try \#.

Steve on iPhone

On 25 Jan 2011, at 07:24, David Ponzone <david.ponzone at ipeva.fr> wrote:

> Your regexp is (\d+)$.
> AFAIK, \d does not match #.
> So your regexp does not match anything up to the #.
> 
> Your regexp should be:
> 
> ^(\d+#\d+)$ (or something like that, I never tried to match #, so there could be a special trick for that).
> 
> Or if you want to be less specific:
> 
> ^(.*)$
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 25/01/2011 ? 04:28, Sam a ?crit :
> 
>> Hi,
>> 
>> This is in the dialplan.
>> 
>> http://pastebin.freeswitch.org/15131
>> 
>> Also Brian , how can i ignore 183 without sdp,
>> what happens is the provider sends 183 without sdp and by applying "<action application="set" data="sip_ignore_183nosdp=true"/>" the FS sends 180 to the customer.
>> Here i want to block the 183 with SDP just like router as b2bua and send nothing to customer, and when actual 183 with sdp comes it should send .
>> 
>> Regds
>> Sam
>> 
>> On Tue, Jan 25, 2011 at 8:45 AM, Brian West <brian at freeswitch.org> wrote:
>> We aren't a proxy.. what exactly are you talking bout?  Can you paste us a dialplan example?
>> 
>> /b
>> 
>> On Jan 24, 2011, at 9:11 PM, Sam wrote:
>> 
>> > The more info on FS is that it is working in proxy mode , so what ever customer sends it sends to the next hop,
>> > I will try using %23 for # while sending calls.
>> >
>> > Regards
>> > Sam
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From u2nsam at gmail.com  Tue Jan 25 11:47:31 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 25 Jan 2011 14:17:31 +0530
Subject: [Freeswitch-users] deflect
In-Reply-To: <AANLkTim6h42+8ZDmFL00dYMNOmAz-3JtG43q+2oMMZ36@mail.gmail.com>
References: <AANLkTiksYLOeki2_XDJC5YwMW6hjUxvdpT7-Mvg7vmKZ@mail.gmail.com>
	<AANLkTi=EMn55D1xGBrWuGgVE_dW_5itoJ4WpzByyagTE@mail.gmail.com>
	<AANLkTi=F5Lhsk1tjcG5z5LTUjZb4e-cQdNXQjK3OsVD7@mail.gmail.com>
	<AANLkTik9j3-o7GGr0=uudVgxLwnmcdTwC8ue8bVrv264@mail.gmail.com>
	<AANLkTim6h42+8ZDmFL00dYMNOmAz-3JtG43q+2oMMZ36@mail.gmail.com>
Message-ID: <AANLkTimLF7GWShxzoPii9Njeci_3KZQZUZMZx=7Y7n=p@mail.gmail.com>

Is it possible to pass Headers on refer message by any means, just like we
can pass headers on invite,
i have done <action application="set" data="sip_h_X-Code=${digits}"/> which
passed successfully on invite but not on refer is there any method to pass
header someting like refer to and refer by, can we have some more here .

Regards
Sam



On Mon, Jan 24, 2011 at 4:38 PM, Steven Ayre <steveayre at gmail.com> wrote:

> No, bridge application.
> <action application="bridge" data="{api_on_answer='uuid_transfer  ${uuid}
> both'}sofia/external/9999 at 192.168.2.130"/>
> The call will be setup going through FS, then when it's answered FS will
> try to do a reinvite to remove itself from the call path at which point both
> signalling and media will not go through FS. It'll only work if both
> endpoints can see each other though.
>
> Perhaps I misunderstood what you're trying to do though?
>
> The deflect app sends a REFER request on the a-leg for an answered call
> that tells the caller to redirect to another server. FS won't be in the call
> path for signalling or media after the redirect. There's no bleg in this
> scenario.
> <action application="deflect" data="sip:9999 at 192.168.2.130<sip%3A9999 at 192.168.2.130>
> "/>
>
> So for example you can answer the call, do IVR, then redirect the caller to
> an extension on another server without the call going through FS.
>
> -Steve
>
>
>
>
> On 24 January 2011 10:01, Sam <u2nsam at gmail.com> wrote:
>
>> So by this ,
>>
>> <action application="deflect" data="{api_on_answer='uuid_transfer  ${uuid}
>> both'}sofia/external/9999 at 192.168.2.130"/>
>>
>> Will transfer both legs of call to 192.168.2.130  ?
>>
>> Regards
>> Sam
>>
>>
>>
>>
>>
>> On Mon, Jan 24, 2011 at 3:12 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> You could try uuid_simplify with the api_on_answer hook
>>>
>>> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_simplify
>>> http://wiki.freeswitch.org/wiki/Variable_api_on_answer
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 24 January 2011 09:05, Sam <u2nsam at gmail.com> wrote:
>>>
>>>>  Hi,
>>>>
>>>> Is it possible by having b2bua in between , would the leg A be deflected
>>>> to the another FS server from first server ?
>>>>
>>>> Regds
>>>> Sam
>>>>
>>>>
>>>> On Wed, Jan 12, 2011 at 11:42 AM, Sam <u2nsam at gmail.com> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> When call comes on 1 server and plays an application and after
>>>>> execution of the
>>>>> application the call is bridge to the other server ,but here after
>>>>> bridging the call
>>>>> should refer/deflect to other server, how this can be done ?
>>>>>
>>>>> Here just using the deflect variable is not recommended as there is
>>>>> proxy in between,
>>>>> so once the call is bridge the next step would be deflect the leg
>>>>> totally to another server via proxy.
>>>>>
>>>>> Regards
>>>>> Sam
>>>>>
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From erik.dekkers at wvds.nl  Tue Jan 25 11:57:51 2011
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Tue, 25 Jan 2011 09:57:51 +0100
Subject: [Freeswitch-users] Javascript IVR session question
In-Reply-To: <AANLkTimaJDm8Fc1EKk1hKrnF60cWBXyg4bzdQ+77NMG=@mail.gmail.com>
References: <CDF5E00590C9BA4AB96BE7CDD28C86AE07872CFA48@FS1.wvds.local>
	<AANLkTimaJDm8Fc1EKk1hKrnF60cWBXyg4bzdQ+77NMG=@mail.gmail.com>
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE0787301809@FS1.wvds.local>

Michael,

It's like this. I would like to let the customer records a message. Then a technician should get called and the previous recorded message should be played.
So after the customer has records his message, the line should be hung up. Then a new call should be made from the IVR to the technician.

Regards,

Erik

Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Michael Collins
Verzonden: maandag 24 januari 2011 21:14
Aan: FreeSWITCH Users Help
Onderwerp: Re: [Freeswitch-users] Javascript IVR session question

If I read this correctly you are hanging up the channel that you later wish to bridge to user/202. Why do you need to hangup? Perhaps you could describe a little more about the application? I'm sure we can help you iron out the details.

-MC
On Sun, Jan 23, 2011 at 12:48 PM, Erik Dekkers <erik.dekkers at wvds.nl<mailto:erik.dekkers at wvds.nl>> wrote:
Hey ppl,

At the moment im building a Javascript based IVR but im kind of stuck on a part.

The IVR should do this:

- Answer session (working)
- Play some wav files (working)
- Record a message to file (working)
- Hang up the first session (working)
- Call the second session (not working)
- Play the previous recorded file (not working)

After I dial the second session, the console says "channel is hungup already". How should i do this?

Kind regards,

Erik Dekkers (wvds-nl on IRC)



my script:

var allDigits = "";
function on_dtmf(session, type, digits, arg)
{
    if (digits.digit == "#") {
        return allDigits;
    }
    if (digits.digit == "*") {
        return false; //stop the recording.
    }
    console_log("digit: " + digits.digit + "\n");
    allDigits += digits.digit;
    return(allDigits);
}
session.answer();
if (session.ready()) {
    allDigits = "";
    var rtn;
    rtn = session.streamFile("/home/edekkers/sounds/10_spreek_in.wav", on_dtmf, "");
    if (session.ready()) {
        var tmp_Filename = "/tmp/test.wav";
        if (session.ready()) {
            rtn = session.recordFile(tmp_Filename, on_dtmf, "", 120);
            }
        rtn = session.streamFile("/home/edekkers/sounds/11_bericht_is_ontvangen.wav", on_dtmf, "");
        if (session.ready()) {
            session.hangup();
        }
    }
}
session.execute("bridge","user/202")
if (session.ready()) {
    session.streamFile("/tmp/test.wav");
}



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From bernhard.suttner at winet.ch  Tue Jan 25 12:07:23 2011
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Tue, 25 Jan 2011 10:07:23 +0100
Subject: [Freeswitch-users] Outgoing registrations within a fs pool
In-Reply-To: <A820A9F6-1776-4B1C-8B17-BAC7A4AD4306@gmail.com>
References: <20110125001130.74166c5a@mail.winet.ch><AANLkTikQCgc=g3ThyKFsnCK280N+unJjEQPZmOzg43oh@mail.gmail.com><e6dc88a8-2744-4fbe-84dc-e52255e0de61@winet.ch>
	<A820A9F6-1776-4B1C-8B17-BAC7A4AD4306@gmail.com>
Message-ID: <43c4993f-c2aa-47d2-bf31-8a278cc1b6d7@winet.ch>

Hi,

I see. Thanks a lot.

Best regards,
Bernhard

-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Steven Ayre
Gesendet: Dienstag, 25. Januar 2011 09:42
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Outgoing registrations within a fs pool

You misunderstand. Each one is registered from a different ip and port, so a registration from one isn't valid for another.

Steve on iPhone

On 25 Jan 2011, at 07:45, "Bernhard Suttner" <bernhard.suttner at winet.ch> wrote:

> Hi,
> 
> ok. I understand. It's maybe a feature request to have "shared outgoing registrations".
> 
> Best regards,
> Bernhard
> 
> -----Urspr?ngliche Nachricht-----
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Steven Ayre
> Gesendet: Dienstag, 25. Januar 2011 00:50
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] Outgoing registrations within a fs pool
> 
> They will all register. They have to because they're all different
> endpoints.
> 
> -Steve
> 
> 
> On 24 January 2011 23:11, Bernhard Suttner <bernhard.suttner at winet.ch>wrote:
> 
>> Hi,
>> 
>> thanks for your answer. If there are 5 freeswitch servers within the pool
>> (shared db) I would add the gateway to all of the freeswitch server. Will
>> all of them try to register or would the first freeswitch server register
>> and the other servers would check if the (re)-registration was already done?
>> 
>> The point is, that all the freeswitch servers do have the same
>> configuration and it does not matter which freeswitch server registers
>> because a incoming call would be handled the same way on all of the servers
>> with shared database.
>> 
>> Best regards,
>> Bernhard
>> 
>> ----- Original Message -----
>> From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
>> To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
>> Sent: Mon, 24 Jan 2011 23:31:48 +0100
>> Subject: Re: [Freeswitch-users] Outgoing registrations within a fs pool
>> 
>> 
>>> I think gateways are all in-memory so it should be ok
>>> 
>>> On Mon, Jan 24, 2011 at 2:58 PM, Bernhard Suttner
>>> <bernhard.suttner at winet.ch> wrote:
>>>> Hi,
>>>> 
>>>> does someone has an idea what happens if I would add the gateway (with
>>> registration) to all the freeswitch servers of the pool with shared
>>> database?
>>>> 
>>>> Best regards,
>>>> Bernhard
>>>> 
>>>> ----- Original Message -----
>>>> From: Bernhard Suttner [mailto:bernhard.suttner at winet.ch]
>>>> To: 'FreeSWITCH Users Help' [mailto:
>> freeswitch-users at lists.freeswitch.org]
>>>> Sent: Fri, 14 Jan 2011 10:42:10 +0100
>>>> Subject: Re: [Freeswitch-users] Outgoing registrations within a fs pool
>>>> 
>>>> 
>>>>> Hi,
>>>>> 
>>>>> Kamailio is able to do a registration with the uacreg module, but it
>> is
>>> not
>>>>> as stable as on FreeSWITCH (its not the main task of Kamailio to
>>> implement
>>>>> B2BUA features). Therefore I asked if FreeSWITCH is able to register
>>> "from a
>>>>> pool of FS servers".
>>>>> 
>>>>> Best regards,
>>>>> Bernhard
>>>>> 
>>>>> -----Urspr?ngliche Nachricht-----
>>>>> Von: freeswitch-users-bounces at lists.freeswitch.org
>>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
>>> Paul
>>>>> Cupis
>>>>> Gesendet: Donnerstag, 13. Januar 2011 21:59
>>>>> An: FreeSWITCH Users Help
>>>>> Betreff: Re: [Freeswitch-users] Outgoing registrations within a fs
>> pool
>>>>> 
>>>>> Perhaps look at getting the kamailio server to register to the
>> provider
>>>>> rather than one of the FS from the pool?
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>> 
>>> 
>>> 
>>> --
>>> Anthony Minessale II
>>> 
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>> 
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>> 
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

_______________________________________________
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From u2nsam at gmail.com  Tue Jan 25 12:26:58 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 25 Jan 2011 14:56:58 +0530
Subject: [Freeswitch-users] # in prefix
In-Reply-To: <0337AF2E-D3BD-4D77-B7C7-5993A6B9EAF4@ipeva.fr>
References: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
	<2BAF59AA-B828-4AF6-9BD8-F4E2EAB390A8@freeswitch.org>
	<AANLkTim2Fd_t0GWqjip7jjKarvkqxqMM_P=gjVbQS0JZ@mail.gmail.com>
	<29DBF085-91BA-4110-AE87-CF8F040B6280@freeswitch.org>
	<AANLkTim3dcKu_PO_ZXeTKViN66wEL-rVuXaME48QPNmP@mail.gmail.com>
	<0337AF2E-D3BD-4D77-B7C7-5993A6B9EAF4@ipeva.fr>
Message-ID: <AANLkTimhfx7RP67h-z9J7g7t0r5Orsk_EB7LuXA9K4bG@mail.gmail.com>

Thnx all,

I did ^(.*)$

Regds
Sam

On Tue, Jan 25, 2011 at 12:54 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Your regexp is (\d+)$.
> AFAIK, \d does not match #.
> So your regexp does not match anything up to the #.
>
> Your regexp should be:
>
> ^(\d+#\d+)$ (or something like that, I never tried to match #, so there
> could be a special trick for that).
>
> Or if you want to be less specific:
>
> ^(.*)$
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 25/01/2011 ? 04:28, Sam a ?crit :
>
> Hi,
>
> This is in the dialplan.
>
> http://pastebin.freeswitch.org/15131
>
> Also Brian , how can i ignore 183 without sdp,
> what happens is the provider sends 183 without sdp and by applying "<action
> application="set" data="sip_ignore_183nosdp=true"/>"  the FS sends 180 to
> the customer.
> Here i want to block the 183 with SDP just like router as b2bua and send
> nothing to customer, and when actual 183 with sdp comes it should send .
>
> Regds
> Sam
>
> On Tue, Jan 25, 2011 at 8:45 AM, Brian West <brian at freeswitch.org> wrote:
>
>> We aren't a proxy.. what exactly are you talking bout?  Can you paste us a
>> dialplan example?
>>
>> /b
>>
>> On Jan 24, 2011, at 9:11 PM, Sam wrote:
>>
>> > The more info on FS is that it is working in proxy mode , so what ever
>> customer sends it sends to the next hop,
>> > I will try using %23 for # while sending calls.
>> >
>> > Regards
>> > Sam
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Tue Jan 25 12:28:22 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 25 Jan 2011 14:58:22 +0530
Subject: [Freeswitch-users] # in prefix
In-Reply-To: <AANLkTim3dcKu_PO_ZXeTKViN66wEL-rVuXaME48QPNmP@mail.gmail.com>
References: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
	<2BAF59AA-B828-4AF6-9BD8-F4E2EAB390A8@freeswitch.org>
	<AANLkTim2Fd_t0GWqjip7jjKarvkqxqMM_P=gjVbQS0JZ@mail.gmail.com>
	<29DBF085-91BA-4110-AE87-CF8F040B6280@freeswitch.org>
	<AANLkTim3dcKu_PO_ZXeTKViN66wEL-rVuXaME48QPNmP@mail.gmail.com>
Message-ID: <AANLkTi=Ngfc2GPRVZQJWWyrEVq9-htXQmA61WX4Y=01N@mail.gmail.com>

Here now, can 183 without sdp be blocked ?

Regds
Sam



On Tue, Jan 25, 2011 at 8:58 AM, Sam <u2nsam at gmail.com> wrote:

> Hi,
>
> This is in the dialplan.
>
> http://pastebin.freeswitch.org/15131
>
> Also Brian , how can i ignore 183 without sdp,
> what happens is the provider sends 183 without sdp and by applying "<action
> application="set" data="sip_ignore_183nosdp=true"/>"  the FS sends 180 to
> the customer.
> Here i want to block the 183 with SDP just like router as b2bua and send
> nothing to customer, and when actual 183 with sdp comes it should send .
>
> Regds
> Sam
>
>
> On Tue, Jan 25, 2011 at 8:45 AM, Brian West <brian at freeswitch.org> wrote:
>
>> We aren't a proxy.. what exactly are you talking bout?  Can you paste us a
>> dialplan example?
>>
>> /b
>>
>> On Jan 24, 2011, at 9:11 PM, Sam wrote:
>>
>> > The more info on FS is that it is working in proxy mode , so what ever
>> customer sends it sends to the next hop,
>> > I will try using %23 for # while sending calls.
>> >
>> > Regards
>> > Sam
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From david.ponzone at ipeva.fr  Tue Jan 25 12:41:04 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 25 Jan 2011 10:41:04 +0100
Subject: [Freeswitch-users] # in prefix
In-Reply-To: <AANLkTi=Ngfc2GPRVZQJWWyrEVq9-htXQmA61WX4Y=01N@mail.gmail.com>
References: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
	<2BAF59AA-B828-4AF6-9BD8-F4E2EAB390A8@freeswitch.org>
	<AANLkTim2Fd_t0GWqjip7jjKarvkqxqMM_P=gjVbQS0JZ@mail.gmail.com>
	<29DBF085-91BA-4110-AE87-CF8F040B6280@freeswitch.org>
	<AANLkTim3dcKu_PO_ZXeTKViN66wEL-rVuXaME48QPNmP@mail.gmail.com>
	<AANLkTi=Ngfc2GPRVZQJWWyrEVq9-htXQmA61WX4Y=01N@mail.gmail.com>
Message-ID: <0A087DFE-3B5B-4AAA-A6B8-857262EB9E31@ipeva.fr>

Can you explain why you want to do that ?
Maybe some people around will give you a better way to achieve what you wanna do, or will recommend you not to do it.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 25/01/2011 ? 10:28, Sam a ?crit :

> Here now, can 183 without sdp be blocked ?
> 
> Regds
> Sam
> 
> 
> 
> On Tue, Jan 25, 2011 at 8:58 AM, Sam <u2nsam at gmail.com> wrote:
> Hi,
> 
> This is in the dialplan.
> 
> http://pastebin.freeswitch.org/15131
> 
> Also Brian , how can i ignore 183 without sdp,
> what happens is the provider sends 183 without sdp and by applying "<action application="set" data="sip_ignore_183nosdp=true"/>"  the FS sends 180 to the customer.
> Here i want to block the 183 with SDP just like router as b2bua and send nothing to customer, and when actual 183 with sdp comes it should send .
> 
> Regds
> Sam
> 
> 
> On Tue, Jan 25, 2011 at 8:45 AM, Brian West <brian at freeswitch.org> wrote:
> We aren't a proxy.. what exactly are you talking bout?  Can you paste us a dialplan example?
> 
> /b
> 
> On Jan 24, 2011, at 9:11 PM, Sam wrote:
> 
> > The more info on FS is that it is working in proxy mode , so what ever customer sends it sends to the next hop,
> > I will try using %23 for # while sending calls.
> >
> > Regards
> > Sam
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From tculjaga at gmail.com  Tue Jan 25 12:41:44 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Jan 2011 10:41:44 +0100
Subject: [Freeswitch-users] FS HA
In-Reply-To: <AANLkTimJLMjPJwthN8AhDq=1_JnDbdmocbgd=-=usyTT@mail.gmail.com>
References: <AANLkTi=_BPuE5KX4D48JDoOW9U3netsr9uDU9Lo5CgYO@mail.gmail.com>
	<FACE40F5-BB80-479A-BFFC-F07F6BA681D2@freeswitch.org>
	<350D44B3-A176-47A9-84F4-72EB62F08299@gmail.com>
	<AANLkTimJLMjPJwthN8AhDq=1_JnDbdmocbgd=-=usyTT@mail.gmail.com>
Message-ID: <AANLkTi=ksT-ZkdOyXiVQkir7a9z2-58PcVirmv4xs5Kg@mail.gmail.com>

Here is the debug: http://pastebin.freeswitch.org/15133

i have set verbose_sdp=true in vars.xml as.
  <X-PRE-PROCESS cmd="set" data="verbose_sdp=true"/>

but not much to be seen of the verbose thing in the debug...

Still, FS is sending a re-INVITE with wrong SDP. The call to be recovered is
using ALAW... and ULAW is not supported.
FS sends a re-INVITE with ULAW and CN in SDP. The re-INVITE is rejected due
to incompatible SDP.

Where does FS get the information for the SDP in re-INVITE message?


please advice,
T.



On Tue, Jan 25, 2011 at 12:01 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:

> yap, i do have PCMA ... and the debug shows it correctly :=)
>
> i will try to see what it does with verbose. Post new debug tomorrow.
>
> ty.
>
>
>
> On Mon, Jan 24, 2011 at 11:29 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Brian, it has PCMU in the sdp, but Sofia thinks it's setting PCMA?
>>
>> Steve on iPhone
>>
>> On 24 Jan 2011, at 20:17, Brian West <brian at freeswitch.org> wrote:
>>
>> > What makes you think that fails?  It has ULAW and CN in the codec list!
>>  Sounds like you need the verbose sdp... set the global variable
>> "verbose_sdp=true"
>> >
>> > /b
>> >
>> > On Jan 24, 2011, at 1:50 PM, Tihomir Culjaga wrote:
>> >
>> >>
>> >>
>> >> i configured FS HA and looks like its trying to recover the call .. but
>> the re-INVITE fails due to "wrong/missed" codec capability.
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From thomas at chaschperli.ch  Tue Jan 25 12:53:52 2011
From: thomas at chaschperli.ch (Thomas Mueller)
Date: Tue, 25 Jan 2011 10:53:52 +0100
Subject: [Freeswitch-users] mod_cepstral patch  for umlauts  (FS-3001)
Message-ID: <4D3E9DB0.9050103@chaschperli.ch>

hi

mod_cepstral wasn't working with umlauts, neither did  swift on cli.

cepstral support told me to use "-e utf-8" on cli to get it working. 
Indeed it worked. The default  is LATIN1/iso8859-1 encoding and can't be 
changed in a global config.

mod_cepstral did not provide the ability to define the encoding of the 
text. I wrote a patch for mod_cepstral and opened a jira bug:

http://jira.freeswitch.org/browse/FS-3001

- Thomas


From tculjaga at gmail.com  Tue Jan 25 12:57:43 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Jan 2011 10:57:43 +0100
Subject: [Freeswitch-users] # in prefix
In-Reply-To: <0A087DFE-3B5B-4AAA-A6B8-857262EB9E31@ipeva.fr>
References: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
	<2BAF59AA-B828-4AF6-9BD8-F4E2EAB390A8@freeswitch.org>
	<AANLkTim2Fd_t0GWqjip7jjKarvkqxqMM_P=gjVbQS0JZ@mail.gmail.com>
	<29DBF085-91BA-4110-AE87-CF8F040B6280@freeswitch.org>
	<AANLkTim3dcKu_PO_ZXeTKViN66wEL-rVuXaME48QPNmP@mail.gmail.com>
	<AANLkTi=Ngfc2GPRVZQJWWyrEVq9-htXQmA61WX4Y=01N@mail.gmail.com>
	<0A087DFE-3B5B-4AAA-A6B8-857262EB9E31@ipeva.fr>
Message-ID: <AANLkTi=siwSeE3Jy98nUh3oEYZN0LLdA_Rax_rNkMUFR@mail.gmail.com>

this is a legal message .. why should it be blocked ?

the SDP could be included earlier .. in 180 and after that you get 183
without SDP ... why to block the 183 message?


T.

On Tue, Jan 25, 2011 at 10:41 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Can you explain why you want to do that ?
> Maybe some people around will give you a better way to achieve what you
> wanna do, or will recommend you not to do it.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 25/01/2011 ? 10:28, Sam a ?crit :
>
> Here now, can 183 without sdp be blocked ?
>
> Regds
> Sam
>
>
>
> On Tue, Jan 25, 2011 at 8:58 AM, Sam <u2nsam at gmail.com> wrote:
>
>> Hi,
>>
>> This is in the dialplan.
>>
>> http://pastebin.freeswitch.org/15131
>>
>> Also Brian , how can i ignore 183 without sdp,
>> what happens is the provider sends 183 without sdp and by applying "<action
>> application="set" data="sip_ignore_183nosdp=true"/>"  the FS sends 180 to
>> the customer.
>> Here i want to block the 183 with SDP just like router as b2bua and send
>> nothing to customer, and when actual 183 with sdp comes it should send .
>>
>> Regds
>> Sam
>>
>>
>> On Tue, Jan 25, 2011 at 8:45 AM, Brian West <brian at freeswitch.org> wrote:
>>
>>> We aren't a proxy.. what exactly are you talking bout?  Can you paste us
>>> a dialplan example?
>>>
>>> /b
>>>
>>> On Jan 24, 2011, at 9:11 PM, Sam wrote:
>>>
>>> > The more info on FS is that it is working in proxy mode , so what ever
>>> customer sends it sends to the next hop,
>>> > I will try using %23 for # while sending calls.
>>> >
>>> > Regards
>>> > Sam
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Tue Jan 25 13:11:18 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 25 Jan 2011 15:41:18 +0530
Subject: [Freeswitch-users] # in prefix
In-Reply-To: <0A087DFE-3B5B-4AAA-A6B8-857262EB9E31@ipeva.fr>
References: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
	<2BAF59AA-B828-4AF6-9BD8-F4E2EAB390A8@freeswitch.org>
	<AANLkTim2Fd_t0GWqjip7jjKarvkqxqMM_P=gjVbQS0JZ@mail.gmail.com>
	<29DBF085-91BA-4110-AE87-CF8F040B6280@freeswitch.org>
	<AANLkTim3dcKu_PO_ZXeTKViN66wEL-rVuXaME48QPNmP@mail.gmail.com>
	<AANLkTi=Ngfc2GPRVZQJWWyrEVq9-htXQmA61WX4Y=01N@mail.gmail.com>
	<0A087DFE-3B5B-4AAA-A6B8-857262EB9E31@ipeva.fr>
Message-ID: <AANLkTikJw5=PRE5E2UREpg4=4ejpe0+77x37zcCR7P7t@mail.gmail.com>

Its because, providers are sending false signaling by sending 183 without
sdp,and it hampers while @ production,
Although by cisco sbc i have done this but i want to do it by FS,
Take a scenario, when call is send 183 without sdp for 10 secs and then
followed by 183 with sdp ( actual signal),
but when some one dials invalid number it rings for 10 secs and then gives
SIP cause 404, which is bad from the providers.
So this is the reason i want to block it.

Most of the providers do this, the way out is blocking.

Regards
Sam



On Tue, Jan 25, 2011 at 3:11 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Can you explain why you want to do that ?
> Maybe some people around will give you a better way to achieve what you
> wanna do, or will recommend you not to do it.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 25/01/2011 ? 10:28, Sam a ?crit :
>
> Here now, can 183 without sdp be blocked ?
>
> Regds
> Sam
>
>
>
> On Tue, Jan 25, 2011 at 8:58 AM, Sam <u2nsam at gmail.com> wrote:
>
>> Hi,
>>
>> This is in the dialplan.
>>
>> http://pastebin.freeswitch.org/15131
>>
>> Also Brian , how can i ignore 183 without sdp,
>> what happens is the provider sends 183 without sdp and by applying "<action
>> application="set" data="sip_ignore_183nosdp=true"/>"  the FS sends 180 to
>> the customer.
>> Here i want to block the 183 with SDP just like router as b2bua and send
>> nothing to customer, and when actual 183 with sdp comes it should send .
>>
>> Regds
>> Sam
>>
>>
>> On Tue, Jan 25, 2011 at 8:45 AM, Brian West <brian at freeswitch.org> wrote:
>>
>>> We aren't a proxy.. what exactly are you talking bout?  Can you paste us
>>> a dialplan example?
>>>
>>> /b
>>>
>>> On Jan 24, 2011, at 9:11 PM, Sam wrote:
>>>
>>> > The more info on FS is that it is working in proxy mode , so what ever
>>> customer sends it sends to the next hop,
>>> > I will try using %23 for # while sending calls.
>>> >
>>> > Regards
>>> > Sam
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Tue Jan 25 13:50:12 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 25 Jan 2011 11:50:12 +0100
Subject: [Freeswitch-users] # in prefix
In-Reply-To: <AANLkTikJw5=PRE5E2UREpg4=4ejpe0+77x37zcCR7P7t@mail.gmail.com>
References: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
	<2BAF59AA-B828-4AF6-9BD8-F4E2EAB390A8@freeswitch.org>
	<AANLkTim2Fd_t0GWqjip7jjKarvkqxqMM_P=gjVbQS0JZ@mail.gmail.com>
	<29DBF085-91BA-4110-AE87-CF8F040B6280@freeswitch.org>
	<AANLkTim3dcKu_PO_ZXeTKViN66wEL-rVuXaME48QPNmP@mail.gmail.com>
	<AANLkTi=Ngfc2GPRVZQJWWyrEVq9-htXQmA61WX4Y=01N@mail.gmail.com>
	<0A087DFE-3B5B-4AAA-A6B8-857262EB9E31@ipeva.fr>
	<AANLkTikJw5=PRE5E2UREpg4=4ejpe0+77x37zcCR7P7t@mail.gmail.com>
Message-ID: <E811C48B-9E86-43B3-9E1A-0CAE4B75D196@ipeva.fr>

If there is no SDP in the 183, it cannot ring for 10 seconds as early media is not possible without a SDP.
If you just want to ignore the early media, do ignore_early_media=true, but it's going to be worse as most Tier1 carriers do send the ringback through early media.
There could be a way in FreeSWITCH to ignore early media while sending back a 180 to leg A.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 25/01/2011 ? 11:11, Sam a ?crit :

> Its because, providers are sending false signaling by sending 183 without sdp,and it hampers while @ production, 
> Although by cisco sbc i have done this but i want to do it by FS,
> Take a scenario, when call is send 183 without sdp for 10 secs and then followed by 183 with sdp ( actual signal),
> but when some one dials invalid number it rings for 10 secs and then gives SIP cause 404, which is bad from the providers.
> So this is the reason i want to block it.
> 
> Most of the providers do this, the way out is blocking.
> 
> Regards
> Sam
> 
> 
> 
> On Tue, Jan 25, 2011 at 3:11 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
> Can you explain why you want to do that ?
> Maybe some people around will give you a better way to achieve what you wanna do, or will recommend you not to do it.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 25/01/2011 ? 10:28, Sam a ?crit :
> 
>> Here now, can 183 without sdp be blocked ?
>> 
>> Regds
>> Sam
>> 
>> 
>> 
>> On Tue, Jan 25, 2011 at 8:58 AM, Sam <u2nsam at gmail.com> wrote:
>> Hi,
>> 
>> This is in the dialplan.
>> 
>> http://pastebin.freeswitch.org/15131
>> 
>> Also Brian , how can i ignore 183 without sdp,
>> what happens is the provider sends 183 without sdp and by applying "<action application="set" data="sip_ignore_183nosdp=true"/>"  the FS sends 180 to the customer.
>> Here i want to block the 183 with SDP just like router as b2bua and send nothing to customer, and when actual 183 with sdp comes it should send .
>> 
>> Regds
>> Sam
>> 
>> 
>> On Tue, Jan 25, 2011 at 8:45 AM, Brian West <brian at freeswitch.org> wrote:
>> We aren't a proxy.. what exactly are you talking bout?  Can you paste us a dialplan example?
>> 
>> /b
>> 
>> On Jan 24, 2011, at 9:11 PM, Sam wrote:
>> 
>> > The more info on FS is that it is working in proxy mode , so what ever customer sends it sends to the next hop,
>> > I will try using %23 for # while sending calls.
>> >
>> > Regards
>> > Sam
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From fabio.bigliardi at gmail.com  Tue Jan 25 13:53:17 2011
From: fabio.bigliardi at gmail.com (Fabio Bigliardi)
Date: Tue, 25 Jan 2011 11:53:17 +0100
Subject: [Freeswitch-users] How to change global variable at runtime
In-Reply-To: <AANLkTikRXU1Ftdbr8tRDsY9o-2RbuysPK09eKMQZeBkO@mail.gmail.com>
References: <AANLkTikmTO3i=gT7qtZGZCEkYtzA=HsX-N7OnrK_vjSA@mail.gmail.com>
	<AANLkTikRXU1Ftdbr8tRDsY9o-2RbuysPK09eKMQZeBkO@mail.gmail.com>
Message-ID: <AANLkTi=mQSWi+o03zXqMKTrj5VfpTkTcBDcG3ZDFezPj@mail.gmail.com>

And how could I set a new on hook consumer after changing the value of the
global variable so that it will ring when a caller is in the fifo?

Thanks a lot.

F. Bigliardi



2011/1/24 Anthony Minessale <anthony.minessale at gmail.com>

> use only one $ not 2.
>
> $$ does not mean global, it means eval only once when you load xml.
> use ${var} and global_setvar together
>
> On Mon, Jan 24, 2011 at 8:26 AM, Fabio Bigliardi
> <fabio.bigliardi at gmail.com> wrote:
> > Hi all,
> > I would like to define a global variable in vars.xml and then change its
> > value at runtime. This value has to be read in dialplan and in
> > fifo.conf.xml.
> > I tried global_setvar from the CLI but from the log I can see that the
> > $${var} is expanded to the old value set in vars.xml, not to the new one.
> >
> >
> > What is the real effect of global_setvar command?
> >
> > Have I got to somehow write vars.xml and then issue reloadxml in order
> the
> > change to take effect?
> >
> > Thanks a lot for your support.
> >
> > Best regards,
> >
> > F. Bigliardi
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Tue Jan 25 14:41:55 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Jan 2011 12:41:55 +0100
Subject: [Freeswitch-users] # in prefix
In-Reply-To: <E811C48B-9E86-43B3-9E1A-0CAE4B75D196@ipeva.fr>
References: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
	<2BAF59AA-B828-4AF6-9BD8-F4E2EAB390A8@freeswitch.org>
	<AANLkTim2Fd_t0GWqjip7jjKarvkqxqMM_P=gjVbQS0JZ@mail.gmail.com>
	<29DBF085-91BA-4110-AE87-CF8F040B6280@freeswitch.org>
	<AANLkTim3dcKu_PO_ZXeTKViN66wEL-rVuXaME48QPNmP@mail.gmail.com>
	<AANLkTi=Ngfc2GPRVZQJWWyrEVq9-htXQmA61WX4Y=01N@mail.gmail.com>
	<0A087DFE-3B5B-4AAA-A6B8-857262EB9E31@ipeva.fr>
	<AANLkTikJw5=PRE5E2UREpg4=4ejpe0+77x37zcCR7P7t@mail.gmail.com>
	<E811C48B-9E86-43B3-9E1A-0CAE4B75D196@ipeva.fr>
Message-ID: <AANLkTik=OP7KD1XXpq1-v1W+gO-N0Ow5Kq2cgaLqP1AF@mail.gmail.com>

David, the terminal generates the ring back when getting 183 without SDP ...


Sam, is this your only concern ? Just the false ring-back?

there are plenty of ways to fix this ... perhaps you can generate within
execute_on_ring application.
Also, you can parse the ring message (180 or 183) and look for DSP.

cheers,
T.



On Tue, Jan 25, 2011 at 11:50 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> If there is no SDP in the 183, it cannot ring for 10 seconds as early media
> is not possible without a SDP.
> If you just want to ignore the early media, do ignore_early_media=true, but
> it's going to be worse as most Tier1 carriers do send the ringback through
> early media.
> There could be a way in FreeSWITCH to ignore early media while sending back
> a 180 to leg A.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 25/01/2011 ? 11:11, Sam a ?crit :
>
> Its because, providers are sending false signaling by sending 183 without
> sdp,and it hampers while @ production,
> Although by cisco sbc i have done this but i want to do it by FS,
> Take a scenario, when call is send 183 without sdp for 10 secs and then
> followed by 183 with sdp ( actual signal),
> but when some one dials invalid number it rings for 10 secs and then gives
> SIP cause 404, which is bad from the providers.
> So this is the reason i want to block it.
>
> Most of the providers do this, the way out is blocking.
>
> Regards
> Sam
>
>
>
> On Tue, Jan 25, 2011 at 3:11 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>
>> Can you explain why you want to do that ?
>> Maybe some people around will give you a better way to achieve what you
>> wanna do, or will recommend you not to do it.
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 25/01/2011 ? 10:28, Sam a ?crit :
>>
>> Here now, can 183 without sdp be blocked ?
>>
>> Regds
>> Sam
>>
>>
>>
>> On Tue, Jan 25, 2011 at 8:58 AM, Sam <u2nsam at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> This is in the dialplan.
>>>
>>> http://pastebin.freeswitch.org/15131
>>>
>>> Also Brian , how can i ignore 183 without sdp,
>>> what happens is the provider sends 183 without sdp and by applying "<action
>>> application="set" data="sip_ignore_183nosdp=true"/>"  the FS sends 180
>>> to the customer.
>>> Here i want to block the 183 with SDP just like router as b2bua and send
>>> nothing to customer, and when actual 183 with sdp comes it should send .
>>>
>>> Regds
>>> Sam
>>>
>>>
>>> On Tue, Jan 25, 2011 at 8:45 AM, Brian West <brian at freeswitch.org>wrote:
>>>
>>>> We aren't a proxy.. what exactly are you talking bout?  Can you paste us
>>>> a dialplan example?
>>>>
>>>> /b
>>>>
>>>> On Jan 24, 2011, at 9:11 PM, Sam wrote:
>>>>
>>>> > The more info on FS is that it is working in proxy mode , so what ever
>>>> customer sends it sends to the next hop,
>>>> > I will try using %23 for # while sending calls.
>>>> >
>>>> > Regards
>>>> > Sam
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Tue Jan 25 14:52:59 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 25 Jan 2011 12:52:59 +0100
Subject: [Freeswitch-users] # in prefix
In-Reply-To: <AANLkTik=OP7KD1XXpq1-v1W+gO-N0Ow5Kq2cgaLqP1AF@mail.gmail.com>
References: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
	<2BAF59AA-B828-4AF6-9BD8-F4E2EAB390A8@freeswitch.org>
	<AANLkTim2Fd_t0GWqjip7jjKarvkqxqMM_P=gjVbQS0JZ@mail.gmail.com>
	<29DBF085-91BA-4110-AE87-CF8F040B6280@freeswitch.org>
	<AANLkTim3dcKu_PO_ZXeTKViN66wEL-rVuXaME48QPNmP@mail.gmail.com>
	<AANLkTi=Ngfc2GPRVZQJWWyrEVq9-htXQmA61WX4Y=01N@mail.gmail.com>
	<0A087DFE-3B5B-4AAA-A6B8-857262EB9E31@ipeva.fr>
	<AANLkTikJw5=PRE5E2UREpg4=4ejpe0+77x37zcCR7P7t@mail.gmail.com>
	<E811C48B-9E86-43B3-9E1A-0CAE4B75D196@ipeva.fr>
	<AANLkTik=OP7KD1XXpq1-v1W+gO-N0Ow5Kq2cgaLqP1AF@mail.gmail.com>
Message-ID: <B1B8B484-B40F-4B54-8FA5-1DC26EAA97BF@ipeva.fr>

Sure, but then I am not sure to undertand what he wants to do.

Sam, to summarize, do you want to ignore the first 183 because you suspect/know it's fake, and wait for the trustable ringback in early media ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 25/01/2011 ? 12:41, Tihomir Culjaga a ?crit :

> David, the terminal generates the ring back when getting 183 without SDP ...
> 
> 
> Sam, is this your only concern ? Just the false ring-back?
> 
> there are plenty of ways to fix this ... perhaps you can generate within execute_on_ring application. 
> Also, you can parse the ring message (180 or 183) and look for DSP.
> 
> cheers,
> T.
> 
> 
> 
> On Tue, Jan 25, 2011 at 11:50 AM, David Ponzone <david.ponzone at ipeva.fr> wrote:
> If there is no SDP in the 183, it cannot ring for 10 seconds as early media is not possible without a SDP.
> If you just want to ignore the early media, do ignore_early_media=true, but it's going to be worse as most Tier1 carriers do send the ringback through early media.
> There could be a way in FreeSWITCH to ignore early media while sending back a 180 to leg A.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 25/01/2011 ? 11:11, Sam a ?crit :
> 
>> Its because, providers are sending false signaling by sending 183 without sdp,and it hampers while @ production, 
>> Although by cisco sbc i have done this but i want to do it by FS,
>> Take a scenario, when call is send 183 without sdp for 10 secs and then followed by 183 with sdp ( actual signal),
>> but when some one dials invalid number it rings for 10 secs and then gives SIP cause 404, which is bad from the providers.
>> So this is the reason i want to block it.
>> 
>> Most of the providers do this, the way out is blocking.
>> 
>> Regards
>> Sam
>> 
>> 
>> 
>> On Tue, Jan 25, 2011 at 3:11 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
>> Can you explain why you want to do that ?
>> Maybe some people around will give you a better way to achieve what you wanna do, or will recommend you not to do it.
>> 
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> 
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> 
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> 
>> 
>> 
>> 
>> Le 25/01/2011 ? 10:28, Sam a ?crit :
>> 
>>> Here now, can 183 without sdp be blocked ?
>>> 
>>> Regds
>>> Sam
>>> 
>>> 
>>> 
>>> On Tue, Jan 25, 2011 at 8:58 AM, Sam <u2nsam at gmail.com> wrote:
>>> Hi,
>>> 
>>> This is in the dialplan.
>>> 
>>> http://pastebin.freeswitch.org/15131
>>> 
>>> Also Brian , how can i ignore 183 without sdp,
>>> what happens is the provider sends 183 without sdp and by applying "<action application="set" data="sip_ignore_183nosdp=true"/>"  the FS sends 180 to the customer.
>>> Here i want to block the 183 with SDP just like router as b2bua and send nothing to customer, and when actual 183 with sdp comes it should send .
>>> 
>>> Regds
>>> Sam
>>> 
>>> 
>>> On Tue, Jan 25, 2011 at 8:45 AM, Brian West <brian at freeswitch.org> wrote:
>>> We aren't a proxy.. what exactly are you talking bout?  Can you paste us a dialplan example?
>>> 
>>> /b
>>> 
>>> On Jan 24, 2011, at 9:11 PM, Sam wrote:
>>> 
>>> > The more info on FS is that it is working in proxy mode , so what ever customer sends it sends to the next hop,
>>> > I will try using %23 for # while sending calls.
>>> >
>>> > Regards
>>> > Sam
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at ipeva.fr  Tue Jan 25 15:00:46 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 25 Jan 2011 13:00:46 +0100
Subject: [Freeswitch-users] # in prefix
In-Reply-To: <AANLkTim3dcKu_PO_ZXeTKViN66wEL-rVuXaME48QPNmP@mail.gmail.com>
References: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
	<2BAF59AA-B828-4AF6-9BD8-F4E2EAB390A8@freeswitch.org>
	<AANLkTim2Fd_t0GWqjip7jjKarvkqxqMM_P=gjVbQS0JZ@mail.gmail.com>
	<29DBF085-91BA-4110-AE87-CF8F040B6280@freeswitch.org>
	<AANLkTim3dcKu_PO_ZXeTKViN66wEL-rVuXaME48QPNmP@mail.gmail.com>
Message-ID: <01ACD465-F2D2-4827-B0DD-91AF6BF6B5A3@ipeva.fr>

Sam,

you actually found the answer to your question, as sip_ignore_183nosdp was implented by Anthony to achieve this:

http://jira.freeswitch.org/browse/FS-1978

So what you meant is that you tried it and it does not work ?
Did it actually change something in the callflow when you set it ?
Have you tried to export it rather than set ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 25/01/2011 ? 04:28, Sam a ?crit :

> Hi,
> 
> This is in the dialplan.
> 
> http://pastebin.freeswitch.org/15131
> 
> Also Brian , how can i ignore 183 without sdp,
> what happens is the provider sends 183 without sdp and by applying "<action application="set" data="sip_ignore_183nosdp=true"/>"  the FS sends 180 to the customer.
> Here i want to block the 183 with SDP just like router as b2bua and send nothing to customer, and when actual 183 with sdp comes it should send .
> 
> Regds
> Sam
> 
> On Tue, Jan 25, 2011 at 8:45 AM, Brian West <brian at freeswitch.org> wrote:
> We aren't a proxy.. what exactly are you talking bout?  Can you paste us a dialplan example?
> 
> /b
> 
> On Jan 24, 2011, at 9:11 PM, Sam wrote:
> 
> > The more info on FS is that it is working in proxy mode , so what ever customer sends it sends to the next hop,
> > I will try using %23 for # while sending calls.
> >
> > Regards
> > Sam
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From tculjaga at gmail.com  Tue Jan 25 15:19:08 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Jan 2011 13:19:08 +0100
Subject: [Freeswitch-users] # in prefix
In-Reply-To: <01ACD465-F2D2-4827-B0DD-91AF6BF6B5A3@ipeva.fr>
References: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
	<2BAF59AA-B828-4AF6-9BD8-F4E2EAB390A8@freeswitch.org>
	<AANLkTim2Fd_t0GWqjip7jjKarvkqxqMM_P=gjVbQS0JZ@mail.gmail.com>
	<29DBF085-91BA-4110-AE87-CF8F040B6280@freeswitch.org>
	<AANLkTim3dcKu_PO_ZXeTKViN66wEL-rVuXaME48QPNmP@mail.gmail.com>
	<01ACD465-F2D2-4827-B0DD-91AF6BF6B5A3@ipeva.fr>
Message-ID: <AANLkTik0az0d4E_NH+sbEa4-o-VaU9oMts9zAGsUJJFY@mail.gmail.com>

cool, that would certainly do :=)

T.

On Tue, Jan 25, 2011 at 1:00 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Sam,
>
> you actually found the answer to your question, as sip_ignore_183nosdp was
> implented by Anthony to achieve this:
>
> http://jira.freeswitch.org/browse/FS-1978
>
> So what you meant is that you tried it and it does not work ?
> Did it actually change something in the callflow when you set it ?
> Have you tried to export it rather than set ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 25/01/2011 ? 04:28, Sam a ?crit :
>
> Hi,
>
> This is in the dialplan.
>
> http://pastebin.freeswitch.org/15131
>
> Also Brian , how can i ignore 183 without sdp,
> what happens is the provider sends 183 without sdp and by applying "<action
> application="set" data="sip_ignore_183nosdp=true"/>"  the FS sends 180 to
> the customer.
> Here i want to block the 183 with SDP just like router as b2bua and send
> nothing to customer, and when actual 183 with sdp comes it should send .
>
> Regds
> Sam
>
> On Tue, Jan 25, 2011 at 8:45 AM, Brian West <brian at freeswitch.org> wrote:
>
>> We aren't a proxy.. what exactly are you talking bout?  Can you paste us a
>> dialplan example?
>>
>> /b
>>
>> On Jan 24, 2011, at 9:11 PM, Sam wrote:
>>
>> > The more info on FS is that it is working in proxy mode , so what ever
>> customer sends it sends to the next hop,
>> > I will try using %23 for # while sending calls.
>> >
>> > Regards
>> > Sam
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Tue Jan 25 15:29:34 2011
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 25 Jan 2011 20:29:34 +0800
Subject: [Freeswitch-users] How to change global variable at runtime
In-Reply-To: <AANLkTi=mQSWi+o03zXqMKTrj5VfpTkTcBDcG3ZDFezPj@mail.gmail.com>
References: <AANLkTikmTO3i=gT7qtZGZCEkYtzA=HsX-N7OnrK_vjSA@mail.gmail.com>
	<AANLkTikRXU1Ftdbr8tRDsY9o-2RbuysPK09eKMQZeBkO@mail.gmail.com>
	<AANLkTi=mQSWi+o03zXqMKTrj5VfpTkTcBDcG3ZDFezPj@mail.gmail.com>
Message-ID: <AANLkTi=11P3Tu631P4wT6UH6GzpwKXha=V4ezhoMwsv1@mail.gmail.com>

API: fifo_member add

On Tue, Jan 25, 2011 at 6:53 PM, Fabio Bigliardi
<fabio.bigliardi at gmail.com> wrote:
> And how could I set a new on hook consumer after changing the value of the
> global variable so that it will ring when a caller is in the fifo?
>
> Thanks a lot.
>
> F. Bigliardi
>
>
>
> 2011/1/24 Anthony Minessale <anthony.minessale at gmail.com>
>>
>> use only one $ not 2.
>>
>> $$ does not mean global, it means eval only once when you load xml.
>> use ${var} and global_setvar together
>>
>> On Mon, Jan 24, 2011 at 8:26 AM, Fabio Bigliardi
>> <fabio.bigliardi at gmail.com> wrote:
>> > Hi all,
>> > I would like to define a global variable in vars.xml and then change its
>> > value at runtime. This value has to be read in dialplan and in
>> > fifo.conf.xml.
>> > I tried global_setvar from the CLI but from the log I can see that the
>> > $${var} is expanded to the old value set in vars.xml, not to the new
>> > one.
>> >
>> >
>> > What is the real effect of global_setvar command?
>> >
>> > Have I got to somehow write vars.xml and then issue reloadxml in order
>> > the
>> > change to take effect?
>> >
>> > Thanks a lot for your support.
>> >
>> > Best regards,
>> >
>> > F. Bigliardi
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From avi at avimarcus.net  Tue Jan 25 16:03:38 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 25 Jan 2011 15:03:38 +0200
Subject: [Freeswitch-users] FS HA
In-Reply-To: <AANLkTi=ksT-ZkdOyXiVQkir7a9z2-58PcVirmv4xs5Kg@mail.gmail.com>
References: <AANLkTi=_BPuE5KX4D48JDoOW9U3netsr9uDU9Lo5CgYO@mail.gmail.com>
	<FACE40F5-BB80-479A-BFFC-F07F6BA681D2@freeswitch.org>
	<350D44B3-A176-47A9-84F4-72EB62F08299@gmail.com>
	<AANLkTimJLMjPJwthN8AhDq=1_JnDbdmocbgd=-=usyTT@mail.gmail.com>
	<AANLkTi=ksT-ZkdOyXiVQkir7a9z2-58PcVirmv4xs5Kg@mail.gmail.com>
Message-ID: <AANLkTikeU5FAgE-HU=0a5whf3JiD5VUCkYkCCRt4u=R0@mail.gmail.com>

Are you using two machines for the HA? do both have the same configs?
-Avi

On Tue, Jan 25, 2011 at 11:41 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
> Here is the debug: http://pastebin.freeswitch.org/15133
>
> i have set verbose_sdp=true in vars.xml as.
> ? <X-PRE-PROCESS cmd="set" data="verbose_sdp=true"/>
>
> but not much to be seen of the verbose thing in the debug...
>
> Still, FS is sending a re-INVITE with wrong SDP. The call to be recovered is
> using ALAW... and ULAW is not supported.
> FS sends a re-INVITE with ULAW and CN in SDP. The re-INVITE is rejected due
> to incompatible SDP.
>
> Where does FS get the information for the SDP in re-INVITE message?
>
>
> please advice,
> T.
>
>
>
> On Tue, Jan 25, 2011 at 12:01 AM, Tihomir Culjaga <tculjaga at gmail.com>
> wrote:
>>
>> yap, i do have PCMA ... and the debug shows it correctly :=)
>>
>> i will try to see what it does with verbose. Post new debug tomorrow.
>>
>> ty.
>>
>>
>> On Mon, Jan 24, 2011 at 11:29 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>>
>>> Brian, it has PCMU in the sdp, but Sofia thinks it's setting PCMA?
>>>
>>> Steve on iPhone
>>>
>>> On 24 Jan 2011, at 20:17, Brian West <brian at freeswitch.org> wrote:
>>>
>>> > What makes you think that fails? ?It has ULAW and CN in the codec list!
>>> > ?Sounds like you need the verbose sdp... set the global variable
>>> > "verbose_sdp=true"
>>> >
>>> > /b
>>> >
>>> > On Jan 24, 2011, at 1:50 PM, Tihomir Culjaga wrote:
>>> >
>>> >>
>>> >>
>>> >> i configured FS HA and looks like its trying to recover the call ..
>>> >> but the re-INVITE fails due to "wrong/missed" codec capability.
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From tculjaga at gmail.com  Tue Jan 25 16:24:44 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Jan 2011 14:24:44 +0100
Subject: [Freeswitch-users] FS HA
In-Reply-To: <AANLkTikeU5FAgE-HU=0a5whf3JiD5VUCkYkCCRt4u=R0@mail.gmail.com>
References: <AANLkTi=_BPuE5KX4D48JDoOW9U3netsr9uDU9Lo5CgYO@mail.gmail.com>
	<FACE40F5-BB80-479A-BFFC-F07F6BA681D2@freeswitch.org>
	<350D44B3-A176-47A9-84F4-72EB62F08299@gmail.com>
	<AANLkTimJLMjPJwthN8AhDq=1_JnDbdmocbgd=-=usyTT@mail.gmail.com>
	<AANLkTi=ksT-ZkdOyXiVQkir7a9z2-58PcVirmv4xs5Kg@mail.gmail.com>
	<AANLkTikeU5FAgE-HU=0a5whf3JiD5VUCkYkCCRt4u=R0@mail.gmail.com>
Message-ID: <AANLkTim3=HVjX5GZVfcVjtSCm-2kuWatyVS+YV0RpNCB@mail.gmail.com>

yap, both using the same config


can you advice where is FS getting the SDP info for the re-INVITE ?


On Tue, Jan 25, 2011 at 2:03 PM, Avi Marcus <avi at avimarcus.net> wrote:

> Are you using two machines for the HA? do both have the same configs?
> -Avi
>
> On Tue, Jan 25, 2011 at 11:41 AM, Tihomir Culjaga <tculjaga at gmail.com>
> wrote:
> > Here is the debug: http://pastebin.freeswitch.org/15133
> >
> > i have set verbose_sdp=true in vars.xml as.
> >   <X-PRE-PROCESS cmd="set" data="verbose_sdp=true"/>
> >
> > but not much to be seen of the verbose thing in the debug...
> >
> > Still, FS is sending a re-INVITE with wrong SDP. The call to be recovered
> is
> > using ALAW... and ULAW is not supported.
> > FS sends a re-INVITE with ULAW and CN in SDP. The re-INVITE is rejected
> due
> > to incompatible SDP.
> >
> > Where does FS get the information for the SDP in re-INVITE message?
> >
> >
> > please advice,
> > T.
> >
> >
> >
> > On Tue, Jan 25, 2011 at 12:01 AM, Tihomir Culjaga <tculjaga at gmail.com>
> > wrote:
> >>
> >> yap, i do have PCMA ... and the debug shows it correctly :=)
> >>
> >> i will try to see what it does with verbose. Post new debug tomorrow.
> >>
> >> ty.
> >>
> >>
> >> On Mon, Jan 24, 2011 at 11:29 PM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >>>
> >>> Brian, it has PCMU in the sdp, but Sofia thinks it's setting PCMA?
> >>>
> >>> Steve on iPhone
> >>>
> >>> On 24 Jan 2011, at 20:17, Brian West <brian at freeswitch.org> wrote:
> >>>
> >>> > What makes you think that fails?  It has ULAW and CN in the codec
> list!
> >>> >  Sounds like you need the verbose sdp... set the global variable
> >>> > "verbose_sdp=true"
> >>> >
> >>> > /b
> >>> >
> >>> > On Jan 24, 2011, at 1:50 PM, Tihomir Culjaga wrote:
> >>> >
> >>> >>
> >>> >>
> >>> >> i configured FS HA and looks like its trying to recover the call ..
> >>> >> but the re-INVITE fails due to "wrong/missed" codec capability.
> >>> >
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From boris at tagnet.ru  Tue Jan 25 17:42:00 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 25 Jan 2011 19:42:00 +0500
Subject: [Freeswitch-users] How exactly execute_extension work?
Message-ID: <4D3EE138.3060500@tagnet.ru>

Hello!

     From one of my extension I call another extension in another 
context. I suppose that exactly this extension should be executed, but 
instead I get all previous extension executed to. So below is my 
configuration. We are calling start_hunting in context tagnet.ru. I 
suppose start_hunting extension should be executed but no, hunting 
begins from the start of context and all extensions before start_hunting 
are executed too. auto_hunt=true solves the problem. So the question is 
- is this normal behaviour of execute_extension or am I doing something 
wrong?

<!-- We are hunting only for TAGNet numbering plan -->
<extension name="ext_hunting">
<condition field="${v_tagnet_dst}" 
expression="^(73435230[0-9]{3}|73435494989|7343549499[0-1])$">
<action application="javascript" data="$${scripts_dir}/toll_check.js 
${toll_allow} local"/>
<action application="set" data="v_tagnet_hunting=true" inline="true"/>
<action application="log" data="NOTICE [top.ctx] - hunting in tagnet.ru"/>
<action application="log" data="NOTICE ${v_tagnet_dst} $1"/>
<action application="execute_extension" data="start_hunting XML tagnet.ru"/>
</condition>
</extension>

Context tagnet.ru:
<!--
     This context is for home networking soft phones and cheap telephony.
-->
<?xml version="1.0" encoding="utf-8"?>
<include>
<context name="tagnet.ru">
<extension name="unloop">
<condition field="${unroll_loops}" expression="^true$"/>
<condition field="${sip_looped_call}" expression="^true$">
<action application="deflect" data="${destination_number}"/>
</condition>
</extension>

<!--
             Destination number normalization
         -->
<extension name="ext_translate_dst" continue="true">
<!-- <condition field="${v_tagnet_hunting}" expression="^true$" 
break="on-true"/> -->
<condition field="${destination_number}" expression="^(1[0-9]{3})$" 
break="on-true">
<action application="set" data="v_tagnet_dst=$1" inline="true"/>
</condition>
<condition field="${destination_number}" expression="^(\d+)$" 
break="on-true">
<action application="set" data="v_tagnet_dst=$1" inline="true"/>
<action application="execute_extension" data="ext_translate_dst XML 
features"/>
</condition>
</extension>

<!--
             Source number normalization
         -->
<extension name="ext_translate_src" continue="true">
<condition field="${caller_id_number}" expression="^2([0-9]{4})$" 
break="on-true">
<action application="set" data="caller_id_number=7343549$1" inline="true"/>
<action application="set" data="ani=7343549$1" inline="true"/>
</condition>
</extension>

<!-- Here we start hunting process -->
<extension name="start_hunting" continue="true">
<condition field="destination_number" expression="^start_hunting$"/>
</extension>

<!--
                 You can place files in the default directory to get 
included.
         -->
<X-PRE-PROCESS cmd="include" data="tagnet.ru/*.xml"/>

<!-- Here we stop hunting process -->
<extension name="stop_hunting">
<condition field="${v_tagnet_hunting}" expression="true"/>
</extension>

<!--
             Transfer call to the TOP context
         -->
<extension name="top.ctx">
<condition field="${v_tagnet_dst}" expression="^(\d+)$">
<action application="log" data="DEBUG [top.ctx] - transfer to top.ctx"/>

<!-- Caller ID normalization -->
<action application="execute_extension" data="ext_translate_src XML 
features"/>

<action application="transfer" data="$1 XML top.ctx"/>
</condition>
</extension>
</context>
</include>




-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From u2nsam at gmail.com  Tue Jan 25 17:59:10 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 25 Jan 2011 20:29:10 +0530
Subject: [Freeswitch-users] # in prefix
In-Reply-To: <01ACD465-F2D2-4827-B0DD-91AF6BF6B5A3@ipeva.fr>
References: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
	<2BAF59AA-B828-4AF6-9BD8-F4E2EAB390A8@freeswitch.org>
	<AANLkTim2Fd_t0GWqjip7jjKarvkqxqMM_P=gjVbQS0JZ@mail.gmail.com>
	<29DBF085-91BA-4110-AE87-CF8F040B6280@freeswitch.org>
	<AANLkTim3dcKu_PO_ZXeTKViN66wEL-rVuXaME48QPNmP@mail.gmail.com>
	<01ACD465-F2D2-4827-B0DD-91AF6BF6B5A3@ipeva.fr>
Message-ID: <AANLkTi=9YQTJjbg_HaHeSCXJv+-708rYxefZp9pQHmQJ@mail.gmail.com>

Yes tried that what Anthony said, its true it ignores the 183 w/o SDP but
sends 180 to leg a instead for that duration,
i want to just ignore the 183 without SDP and not 183 with sdp; 183 with SDP
should go to the leg a but not 183 w/o sdp.

Yes Tihomir false ring-back is a concern (i.e. 183 w/o sdp) and this is
generated by one of the tier 1 provider having nextone sbc :(

Regards
Sam

On Tue, Jan 25, 2011 at 5:30 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Sam,
>
> you actually found the answer to your question, as sip_ignore_183nosdp was
> implented by Anthony to achieve this:
>
> http://jira.freeswitch.org/browse/FS-1978
>
> So what you meant is that you tried it and it does not work ?
> Did it actually change something in the callflow when you set it ?
> Have you tried to export it rather than set ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 25/01/2011 ? 04:28, Sam a ?crit :
>
> Hi,
>
> This is in the dialplan.
>
> http://pastebin.freeswitch.org/15131
>
> Also Brian , how can i ignore 183 without sdp,
> what happens is the provider sends 183 without sdp and by applying "<action
> application="set" data="sip_ignore_183nosdp=true"/>"  the FS sends 180 to
> the customer.
> Here i want to block the 183 with SDP just like router as b2bua and send
> nothing to customer, and when actual 183 with sdp comes it should send .
>
> Regds
> Sam
>
> On Tue, Jan 25, 2011 at 8:45 AM, Brian West <brian at freeswitch.org> wrote:
>
>> We aren't a proxy.. what exactly are you talking bout?  Can you paste us a
>> dialplan example?
>>
>> /b
>>
>> On Jan 24, 2011, at 9:11 PM, Sam wrote:
>>
>> > The more info on FS is that it is working in proxy mode , so what ever
>> customer sends it sends to the next hop,
>> > I will try using %23 for # while sending calls.
>> >
>> > Regards
>> > Sam
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Tue Jan 25 18:20:14 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Jan 2011 16:20:14 +0100
Subject: [Freeswitch-users] # in prefix
In-Reply-To: <AANLkTi=9YQTJjbg_HaHeSCXJv+-708rYxefZp9pQHmQJ@mail.gmail.com>
References: <AANLkTineo=QK6TN78o8F76ZjcKUG3J=D7Lk7BKjjzQjc@mail.gmail.com>
	<2BAF59AA-B828-4AF6-9BD8-F4E2EAB390A8@freeswitch.org>
	<AANLkTim2Fd_t0GWqjip7jjKarvkqxqMM_P=gjVbQS0JZ@mail.gmail.com>
	<29DBF085-91BA-4110-AE87-CF8F040B6280@freeswitch.org>
	<AANLkTim3dcKu_PO_ZXeTKViN66wEL-rVuXaME48QPNmP@mail.gmail.com>
	<01ACD465-F2D2-4827-B0DD-91AF6BF6B5A3@ipeva.fr>
	<AANLkTi=9YQTJjbg_HaHeSCXJv+-708rYxefZp9pQHmQJ@mail.gmail.com>
Message-ID: <AANLkTi=XfBGaMQUp2OFq2xSMGzMEb6hq5Kvur-7AtmCq@mail.gmail.com>

than i advice you to do execute_on_ring and parse your 180 / 183 messages in
search of SDP. If you have an SDP, open media channel.

what if you get 180/183 without SDP ? This is a legal call flow as well.
after that you could get 200 OK with SDP .. what do you need to do in such
cases ?



On Tue, Jan 25, 2011 at 3:59 PM, Sam <u2nsam at gmail.com> wrote:

> Yes tried that what Anthony said, its true it ignores the 183 w/o SDP but
> sends 180 to leg a instead for that duration,
> i want to just ignore the 183 without SDP and not 183 with sdp; 183 with
> SDP should go to the leg a but not 183 w/o sdp.
>
> Yes Tihomir false ring-back is a concern (i.e. 183 w/o sdp) and this is
> generated by one of the tier 1 provider having nextone sbc :(
>
> Regards
> Sam
>
>
> On Tue, Jan 25, 2011 at 5:30 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>
>> Sam,
>>
>> you actually found the answer to your question, as sip_ignore_183nosdp was
>> implented by Anthony to achieve this:
>>
>> http://jira.freeswitch.org/browse/FS-1978
>>
>> So what you meant is that you tried it and it does not work ?
>> Did it actually change something in the callflow when you set it ?
>> Have you tried to export it rather than set ?
>>
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 25/01/2011 ? 04:28, Sam a ?crit :
>>
>> Hi,
>>
>> This is in the dialplan.
>>
>> http://pastebin.freeswitch.org/15131
>>
>> Also Brian , how can i ignore 183 without sdp,
>> what happens is the provider sends 183 without sdp and by applying "<action
>> application="set" data="sip_ignore_183nosdp=true"/>"  the FS sends 180 to
>> the customer.
>> Here i want to block the 183 with SDP just like router as b2bua and send
>> nothing to customer, and when actual 183 with sdp comes it should send .
>>
>> Regds
>> Sam
>>
>> On Tue, Jan 25, 2011 at 8:45 AM, Brian West <brian at freeswitch.org> wrote:
>>
>>> We aren't a proxy.. what exactly are you talking bout?  Can you paste us
>>> a dialplan example?
>>>
>>> /b
>>>
>>> On Jan 24, 2011, at 9:11 PM, Sam wrote:
>>>
>>> > The more info on FS is that it is working in proxy mode , so what ever
>>> customer sends it sends to the next hop,
>>> > I will try using %23 for # while sending calls.
>>> >
>>> > Regards
>>> > Sam
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jeff at jefflenk.com  Tue Jan 25 18:21:06 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 25 Jan 2011 07:21:06 -0800 (PST)
Subject: [Freeswitch-users] mod_cepstral patch  for umlauts  (FS-3001)
In-Reply-To: <4D3E9DB0.9050103@chaschperli.ch>
References: <4D3E9DB0.9050103@chaschperli.ch>
Message-ID: <1295968866685-5959192.post@n2.nabble.com>


Thanks for doing that. I will take a look.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-cepstral-patch-for-umlauts-FS-3001-tp5958140p5959192.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From helmut.kuper at ewetel.de  Tue Jan 25 18:45:58 2011
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 25 Jan 2011 16:45:58 +0100
Subject: [Freeswitch-users] intercom in lua dialplan
Message-ID: <4D3EF036.80701@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

I try to intercept a call in lua script. For the intercepting extension
I do this:


if (dialed_ext:len()==6 and dialed_ext:sub(1,2) == "**") then
	uuid=api:execute("hash", "select/last_caller/"..dialed_ext:sub(3))
        if (uuid and uuid:len()> 2) then
                session:setAutoHangup(false)
                session:execute("intercept", uuid)
		return
        end
end


This results to a hangup of the intercepting extension while intercepted
extension hangs up (completed_elsewhere) and caller/originator's call is
connected.

FS logs shows this for the intercepting extension:

switch_core_state_machine.c:189 sofia/internal_et/8000 at spklw.x has
executed the last dialplan instruction, hanging up.


When I use the ACTIONS table like this:

if (dialed_ext:len()==6 and dialed_ext:sub(1,2) == "**") then
	uuid=api:execute("hash", "select/last_caller/"..dialed_ext:sub(3))
        if (uuid and uuid:len()> 2) then
                session:setAutoHangup(false)
                table.insert(ACTIONS, {"intercept", uuid})
		return
        end
end


then intercept/pickup works fine. Is there a trick to get it work with
"session:execute("intercept", uuid)" ?

regards
helmut
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From kbdfck at gmail.com  Tue Jan 25 19:01:43 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Tue, 25 Jan 2011 19:01:43 +0300
Subject: [Freeswitch-users] Is it possible to use att_xfer on channels
 already bridged via loopback?
In-Reply-To: <AANLkTi=WKADqDk-qqevfAh3vR1mZgZdwEEey_cuFUEz8@mail.gmail.com>
References: <AANLkTimWAPV52349=Yge2zjnJyP40EdgDJXgyH-QcOCd@mail.gmail.com>
	<AANLkTi=WKADqDk-qqevfAh3vR1mZgZdwEEey_cuFUEz8@mail.gmail.com>
Message-ID: <AANLkTimmE9oirGUznj6RTZHYkf=pWZLUg-5SHfRqbnFH@mail.gmail.com>

http://pastebin.freeswitch.org/15141 - dialplan features.xml
http://pastebin.freeswitch.org/15140 - extraction from main dialplan
to only test att_xfers
http://pastebin.freeswitch.org/15138 - freeswitch.xml

Do you also need my debug logs or any other info?
My problem can be reproduced on my hardware with this configuration.
There is a problem with MOH also - if first att_xfer fails for any
reason, there will be no MOH for transferee on subsequent att_xfer
calls, just silence


2011/1/24 Michael Collins <msc at freeswitch.org>:
> Dmitry,
> I'd like to try this myself on one of my boxes. Would you pastebin the
> dialplan you are using?
> Thanks,
> MC
>
> On Mon, Jan 24, 2011 at 8:01 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>
>> Is att_xfer or mod_loopback is broken in FS-current?
>> I use FreeSWITCH Version 1.0.head (git-7eceff4 2011-01-16 22-33-50 +0000)
>> Seems there were no updates of att_xfer or mod_loopback since that.
>>
>> I use loopback channel as destination when doing att_xfer to re-enter
>> dialplan.
>> With loopback_bowout=false and loopback_bowout_on_execute=false this
>> works. But when any of connected parties tries to do att_xfer again,
>> all channels get hangup on transferer hangup.
>>
>> Scenario:
>>
>> A calls B, B answers
>> A launches att_xfer via *7, B listens to MOH
>> A dials C and we do att_xfer to loopback/C
>> C answers, A hangs up to complete transfer
>> C and B are now bridged via loopback, `show channels` shows 4 channels
>> include 2 loopback legs.
>>
>> Now, C also tries to do in-call transfer with *7.
>> C launches att_xfer via *7, B listens to MOH
>> C dials D and do att_xfer to loopback/D
>> D answers, C hangs up to complete transfer
>> B and D are hung up instead of be bridged together.
>>
>> There are also issues with MOH wile running att_xfer, but they are not
>> so important as att_xfer behavior itself.
>>
>>
>>
>> --
>> Best regards,
>>
>> Dmitry Sytchev,
>> IT Engineer
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Best regards,

Dmitry Sytchev,
IT Engineer


From liuyp2 at asiainfo-linkage.com  Tue Jan 25 04:53:36 2011
From: liuyp2 at asiainfo-linkage.com (=?utf-8?B?bGl1eXAy?=)
Date: Tue, 25 Jan 2011 09:53:36 +0800
Subject: [Freeswitch-users] =?utf-8?q?FreeSWITCH_1=2E0=2E7?=
References: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org>,
	<4D3D0EE9.4070200@bksys.co.in>
Message-ID: <201101250953332657154@asiainfo-linkage.com>

What's new features in fs1.0.7?

Thanks! 




liuyp2
2011-01-25



???? abubacker
????? 2011-01-24 14:02:25
???? FreeSWITCH Users Help
??? 
??? Re: [Freeswitch-users] FreeSWITCH 1.0.7

On Friday 14 January 2011 09:55 PM, Brian West wrote: 
http://latest.freeswitch.org/



Enjoy!


/b



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
  
Great job , Thanks to every one ! 


-- 
Best regards, 
N.Abubacker , 
Associate system engineer , 
bk systems pvt ltd , 
Ph : 9144-43902701

Disclaimer: http://www.bksystems.co.in/email-policy
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From liuyp2 at asiainfo-linkage.com  Tue Jan 25 05:40:42 2011
From: liuyp2 at asiainfo-linkage.com (=?utf-8?B?bGl1eXAy?=)
Date: Tue, 25 Jan 2011 10:40:42 +0800
Subject: [Freeswitch-users] =?utf-8?q?Caller_ID_using_Fifo?=
References: <AANLkTik6pdgXgND6kmzZA4qe8vDZQ0_7e47+ciVWPfub@mail.gmail.com>,
	<AANLkTikKDO3qru-9=nuPEQcW+iCN7ccizyJAh9sqUkCN@mail.gmail.com>,
	<AANLkTi=p7_CsUDWS62+weZv50+PBZ8RLop0Q04y-c4hB@mail.gmail.com>,
	<AANLkTi=m_hW9vQV5nbC2jwcos0bKy2rCf7YAr6aA23oz@mail.gmail.com>,
	<AANLkTinX7fZOO9edgHs739RxhcBJwuh70E9f07mM0mzL@mail.gmail.com>
Message-ID: <201101251040421253879@asiainfo-linkage.com>

mod_fifo can't transfer sip header  message which defined by myself(sip_h_X-xxx) to b-leg also.

Is there any solution in latest version?




liuyp2
2011-01-25


 
???? Anthony Minessale
????? 2011-01-25 09:28:25
???? FreeSWITCH Users Help
??? 
??? Re: [Freeswitch-users] Caller ID using Fifo

You should all confer to make sure you are all using fs latest git because that is the version I am talking about.  Fifo has some major new features in latest that do not exist in older versions including showing the customers cid when it calls agents.  The dilemma jm describes used to be true but is no longer the case with the default ringall strategy on latest git.
The customers cid is sent to the agent and if the fifo xml defines outbound_name param that will be included as well.
If you want to override it you must do what you quoted in the wiki in the dialstring contained in the member tag of the xml for that membership not in the dialplan.

On Jan 14, 2011 10:36 AM, "Marc de Corny" <marcdecorny at gmail.com> wrote:
>
> Just to follow up on this subject.
>  
> I have done a lot of testing on the fifo trying to get the caller_id_name changed on the outbound call to the agent and to be honest I cannot understand the explanation.
>  
> If mod_fifo does not know which call it will connect until the agent answers, how come it displays the CLI correctly, jsut won;t let me change it.
>  
> Still seems strange. I am looking into the Mod_callcentre to check if it sends caller_id information. but the same logic if valid could apply
>  
> Also maybe someone should change the Wiki ( I would but do not have enough expertise on the subject) because the following is a bit misleading
>  
>  "Note: If you wish to specify the caller ID presented when a fifo calls an agent, set the origination_caller_id_name and origination_caller_id_num variables to the values desired. These could be set within the {} of the dialstring, or they could be set using the set application in the dialplan which places the caller into the fifo (before the 'fifo in' executed on the caller). "
> thanks
> Marc
> On Thu, Jan 13, 2011 at 10:47 PM, Joao Leme <joaocarlosleme at gmail.com> wrote:
>>
>> What about showing the Caller ID after it is answered? Any way to do that?
>>
>> 2011/1/12 Jo?o Mesquita <jmesquita at freeswitch.org>
>>
>>> Jo?o Leme,
>>>
>>> The caller id is not passed when the phone is ringing because mod_fifo does not know which call is going to be sent to that channel once it is answered until it is really answered. I don't know if mod_callcenter does show anything but you should consider looking at the documentation if you really need this feature.
>>>
>>> Regards,
>>> Jo?o Mesquita
>>>
>>>
>>> On Wed, Jan 12, 2011 at 9:15 PM, Joao Leme <joaocarlosleme at gmail.com> wrote:
>>>>
>>>> Hi there,
>>>> I would like to know if there is a way to see the caller ID on my Sip Client (X-Lite for example) of the caller that I answear from a Fifo queue?
>>>> Thanks,
>>>> John
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From john at 247-talk.co.uk  Tue Jan 25 18:22:10 2011
From: john at 247-talk.co.uk (John Carpenter)
Date: Tue, 25 Jan 2011 15:22:10 +0000
Subject: [Freeswitch-users] IVR Bridged Call Dropping after 2 rings
In-Reply-To: <AANLkTikMTAKiakvQFu20xdMh7yrrynJxXd8X9a9RG911@mail.gmail.com>
References: <1295661432.3014.15.camel@John-Home>
	<AANLkTikMTAKiakvQFu20xdMh7yrrynJxXd8X9a9RG911@mail.gmail.com>
Message-ID: <1295968930.3560.8.camel@John-Home>

Hi, I have now traced the problem down to the SIP tunk provider having a
timeout of 10 seconds. If they receive no signalling or RTP for 10
seconds they drop the call. If I had known this earlier I would not have
signed up with them but its too late now.
So the question is how do I get FS to send RTP back to SIP trunk when a
call is being bridged, it currently dies if extension not answered in 10
seconds. Have tried proxy_media and bypass_media without any success. My
extensions are remote and using NAT.

regards, John

On Mon, 2011-01-24 at 12:28 -0800, Michael Collins wrote:

> Can you pastebin a debug log with a siptrace? Also, pastebin your
> dialplan. I think we can help with this but I want to see what you're
> doing before I suggest anything.
> 
> 
> 
> -MC
> 
> 
> On Fri, Jan 21, 2011 at 5:57 PM, John Carpenter <john at 247-talk.co.uk>
> wrote:
> 
>         Hi, I am trying to setup a very simple IVR using LUA. Call
>         arrives from a DID SIP trunk and is answered and message is
>         played ok, after a particular digit is pressed it bridges the
>         call to an extension which is remotely connected. It works but
>         after 2 rings the call to the extension is dropped with a SIP
>         message "BYE" from DID provider. If I just route the call
>         directly to the extension (no IVR) it works fine. It seems
>         like the DID hangs up when the call is bridged to the
>         extension. Have tried same thing using the XML IVR Engine and
>         get exactly the same result. The IVR script is below
>         
>         pathsep = '/'
>         session:setAutoHangup(false);
>         session:answer()
>         prompt = "ivr" .. pathsep .. "247talk.wav"
>         invalid = "ivr" .. pathsep ..
>         "ivr-that_was_an_invalid_entry.wav"
>         freeswitch.consoleLog("INFO", "Prompt file is '" .. prompt ..
>         "'\n")
>         continue = true
>         
>         while( session:ready() == true and continue == true) do
>                 digits = session:playAndGetDigits(1,1,3,7000,"#",
>         prompt, invalid, "\\d+")
>                 if (digits == "1") then
>                         continue = false
>                         session:execute("bridge","sofia/external/2476%
>         91.xxx.xx.xx")
>                 end
>                 if (digits == "2") then
>                         session:execute("bridge","sofia/external/2475%
>         91.xxx.xx.xx")
>                 end
>                 if (digits == "3") then
>                         continue = false
>                         session:execute("bridge","sofia/external/2475%
>         91.xxx.xx.xx")
>                 end
>         end
>         
>         session:hangup()
>         
>         Any help with this greatly appreciated it is driving me nuts.
>         
>         regards, John Carpenter
>         
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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From u2nsam at gmail.com  Tue Jan 25 19:24:50 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 25 Jan 2011 21:54:50 +0530
Subject: [Freeswitch-users] deflect
In-Reply-To: <AANLkTi=F5Lhsk1tjcG5z5LTUjZb4e-cQdNXQjK3OsVD7@mail.gmail.com>
References: <AANLkTiksYLOeki2_XDJC5YwMW6hjUxvdpT7-Mvg7vmKZ@mail.gmail.com>
	<AANLkTi=EMn55D1xGBrWuGgVE_dW_5itoJ4WpzByyagTE@mail.gmail.com>
	<AANLkTi=F5Lhsk1tjcG5z5LTUjZb4e-cQdNXQjK3OsVD7@mail.gmail.com>
Message-ID: <AANLkTinV=xZ_ahs=8DH7n8hADGL4hTOR05hNCWCasA85@mail.gmail.com>

Hi,

Is it possible in this scenario,

I have a call (leg a) to an IVR on FS1 , after the ivr the below statement
is executed,
<action application="bridge" data="{api_on_answer='uuid_transfer  ${uuid}
both'}sofia/external/12127773456 at 192.168.2.130 <9999 at 192.168.2.130>"/>

As the FS1 sends invite to 192.168.2.130 and the call is connected to the
moviephone IVR,
but here what happens is the call is getting disconnected from leg a and the
movie phone ivr 12127773456.



Regds
Sam



On Mon, Jan 24, 2011 at 3:12 PM, Steven Ayre <steveayre at gmail.com> wrote:

> You could try uuid_simplify with the api_on_answer hook
>
> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_simplify
> http://wiki.freeswitch.org/wiki/Variable_api_on_answer
>
> -Steve
>
>
>
> On 24 January 2011 09:05, Sam <u2nsam at gmail.com> wrote:
>
>> Hi,
>>
>> Is it possible by having b2bua in between , would the leg A be deflected
>> to the another FS server from first server ?
>>
>> Regds
>> Sam
>>
>>
>> On Wed, Jan 12, 2011 at 11:42 AM, Sam <u2nsam at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> When call comes on 1 server and plays an application and after execution
>>> of the
>>> application the call is bridge to the other server ,but here after
>>> bridging the call
>>> should refer/deflect to other server, how this can be done ?
>>>
>>> Here just using the deflect variable is not recommended as there is proxy
>>> in between,
>>> so once the call is bridge the next step would be deflect the leg totally
>>> to another server via proxy.
>>>
>>> Regards
>>> Sam
>>>
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Tue Jan 25 19:26:28 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 25 Jan 2011 17:26:28 +0100
Subject: [Freeswitch-users] IVR Bridged Call Dropping after 2 rings
In-Reply-To: <1295968930.3560.8.camel@John-Home>
References: <1295661432.3014.15.camel@John-Home>
	<AANLkTikMTAKiakvQFu20xdMh7yrrynJxXd8X9a9RG911@mail.gmail.com>
	<1295968930.3560.8.camel@John-Home>
Message-ID: <AF03CB60-ACAC-4146-9CD4-DB2AA5EE5D10@ipeva.fr>

Send an audio ringback to them.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 25/01/2011 ? 16:22, John Carpenter a ?crit :

> Hi, I have now traced the problem down to the SIP tunk provider having a timeout of 10 seconds. If they receive no signalling or RTP for 10 seconds they drop the call. If I had known this earlier I would not have signed up with them but its too late now.
> So the question is how do I get FS to send RTP back to SIP trunk when a call is being bridged, it currently dies if extension not answered in 10 seconds. Have tried proxy_media and bypass_media without any success. My extensions are remote and using NAT.
> 
> regards, John
> 
> On Mon, 2011-01-24 at 12:28 -0800, Michael Collins wrote:
>> Can you pastebin a debug log with a siptrace? Also, pastebin your dialplan. I think we can help with this but I want to see what you're doing before I suggest anything.
>> 
>> 
>> -MC
>> 
>> On Fri, Jan 21, 2011 at 5:57 PM, John Carpenter <john at 247-talk.co.uk> wrote:
>> Hi, I am trying to setup a very simple IVR using LUA. Call arrives from a DID SIP trunk and is answered and message is played ok, after a particular digit is pressed it bridges the call to an extension which is remotely connected. It works but after 2 rings the call to the extension is dropped with a SIP message "BYE" from DID provider. If I just route the call directly to the extension (no IVR) it works fine. It seems like the DID hangs up when the call is bridged to the extension. Have tried same thing using the XML IVR Engine and get exactly the same result. The IVR script is below
>> 
>> pathsep = '/'
>> session:setAutoHangup(false);
>> session:answer()
>> prompt = "ivr" .. pathsep .. "247talk.wav"
>> invalid = "ivr" .. pathsep .. "ivr-that_was_an_invalid_entry.wav"
>> freeswitch.consoleLog("INFO", "Prompt file is '" .. prompt .. "'\n")
>> continue = true
>> 
>> while( session:ready() == true and continue == true) do
>>         digits = session:playAndGetDigits(1,1,3,7000,"#", prompt, invalid, "\\d+")
>>         if (digits == "1") then
>>                 continue = false
>>                 session:execute("bridge","sofia/external/2476%91.xxx.xx.xx")
>>         end
>>         if (digits == "2") then
>>                 session:execute("bridge","sofia/external/2475%91.xxx.xx.xx")
>>         end
>>         if (digits == "3") then
>>                 continue = false
>>                 session:execute("bridge","sofia/external/2475%91.xxx.xx.xx")
>>         end
>> end
>> 
>> session:hangup()
>> 
>> Any help with this greatly appreciated it is driving me nuts.
>> 
>> regards, John Carpenter
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Tue Jan 25 19:19:15 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 25 Jan 2011 10:19:15 -0600
Subject: [Freeswitch-users] IVRD
In-Reply-To: <C35D701C93804A8FB9B2F87E424BCD44@e1705>
References: <B890EB1A467549D48DB330F5795906E9@e1705>
	<AANLkTin54y6CJHhSc8DbEZBnx7oBW+4ppD8E3-AQnSk4@mail.gmail.com>
	<A842C399BF77474D86093E7D4FDF2F24@e1705>
	<AANLkTi=CwgFtn3Xv0-9mM5nzsk4hWUpuEDYjeW7dw97=@mail.gmail.com>
	<4CEE1B8B58EE4E6FB13FC4B626A47AE3@e1705>
	<AANLkTikr5_3U6E1WHpsgNExN9a0M2_+papGWgDcoDZ8O@mail.gmail.com>
	<C35D701C93804A8FB9B2F87E424BCD44@e1705>
Message-ID: <AANLkTikNh7vEpnkZaYK+0ersEkJNzLNqDdkCCgQJsXYY@mail.gmail.com>

I just tried it on my box and it works fine.  I'll give you the steps
I did just so you can compare. This is with unaltered FS git HEAD
Make sure you have at least done "make sounds-install" from the main
FS build root.

1) cd libs/esl
2) make perlmod
3) cd perl
4) open new file called test.pl and paste in the demo script from
http://wiki.freeswitch.org/wiki/Ivrd
5) edit only one line:

use lib '/usr/src/freeswitch/libs/esl/perl';

If it is not already right, change this to match the absolute path to
the perl folder you are currently in.
6) chmod a+rx test.pl
7) ../../../fs_ivrd -h 127.0.0.1 -p 8040

8) configure exten in DP: (make sure the path is also correct matching
the line you hacked in the test.pl)

    <extension name="7002">
      <condition field="destination_number" expression="^7002$">
        <action application="set"
data="ivr_path=/usr/src/freeswitch/libs/esl/perl/test.pl"/>
        <action application="socket" data="127.0.0.1:8040 async full"/>
        <action application="info"/>
      </condition>
    </extension>

9) start FS and call 7002


On Mon, Jan 24, 2011 at 10:33 PM, Madovsky <infos at madovsky.org> wrote:
> I tried also the ivrd-demo.php with fs_ivrd from this thread
>
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg14816.html
>
> and add a mail functio in the script to know if fs_ivrd call the php script
> but it doesn't.
>
> I really don't know what's happen.
>
> thanks
>
> ----- Original Message -----
> From: "Anthony Minessale" <anthony.minessale at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, January 24, 2011 7:02 PM
> Subject: Re: [Freeswitch-users] IVRD
>
>
>> you may want to go over everything again:
>> We use that extensively so it's unlikely there is a problem.
>>
>> search your dir for any stale ESL.so or .pm files and recopy them all
>> from your source tree so they match the version of FS you are on.
>>
>>
>> Did you try the exact test example?
>> I can test it on my end tomorrow if you are still stuck.
>>
>>
>>
>> On Mon, Jan 24, 2011 at 5:55 PM, Madovsky <infos at madovsky.org> wrote:
>>> I compiled ESL from the last source tree of FS I have (git from about 5
>>> days
>>> ago)
>>> and followed the instructions on wiki ESL Perl
>>> ESL stuff are in
>>>
>>> /usr/local/lib64/perl5/site_perl/5.10.0/x86_64-linux-thread-multi/ESL
>>>
>>>
>>>
>>> ----- Original Message -----
>>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>> Sent: Monday, January 24, 2011 6:32 PM
>>> Subject: Re: [Freeswitch-users] IVRD
>>>
>>>
>>>> did you maybe update FS and not update all the ESL stuff?
>>>> Do you have it in a nostandard location?
>>>>
>>>>
>>>> On Mon, Jan 24, 2011 at 5:26 PM, Madovsky <infos at madovsky.org> wrote:
>>>>> Apparently the script stalls at
>>>>>
>>>>> ## Create the connection object which is basically an IVR
>>>>> my $con = new ESL::IVR;
>>>>>
>>>>> I'm looking into IVR.pm to know wha'ts happening
>>>>>
>>>>> ----- Original Message -----
>>>>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>> Sent: Monday, January 24, 2011 3:30 PM
>>>>> Subject: Re: [Freeswitch-users] IVRD
>>>>>
>>>>>
>>>>> did you run the fs_ivrd from a shell and look for output in stderr?
>>>>> it has to be related to the script executing failing etc.
>>>>>
>>>>>
>>>>> On Mon, Jan 24, 2011 at 11:57 AM, Madovsky <infos at madovsky.org> wrote:
>>>>>> Sorry I sent this email by mistake without to finish it.
>>>>>> so in my dialplan I have :
>>>>>> <extension name="gluglu">
>>>>>> <condition field="destination_number"
>>>>>> expression="^999$">
>>>>>> <action application="set"
>>>>>> data="ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl"/>
>>>>>> <action application="socket" data="127.0.0.1:9090
>>>>>> full"/>
>>>>>> <action application="answer"/>
>>>>>> <action application="voicemail" data="default
>>>>>> ${domain_name} ${dialed_extension}"/>
>>>>>> <condition>
>>>>>> </extension>
>>>>>>
>>>>>> and when I call this extension in log level 7 I can see :
>>>>>>
>>>>>> Dialplan: sofia/internal/9999999999999 at default Action
>>>>>> set(ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl)
>>>>>> Dialplan: sofia/internal/9999999999999 at default Action
>>>>>> socket(127.0.0.1:9090
>>>>>> full)
>>>>>> Starting ivrd-hello_world.pl...
>>>>>>
>>>>>> and no ivrd-hello_world.pl code is executed in the dialplan unless the
>>>>>> print
>>>>>> "Starting ivrd-hello_world.pl";
>>>>>>
>>>>>> the same if I replace fs_ivrd with server2.pl for example
>>>>>>
>>>>>> Thanks
>>>>>>
>>>>>>
>>>>>> ----- Original Message -----
>>>>>> From: "Madovsky" <infos at madovsky.org>
>>>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>>> Sent: Monday, January 24, 2011 12:48 PM
>>>>>> Subject: Re: [Freeswitch-users] IVRD
>>>>>>
>>>>>>
>>>>>>> Concerning the use of fs_ivrd:
>>>>>>>
>>>>>>> path is ok
>>>>>>> permission is root 755
>>>>>>> perl -c gives ../scripts/perl/ivrd-hello_world.pl syntax OK
>>>>>>>
>>>>>>> in dialplan I have :
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> ----- Original Message -----
>>>>>>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>>>>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>>>> Sent: Monday, January 24, 2011 11:20 AM
>>>>>>> Subject: Re: [Freeswitch-users] IVRD
>>>>>>>
>>>>>>>
>>>>>>> check for proper path and execute permissions on the file and perl -c
>>>>>>> to make sure it compiles.
>>>>>>>
>>>>>>>
>>>>>>> On Mon, Jan 24, 2011 at 12:13 AM, Madovsky <infos at madovsky.org>
>>>>>>> wrote:
>>>>>>>> I tried to use IVRD from wiki example
>>>>>>>>
>>>>>>>> http://wiki.freeswitch.org/wiki/Ivrd
>>>>>>>>
>>>>>>>> and server2.pl in ESL directory
>>>>>>>> copy and paste in my dialplan ans settings
>>>>>>>> so the daemon is running well, but if I attempt
>>>>>>>> to call nothing happens unless hangup.
>>>>>>>> on the log I can see only
>>>>>>>>
>>>>>>>> EXECUTE sofia/internal/9999999999999 at default socket(127.0.0.1:8084
>>>>>>>> full)
>>>>>>>>
>>>>>>>> I tried the tests of troubleshooting without error
>>>>>>>> I don't understand why the events are not received in the perl
>>>>>>>> script
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Anthony Minessale II
>>>>>>>
>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>
>>>>>>> AIM: anthm
>>>>>>> MSN:anthony_minessale at hotmail.com
>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>>
>>>>>>> FreeSWITCH Developer Conference
>>>>>>> sip:888 at conference.freeswitch.org
>>>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>>>> pstn:+19193869900
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org
>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From infos at madovsky.org  Tue Jan 25 19:43:05 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 25 Jan 2011 11:43:05 -0500
Subject: [Freeswitch-users] IVR Bridged Call Dropping after 2 rings
References: <1295661432.3014.15.camel@John-Home><AANLkTikMTAKiakvQFu20xdMh7yrrynJxXd8X9a9RG911@mail.gmail.com>
	<1295968930.3560.8.camel@John-Home>
Message-ID: <BD7AF63A73014E9EAF0BA279392ACAF4@e1705>

what's happen if you send comfort_noise ?
  ----- Original Message ----- 
  From: John Carpenter 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, January 25, 2011 10:22 AM
  Subject: Re: [Freeswitch-users] IVR Bridged Call Dropping after 2 rings


  Hi, I have now traced the problem down to the SIP tunk provider having a timeout of 10 seconds. If they receive no signalling or RTP for 10 seconds they drop the call. If I had known this earlier I would not have signed up with them but its too late now.
  So the question is how do I get FS to send RTP back to SIP trunk when a call is being bridged, it currently dies if extension not answered in 10 seconds. Have tried proxy_media and bypass_media without any success. My extensions are remote and using NAT.

  regards, John

  On Mon, 2011-01-24 at 12:28 -0800, Michael Collins wrote:

    Can you pastebin a debug log with a siptrace? Also, pastebin your dialplan. I think we can help with this but I want to see what you're doing before I suggest anything. 



    -MC


    On Fri, Jan 21, 2011 at 5:57 PM, John Carpenter <john at 247-talk.co.uk> wrote: 
      Hi, I am trying to setup a very simple IVR using LUA. Call arrives from a DID SIP trunk and is answered and message is played ok, after a particular digit is pressed it bridges the call to an extension which is remotely connected. It works but after 2 rings the call to the extension is dropped with a SIP message "BYE" from DID provider. If I just route the call directly to the extension (no IVR) it works fine. It seems like the DID hangs up when the call is bridged to the extension. Have tried same thing using the XML IVR Engine and get exactly the same result. The IVR script is below

      pathsep = '/'
      session:setAutoHangup(false);
      session:answer()
      prompt = "ivr" .. pathsep .. "247talk.wav"
      invalid = "ivr" .. pathsep .. "ivr-that_was_an_invalid_entry.wav"
      freeswitch.consoleLog("INFO", "Prompt file is '" .. prompt .. "'\n")
      continue = true

      while( session:ready() == true and continue == true) do
              digits = session:playAndGetDigits(1,1,3,7000,"#", prompt, invalid, "\\d+")
              if (digits == "1") then
                      continue = false
                      session:execute("bridge","sofia/external/2476%91.xxx.xx.xx")
              end
              if (digits == "2") then
                      session:execute("bridge","sofia/external/2475%91.xxx.xx.xx")
              end
              if (digits == "3") then
                      continue = false
                      session:execute("bridge","sofia/external/2475%91.xxx.xx.xx")
              end
      end

      session:hangup()

      Any help with this greatly appreciated it is driving me nuts.

      regards, John Carpenter 

      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org





_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From steveayre at gmail.com  Tue Jan 25 19:50:41 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 25 Jan 2011 16:50:41 +0000
Subject: [Freeswitch-users] FreeSWITCH 1.0.7
In-Reply-To: <201101250953332657154@asiainfo-linkage.com>
References: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org>
	<4D3D0EE9.4070200@bksys.co.in>
	<201101250953332657154@asiainfo-linkage.com>
Message-ID: <AANLkTikaODmqGrxA993g+Sb5HfbYNqYNGLw3TwcvLHSs@mail.gmail.com>

Read ChangeLog in the 1.0.7 tarball.

There are around 700 lines, so I won't post them here.

-Steve



On 25 January 2011 01:53, liuyp2 <liuyp2 at asiainfo-linkage.com> wrote:

>  What's new features in fs1.0.7?
>
> Thanks!
>
>  ------------------------------
>  liuyp2
> 2011-01-25
>  ------------------------------
>  *????* abubacker
> *?????* 2011-01-24 14:02:25
> *????* FreeSWITCH Users Help
> *???*
> *???* Re: [Freeswitch-users] FreeSWITCH 1.0.7
>
> On Friday 14 January 2011 09:55 PM, Brian West wrote:
>
> *http://latest.freeswitch.org/*
>
> Enjoy!
>
> /b
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
> Great job , Thanks to every one !
>
> --
> Best regards,
> N.Abubacker ,
> Associate system engineer ,
> bk systems pvt ltd ,
> Ph : 9144-43902701
>
> Disclaimer: http://www.bksystems.co.in/email-policy
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Tue Jan 25 20:37:54 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 25 Jan 2011 12:37:54 -0500
Subject: [Freeswitch-users] IVRD
References: <B890EB1A467549D48DB330F5795906E9@e1705><AANLkTin54y6CJHhSc8DbEZBnx7oBW+4ppD8E3-AQnSk4@mail.gmail.com><A842C399BF77474D86093E7D4FDF2F24@e1705><AANLkTi=CwgFtn3Xv0-9mM5nzsk4hWUpuEDYjeW7dw97=@mail.gmail.com><4CEE1B8B58EE4E6FB13FC4B626A47AE3@e1705><AANLkTikr5_3U6E1WHpsgNExN9a0M2_+papGWgDcoDZ8O@mail.gmail.com><C35D701C93804A8FB9B2F87E424BCD44@e1705>
	<AANLkTikNh7vEpnkZaYK+0ersEkJNzLNqDdkCCgQJsXYY@mail.gmail.com>
Message-ID: <179A0E5EEAA149E689A8206CA6542732@e1705>

ok I retry it , go:

2) make perlmod


----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, January 25, 2011 11:19 AM
Subject: Re: [Freeswitch-users] IVRD


>I just tried it on my box and it works fine.  I'll give you the steps
> I did just so you can compare. This is with unaltered FS git HEAD
> Make sure you have at least done "make sounds-install" from the main
> FS build root.
>
> 1) cd libs/esl
> 2) make perlmod
> 3) cd perl
> 4) open new file called test.pl and paste in the demo script from
> http://wiki.freeswitch.org/wiki/Ivrd
> 5) edit only one line:
>
> use lib '/usr/src/freeswitch/libs/esl/perl';
>
> If it is not already right, change this to match the absolute path to
> the perl folder you are currently in.
> 6) chmod a+rx test.pl
> 7) ../../../fs_ivrd -h 127.0.0.1 -p 8040
>
> 8) configure exten in DP: (make sure the path is also correct matching
> the line you hacked in the test.pl)
>
>    <extension name="7002">
>      <condition field="destination_number" expression="^7002$">
>        <action application="set"
> data="ivr_path=/usr/src/freeswitch/libs/esl/perl/test.pl"/>
>        <action application="socket" data="127.0.0.1:8040 async full"/>
>        <action application="info"/>
>      </condition>
>    </extension>
>
> 9) start FS and call 7002
>
>
> On Mon, Jan 24, 2011 at 10:33 PM, Madovsky <infos at madovsky.org> wrote:
>> I tried also the ivrd-demo.php with fs_ivrd from this thread
>>
>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg14816.html
>>
>> and add a mail functio in the script to know if fs_ivrd call the php 
>> script
>> but it doesn't.
>>
>> I really don't know what's happen.
>>
>> thanks
>>
>> ----- Original Message -----
>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, January 24, 2011 7:02 PM
>> Subject: Re: [Freeswitch-users] IVRD
>>
>>
>>> you may want to go over everything again:
>>> We use that extensively so it's unlikely there is a problem.
>>>
>>> search your dir for any stale ESL.so or .pm files and recopy them all
>>> from your source tree so they match the version of FS you are on.
>>>
>>>
>>> Did you try the exact test example?
>>> I can test it on my end tomorrow if you are still stuck.
>>>
>>>
>>>
>>> On Mon, Jan 24, 2011 at 5:55 PM, Madovsky <infos at madovsky.org> wrote:
>>>> I compiled ESL from the last source tree of FS I have (git from about 5
>>>> days
>>>> ago)
>>>> and followed the instructions on wiki ESL Perl
>>>> ESL stuff are in
>>>>
>>>> /usr/local/lib64/perl5/site_perl/5.10.0/x86_64-linux-thread-multi/ESL
>>>>
>>>>
>>>>
>>>> ----- Original Message -----
>>>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Monday, January 24, 2011 6:32 PM
>>>> Subject: Re: [Freeswitch-users] IVRD
>>>>
>>>>
>>>>> did you maybe update FS and not update all the ESL stuff?
>>>>> Do you have it in a nostandard location?
>>>>>
>>>>>
>>>>> On Mon, Jan 24, 2011 at 5:26 PM, Madovsky <infos at madovsky.org> wrote:
>>>>>> Apparently the script stalls at
>>>>>>
>>>>>> ## Create the connection object which is basically an IVR
>>>>>> my $con = new ESL::IVR;
>>>>>>
>>>>>> I'm looking into IVR.pm to know wha'ts happening
>>>>>>
>>>>>> ----- Original Message -----
>>>>>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>>>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>>> Sent: Monday, January 24, 2011 3:30 PM
>>>>>> Subject: Re: [Freeswitch-users] IVRD
>>>>>>
>>>>>>
>>>>>> did you run the fs_ivrd from a shell and look for output in stderr?
>>>>>> it has to be related to the script executing failing etc.
>>>>>>
>>>>>>
>>>>>> On Mon, Jan 24, 2011 at 11:57 AM, Madovsky <infos at madovsky.org> 
>>>>>> wrote:
>>>>>>> Sorry I sent this email by mistake without to finish it.
>>>>>>> so in my dialplan I have :
>>>>>>> <extension name="gluglu">
>>>>>>> <condition field="destination_number"
>>>>>>> expression="^999$">
>>>>>>> <action application="set"
>>>>>>> data="ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl"/>
>>>>>>> <action application="socket" data="127.0.0.1:9090
>>>>>>> full"/>
>>>>>>> <action application="answer"/>
>>>>>>> <action application="voicemail" data="default
>>>>>>> ${domain_name} ${dialed_extension}"/>
>>>>>>> <condition>
>>>>>>> </extension>
>>>>>>>
>>>>>>> and when I call this extension in log level 7 I can see :
>>>>>>>
>>>>>>> Dialplan: sofia/internal/9999999999999 at default Action
>>>>>>> set(ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl)
>>>>>>> Dialplan: sofia/internal/9999999999999 at default Action
>>>>>>> socket(127.0.0.1:9090
>>>>>>> full)
>>>>>>> Starting ivrd-hello_world.pl...
>>>>>>>
>>>>>>> and no ivrd-hello_world.pl code is executed in the dialplan unless 
>>>>>>> the
>>>>>>> print
>>>>>>> "Starting ivrd-hello_world.pl";
>>>>>>>
>>>>>>> the same if I replace fs_ivrd with server2.pl for example
>>>>>>>
>>>>>>> Thanks
>>>>>>>
>>>>>>>
>>>>>>> ----- Original Message -----
>>>>>>> From: "Madovsky" <infos at madovsky.org>
>>>>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>>>> Sent: Monday, January 24, 2011 12:48 PM
>>>>>>> Subject: Re: [Freeswitch-users] IVRD
>>>>>>>
>>>>>>>
>>>>>>>> Concerning the use of fs_ivrd:
>>>>>>>>
>>>>>>>> path is ok
>>>>>>>> permission is root 755
>>>>>>>> perl -c gives ../scripts/perl/ivrd-hello_world.pl syntax OK
>>>>>>>>
>>>>>>>> in dialplan I have :
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> ----- Original Message -----
>>>>>>>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>>>>>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>>>>> Sent: Monday, January 24, 2011 11:20 AM
>>>>>>>> Subject: Re: [Freeswitch-users] IVRD
>>>>>>>>
>>>>>>>>
>>>>>>>> check for proper path and execute permissions on the file and 
>>>>>>>> perl -c
>>>>>>>> to make sure it compiles.
>>>>>>>>
>>>>>>>>
>>>>>>>> On Mon, Jan 24, 2011 at 12:13 AM, Madovsky <infos at madovsky.org>
>>>>>>>> wrote:
>>>>>>>>> I tried to use IVRD from wiki example
>>>>>>>>>
>>>>>>>>> http://wiki.freeswitch.org/wiki/Ivrd
>>>>>>>>>
>>>>>>>>> and server2.pl in ESL directory
>>>>>>>>> copy and paste in my dialplan ans settings
>>>>>>>>> so the daemon is running well, but if I attempt
>>>>>>>>> to call nothing happens unless hangup.
>>>>>>>>> on the log I can see only
>>>>>>>>>
>>>>>>>>> EXECUTE sofia/internal/9999999999999 at default socket(127.0.0.1:8084
>>>>>>>>> full)
>>>>>>>>>
>>>>>>>>> I tried the tests of troubleshooting without error
>>>>>>>>> I don't understand why the events are not received in the perl
>>>>>>>>> script
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> Anthony Minessale II
>>>>>>>>
>>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>>
>>>>>>>> AIM: anthm
>>>>>>>> MSN:anthony_minessale at hotmail.com
>>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>>>
>>>>>>>> FreeSWITCH Developer Conference
>>>>>>>> sip:888 at conference.freeswitch.org
>>>>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>>>>> pstn:+19193869900
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II
>>>>>>
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>
>>>>>> AIM: anthm
>>>>>> MSN:anthony_minessale at hotmail.com
>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>
>>>>>> FreeSWITCH Developer Conference
>>>>>> sip:888 at conference.freeswitch.org
>>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>>> pstn:+19193869900
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org
>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From infos at madovsky.org  Tue Jan 25 20:49:26 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 25 Jan 2011 12:49:26 -0500
Subject: [Freeswitch-users] IVRD
References: <B890EB1A467549D48DB330F5795906E9@e1705><AANLkTin54y6CJHhSc8DbEZBnx7oBW+4ppD8E3-AQnSk4@mail.gmail.com><A842C399BF77474D86093E7D4FDF2F24@e1705><AANLkTi=CwgFtn3Xv0-9mM5nzsk4hWUpuEDYjeW7dw97=@mail.gmail.com><4CEE1B8B58EE4E6FB13FC4B626A47AE3@e1705><AANLkTikr5_3U6E1WHpsgNExN9a0M2_+papGWgDcoDZ8O@mail.gmail.com><C35D701C93804A8FB9B2F87E424BCD44@e1705>
	<AANLkTikNh7vEpnkZaYK+0ersEkJNzLNqDdkCCgQJsXYY@mail.gmail.com>
Message-ID: <C36AB7DACBAF4E0899E8E636DCC968CA@e1705>

argh sorry for last email.

1) done
2) make perlmod = OK
make perlmod
make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" 
CFLAGS="-I/home/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb 
 -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable 
 -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" 
CXXFLAGS="-I/home/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb 
 -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" 
CXX_CFLAGS="" -C perl
make[1]: Entering directory `/home/src/freeswitch/libs/esl/perl'
make[1]: Nothing to be done for `all'.
make[1]: Leaving directory `/home/src/freeswitch/libs/esl/perl'
3) OK
4) OK
5) OK
6) OK
7) OK
8) OK
9) OK

result :

root at node250 perl]# Starting ivrd-hello_world.pl...
Can't call method "getHeader" on an undefined value at 
/usr/src/freeswitch/libs/esl/perl/ESL/IVR.pm line 11.

it seems to work better now....




----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, January 25, 2011 11:19 AM
Subject: Re: [Freeswitch-users] IVRD


>I just tried it on my box and it works fine.  I'll give you the steps
> I did just so you can compare. This is with unaltered FS git HEAD
> Make sure you have at least done "make sounds-install" from the main
> FS build root.
>
> 1) cd libs/esl
> 2) make perlmod
> 3) cd perl
> 4) open new file called test.pl and paste in the demo script from
> http://wiki.freeswitch.org/wiki/Ivrd
> 5) edit only one line:
>
> use lib '/usr/src/freeswitch/libs/esl/perl';
>
> If it is not already right, change this to match the absolute path to
> the perl folder you are currently in.
> 6) chmod a+rx test.pl
> 7) ../../../fs_ivrd -h 127.0.0.1 -p 8040
>
> 8) configure exten in DP: (make sure the path is also correct matching
> the line you hacked in the test.pl)
>
>    <extension name="7002">
>      <condition field="destination_number" expression="^7002$">
>        <action application="set"
> data="ivr_path=/usr/src/freeswitch/libs/esl/perl/test.pl"/>
>        <action application="socket" data="127.0.0.1:8040 async full"/>
>        <action application="info"/>
>      </condition>
>    </extension>
>
> 9) start FS and call 7002
>
>
> On Mon, Jan 24, 2011 at 10:33 PM, Madovsky <infos at madovsky.org> wrote:
>> I tried also the ivrd-demo.php with fs_ivrd from this thread
>>
>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg14816.html
>>
>> and add a mail functio in the script to know if fs_ivrd call the php 
>> script
>> but it doesn't.
>>
>> I really don't know what's happen.
>>
>> thanks
>>
>> ----- Original Message -----
>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, January 24, 2011 7:02 PM
>> Subject: Re: [Freeswitch-users] IVRD
>>
>>
>>> you may want to go over everything again:
>>> We use that extensively so it's unlikely there is a problem.
>>>
>>> search your dir for any stale ESL.so or .pm files and recopy them all
>>> from your source tree so they match the version of FS you are on.
>>>
>>>
>>> Did you try the exact test example?
>>> I can test it on my end tomorrow if you are still stuck.
>>>
>>>
>>>
>>> On Mon, Jan 24, 2011 at 5:55 PM, Madovsky <infos at madovsky.org> wrote:
>>>> I compiled ESL from the last source tree of FS I have (git from about 5
>>>> days
>>>> ago)
>>>> and followed the instructions on wiki ESL Perl
>>>> ESL stuff are in
>>>>
>>>> /usr/local/lib64/perl5/site_perl/5.10.0/x86_64-linux-thread-multi/ESL
>>>>
>>>>
>>>>
>>>> ----- Original Message -----
>>>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Monday, January 24, 2011 6:32 PM
>>>> Subject: Re: [Freeswitch-users] IVRD
>>>>
>>>>
>>>>> did you maybe update FS and not update all the ESL stuff?
>>>>> Do you have it in a nostandard location?
>>>>>
>>>>>
>>>>> On Mon, Jan 24, 2011 at 5:26 PM, Madovsky <infos at madovsky.org> wrote:
>>>>>> Apparently the script stalls at
>>>>>>
>>>>>> ## Create the connection object which is basically an IVR
>>>>>> my $con = new ESL::IVR;
>>>>>>
>>>>>> I'm looking into IVR.pm to know wha'ts happening
>>>>>>
>>>>>> ----- Original Message -----
>>>>>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>>>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>>> Sent: Monday, January 24, 2011 3:30 PM
>>>>>> Subject: Re: [Freeswitch-users] IVRD
>>>>>>
>>>>>>
>>>>>> did you run the fs_ivrd from a shell and look for output in stderr?
>>>>>> it has to be related to the script executing failing etc.
>>>>>>
>>>>>>
>>>>>> On Mon, Jan 24, 2011 at 11:57 AM, Madovsky <infos at madovsky.org> 
>>>>>> wrote:
>>>>>>> Sorry I sent this email by mistake without to finish it.
>>>>>>> so in my dialplan I have :
>>>>>>> <extension name="gluglu">
>>>>>>> <condition field="destination_number"
>>>>>>> expression="^999$">
>>>>>>> <action application="set"
>>>>>>> data="ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl"/>
>>>>>>> <action application="socket" data="127.0.0.1:9090
>>>>>>> full"/>
>>>>>>> <action application="answer"/>
>>>>>>> <action application="voicemail" data="default
>>>>>>> ${domain_name} ${dialed_extension}"/>
>>>>>>> <condition>
>>>>>>> </extension>
>>>>>>>
>>>>>>> and when I call this extension in log level 7 I can see :
>>>>>>>
>>>>>>> Dialplan: sofia/internal/9999999999999 at default Action
>>>>>>> set(ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl)
>>>>>>> Dialplan: sofia/internal/9999999999999 at default Action
>>>>>>> socket(127.0.0.1:9090
>>>>>>> full)
>>>>>>> Starting ivrd-hello_world.pl...
>>>>>>>
>>>>>>> and no ivrd-hello_world.pl code is executed in the dialplan unless 
>>>>>>> the
>>>>>>> print
>>>>>>> "Starting ivrd-hello_world.pl";
>>>>>>>
>>>>>>> the same if I replace fs_ivrd with server2.pl for example
>>>>>>>
>>>>>>> Thanks
>>>>>>>
>>>>>>>
>>>>>>> ----- Original Message -----
>>>>>>> From: "Madovsky" <infos at madovsky.org>
>>>>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>>>> Sent: Monday, January 24, 2011 12:48 PM
>>>>>>> Subject: Re: [Freeswitch-users] IVRD
>>>>>>>
>>>>>>>
>>>>>>>> Concerning the use of fs_ivrd:
>>>>>>>>
>>>>>>>> path is ok
>>>>>>>> permission is root 755
>>>>>>>> perl -c gives ../scripts/perl/ivrd-hello_world.pl syntax OK
>>>>>>>>
>>>>>>>> in dialplan I have :
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> ----- Original Message -----
>>>>>>>> From: "Anthony Minessale" <anthony.minessale at gmail.com>
>>>>>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>>>>> Sent: Monday, January 24, 2011 11:20 AM
>>>>>>>> Subject: Re: [Freeswitch-users] IVRD
>>>>>>>>
>>>>>>>>
>>>>>>>> check for proper path and execute permissions on the file and 
>>>>>>>> perl -c
>>>>>>>> to make sure it compiles.
>>>>>>>>
>>>>>>>>
>>>>>>>> On Mon, Jan 24, 2011 at 12:13 AM, Madovsky <infos at madovsky.org>
>>>>>>>> wrote:
>>>>>>>>> I tried to use IVRD from wiki example
>>>>>>>>>
>>>>>>>>> http://wiki.freeswitch.org/wiki/Ivrd
>>>>>>>>>
>>>>>>>>> and server2.pl in ESL directory
>>>>>>>>> copy and paste in my dialplan ans settings
>>>>>>>>> so the daemon is running well, but if I attempt
>>>>>>>>> to call nothing happens unless hangup.
>>>>>>>>> on the log I can see only
>>>>>>>>>
>>>>>>>>> EXECUTE sofia/internal/9999999999999 at default socket(127.0.0.1:8084
>>>>>>>>> full)
>>>>>>>>>
>>>>>>>>> I tried the tests of troubleshooting without error
>>>>>>>>> I don't understand why the events are not received in the perl
>>>>>>>>> script
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> Anthony Minessale II
>>>>>>>>
>>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>>
>>>>>>>> AIM: anthm
>>>>>>>> MSN:anthony_minessale at hotmail.com
>>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>>>
>>>>>>>> FreeSWITCH Developer Conference
>>>>>>>> sip:888 at conference.freeswitch.org
>>>>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>>>>> pstn:+19193869900
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II
>>>>>>
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>
>>>>>> AIM: anthm
>>>>>> MSN:anthony_minessale at hotmail.com
>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>
>>>>>> FreeSWITCH Developer Conference
>>>>>> sip:888 at conference.freeswitch.org
>>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>>> pstn:+19193869900
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org
>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From brian at freeswitch.org  Tue Jan 25 20:57:54 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 25 Jan 2011 11:57:54 -0600
Subject: [Freeswitch-users] Is it possible to use att_xfer on channels
	already bridged via loopback?
In-Reply-To: <AANLkTimmE9oirGUznj6RTZHYkf=pWZLUg-5SHfRqbnFH@mail.gmail.com>
References: <AANLkTimWAPV52349=Yge2zjnJyP40EdgDJXgyH-QcOCd@mail.gmail.com>
	<AANLkTi=WKADqDk-qqevfAh3vR1mZgZdwEEey_cuFUEz8@mail.gmail.com>
	<AANLkTimmE9oirGUznj6RTZHYkf=pWZLUg-5SHfRqbnFH@mail.gmail.com>
Message-ID: <F54ED3D2-5938-4E17-8C1D-7695342BE074@freeswitch.org>

I couldn't replicate it on mine but i'll try his configs.

/b

On Jan 25, 2011, at 10:01 AM, Dmitry Sytchev wrote:

> http://pastebin.freeswitch.org/15141 - dialplan features.xml
> http://pastebin.freeswitch.org/15140 - extraction from main dialplan
> to only test att_xfers
> http://pastebin.freeswitch.org/15138 - freeswitch.xml
> 
> Do you also need my debug logs or any other info?
> My problem can be reproduced on my hardware with this configuration.
> There is a problem with MOH also - if first att_xfer fails for any
> reason, there will be no MOH for transferee on subsequent att_xfer
> calls, just silence
> 
> 
> 2011/1/24 Michael Collins <msc at freeswitch.org>:
>> Dmitry,
>> I'd like to try this myself on one of my boxes. Would you pastebin the
>> dialplan you are using?
>> Thanks,
>> MC

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From pipiwei03 at gmail.com  Tue Jan 25 21:27:05 2011
From: pipiwei03 at gmail.com (pipiwei03)
Date: Wed, 26 Jan 2011 02:27:05 +0800
Subject: [Freeswitch-users]  How to make the heard DTMF tone more longer
Message-ID: <AANLkTin4Ubshv1W=JrF7kroq+pXoc7EfrJXawHOJs0Qm@mail.gmail.com>

Dear all:

I try to send DTMF tone.

 I use two method in lua
1.queue_dtm after bridge
2. send_dtmf in a session

these two methods are working and call my phone successfully

But, I got a problem is...
After bridge, I hear soon DTMF tones.
The heard DTMF tones are so short. It is very strange to feel as broken.

Even though I set and increase the tone_duration as below:
      queue_dtmf "12345678 at 5000"
  or send_dtmf "12345678 at 5000"

The heard DTMF tones also very short, but the duration between inter-digit
become longer...
Somebody tell me how to make "heard DTMF tone" longer...or silent the dial
tone and sent DTMF tones.

Thanks a lot for your support.


Best regards,

Joy
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From brian at freeswitch.org  Tue Jan 25 21:41:09 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 25 Jan 2011 12:41:09 -0600
Subject: [Freeswitch-users] How to make the heard DTMF tone more longer
In-Reply-To: <AANLkTin4Ubshv1W=JrF7kroq+pXoc7EfrJXawHOJs0Qm@mail.gmail.com>
References: <AANLkTin4Ubshv1W=JrF7kroq+pXoc7EfrJXawHOJs0Qm@mail.gmail.com>
Message-ID: <7BF6DEBF-AF5B-4A17-B0E6-03F9B3B83E77@freeswitch.org>

This is all going to depend on your gateway or device that turns the DTMF back into tones .

/b

On Jan 25, 2011, at 12:27 PM, pipiwei03 wrote:

> Dear all:
> 
> I try to send DTMF tone. 
> 
>  I use two method in lua 
> 1.queue_dtm after bridge 
> 2. send_dtmf in a session
> 
> these two methods are working and call my phone successfully
> 
> But, I got a problem is...
> After bridge, I hear soon DTMF tones. 
> The heard DTMF tones are so short. It is very strange to feel as broken.
> 
> Even though I set and increase the tone_duration as below:
>       queue_dtmf "12345678 at 5000" 
>   or send_dtmf "12345678 at 5000"
> 
> The heard DTMF tones also very short, but the duration between inter-digit become longer...
> Somebody tell me how to make "heard DTMF tone" longer...or silent the dial tone and sent DTMF tones.
> 
> Thanks a lot for your support.
> 
> 
> Best regards,
> 
> Joy
> ________

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From jerry.richards at teotech.com  Tue Jan 25 21:46:51 2011
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 25 Jan 2011 10:46:51 -0800
Subject: [Freeswitch-users] Bridge String Length Too Short
Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D6960660335E94@VA3DIAXVS351.RED001.local>

Hello All,

I am seeing my bridge statement getting truncated.  How may I extend the length of the string it will accept?

Here is the actual bridge log:
EXECUTE sofia/internal/2003 at 192.168.72.144:5060 bridge({ignore_early_media=true}{presence_id=2002 at 192.168.72.144}sofia/internal/sip:2002 at 192.168.72.140:5060;transport=udp,sofia/internal/sip:2001 at 192.168.72.93:5060;transport=udp,[lcr_carrier=Carrier / Location 2/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]op)

But it should be:
EXECUTE sofia/internal/2003 at 192.168.72.144:5060 bridge({ignore_early_media=true}{presence_id=2002 at 192.168.72.144}sofia/internal/sip:2002 at 192.168.72.140:5060;transport=udp,sofia/internal/sip:2001 at 192.168.72.93:5060;transport=udp,[lcr_carrier=Carrier / Location 2/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]openzap/smg_prid/a/15248152341 at g1)

Thanks,
Jerry
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From msc at freeswitch.org  Tue Jan 25 21:49:28 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Jan 2011 10:49:28 -0800
Subject: [Freeswitch-users] IVR Bridged Call Dropping after 2 rings
In-Reply-To: <1295968930.3560.8.camel@John-Home>
References: <1295661432.3014.15.camel@John-Home>
	<AANLkTikMTAKiakvQFu20xdMh7yrrynJxXd8X9a9RG911@mail.gmail.com>
	<1295968930.3560.8.camel@John-Home>
Message-ID: <AANLkTi=-RD1adnuYSkdUagR5PA685dALdnX3xAtfOG=D@mail.gmail.com>

Do they absolutely require RTP? If so then add a "pre_answer" app to your
dialplan before you bridge to your local endpoint. See
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_pre_answer for more
information.

-MC

On Tue, Jan 25, 2011 at 7:22 AM, John Carpenter <john at 247-talk.co.uk> wrote:

>  Hi, I have now traced the problem down to the SIP tunk provider having a
> timeout of 10 seconds. If they receive no signalling or RTP for 10 seconds
> they drop the call. If I had known this earlier I would not have signed up
> with them but its too late now.
> So the question is how do I get FS to send RTP back to SIP trunk when a
> call is being bridged, it currently dies if extension not answered in 10
> seconds. Have tried proxy_media and bypass_media without any success. My
> extensions are remote and using NAT.
>
> regards, John
>
>
> On Mon, 2011-01-24 at 12:28 -0800, Michael Collins wrote:
>
> Can you pastebin a debug log with a siptrace? Also, pastebin your dialplan.
> I think we can help with this but I want to see what you're doing before I
> suggest anything.
>
>
>
>  -MC
>
>  On Fri, Jan 21, 2011 at 5:57 PM, John Carpenter <john at 247-talk.co.uk>
> wrote:
>
>  Hi, I am trying to setup a very simple IVR using LUA. Call arrives from a
> DID SIP trunk and is answered and message is played ok, after a particular
> digit is pressed it bridges the call to an extension which is remotely
> connected. It works but after 2 rings the call to the extension is dropped
> with a SIP message "BYE" from DID provider. If I just route the call
> directly to the extension (no IVR) it works fine. It seems like the DID
> hangs up when the call is bridged to the extension. Have tried same thing
> using the XML IVR Engine and get exactly the same result. The IVR script is
> below
>
> pathsep = '/'
> session:setAutoHangup(false);
> session:answer()
> prompt = "ivr" .. pathsep .. "247talk.wav"
> invalid = "ivr" .. pathsep .. "ivr-that_was_an_invalid_entry.wav"
> freeswitch.consoleLog("INFO", "Prompt file is '" .. prompt .. "'\n")
> continue = true
>
> while( session:ready() == true and continue == true) do
>         digits = session:playAndGetDigits(1,1,3,7000,"#", prompt, invalid,
> "\\d+")
>         if (digits == "1") then
>                 continue = false
>
> session:execute("bridge","sofia/external/2476%91.xxx.xx.xx")
>         end
>         if (digits == "2") then
>
> session:execute("bridge","sofia/external/2475%91.xxx.xx.xx")
>         end
>         if (digits == "3") then
>                 continue = false
>
> session:execute("bridge","sofia/external/2475%91.xxx.xx.xx")
>         end
> end
>
> session:hangup()
>
> Any help with this greatly appreciated it is driving me nuts.
>
> regards, John Carpenter
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>  _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Jan 25 21:50:31 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Jan 2011 10:50:31 -0800
Subject: [Freeswitch-users] Bridge String Length Too Short
In-Reply-To: <2BF7FB90DF25EA4485949F3AF2B9D6960660335E94@VA3DIAXVS351.RED001.local>
References: <2BF7FB90DF25EA4485949F3AF2B9D6960660335E94@VA3DIAXVS351.RED001.local>
Message-ID: <AANLkTi=KgLZvAudkCSvnrLtsrGwwdCYiS8rynCZenjAQ@mail.gmail.com>

How are you generating this call? Is it by chance using fs_cli -x ?
-MC

On Tue, Jan 25, 2011 at 10:46 AM, Jerry Richards <jerry.richards at teotech.com
> wrote:

> Hello All,
>
>
>
> I am seeing my bridge statement getting truncated.  How may I extend the
> length of the string it will accept?
>
>
>
> Here is the actual bridge log:
>
> EXECUTE sofia/internal/2003 at 192.168.72.144:5060bridge({ignore_early_media=true}{presence_id=
> 2002 at 192.168.72.144}sofia/internal/sip:2002 at 192.168.72.140:5060
> ;transport=udp,sofia/internal/sip:2001 at 192.168.72.93:5060;transport=udp,[lcr_carrier=Carrier
> / Location 2/INTERNAL PRI
> TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]op)
>
>
>
> But it should be:
>
> EXECUTE sofia/internal/2003 at 192.168.72.144:5060bridge({ignore_early_media=true}{presence_id=
> 2002 at 192.168.72.144}sofia/internal/sip:2002 at 192.168.72.140:5060
> ;transport=udp,sofia/internal/sip:2001 at 192.168.72.93:5060;transport=udp,[lcr_carrier=Carrier
> / Location 2/INTERNAL PRI
> TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]openzap/smg_prid/a/15248152341 at g1
> )
>
>
>
> Thanks,
>
> Jerry
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Jan 25 21:53:56 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Jan 2011 10:53:56 -0800
Subject: [Freeswitch-users] deflect
In-Reply-To: <AANLkTinV=xZ_ahs=8DH7n8hADGL4hTOR05hNCWCasA85@mail.gmail.com>
References: <AANLkTiksYLOeki2_XDJC5YwMW6hjUxvdpT7-Mvg7vmKZ@mail.gmail.com>
	<AANLkTi=EMn55D1xGBrWuGgVE_dW_5itoJ4WpzByyagTE@mail.gmail.com>
	<AANLkTi=F5Lhsk1tjcG5z5LTUjZb4e-cQdNXQjK3OsVD7@mail.gmail.com>
	<AANLkTinV=xZ_ahs=8DH7n8hADGL4hTOR05hNCWCasA85@mail.gmail.com>
Message-ID: <AANLkTimfCVsJFCVDv1Az-8ERh8UZRqx1cKG1hY+GfKwd@mail.gmail.com>

I strongly recommend that you capture the debug output and drop it into a
pastebin at pastebin.freeswitch.org. You may also wish to capture the sip
traffic as well. If you are using fs_cli then you already see the debug
level console output. To get the sip traffic inline with the debug output
just do "sofia global siptrace on".

-MC

On Tue, Jan 25, 2011 at 8:24 AM, Sam <u2nsam at gmail.com> wrote:

> Hi,
>
> Is it possible in this scenario,
>
> I have a call (leg a) to an IVR on FS1 , after the ivr the below statement
> is executed,
> <action application="bridge" data="{api_on_answer='uuid_transfer  ${uuid}
> both'}sofia/external/12127773456 at 192.168.2.130 <9999 at 192.168.2.130>"/>
>
> As the FS1 sends invite to 192.168.2.130 and the call is connected to the
> moviephone IVR,
> but here what happens is the call is getting disconnected from leg a and
> the movie phone ivr 12127773456.
>
>
>
> Regds
>
> Sam
>
>
>
> On Mon, Jan 24, 2011 at 3:12 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> You could try uuid_simplify with the api_on_answer hook
>>
>> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_simplify
>> http://wiki.freeswitch.org/wiki/Variable_api_on_answer
>>
>> -Steve
>>
>>
>>
>> On 24 January 2011 09:05, Sam <u2nsam at gmail.com> wrote:
>>
>>>  Hi,
>>>
>>> Is it possible by having b2bua in between , would the leg A be deflected
>>> to the another FS server from first server ?
>>>
>>> Regds
>>> Sam
>>>
>>>
>>> On Wed, Jan 12, 2011 at 11:42 AM, Sam <u2nsam at gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> When call comes on 1 server and plays an application and after execution
>>>> of the
>>>> application the call is bridge to the other server ,but here after
>>>> bridging the call
>>>> should refer/deflect to other server, how this can be done ?
>>>>
>>>> Here just using the deflect variable is not recommended as there is
>>>> proxy in between,
>>>> so once the call is bridge the next step would be deflect the leg
>>>> totally to another server via proxy.
>>>>
>>>> Regards
>>>> Sam
>>>>
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jerry.richards at teotech.com  Tue Jan 25 22:29:34 2011
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 25 Jan 2011 11:29:34 -0800
Subject: [Freeswitch-users] Bridge String Length Too Short
In-Reply-To: <AANLkTi=KgLZvAudkCSvnrLtsrGwwdCYiS8rynCZenjAQ@mail.gmail.com>
References: <2BF7FB90DF25EA4485949F3AF2B9D6960660335E94@VA3DIAXVS351.RED001.local>
	<AANLkTi=KgLZvAudkCSvnrLtsrGwwdCYiS8rynCZenjAQ@mail.gmail.com>
Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D6960660335ECB@VA3DIAXVS351.RED001.local>

No, I have a C-App to invoke the bridge statement:
switch_core_session_execute_application(session, "bridge", results.dialstring);

Thanks,
Jerry


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Tuesday, January 25, 2011 10:51 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Bridge String Length Too Short

How are you generating this call? Is it by chance using fs_cli -x ?
-MC
On Tue, Jan 25, 2011 at 10:46 AM, Jerry Richards <jerry.richards at teotech.com<mailto:jerry.richards at teotech.com>> wrote:
Hello All,

I am seeing my bridge statement getting truncated.  How may I extend the length of the string it will accept?

Here is the actual bridge log:
EXECUTE sofia/internal/2003 at 192.168.72.144:5060<http://2003 at 192.168.72.144:5060> bridge({ignore_early_media=true}{presence_id=2002 at 192.168.72.144<mailto:2002 at 192.168.72.144>}sofia/internal/sip:2002 at 192.168.72.140:5060;transport=udp,sofia/internal/sip:2001 at 192.168.72.93:5060;transport=udp,[lcr_carrier=Carrier / Location 2/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]op)

But it should be:
EXECUTE sofia/internal/2003 at 192.168.72.144:5060<http://2003 at 192.168.72.144:5060> bridge({ignore_early_media=true}{presence_id=2002 at 192.168.72.144<mailto:2002 at 192.168.72.144>}sofia/internal/sip:2002 at 192.168.72.140:5060;transport=udp,sofia/internal/sip:2001 at 192.168.72.93:5060;transport=udp,[lcr_carrier=Carrier / Location 2/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]openzap/smg_prid/a/15248152341 at g1)

Thanks,
Jerry

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From anthony.minessale at gmail.com  Tue Jan 25 22:35:52 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 25 Jan 2011 13:35:52 -0600
Subject: [Freeswitch-users] Bridge String Length Too Short
In-Reply-To: <2BF7FB90DF25EA4485949F3AF2B9D6960660335ECB@VA3DIAXVS351.RED001.local>
References: <2BF7FB90DF25EA4485949F3AF2B9D6960660335E94@VA3DIAXVS351.RED001.local>
	<AANLkTi=KgLZvAudkCSvnrLtsrGwwdCYiS8rynCZenjAQ@mail.gmail.com>
	<2BF7FB90DF25EA4485949F3AF2B9D6960660335ECB@VA3DIAXVS351.RED001.local>
Message-ID: <AANLkTi==dKu9EdkyNmCJRXUs-RrQ0_FDciSXk8mugtBV@mail.gmail.com>

looks like a custom app, check the number of bytes in results.dialstring

On Tue, Jan 25, 2011 at 1:29 PM, Jerry Richards
<jerry.richards at teotech.com> wrote:
> No, I have a C-App to invoke the bridge statement:
>
> switch_core_session_execute_application(session, "bridge",
> results.dialstring);
>
>
>
> Thanks,
>
> Jerry
>
>
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
> Collins
> Sent: Tuesday, January 25, 2011 10:51 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Bridge String Length Too Short
>
>
>
> How are you generating this call? Is it by chance using fs_cli -x ?
>
> -MC
>
> On Tue, Jan 25, 2011 at 10:46 AM, Jerry Richards
> <jerry.richards at teotech.com> wrote:
>
> Hello All,
>
>
>
> I am seeing my bridge statement getting truncated.? How may I extend the
> length of the string it will accept?
>
>
>
> Here is the actual bridge log:
>
> EXECUTE sofia/internal/2003 at 192.168.72.144:5060
> bridge({ignore_early_media=true}{presence_id=2002 at 192.168.72.144}sofia/internal/sip:2002 at 192.168.72.140:5060;transport=udp,sofia/internal/sip:2001 at 192.168.72.93:5060;transport=udp,[lcr_carrier=Carrier
> / Location 2/INTERNAL PRI
> TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]op)
>
>
>
> But it should be:
>
> EXECUTE sofia/internal/2003 at 192.168.72.144:5060
> bridge({ignore_early_media=true}{presence_id=2002 at 192.168.72.144}sofia/internal/sip:2002 at 192.168.72.140:5060;transport=udp,sofia/internal/sip:2001 at 192.168.72.93:5060;transport=udp,[lcr_carrier=Carrier
> / Location 2/INTERNAL PRI
> TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]openzap/smg_prid/a/15248152341 at g1)
>
>
>
> Thanks,
>
> Jerry
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From segmond at gmail.com  Tue Jan 25 22:20:47 2011
From: segmond at gmail.com (Segmond Yunsai)
Date: Tue, 25 Jan 2011 14:20:47 -0500
Subject: [Freeswitch-users] Is this possible with google voice and freeswitch
Message-ID: <AANLkTikrLPe5a=8cox-hpL4rpdMtZT+Vs4C0x4bLjYT8@mail.gmail.com>

Hello,

I want to have multiple jingle profiles.

when a call is made to a google voice number in the jingle profiles

the freeswitch voice will intercept it with a lua script
the script will somehow figure out which google voice number/jingle profile
was called
hold the call, so it doesn't disconnect/get forwarded to voicemail

do some things in the lua script

then make an outgoing call using google voice or a gateway and bridge em
with the incoming call

is this possible?

i've been able to answer google calls to an extension
i've been able to intercept with lua
i haven't been able to get dialing out with google voice
and i also can't bridge google voice incoming call without an outgoing call
through my gateway.
the outgoing phone rings, and the phone calling the google voice number
rings and then goes to the voicemail.

freeswitch gurus please help.
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From msc at freeswitch.org  Tue Jan 25 22:44:45 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Jan 2011 11:44:45 -0800
Subject: [Freeswitch-users] intercom in lua dialplan
In-Reply-To: <4D3EF036.80701@ewetel.de>
References: <4D3EF036.80701@ewetel.de>
Message-ID: <AANLkTik8jrN_hzom=VEQZfx39YqMgZ15OE8kzZWqJ8iO@mail.gmail.com>

What are the log lines leading up to the "executed last dialplan
instruction"?
-MC

On Tue, Jan 25, 2011 at 7:45 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> I try to intercept a call in lua script. For the intercepting extension
> I do this:
>
>
> if (dialed_ext:len()==6 and dialed_ext:sub(1,2) == "**") then
>        uuid=api:execute("hash", "select/last_caller/"..dialed_ext:sub(3))
>        if (uuid and uuid:len()> 2) then
>                session:setAutoHangup(false)
>                session:execute("intercept", uuid)
>                return
>        end
> end
>
>
> This results to a hangup of the intercepting extension while intercepted
> extension hangs up (completed_elsewhere) and caller/originator's call is
> connected.
>
> FS logs shows this for the intercepting extension:
>
> switch_core_state_machine.c:189 sofia/internal_et/8000 at spklw.x has
> executed the last dialplan instruction, hanging up.
>
>
> When I use the ACTIONS table like this:
>
> if (dialed_ext:len()==6 and dialed_ext:sub(1,2) == "**") then
>        uuid=api:execute("hash", "select/last_caller/"..dialed_ext:sub(3))
>        if (uuid and uuid:len()> 2) then
>                session:setAutoHangup(false)
>                table.insert(ACTIONS, {"intercept", uuid})
>                return
>        end
> end
>
>
> then intercept/pickup works fine. Is there a trick to get it work with
> "session:execute("intercept", uuid)" ?
>
> regards
> helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.10 (MingW32)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/
>
> iEYEARECAAYFAk0+8DYACgkQ4tZeNddg3dyKwACeKR4lsxUv3G26pdNp8GGsbWNg
> /OIAn2EIl2Vq+qibtodGd9vjjLfqKv8s
> =2fTE
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From msc at freeswitch.org  Tue Jan 25 23:03:42 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Jan 2011 12:03:42 -0800
Subject: [Freeswitch-users] Javascript IVR session question
In-Reply-To: <CDF5E00590C9BA4AB96BE7CDD28C86AE0787301809@FS1.wvds.local>
References: <CDF5E00590C9BA4AB96BE7CDD28C86AE07872CFA48@FS1.wvds.local>
	<AANLkTimaJDm8Fc1EKk1hKrnF60cWBXyg4bzdQ+77NMG=@mail.gmail.com>
	<CDF5E00590C9BA4AB96BE7CDD28C86AE0787301809@FS1.wvds.local>
Message-ID: <AANLkTi=DAkwwfQ10WGnkW0CaFmZ-9QEXV-VC8UJoJWq8@mail.gmail.com>

So the customer records a message, then hangs up. Then you want a separate
call made to the technician. So the call made to the tech (which is user/202
I presume) is just another IVR?

>From what I can see, you've got a one-legged call between the caller and the
IVR, and you disconnect that when the caller is done recording his message.
The "proper" thing to do would be to generate a new call (instead of
attempting to bridge a non-existent call) to the tech. Check out this wiki
page for the syntax:
http://wiki.freeswitch.org/wiki/Session_originate

-MC

On Tue, Jan 25, 2011 at 12:57 AM, Erik Dekkers <erik.dekkers at wvds.nl> wrote:

> Michael,
>
>
>
> It?s like this. I would like to let the customer records a message. Then a
> technician should get called and the previous recorded message should be
> played.
>
> So after the customer has records his message, the line should be hung up.
> Then a new call should be made from the IVR to the technician.
>
>
>
> Regards,
>
>
>
> Erik
>
>
>
> *Van:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Namens *Michael Collins
> *Verzonden:* maandag 24 januari 2011 21:14
> *Aan:* FreeSWITCH Users Help
> *Onderwerp:* Re: [Freeswitch-users] Javascript IVR session question
>
>
>
> If I read this correctly you are hanging up the channel that you later wish
> to bridge to user/202. Why do you need to hangup? Perhaps you could describe
> a little more about the application? I'm sure we can help you iron out the
> details.
>
>
>
> -MC
>
> On Sun, Jan 23, 2011 at 12:48 PM, Erik Dekkers <erik.dekkers at wvds.nl>
> wrote:
>
> Hey ppl,
>
>
>
> At the moment im building a Javascript based IVR but im kind of stuck on a
> part.
>
>
>
> The IVR should do this:
>
>
>
> - Answer session (working)
>
> - Play some wav files (working)
>
> - Record a message to file (working)
>
> - Hang up the first session (working)
>
> - Call the second session (not working)
>
> - Play the previous recorded file (not working)
>
>
>
> After I dial the second session, the console says "channel is hungup
> already". How should i do this?
>
>
>
> Kind regards,
>
>
>
> Erik Dekkers (wvds-nl on IRC)
>
>
>
>
>
>
>
> my script:
>
>
>
> var allDigits = "";
>
> function on_dtmf(session, type, digits, arg)
> {
>     if (digits.digit == "#") {
>         return allDigits;
>     }
>
>     if (digits.digit == "*") {
>         return false; //stop the recording.
>     }
>
>     console_log("digit: " + digits.digit + "\n");
>     allDigits += digits.digit;
>     return(allDigits);
> }
>
> session.answer();
>
> if (session.ready()) {
>     allDigits = "";
>     var rtn;
>
>     rtn = session.streamFile("/home/edekkers/sounds/10_spreek_in.wav",
> on_dtmf, "");
>
>     if (session.ready()) {
>         var tmp_Filename = "/tmp/test.wav";
>
>         if (session.ready()) {
>             rtn = session.recordFile(tmp_Filename, on_dtmf, "", 120);
>             }
>
>         rtn =
> session.streamFile("/home/edekkers/sounds/11_bericht_is_ontvangen.wav",
> on_dtmf, "");
>
>         if (session.ready()) {
>             session.hangup();
>         }
>     }
> }
>
> session.execute("bridge","user/202")
> if (session.ready()) {
>     session.streamFile("/tmp/test.wav");
> }
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Jan 25 23:32:43 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Jan 2011 12:32:43 -0800
Subject: [Freeswitch-users] How exactly execute_extension work?
In-Reply-To: <4D3EE138.3060500@tagnet.ru>
References: <4D3EE138.3060500@tagnet.ru>
Message-ID: <AANLkTik7NS3zYenMhC7qP_ANk2h61UA8MwcepgCsujBe@mail.gmail.com>

The execute_extension app is similar to a GOSUB command in BASIC: it goes
out, executes some stuff, and then comes back. However, keep in mind that
when you execute an extension it does loop through the dialplan looking for
the destination_number that you specified. Depending on your needs you
probably just need to create a distinct dialplan context for this special
extension. Then you execute the extension in a separate context and only the
extensions within that context will be checked against destination_number.

-MC

On Tue, Jan 25, 2011 at 6:42 AM, Boris Kovalenko <boris at tagnet.ru> wrote:

> Hello!
>
>     From one of my extension I call another extension in another
> context. I suppose that exactly this extension should be executed, but
> instead I get all previous extension executed to. So below is my
> configuration. We are calling start_hunting in context tagnet.ru. I
> suppose start_hunting extension should be executed but no, hunting
> begins from the start of context and all extensions before start_hunting
> are executed too. auto_hunt=true solves the problem. So the question is
> - is this normal behaviour of execute_extension or am I doing something
> wrong?
>
> <!-- We are hunting only for TAGNet numbering plan -->
> <extension name="ext_hunting">
> <condition field="${v_tagnet_dst}"
> expression="^(73435230[0-9]{3}|73435494989|7343549499[0-1])$">
> <action application="javascript" data="$${scripts_dir}/toll_check.js
> ${toll_allow} local"/>
> <action application="set" data="v_tagnet_hunting=true" inline="true"/>
> <action application="log" data="NOTICE [top.ctx] - hunting in tagnet.ru"/>
> <action application="log" data="NOTICE ${v_tagnet_dst} $1"/>
> <action application="execute_extension" data="start_hunting XML tagnet.ru
> "/>
> </condition>
> </extension>
>
> Context tagnet.ru:
> <!--
>     This context is for home networking soft phones and cheap telephony.
> -->
> <?xml version="1.0" encoding="utf-8"?>
> <include>
> <context name="tagnet.ru">
> <extension name="unloop">
> <condition field="${unroll_loops}" expression="^true$"/>
> <condition field="${sip_looped_call}" expression="^true$">
> <action application="deflect" data="${destination_number}"/>
> </condition>
> </extension>
>
> <!--
>             Destination number normalization
>         -->
> <extension name="ext_translate_dst" continue="true">
> <!-- <condition field="${v_tagnet_hunting}" expression="^true$"
> break="on-true"/> -->
> <condition field="${destination_number}" expression="^(1[0-9]{3})$"
> break="on-true">
> <action application="set" data="v_tagnet_dst=$1" inline="true"/>
> </condition>
> <condition field="${destination_number}" expression="^(\d+)$"
> break="on-true">
> <action application="set" data="v_tagnet_dst=$1" inline="true"/>
> <action application="execute_extension" data="ext_translate_dst XML
> features"/>
> </condition>
> </extension>
>
> <!--
>             Source number normalization
>         -->
> <extension name="ext_translate_src" continue="true">
> <condition field="${caller_id_number}" expression="^2([0-9]{4})$"
> break="on-true">
> <action application="set" data="caller_id_number=7343549$1" inline="true"/>
> <action application="set" data="ani=7343549$1" inline="true"/>
> </condition>
> </extension>
>
> <!-- Here we start hunting process -->
> <extension name="start_hunting" continue="true">
> <condition field="destination_number" expression="^start_hunting$"/>
> </extension>
>
> <!--
>                 You can place files in the default directory to get
> included.
>         -->
> <X-PRE-PROCESS cmd="include" data="tagnet.ru/*.xml"/>
>
> <!-- Here we stop hunting process -->
> <extension name="stop_hunting">
> <condition field="${v_tagnet_hunting}" expression="true"/>
> </extension>
>
> <!--
>             Transfer call to the TOP context
>         -->
> <extension name="top.ctx">
> <condition field="${v_tagnet_dst}" expression="^(\d+)$">
> <action application="log" data="DEBUG [top.ctx] - transfer to top.ctx"/>
>
> <!-- Caller ID normalization -->
> <action application="execute_extension" data="ext_translate_src XML
> features"/>
>
> <action application="transfer" data="$1 XML top.ctx"/>
> </condition>
> </extension>
> </context>
> </include>
>
>
>
>
> --
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   (3435) 494991
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jerry.richards at teotech.com  Wed Jan 26 02:32:42 2011
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 25 Jan 2011 15:32:42 -0800
Subject: [Freeswitch-users] Bridge String Length Too Short (Resolved)
Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D6960660336014@VA3DIAXVS351.RED001.local>

FYI, I found the problem.  I had an sprintf in the C-App that was only copying 256 characters.

Thanks,
Jerry


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jerry Richards
Sent: Tuesday, January 25, 2011 11:30 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Bridge String Length Too Short

No, I have a C-App to invoke the bridge statement:
switch_core_session_execute_application(session, "bridge", results.dialstring);

Thanks,
Jerry


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Tuesday, January 25, 2011 10:51 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Bridge String Length Too Short

How are you generating this call? Is it by chance using fs_cli -x ?
-MC
On Tue, Jan 25, 2011 at 10:46 AM, Jerry Richards <jerry.richards at teotech.com<mailto:jerry.richards at teotech.com>> wrote:
Hello All,

I am seeing my bridge statement getting truncated.  How may I extend the length of the string it will accept?

Here is the actual bridge log:
EXECUTE sofia/internal/2003 at 192.168.72.144:5060<http://2003 at 192.168.72.144:5060> bridge({ignore_early_media=true}{presence_id=2002 at 192.168.72.144<mailto:2002 at 192.168.72.144>}sofia/internal/sip:2002 at 192.168.72.140:5060;transport=udp,sofia/internal/sip:2001 at 192.168.72.93:5060;transport=udp,[lcr_carrier=Carrier / Location 2/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]op)

But it should be:
EXECUTE sofia/internal/2003 at 192.168.72.144:5060<http://2003 at 192.168.72.144:5060> bridge({ignore_early_media=true}{presence_id=2002 at 192.168.72.144<mailto:2002 at 192.168.72.144>}sofia/internal/sip:2002 at 192.168.72.140:5060;transport=udp,sofia/internal/sip:2001 at 192.168.72.93:5060;transport=udp,[lcr_carrier=Carrier / Location 2/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]openzap/smg_prid/a/15248152341 at g1)

Thanks,
Jerry

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
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From jerry.richards at teotech.com  Wed Jan 26 02:39:52 2011
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 25 Jan 2011 15:39:52 -0800
Subject: [Freeswitch-users] Difference Between "realm" and "challenge-realm"
	in sip_profiles
Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D696066033601A@VA3DIAXVS351.RED001.local>

Hello All,

What is the difference between the parameters "realm" and "challenge-realm" in the conf/sip_profiles tree?

Thanks,
Jerry

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From anthony.minessale at gmail.com  Wed Jan 26 03:12:36 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 25 Jan 2011 18:12:36 -0600
Subject: [Freeswitch-users] FS HA
In-Reply-To: <AANLkTim3=HVjX5GZVfcVjtSCm-2kuWatyVS+YV0RpNCB@mail.gmail.com>
References: <AANLkTi=_BPuE5KX4D48JDoOW9U3netsr9uDU9Lo5CgYO@mail.gmail.com>
	<FACE40F5-BB80-479A-BFFC-F07F6BA681D2@freeswitch.org>
	<350D44B3-A176-47A9-84F4-72EB62F08299@gmail.com>
	<AANLkTimJLMjPJwthN8AhDq=1_JnDbdmocbgd=-=usyTT@mail.gmail.com>
	<AANLkTi=ksT-ZkdOyXiVQkir7a9z2-58PcVirmv4xs5Kg@mail.gmail.com>
	<AANLkTikeU5FAgE-HU=0a5whf3JiD5VUCkYkCCRt4u=R0@mail.gmail.com>
	<AANLkTim3=HVjX5GZVfcVjtSCm-2kuWatyVS+YV0RpNCB@mail.gmail.com>
Message-ID: <AANLkTimB-jxEjrSD7pp45DcAmCmZH_q-dycT25c1gn06@mail.gmail.com>

There is an xml_cdr snapshot in sip_recovery sql table that contains
the data used for the recovery.
Also based on your logs, you are not using latest git.  I would try
that first before anything else.



On Tue, Jan 25, 2011 at 7:24 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
> yap, both using the same config
>
>
> can you advice where is FS getting the SDP info for the re-INVITE ?
>
>
> On Tue, Jan 25, 2011 at 2:03 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>> Are you using two machines for the HA? do both have the same configs?
>> -Avi
>>
>> On Tue, Jan 25, 2011 at 11:41 AM, Tihomir Culjaga <tculjaga at gmail.com>
>> wrote:
>> > Here is the debug: http://pastebin.freeswitch.org/15133
>> >
>> > i have set verbose_sdp=true in vars.xml as.
>> > ? <X-PRE-PROCESS cmd="set" data="verbose_sdp=true"/>
>> >
>> > but not much to be seen of the verbose thing in the debug...
>> >
>> > Still, FS is sending a re-INVITE with wrong SDP. The call to be
>> > recovered is
>> > using ALAW... and ULAW is not supported.
>> > FS sends a re-INVITE with ULAW and CN in SDP. The re-INVITE is rejected
>> > due
>> > to incompatible SDP.
>> >
>> > Where does FS get the information for the SDP in re-INVITE message?
>> >
>> >
>> > please advice,
>> > T.
>> >
>> >
>> >
>> > On Tue, Jan 25, 2011 at 12:01 AM, Tihomir Culjaga <tculjaga at gmail.com>
>> > wrote:
>> >>
>> >> yap, i do have PCMA ... and the debug shows it correctly :=)
>> >>
>> >> i will try to see what it does with verbose. Post new debug tomorrow.
>> >>
>> >> ty.
>> >>
>> >>
>> >> On Mon, Jan 24, 2011 at 11:29 PM, Steven Ayre <steveayre at gmail.com>
>> >> wrote:
>> >>>
>> >>> Brian, it has PCMU in the sdp, but Sofia thinks it's setting PCMA?
>> >>>
>> >>> Steve on iPhone
>> >>>
>> >>> On 24 Jan 2011, at 20:17, Brian West <brian at freeswitch.org> wrote:
>> >>>
>> >>> > What makes you think that fails? ?It has ULAW and CN in the codec
>> >>> > list!
>> >>> > ?Sounds like you need the verbose sdp... set the global variable
>> >>> > "verbose_sdp=true"
>> >>> >
>> >>> > /b
>> >>> >
>> >>> > On Jan 24, 2011, at 1:50 PM, Tihomir Culjaga wrote:
>> >>> >
>> >>> >>
>> >>> >>
>> >>> >> i configured FS HA and looks like its trying to recover the call ..
>> >>> >> but the re-INVITE fails due to "wrong/missed" codec capability.
>> >>> >
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >
>> >>> >
>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From msc at freeswitch.org  Wed Jan 26 03:30:33 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Jan 2011 16:30:33 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Wednesday Jan 26 -
	Vestec Speech Recognition
Message-ID: <AANLkTinQHXs7w=+3=a4S+ZA5=3O6Dd-B3iBTjuATfeY2@mail.gmail.com>

Hello!

I wanted to let everyone know that we have special guests on the conference
call this Wednesday: Kashif Kahn and Dr. Jin-Myung Won, both from Vestec.
Dr. Won has been deeply involved in ASR development and will be making
himself available to answer your technical questions. Please join the
conference call and be part of this great discussion about advanced ASR
technology!

-Michael

Agenda page: http://wiki.freeswitch.org/wiki/FS_weekly_2011_01_26
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From john at 247-talk.co.uk  Wed Jan 26 03:35:50 2011
From: john at 247-talk.co.uk (John Carpenter)
Date: Wed, 26 Jan 2011 00:35:50 +0000
Subject: [Freeswitch-users] IVR Bridged Call Dropping after 2 rings
In-Reply-To: <AF03CB60-ACAC-4146-9CD4-DB2AA5EE5D10@ipeva.fr>
References: <1295661432.3014.15.camel@John-Home>
	<AANLkTikMTAKiakvQFu20xdMh7yrrynJxXd8X9a9RG911@mail.gmail.com>
	<1295968930.3560.8.camel@John-Home>
	<AF03CB60-ACAC-4146-9CD4-DB2AA5EE5D10@ipeva.fr>
Message-ID: <1296002150.3560.10.camel@John-Home>

Thanks for all the suggestions, will not be able to test until tomorrow,
will post results then

regards, John

On Tue, 2011-01-25 at 17:26 +0100, David Ponzone wrote:

> Send an audio ringback to them.
> 
> 
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
> ? l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion non autoris?e est interdite. Tout message ?lectronique est
> susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre
> de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes
> pas destinataire de ce message, merci de le d?truire imm?diatement et
> d'avertir l'exp?diteur.
> 
> 
> 
> 
> 
> 
> 
> 
> 
> Le 25/01/2011 ? 16:22, John Carpenter a ?crit :
> 
> 
> 
> > Hi, I have now traced the problem down to the SIP tunk provider
> > having a timeout of 10 seconds. If they receive no signalling or RTP
> > for 10 seconds they drop the call. If I had known this earlier I
> > would not have signed up with them but its too late now.
> > So the question is how do I get FS to send RTP back to SIP trunk
> > when a call is being bridged, it currently dies if extension not
> > answered in 10 seconds. Have tried proxy_media and bypass_media
> > without any success. My extensions are remote and using NAT.
> > 
> > regards, John
> > 
> > On Mon, 2011-01-24 at 12:28 -0800, Michael Collins wrote:
> > 
> > > Can you pastebin a debug log with a siptrace? Also, pastebin your
> > > dialplan. I think we can help with this but I want to see what
> > > you're doing before I suggest anything.
> > > 
> > > 
> > > -MC
> > > 
> > > On Fri, Jan 21, 2011 at 5:57 PM, John Carpenter
> > > <john at 247-talk.co.uk> wrote:
> > > 
> > >         Hi, I am trying to setup a very simple IVR using LUA. Call
> > >         arrives from a DID SIP trunk and is answered and message
> > >         is played ok, after a particular digit is pressed it
> > >         bridges the call to an extension which is remotely
> > >         connected. It works but after 2 rings the call to the
> > >         extension is dropped with a SIP message "BYE" from DID
> > >         provider. If I just route the call directly to the
> > >         extension (no IVR) it works fine. It seems like the DID
> > >         hangs up when the call is bridged to the extension. Have
> > >         tried same thing using the XML IVR Engine and get exactly
> > >         the same result. The IVR script is below
> > >         
> > >         pathsep = '/'
> > >         session:setAutoHangup(false);
> > >         session:answer()
> > >         prompt = "ivr" .. pathsep .. "247talk.wav"
> > >         invalid = "ivr" .. pathsep ..
> > >         "ivr-that_was_an_invalid_entry.wav"
> > >         freeswitch.consoleLog("INFO", "Prompt file is '" ..
> > >         prompt .. "'\n")
> > >         continue = true
> > >         
> > >         while( session:ready() == true and continue == true) do
> > >                 digits = session:playAndGetDigits(1,1,3,7000,"#",
> > >         prompt, invalid, "\\d+")
> > >                 if (digits == "1") then
> > >                         continue = false
> > >         
> > >         session:execute("bridge","sofia/external/2476%
> > >         91.xxx.xx.xx")
> > >                 end
> > >                 if (digits == "2") then
> > >         
> > >         session:execute("bridge","sofia/external/2475%
> > >         91.xxx.xx.xx")
> > >                 end
> > >                 if (digits == "3") then
> > >                         continue = false
> > >         
> > >         session:execute("bridge","sofia/external/2475%
> > >         91.xxx.xx.xx")
> > >                 end
> > >         end
> > >         
> > >         session:hangup()
> > >         
> > >         Any help with this greatly appreciated it is driving me
> > >         nuts.
> > >         
> > >         regards, John Carpenter 
> > >         
> > >         _______________________________________________
> > >         FreeSWITCH-users mailing list
> > >         FreeSWITCH-users at lists.freeswitch.org
> > >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >         http://www.freeswitch.org
> > >         
> > > 
> > > 
> > > 
> > > 
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > 
> > 
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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From arun.chinnachamy at cognizant.com  Wed Jan 26 06:15:18 2011
From: arun.chinnachamy at cognizant.com (Arun Chinnachamy)
Date: Tue, 25 Jan 2011 22:15:18 -0500
Subject: [Freeswitch-users] Issue when Transferring Outgoing call.
Message-ID: <C964FBF6.36F6%arun.chinnachamy@cognizant.com>

I am facing an issue in FS. I am able to transfer an incoming call from
Gateway to local registered user but i am not able to do the same for
outgoing call to gateway. FS is trying to call the local user through
gateway. any suggestions?

This e-mail and any files transmitted with it are for the sole use of the intended recipient(s) and may contain confidential and privileged information.
If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies of the original message. 
Any unauthorised review, use, disclosure, dissemination, forwarding, printing or copying of this email or any action taken in reliance on this e-mail is strictly prohibited and may be unlawful.
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From brian at freeswitch.org  Wed Jan 26 06:31:10 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 25 Jan 2011 21:31:10 -0600
Subject: [Freeswitch-users] Issue when Transferring Outgoing call.
In-Reply-To: <C964FBF6.36F6%arun.chinnachamy@cognizant.com>
References: <C964FBF6.36F6%arun.chinnachamy@cognizant.com>
Message-ID: <52F15585-1C90-4973-9052-024A97301610@freeswitch.org>

Can you elaborate on this logs or something would help?  Also remove that annoying unenforceable signature when emailing a public mailing list please.

/b

On Jan 25, 2011, at 9:15 PM, Arun Chinnachamy wrote:

> I am facing an issue in FS. I am able to transfer an incoming call from Gateway to local registered user but i am not able to do the same for outgoing call to gateway. FS is trying to call the local user through gateway. any suggestions?
> This e-mail and any files transmitted with it are for the sole use of the intended recipient(s) and may contain confidential and privileged information.
> If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies of the original message. 
> Any unauthorised review, use, disclosure, dissemination, forwarding, printing or copying of this email or any action taken in reliance on this e-mail is strictly prohibited and may be unlawful.
> ____

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From infos at madovsky.org  Wed Jan 26 08:03:10 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 26 Jan 2011 00:03:10 -0500
Subject: [Freeswitch-users] ESL php client
Message-ID: <0EE97D79ED6E47B783526E4C4E084D56@e1705>

is it safe to remove warnings as error ?
I'd like to correct it but don't know enough C language
to do it.
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, January 25, 2011 12:28 AM
  Subject: ESL php client


  no luck from libs/esl
  make phpmod

  make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/home/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/home/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C php
  make[1]: Entering directory `/home/src/freeswitch/libs/esl/php'
  g++  -I/home/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable -I/usr/local/include/php -I/usr/local/include/php/main -I/usr/local/include/php/TSRM -I/usr/local/include/php/Zend -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o
  cc1plus: warnings being treated as errors
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  esl_wrap.cpp:2565: error: deprecated conversion from string constant to ?char*?
  make[1]: *** [esl_wrap.o] Error 1
  make[1]: Leaving directory `/home/src/freeswitch/libs/esl/php'
  make: *** [phpmod] Error 2

  PHP 4.4.9 is instlalled on my server

  Thanks
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From infos at madovsky.org  Wed Jan 26 08:18:37 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 26 Jan 2011 00:18:37 -0500
Subject: [Freeswitch-users] make phpmod and test.php
Message-ID: <ECEFD2A90E5C43BCAA276827632D1533@e1705>

after removed warn as errors and compile ESL.so
and did on bash command

php test.php

Parse error: syntax error, unexpected T_CLASS in /home/src/freeswitch/libs/esl/php/ESL.php on line 30


I just want to try to listen ESL events.... ;)
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From infos at madovsky.org  Wed Jan 26 08:26:39 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 26 Jan 2011 00:26:39 -0500
Subject: [Freeswitch-users] make phpmod and test.php
Message-ID: <70DF9C14348A4760BCBB0944910D9E22@e1705>

Seems that ESL.php is for PHP5 only ?
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, January 26, 2011 12:18 AM
  Subject: make phpmod and test.php


  after removed warn as errors and compile ESL.so
  and did on bash command

  php test.php

  Parse error: syntax error, unexpected T_CLASS in /home/src/freeswitch/libs/esl/php/ESL.php on line 30


  I just want to try to listen ESL events.... ;)
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From u2nsam at gmail.com  Wed Jan 26 08:43:25 2011
From: u2nsam at gmail.com (Sam)
Date: Wed, 26 Jan 2011 11:13:25 +0530
Subject: [Freeswitch-users] hi David
Message-ID: <AANLkTimZPu6fGRq_Y-73P06T_vnZaQcnejAR7L51jT+w@mail.gmail.com>

Hi David,

the call is not processing, can you check why is the error.

http://pastebin.freeswitch.org/15149

Regards
Samir
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From chat2jesse at gmail.com  Wed Jan 26 09:27:52 2011
From: chat2jesse at gmail.com (jesse)
Date: Tue, 25 Jan 2011 22:27:52 -0800
Subject: [Freeswitch-users] ESL php client
In-Reply-To: <0EE97D79ED6E47B783526E4C4E084D56@e1705>
References: <0EE97D79ED6E47B783526E4C4E084D56@e1705>
Message-ID: <AANLkTikkEfHEvpSmu9sAed0J_1ufy2tHm34t1Lo0zRp7@mail.gmail.com>

yes!

you need to remove -Werror -Wno-unused-variable from PICKY and
CXXFLAGS, then build will work


On Tue, Jan 25, 2011 at 9:03 PM, Madovsky <infos at madovsky.org> wrote:
> is it safe to remove warnings as error ?
> I'd like to correct it but don't know enough C language
> to do it.
>
> ----- Original Message -----
> From: Madovsky
> To: freeswitch-users at lists.freeswitch.org
> Sent: Tuesday, January 25, 2011 12:28 AM
> Subject: ESL php client
> no luck from libs/esl
> make phpmod
>
> make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
> CFLAGS="-I/home/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb
> -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror
> -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes"
> CXXFLAGS="-I/home/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g
> -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable"
> CXX_CFLAGS="" -C php
> make[1]: Entering directory `/home/src/freeswitch/libs/esl/php'
> g++? -I/home/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb
> -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable
> -I/usr/local/include/php -I/usr/local/include/php/main
> -I/usr/local/include/php/TSRM -I/usr/local/include/php/Zend
> -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o
> cc1plus: warnings being treated as errors
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> esl_wrap.cpp:2565: error: deprecated conversion from string constant to
> ?char*?
> make[1]: *** [esl_wrap.o] Error 1
> make[1]: Leaving directory `/home/src/freeswitch/libs/esl/php'
> make: *** [phpmod] Error 2
> PHP 4.4.9 is instlalled on my server
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From david.ponzone at ipeva.fr  Wed Jan 26 10:06:10 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 26 Jan 2011 08:06:10 +0100
Subject: [Freeswitch-users] hi David
In-Reply-To: <AANLkTimZPu6fGRq_Y-73P06T_vnZaQcnejAR7L51jT+w@mail.gmail.com>
References: <AANLkTimZPu6fGRq_Y-73P06T_vnZaQcnejAR7L51jT+w@mail.gmail.com>
Message-ID: <F4034B63-AC44-46D5-8EA0-44D5F2B1F768@ipeva.fr>

David Who ?
Which issue are you talking about ?
You should have replied to the original thread, it would help.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 26/01/2011 ? 06:43, Sam a ?crit :

> Hi David,
> 
> the call is not processing, can you check why is the error.
> 
> http://pastebin.freeswitch.org/15149
> 
> Regards
> Samir
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From helmut.kuper at ewetel.de  Wed Jan 26 10:48:41 2011
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 26 Jan 2011 08:48:41 +0100
Subject: [Freeswitch-users] intercom in lua dialplan
In-Reply-To: <AANLkTik8jrN_hzom=VEQZfx39YqMgZ15OE8kzZWqJ8iO@mail.gmail.com>
References: <4D3EF036.80701@ewetel.de>
	<AANLkTik8jrN_hzom=VEQZfx39YqMgZ15OE8kzZWqJ8iO@mail.gmail.com>
Message-ID: <4D3FD1D9.2@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Michael,

you can find the complete log of this process in pastebin:
http://pastebin.freeswitch.org/15150

44180000 is A-Party calling 8367
8000 is C-Party which tries to intercept via **8367


Am 25.01.2011 20:44, schrieb Michael Collins:
> What are the log lines leading up to the "executed last dialplan
> instruction"?


regards
helmut
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.10 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/

iEYEARECAAYFAk0/0dkACgkQ4tZeNddg3dx/kQCgk96NGvr777y6EC1Jz9vtsP8L
9AUAmwaBJZte6PmV13gsypaWnEHD+ty7
=xuM4
-----END PGP SIGNATURE-----


From erik.dekkers at wvds.nl  Wed Jan 26 11:00:33 2011
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Wed, 26 Jan 2011 09:00:33 +0100
Subject: [Freeswitch-users] Javascript IVR session question
In-Reply-To: <AANLkTi=DAkwwfQ10WGnkW0CaFmZ-9QEXV-VC8UJoJWq8@mail.gmail.com>
References: <CDF5E00590C9BA4AB96BE7CDD28C86AE07872CFA48@FS1.wvds.local>
	<AANLkTimaJDm8Fc1EKk1hKrnF60cWBXyg4bzdQ+77NMG=@mail.gmail.com>
	<CDF5E00590C9BA4AB96BE7CDD28C86AE0787301809@FS1.wvds.local>
	<AANLkTi=DAkwwfQ10WGnkW0CaFmZ-9QEXV-VC8UJoJWq8@mail.gmail.com>
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE078730181C@FS1.wvds.local>

Michael,

This is where I was looking for! While reading your answer I remember it again how to do this :)
Thank you very much.

Kind regards,

Erik

Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Michael Collins
Verzonden: dinsdag 25 januari 2011 21:04
Aan: FreeSWITCH Users Help
Onderwerp: Re: [Freeswitch-users] Javascript IVR session question

So the customer records a message, then hangs up. Then you want a separate call made to the technician. So the call made to the tech (which is user/202 I presume) is just another IVR?

>From what I can see, you've got a one-legged call between the caller and the IVR, and you disconnect that when the caller is done recording his message. The "proper" thing to do would be to generate a new call (instead of attempting to bridge a non-existent call) to the tech. Check out this wiki page for the syntax:
http://wiki.freeswitch.org/wiki/Session_originate

-MC

On Tue, Jan 25, 2011 at 12:57 AM, Erik Dekkers <erik.dekkers at wvds.nl<mailto:erik.dekkers at wvds.nl>> wrote:
Michael,

It's like this. I would like to let the customer records a message. Then a technician should get called and the previous recorded message should be played.
So after the customer has records his message, the line should be hung up. Then a new call should be made from the IVR to the technician.

Regards,

Erik

Van: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] Namens Michael Collins
Verzonden: maandag 24 januari 2011 21:14
Aan: FreeSWITCH Users Help
Onderwerp: Re: [Freeswitch-users] Javascript IVR session question

If I read this correctly you are hanging up the channel that you later wish to bridge to user/202. Why do you need to hangup? Perhaps you could describe a little more about the application? I'm sure we can help you iron out the details.

-MC
On Sun, Jan 23, 2011 at 12:48 PM, Erik Dekkers <erik.dekkers at wvds.nl<mailto:erik.dekkers at wvds.nl>> wrote:
Hey ppl,

At the moment im building a Javascript based IVR but im kind of stuck on a part.

The IVR should do this:

- Answer session (working)
- Play some wav files (working)
- Record a message to file (working)
- Hang up the first session (working)
- Call the second session (not working)
- Play the previous recorded file (not working)

After I dial the second session, the console says "channel is hungup already". How should i do this?

Kind regards,

Erik Dekkers (wvds-nl on IRC)



my script:

var allDigits = "";
function on_dtmf(session, type, digits, arg)
{
    if (digits.digit == "#") {
        return allDigits;
    }
    if (digits.digit == "*") {
        return false; //stop the recording.
    }
    console_log("digit: " + digits.digit + "\n");
    allDigits += digits.digit;
    return(allDigits);
}
session.answer();
if (session.ready()) {
    allDigits = "";
    var rtn;
    rtn = session.streamFile("/home/edekkers/sounds/10_spreek_in.wav", on_dtmf, "");
    if (session.ready()) {
        var tmp_Filename = "/tmp/test.wav";
        if (session.ready()) {
            rtn = session.recordFile(tmp_Filename, on_dtmf, "", 120);
            }
        rtn = session.streamFile("/home/edekkers/sounds/11_bericht_is_ontvangen.wav", on_dtmf, "");
        if (session.ready()) {
            session.hangup();
        }
    }
}
session.execute("bridge","user/202")
if (session.ready()) {
    session.streamFile("/tmp/test.wav");
}



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
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From arun.chinnachamy at cognizant.com  Wed Jan 26 08:14:28 2011
From: arun.chinnachamy at cognizant.com (Arun Chinnachamy)
Date: Wed, 26 Jan 2011 00:14:28 -0500
Subject: [Freeswitch-users] Issue when Transferring Outgoing call.
In-Reply-To: <52F15585-1C90-4973-9052-024A97301610@freeswitch.org>
Message-ID: <C96517E4.3707%arun.chinnachamy@cognizant.com>

> This message is in MIME format. Since your mail reader does not understand
this format, some or all of this message may not be legible.

--B_3378845668_10517720
Content-type: text/plain;
	charset="US-ASCII"
Content-transfer-encoding: 7bit

Thanks Brian. Please find more information below.

I have two clients 1000 and 1001 registered to FS under domain 192.168.1.10.
When I received a call from Gateway (say extension 102 and IP Address
192.168.1.50) via FS, I can receive the call at 1000 and I am successful at
transferring (Refer) the call from 1000 to 1001. Everything works fine.

But when I call the gateway 102 from the client 1000, the call is
established fine. When I tried to transfer the call from 1000 to 1001, the
transfer failed. I checked the logs and also captured the packets using wire
shark. 

FS is receiving the Refer packet. Instead of sending the Invite to
1001 at 192.168.1.10, the invite was sent to 1001 at 192.168.1.50 for which the
Gateway is replying with User not registered. I found in this link (
http://lists.freeswitch.org/pipermail/freeswitch-users/2010-June/058984.html
) that It is an issue fixed using dialplan but even after using the latest
dial plan I am getting the same behavior from FS.

Please suggest.

P.S: the Signature is being appended by my server which I do not have
control over. It annoys me but can not help it.


On 1/25/11 10:31 PM, "Brian West" <brian at freeswitch.org> wrote:

> Can you elaborate on this logs or something would help?  Also remove that
> annoying unenforceable signature when emailing a public mailing list please.
> 
> /b
> 
> On Jan 25, 2011, at 9:15 PM, Arun Chinnachamy wrote:
> 
>> I am facing an issue in FS. I am able to transfer an incoming call from
>> Gateway to local registered user but i am not able to do the same for
>> outgoing call to gateway. FS is trying to call the local user through
>> gateway. any suggestions?
>> 
>> ____
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


This e-mail and any files transmitted with it are for the sole use of the intended recipient(s) and may contain confidential and privileged information.
If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies of the original message. 
Any unauthorised review, use, disclosure, dissemination, forwarding, printing or copying of this email or any action taken in reliance on this e-mail is strictly prohibited and may be unlawful.

--B_3378845668_10517720
Content-type: text/html;
	charset="utf-8"
Content-transfer-encoding: quoted-printable

<HTML>
<HEAD>
<TITLE>Re:=20[Freeswitch-users]=20Issue=20when=20Transferring=20Outgoing=
=20call.</TITLE>
</HEAD>
<BODY>
<FONT=20FACE=3D"Calibri,=20Verdana,=20Helvetica,=20Arial"><SPAN=20STYLE=
=3D'font-size:11pt'>Thanks=20Brian.=20Please=20find=20more=20information=
=20below.<BR>
<BR>
I=20have=20two=20clients=201000=20and=201001=20registered=20to=20FS=20und=
er=20domain=20192.168.1.10.=20When=20I=20received=20a=20call=20from=20Gat=
eway=20(say=20extension=20102=20and=20IP=20Address=20192.168.1.50)=20via=
=20FS,=20I=20can=20receive=20the=20call=20at=201000=20and=20I=20am=20succ=
essful=20at=20transferring=20(Refer)=20the=20call=20from=201000=20to=2010=
01.=20Everything=20works=20fine.<BR>
<BR>
But=20when=20I=20call=20the=20gateway=20102=20from=20the=20client=201000,=
=20the=20call=20is=20established=20fine.=20When=20I=20tried=20to=20transf=
er=20the=20call=20from=201000=20to=201001,=20the=20transfer=20failed.=20I=
=20checked=20the=20logs=20and=20also=20captured=20the=20packets=20using=
=20wire=20shark.=20<BR>
<BR>
FS=20is=20receiving=20the=20Refer=20packet.=20Instead=20of=20sending=20th=
e=20Invite=20to=20<a=20href=3D"1001 at 192.168.1.10">1001 at 192.168.1.10</a>,=
=20the=20invite=20was=20sent=20to=20<a=20href=3D"1001 at 192.168.1.50">1001@=
192.168.1.50</a>=20for=20which=20the=20Gateway=20is=20replying=20with=20U=
ser=20not=20registered.=20I=20found=20in=20this=20link=20(=20<a=20href=3D=
"http://lists.freeswitch.org/pipermail/freeswitch-users/2010-June/058984.=
html">http://lists.freeswitch.org/pipermail/freeswitch-users/2010-June/05=
8984.html</a>)=20that=20It=20is=20an=20issue=20fixed=20using=20dialplan=
=20but=20even=20after=20using=20the=20latest=20dial=20plan=20I=20am=20get=
ting=20the=20same=20behavior=20from=20FS.=20<BR>
<BR>
Please=20suggest.<BR>
<BR>
P.S:=20the=20Signature=20is=20being=20appended=20by=20my=20server=20which=
=20I=20do=20not=20have=20control=20over.=20It=20annoys=20me=20but=20can=
=20not=20help=20it.=20<BR>
<BR>
<BR>
On=201/25/11=2010:31=20PM,=20&quot;Brian=20West&quot;=20&lt;<a=20href=3D"=
brian at freeswitch.org">brian at freeswitch.org</a>&gt;=20wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT=20FACE=3D"Calibri,=20Verdana,=20Helvetica=
,=20Arial"><SPAN=20STYLE=3D'font-size:11pt'>Can=20you=20elaborate=20on=20=
this=20logs=20or=20something=20would=20help?=20&nbsp;Also=20remove=20that=
=20annoying=20unenforceable=20signature=20when=20emailing=20a=20public=20=
mailing=20list=20please.<BR>
<BR>
/b<BR>
<BR>
On=20Jan=2025,=202011,=20at=209:15=20PM,=20Arun=20Chinnachamy=20wrote:<BR=
>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT=20COLOR=3D"#131313"><FONT=20SIZE=3D"1"><F=
ONT=20FACE=3D"Lucida=20Grande"><SPAN=20STYLE=3D'font-size:8pt'>I=20am=20f=
acing=20an=20issue=20in=20FS.=20I=20am=20able=20to=20transfer=20an=20inco=
ming=20call=20from=20Gateway=20to=20local=20registered=20user=20but=20i=
=20am=20not=20able=20to=20do=20the=20same=20for=20outgoing=20call=20to=20=
gateway.=20FS=20is=20trying=20to=20call=20the=20local=20user=20through=20=
gateway.=20any=20suggestions?<BR>
</SPAN></FONT></FONT></FONT><FONT=20FACE=3D"Helvetica,=20Verdana,=20Arial=
"><SPAN=20STYLE=3D'font-size:12pt'><BR>
____<BR>
</SPAN></FONT></BLOCKQUOTE><FONT=20FACE=3D"Calibri,=20Verdana,=20Helvetic=
a,=20Arial"><SPAN=20STYLE=3D'font-size:11pt'><BR>
<BR>
<HR=20ALIGN=3DCENTER=20SIZE=3D"3"=20WIDTH=3D"95%"></SPAN></FONT><FONT=20S=
IZE=3D"2"><FONT=20FACE=3D"Consolas,=20Courier=20New,=20Courier"><SPAN=20S=
TYLE=3D'font-size:10pt'>_______________________________________________<B=
R>
FreeSWITCH-users=20mailing=20list<BR>
<a=20href=3D"FreeSWITCH-users at lists.freeswitch.org">FreeSWITCH-users at list=
s.freeswitch.org</a><BR>
<a=20href=3D"http://lists.freeswitch.org/mailman/listinfo/freeswitch-user=
s">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><BR>
UNSUBSCRIBE:<a=20href=3D"http://lists.freeswitch.org/mailman/options/free=
switch-users">http://lists.freeswitch.org/mailman/options/freeswitch-user=
s</a><BR>
<a=20href=3D"http://www.freeswitch.org">http://www.freeswitch.org</a><BR>
</SPAN></FONT></FONT></BLOCKQUOTE>
</BODY>
</HTML>


<table><tr><td=20bgcolor=3D#ffffff><font=20color=3D#000000><pre>This=20e-=
mail=20and=20any=20files=20transmitted=20with=20it=20are=20for=20the=20so=
le=20use=20of=20the=20intended=20recipient(s)=20and=20may=20contain=20con=
fidential=20and=20privileged=20information.
If=20you=20are=20not=20the=20intended=20recipient,=20please=20contact=20t=
he=20sender=20by=20reply=20e-mail=20and=20destroy=20all=20copies=20of=20t=
he=20original=20message.=20
Any=20unauthorised=20review,=20use,=20disclosure,=20dissemination,=20forw=
arding,=20printing=20or=20copying=20of=20this=20email=20or=20any=20action=
=20taken=20in=20reliance=20on=20this=20e-mail=20is=20strictly=20prohibite=
d=20and=20may=20be=20unlawful.</pre></font></td></tr></table>

--B_3378845668_10517720--



From lrobot.qq at gmail.com  Wed Jan 26 12:57:10 2011
From: lrobot.qq at gmail.com (=?UTF-8?B?572X5riF5rOJ?=)
Date: Wed, 26 Jan 2011 17:57:10 +0800
Subject: [Freeswitch-users] Need park a call with out pre-answer the
	call(not reply 183)
Message-ID: <AANLkTi=5N06u+Tw5uJA_ZxXo0jv+nwXxxKyHtjfsqm5p@mail.gmail.com>

Hi all,

  I am try use freeswitch. I need run freeswitch in "bypass-media" mode. So
when execute park application. I want the application will not reply 183
message to per answer it.
  How could I configure or modify code to get this?

  Thanks

Best Regards

-Qingquan
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From mkopacki at gmail.com  Wed Jan 26 10:56:32 2011
From: mkopacki at gmail.com (Michal Kopacki)
Date: Wed, 26 Jan 2011 08:56:32 +0100
Subject: [Freeswitch-users] network scenario doubts
Message-ID: <4D3FD3B0.4080309@gmail.com>

       Hello,

   This is my first post to this list, so hello everyone.

   I'm at the very beginning of freeswitch journey and i have a problem 
with fit FS to my network scenario.

OS: fedora 13 (x86_64)
FS: 1.0.7 (compiled from sources), default config

Desired scenario:

softphone -> mydomain.com (with ip 193.59.72.xx) -> my isp network -> 
10.25.48.xx on eth1 -> FS -> 192.168.0.1 on eth0 -> softphone

And now, I'm able to connect to FS from my local network, but I'm not 
able to connect from outside (neither domain nor ip). In internal.xml I 
set internal ip of server and i wanted to set external ip in 
external.xml, but there is a question: which one ?  In case of 
193.59.72.xx external profile didnt' start and with 10.25.48.xx outside 
softphone didn't register.  I checked with netstat and realized that 
port 5060 is bind to internal nic only and 5080 to external nic 
(10.25.48.xx) and I have no idea what next.

Is it even possible to work with such network scenario ? I would be 
grateful for pointing me to right direction or maybe propose different 
approach.

-- 
Best regards,
Michal



From patrick.plattes at niemann-frey.info  Wed Jan 26 11:33:47 2011
From: patrick.plattes at niemann-frey.info (Patrick Plattes)
Date: Wed, 26 Jan 2011 09:33:47 +0100
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please remember
	the BLF state)
Message-ID: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>

Hi List :-),

I'm currently switching from Asterisk to FreeSWITCH. It's really hard
work for an Asterisk user, but using Asterisk becomes more and more
painful even for small installations (less than 100 sip users). I know
FreeSWITCH is not a drop-in replacement for the Asterisk PBX, but I
don't want to change the behaviour of the PBX for the users.

It's a preconception in Germany that the American people like
(especially for the X-mas time) kitsch. Everyone here know the
American houses with a hole bunch of blinking lights. But those
decorated hoses are nothing against our offices! Our phones have up to
136 lights (BLFs). You often have to wear sunglasses at the office ;-)

A typical usage of BLFs is to check if an agent is a member of the
queue. I've build a simple extension to add and delete a member. I can
user "presence in" and "presence out" to enable or disable the BLF,
but there is one big issue. After a reboot of the phone the the user
is still a member of the queue, but the BLF is off. We are using hints
at Asterisk to show the user if he is a member and it works even after
a reboot - with "presence" at FreeSWITCH it works (of cause) not. Does
anyone have an idea how to implement it?

My current extension is just for testing and so I use mod_fifo. It
shouldn't be a problem to use mod_callcenter. The phone calls
"queue-the_name_of_the_queue-the_name_of_the_user at pbxdomain" eg.
"queue-sales_de-1000 at freeswitch.cust"

So how implement BLF persistence?

Thanks,
 Patrick


Colourized version: http://pastebin.com/ufJ7U930



<extension name="Queue Management">

 <condition field="destination_number"
expression="^queue-(\w+)-(\d+)$" break="on-false">
   <action application="set"
data="queue_name=${regex(${destination_number}|^queue-(\w+)-(\d+)$|$1)}"
inline="true"/>
   <action application="set"
data="queue_user=${regex(${destination_number}|^queue-(\w+)-(\d+)$|$2)}"
inline="true"/>
 </condition>

 <condition field="${fifo(list ${queue_name}@${domain_name})}"
expression="${queue_user}">
   <action application="answer"/>
   <action application="set" data="result=${fifo_member(del
${queue_name}@${domain_name}
{fifo_member_wait=nowait}user/${queue_user} )}"/>
   <action application="log" data="INFO Delete FIFO agent result: ${result}"/>
   <action application="log" data="INFO User Logout: ${user_name}"/>
   <action application="sleep" data="600"/>
   <action application="playback" data="ivr/ivr-you_are_now_logged_out.wav"/>
   <action application="set" data="result=${presence(out
queue-${queue_name}-${queue_user}@${domain_name} inuse inuse)}"/>
   <action application="log" data="INFO Presence OUT ${result}"/>
   <action application="hangup"/>

   <anti-action application="answer"/>
   <anti-action application="set" data="result=${fifo_member(add
${queue_name}@${domain_name}
{fifo_member_wait=nowait}user/${queue_user} )}"/>
   <anti-action application="log" data="INFO Add FIFO agent result: ${result}"/>
   <anti-action application="log" data="INFO User Login: ${user_name}"/>
   <anti-action application="sleep" data="600"/>
   <anti-action application="playback"
data="ivr/ivr-you_are_now_logged_in.wav"/>
   <anti-action application="set" data="result=${presence(in
queue-${queue_name}-${queue_user}@${domain_name} inuse inuse)}"/>
   <anti-action application="log" data="INFO Presence IN ${result}"/>
   <anti-action application="hangup"/>
 </condition>
</extension>


From tculjaga at gmail.com  Wed Jan 26 13:53:59 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 26 Jan 2011 11:53:59 +0100
Subject: [Freeswitch-users] FS HA
In-Reply-To: <AANLkTimB-jxEjrSD7pp45DcAmCmZH_q-dycT25c1gn06@mail.gmail.com>
References: <AANLkTi=_BPuE5KX4D48JDoOW9U3netsr9uDU9Lo5CgYO@mail.gmail.com>
	<FACE40F5-BB80-479A-BFFC-F07F6BA681D2@freeswitch.org>
	<350D44B3-A176-47A9-84F4-72EB62F08299@gmail.com>
	<AANLkTimJLMjPJwthN8AhDq=1_JnDbdmocbgd=-=usyTT@mail.gmail.com>
	<AANLkTi=ksT-ZkdOyXiVQkir7a9z2-58PcVirmv4xs5Kg@mail.gmail.com>
	<AANLkTikeU5FAgE-HU=0a5whf3JiD5VUCkYkCCRt4u=R0@mail.gmail.com>
	<AANLkTim3=HVjX5GZVfcVjtSCm-2kuWatyVS+YV0RpNCB@mail.gmail.com>
	<AANLkTimB-jxEjrSD7pp45DcAmCmZH_q-dycT25c1gn06@mail.gmail.com>
Message-ID: <AANLkTimwD=847nJ6Lz0gDrUh=nmELW261ch3QBAhDhf-@mail.gmail.com>

hello Anthony,

ya, i was afraid of that... this means moving all my modules & patches to
the latest git...

radius cdr interface
odbc cdr interface

within mod_xml_cdr <= i can share this as a patch... if you are interested.


mod_say_hr - really bad programming .. needs big re-factoring but lack of
time :(
patches for mod_say_de & mod_say_fr - because wrong playing in some
scenarios

and some small stuff i made within mod_commands... anyhow this was on the
road.


anyhow i was really into understanding the way the call context replication
works... this may be the DB connection issue as the ODBC connection is reset
on switchover (im switching entire resource .. floating IP, & database)


so what really happens is the FS on 2nd node just getting sessions from
sip_recovery

from sofia_glue_recover: i see you are selecting sip_recovery table.


select * from sip_recovery;
   runtime_uuid profile_name hostname uuid metadata
24f77a1f-f315-4beb-904f-f779e8767c75 internal cxss01
102b2c7f-466f-4b6d-a795-e2cc25630e78    24f77a1f-f315-4beb-904f-f779e8767c75
external cxss01 52834d99-c1be-4832-bee9-7ba05238871d



so, what do you do afterward?


on the recovering node i see some portion of dialplan is executed and
re-INVITEs being sent ... where do you get the DSP info for the recovering
re-INVITE ?




thanks for your help,
Tihomir.




On Wed, Jan 26, 2011 at 1:12 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> There is an xml_cdr snapshot in sip_recovery sql table that contains
> the data used for the recovery.
> Also based on your logs, you are not using latest git.  I would try
> that first before anything else.
>
>
>
> On Tue, Jan 25, 2011 at 7:24 AM, Tihomir Culjaga <tculjaga at gmail.com>
> wrote:
> > yap, both using the same config
> >
> >
> > can you advice where is FS getting the SDP info for the re-INVITE ?
> >
> >
> > On Tue, Jan 25, 2011 at 2:03 PM, Avi Marcus <avi at avimarcus.net> wrote:
> >>
> >> Are you using two machines for the HA? do both have the same configs?
> >> -Avi
> >>
> >> On Tue, Jan 25, 2011 at 11:41 AM, Tihomir Culjaga <tculjaga at gmail.com>
> >> wrote:
> >> > Here is the debug: http://pastebin.freeswitch.org/15133
> >> >
> >> > i have set verbose_sdp=true in vars.xml as.
> >> >   <X-PRE-PROCESS cmd="set" data="verbose_sdp=true"/>
> >> >
> >> > but not much to be seen of the verbose thing in the debug...
> >> >
> >> > Still, FS is sending a re-INVITE with wrong SDP. The call to be
> >> > recovered is
> >> > using ALAW... and ULAW is not supported.
> >> > FS sends a re-INVITE with ULAW and CN in SDP. The re-INVITE is
> rejected
> >> > due
> >> > to incompatible SDP.
> >> >
> >> > Where does FS get the information for the SDP in re-INVITE message?
> >> >
> >> >
> >> > please advice,
> >> > T.
> >> >
> >> >
> >> >
> >> > On Tue, Jan 25, 2011 at 12:01 AM, Tihomir Culjaga <tculjaga at gmail.com
> >
> >> > wrote:
> >> >>
> >> >> yap, i do have PCMA ... and the debug shows it correctly :=)
> >> >>
> >> >> i will try to see what it does with verbose. Post new debug tomorrow.
> >> >>
> >> >> ty.
> >> >>
> >> >>
> >> >> On Mon, Jan 24, 2011 at 11:29 PM, Steven Ayre <steveayre at gmail.com>
> >> >> wrote:
> >> >>>
> >> >>> Brian, it has PCMU in the sdp, but Sofia thinks it's setting PCMA?
> >> >>>
> >> >>> Steve on iPhone
> >> >>>
> >> >>> On 24 Jan 2011, at 20:17, Brian West <brian at freeswitch.org> wrote:
> >> >>>
> >> >>> > What makes you think that fails?  It has ULAW and CN in the codec
> >> >>> > list!
> >> >>> >  Sounds like you need the verbose sdp... set the global variable
> >> >>> > "verbose_sdp=true"
> >> >>> >
> >> >>> > /b
> >> >>> >
> >> >>> > On Jan 24, 2011, at 1:50 PM, Tihomir Culjaga wrote:
> >> >>> >
> >> >>> >>
> >> >>> >>
> >> >>> >> i configured FS HA and looks like its trying to recover the call
> ..
> >> >>> >> but the re-INVITE fails due to "wrong/missed" codec capability.
> >> >>> >
> >> >>> >
> >> >>> > _______________________________________________
> >> >>> > FreeSWITCH-users mailing list
> >> >>> > FreeSWITCH-users at lists.freeswitch.org
> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >
> >> >>> >
> >> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> > http://www.freeswitch.org
> >> >>>
> >> >>> _______________________________________________
> >> >>> FreeSWITCH-users mailing list
> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>
> >> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> http://www.freeswitch.org
> >> >>
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
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>
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From christian.loeschenkohl at xpirio.com  Wed Jan 26 14:05:08 2011
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Wed, 26 Jan 2011 12:05:08 +0100
Subject: [Freeswitch-users] php esl with outbound socket connection
Message-ID: <4D3FFFE4.7030404@xpirio.com>

hello list

has somebody a working example of an php esl outbound socket connection?
the example at http://wiki.freeswitch.org/wiki/PHP_ESL shows only sending manual
commands.
i have written my own php class here (not based on the esl module) that is in use for
a year now, but i wonder if it's possible with the esl module too (in combination with ivrd).
the advantage would be to communicate more directly, like in mod_perl (setting and getting
switch variables within the current connection and so on).

the comeback of mod_php would be my dream here, but it doesn't seem to be possible (high costs and
no maintainer).

please share your thoughts on this topic

br

-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 5 77 11 - 1000
F  +43 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From peter.olsson at visionutveckling.se  Wed Jan 26 14:22:56 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 26 Jan 2011 12:22:56 +0100
Subject: [Freeswitch-users] Need park a call with out pre-answer
	the	call(not reply 183)
Message-ID: <235E565A-8835-4991-BF59-70D64CCB4466@visionutveckling.se>

I think this should work already, have you tried? Park does not require media.

Peter


----- Reply message -----
Fr?n: "???" <lrobot.qq at gmail.com>
Datum: ons, jan 26, 2011 17:16
Rubrik: [Freeswitch-users] Need park a call with out pre-answer the call(not reply 183)
Till: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>

Hi all,

  I am try use freeswitch. I need run freeswitch in "bypass-media" mode. So when execute park application. I want the application will not reply 183 message to per answer it.
  How could I configure or modify code to get this?

  Thanks

Best Regards

-Qingquan
!DSPAM:4d3ff40d32767902010757!

From avi at avimarcus.net  Wed Jan 26 15:00:46 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 26 Jan 2011 14:00:46 +0200
Subject: [Freeswitch-users] FS HA
In-Reply-To: <AANLkTimwD=847nJ6Lz0gDrUh=nmELW261ch3QBAhDhf-@mail.gmail.com>
References: <AANLkTi=_BPuE5KX4D48JDoOW9U3netsr9uDU9Lo5CgYO@mail.gmail.com>
	<FACE40F5-BB80-479A-BFFC-F07F6BA681D2@freeswitch.org>
	<350D44B3-A176-47A9-84F4-72EB62F08299@gmail.com>
	<AANLkTimJLMjPJwthN8AhDq=1_JnDbdmocbgd=-=usyTT@mail.gmail.com>
	<AANLkTi=ksT-ZkdOyXiVQkir7a9z2-58PcVirmv4xs5Kg@mail.gmail.com>
	<AANLkTikeU5FAgE-HU=0a5whf3JiD5VUCkYkCCRt4u=R0@mail.gmail.com>
	<AANLkTim3=HVjX5GZVfcVjtSCm-2kuWatyVS+YV0RpNCB@mail.gmail.com>
	<AANLkTimB-jxEjrSD7pp45DcAmCmZH_q-dycT25c1gn06@mail.gmail.com>
	<AANLkTimwD=847nJ6Lz0gDrUh=nmELW261ch3QBAhDhf-@mail.gmail.com>
Message-ID: <AANLkTinDjjDMDKykSKaBJKUsKs_1A+uJo=WXgJ60ix7T@mail.gmail.com>

The sip recovery table should have a LOT more data than that, all stuffed
into a single column.
Do you have track calls on in all the involved profiles? I suppose it really
wouldn't be here otherwise.

e.g. my sip_recovery table has this in the metadata field for a "simple" MOH
call. Does yours?


<cdr>
  <channel_data>
    <state>CS_EXECUTE</state>
    <direction>inbound</direction>
    <state_number>4</state_number>
    <flags>0=1;1=1;3=1;35=1;41=1;51=1</flags>
    <caps>1=1;2=1;3=1;4=1;5=1</caps>
  </channel_data>
  <variables>
    <direction>inbound</direction>
    <uuid>6cda8922-2943-11e0-b622-639ef8e43974</uuid>
    <sip_local_network_addr>178.79.147.47</sip_local_network_addr>
    <sip_network_ip>79.176.185.32</sip_network_ip>
    <sip_network_port>5072</sip_network_port>
    <sip_received_ip>79.176.yyy.32</sip_received_ip>
    <sip_received_port>5072</sip_received_port>
    <sip_via_protocol>udp</sip_via_protocol>
    <sip_authorized>true</sip_authorized>
    <sip_number_alias>1000</sip_number_alias>
    <sip_auth_username>1000</sip_auth_username>
    <sip_auth_realm>sip-ha.yyy.com</sip_auth_realm>
    <number_alias>1000</number_alias>
    <user_name>1000</user_name>
    <domain_name>sip.yyy.com</domain_name>
    <record_stereo>true</record_stereo>
    <default_areacode>208</default_areacode>
    <transfer_fallback_extension>operator</transfer_fallback_extension>
    <accountcode>1000</accountcode>
    <user_context>default</user_context>
    <effective_caller_id_name>Avi%20Marcus</effective_caller_id_name>
    <effective_caller_id_number>12013554419</effective_caller_id_number>
    <outbound_caller_id_name>Avi%20Marcus</outbound_caller_id_name>
    <outbound_caller_id_number>12013554419</outbound_caller_id_number>
    <nibble_account>1000</nibble_account>
    <sip_from_user>1000</sip_from_user>
    <sip_from_uri>1000%40sip-ha.yyy.com</sip_from_uri>
    <sip_from_host>sip-ha.yyy.com</sip_from_host>
    <sip_from_user_stripped>1000</sip_from_user_stripped>
    <sofia_profile_name>internal</sofia_profile_name>
    <sip_Remote-Party-ID>%3Csip%3A1000%40sip-ha.yyy.com
%3E%3Bscreen%3Dyes%3Bparty%3Dcalling</sip_Remote-Party-ID>
    <sip_cid_type>rpid</sip_cid_type>

 <sip_full_via>SIP/2.0/UDP%20192.168.1.5%3A5072%3Bbranch%3Dz9hG4bK-329da816%3Brport%3D5072%3Breceived%3D79.176.185.32</sip_full_via>
    <sip_req_user>*9664</sip_req_user>
    <sip_req_uri>*9664%40sip-ha.yyy.com</sip_req_uri>
    <sip_req_host>sip-ha.yyy.com</sip_req_host>
    <sip_to_user>*9664</sip_to_user>
    <sip_to_uri>*9664%40sip-ha.yyy.com</sip_to_uri>
    <sip_to_host>sip-ha.yyy.com</sip_to_host>
    <sip_contact_user>1000</sip_contact_user>
    <sip_contact_port>5072</sip_contact_port>
    <sip_contact_uri>1000%4079.176.yyy.32%3A5072</sip_contact_uri>
    <sip_contact_host>79.176.yyy.32</sip_contact_host>
    <channel_name>sofia/internal/1000%40sip-ha.yyy.com</channel_name>
    <sip_user_agent>Linksys/SPA2102-5.2.10</sip_user_agent>
    <sip_via_host>192.168.1.5</sip_via_host>
    <sip_via_port>5072</sip_via_port>
    <sip_via_rport>5072</sip_via_rport>
    <max_forwards>70</max_forwards>
    <presence_id>1000%40sip-ha.yyy.com</presence_id>

 <switch_r_sdp>v%3D0%0D%0Ao%3D-%2033038436%2033038436%20IN%20IP4%2079.176.185.32%0D%0As%3D-%0D%0Ac%3DIN%20IP4%2079.176.185.32%0D%0At%3D0%200%0D%0Am%3Daudio%2016422%20RTP/AVP%2018%200%202%204%208%2096%2097%2098%20100%20101%0D%0Aa%3Drtpmap%3A18%20G729/8000%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A2%20G726-32/8000%0D%0Aa%3Drtpmap%3A4%20G723/8000%0D%0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Drtpmap%3A96%20G726-40/8000%0D%0Aa%3Drtpmap%3A97%20G726-24/8000%0D%0Aa%3Drtpmap%3A98%20G726-16/8000%0D%0Aa%3Drtpmap%3A100%20NSE/8000%0D%0Aa%3Dfmtp%3A100%20192-193%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0Aa%3Dptime%3A20%0D%0A</switch_r_sdp>

 <ep_codec_string>PCMU%408000h%4020i%4064000b,PCMA%408000h%4020i%4064000b</ep_codec_string>
    <nibble_current_balance>0.000000</nibble_current_balance>

 <RFC2822_DATE>Wed,%2026%20Jan%202011%2013%3A57%3A18%20%2B0200</RFC2822_DATE>
    <zrtp_secure_media>true</zrtp_secure_media>
    <sip_use_codec_name>PCMU</sip_use_codec_name>
    <sip_use_codec_rate>8000</sip_use_codec_rate>
    <sip_use_codec_ptime>20</sip_use_codec_ptime>
    <read_codec>PCMU</read_codec>
    <read_rate>8000</read_rate>
    <write_codec>PCMU</write_codec>
    <write_rate>8000</write_rate>
    <local_media_ip>178.79.yyy.47</local_media_ip>
    <local_media_port>19600</local_media_port>
    <sip_use_pt>0</sip_use_pt>
    <rtp_use_ssrc>922702377</rtp_use_ssrc>
    <remote_media_ip>79.176.185.32</remote_media_ip>
    <remote_media_port>16422</remote_media_port>

 <sip_local_sdp_str>v%3D0%0Ao%3DFreeSWITCH%201296023438%201296023439%20IN%20IP4%20178.79.147.47%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%20178.79.147.47%0At%3D0%200%0Am%3Daudio%2019600%20RTP/AVP%200%20101%0Aa%3Drtpmap%3A0%20PCMU/8000%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A</sip_local_sdp_str>
    <endpoint_disposition>ANSWER</endpoint_disposition>
    <sip_to_tag>vgcH82ctg6QaD</sip_to_tag>
    <sip_from_tag>9241a2573e4f702o0</sip_from_tag>
    <sip_cseq>102</sip_cseq>
    <sip_call_id>17217721-3d90de36%40192.168.1.5</sip_call_id>
    <sip_full_from>%3Csip%3A1000%40sip-ha.yyy.com
%3E%3Btag%3D9241a2573e4f702o0</sip_full_from>
    <sip_full_to>%3Csip%3A*9664%40sip-ha.yyy.com
%3E%3Btag%3DvgcH82ctg6QaD</sip_full_to>
    <playback_seconds>2</playback_seconds>
    <playback_ms>2000</playback_ms>
    <playback_samples>16000</playback_samples>

 <current_application_data>local_stream%3A//moh</current_application_data>
    <current_application>playback</current_application>
  </variables>
  <app_log>
    <application app_name="hash"
app_data="insert/sip.yyy.com-spymap/1000/6cda8922-2943-11e0-b622-639ef8e43974"></application>
    <application app_name="hash"
app_data="insert/sip.yyy.com-last_dial/1000/*9664"></application>
    <application app_name="hash"
app_data="insert/sip.yyy.com-last_dial/global/6cda8922-2943-11e0-b622-639ef8e43974"></application>
    <application app_name="set" app_data="RFC2822_DATE=Wed, 26 Jan 2011
13:57:18 +0200"></application>
    <application app_name="set"
app_data="zrtp_secure_media=true"></application>
    <application app_name="answer" app_data=""></application>
    <application app_name="playback"
app_data="silence_stream://2000"></application>
    <application app_name="execute_extension" app_data="is_zrtp_secure XML
features"></application>
    <application app_name="eval" app_data="not_secure"></application>
    <application app_name="playback"
app_data="local_stream://moh"></application>
  </app_log>
  <callflow dialplan="XML" profile_index="1">
    <extension name="global" number="*9664">
      <application app_name="hash"
app_data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"></application>
      <application app_name="hash"
app_data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"></application>
      <application app_name="hash"
app_data="insert/${domain_name}-last_dial/global/${uuid}"></application>
      <application app_name="set" app_data="RFC2822_DATE=${strftime(%a, %d
%b %Y %T %z)}"></application>
      <application app_name="set"
app_data="zrtp_secure_media=true"></application>
      <application app_name="answer" app_data=""></application>
      <application app_name="playback"
app_data="silence_stream://2000"></application>
      <application app_name="execute_extension" app_data="is_zrtp_secure XML
features"></application>
      <application app_name="playback"
app_data="local_stream://moh"></application>
      <sub_extensions>
        <extension name="is_zrtp_secure" number="is_zrtp_secure"
dialplan="XML">
          <application app_name="eval" app_data="not_secure"></application>
        </extension>
      </sub_extensions>
    </extension>
    <caller_profile>
      <username>1000</username>
      <dialplan>XML</dialplan>
      <caller_id_name>1000</caller_id_name>
      <ani>1000</ani>
      <aniii></aniii>
      <caller_id_number>1000</caller_id_number>
      <network_addr>79.176.yyy.32</network_addr>
      <rdnis></rdnis>
      <destination_number>*9664</destination_number>
      <uuid>6cda8922-2943-11e0-b622-639ef8e43974</uuid>
      <source>mod_sofia</source>
      <context>default</context>
      <chan_name>sofia/internal/1000 at sip-ha.yyy.com</chan_name>
    </caller_profile>
    <times>
      <created_time>1296043038692089</created_time>
      <profile_created_time>1296043038692089</profile_created_time>
      <progress_time>0</progress_time>
      <progress_media_time>1296043038708572</progress_media_time>
      <answered_time>1296043038708572</answered_time>
      <hangup_time>0</hangup_time>
      <resurrect_time>0</resurrect_time>
      <transfer_time>0</transfer_time>
    </times>
  </callflow>
</cdr>




On Wed, Jan 26, 2011 at 12:53 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:

> hello Anthony,
>
> ya, i was afraid of that... this means moving all my modules & patches to
> the latest git...
>
> radius cdr interface
> odbc cdr interface
>
> within mod_xml_cdr <= i can share this as a patch... if you are interested.
>
>
> mod_say_hr - really bad programming .. needs big re-factoring but lack of
> time :(
> patches for mod_say_de & mod_say_fr - because wrong playing in some
> scenarios
>
> and some small stuff i made within mod_commands... anyhow this was on the
> road.
>
>
> anyhow i was really into understanding the way the call context replication
> works... this may be the DB connection issue as the ODBC connection is reset
> on switchover (im switching entire resource .. floating IP, & database)
>
>
> so what really happens is the FS on 2nd node just getting sessions from
> sip_recovery
>
> from sofia_glue_recover: i see you are selecting sip_recovery table.
>
>
> select * from sip_recovery;
>    runtime_uuid profile_name hostname uuid metadata
> 24f77a1f-f315-4beb-904f-f779e8767c75 internal cxss01
> 102b2c7f-466f-4b6d-a795-e2cc25630e78
> 24f77a1f-f315-4beb-904f-f779e8767c75 external cxss01
> 52834d99-c1be-4832-bee9-7ba05238871d
>
>
>
> so, what do you do afterward?
>
>
> on the recovering node i see some portion of dialplan is executed and
> re-INVITEs being sent ... where do you get the DSP info for the recovering
> re-INVITE ?
>
>
>
>
> thanks for your help,
> Tihomir.
>
>
>
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From tculjaga at gmail.com  Wed Jan 26 15:20:45 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 26 Jan 2011 13:20:45 +0100
Subject: [Freeswitch-users] FS HA
In-Reply-To: <AANLkTinDjjDMDKykSKaBJKUsKs_1A+uJo=WXgJ60ix7T@mail.gmail.com>
References: <AANLkTi=_BPuE5KX4D48JDoOW9U3netsr9uDU9Lo5CgYO@mail.gmail.com>
	<FACE40F5-BB80-479A-BFFC-F07F6BA681D2@freeswitch.org>
	<350D44B3-A176-47A9-84F4-72EB62F08299@gmail.com>
	<AANLkTimJLMjPJwthN8AhDq=1_JnDbdmocbgd=-=usyTT@mail.gmail.com>
	<AANLkTi=ksT-ZkdOyXiVQkir7a9z2-58PcVirmv4xs5Kg@mail.gmail.com>
	<AANLkTikeU5FAgE-HU=0a5whf3JiD5VUCkYkCCRt4u=R0@mail.gmail.com>
	<AANLkTim3=HVjX5GZVfcVjtSCm-2kuWatyVS+YV0RpNCB@mail.gmail.com>
	<AANLkTimB-jxEjrSD7pp45DcAmCmZH_q-dycT25c1gn06@mail.gmail.com>
	<AANLkTimwD=847nJ6Lz0gDrUh=nmELW261ch3QBAhDhf-@mail.gmail.com>
	<AANLkTinDjjDMDKykSKaBJKUsKs_1A+uJo=WXgJ60ix7T@mail.gmail.com>
Message-ID: <AANLkTi=QkaTTBPpXfWs3bMEZqGs=9L04Mda8JU+rWpBb@mail.gmail.com>

dam,  your metadata is awesome ... now i see the difference.

i do have track calls enabled on both profiles (internal & external) and i
have odbc configured as well.



but my version is:

freeswitch at cxss01> version

FreeSWITCH Version 1.0.6 (svn-exported)

freeswitch at cxss01>



how can i make sure it is because of the version ?





freeswitch at cxss01>
freeswitch at cxss01>
freeswitch at cxss01> sofia profile external start
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 debug [0]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 sip-trace [no]
*2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 track-calls [true]*
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 rfc2833-pt [101]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 sip-port [5080]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 dialplan [XML]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 context [public]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 dtmf-duration [70]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:3279 Duration out of bounds,
using default of 2000!
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 dtmf-type [inband]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 inbound-codec-prefs
[PCMA,GSM]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 outbound-codec-prefs
[PCMA,GSM]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 hold-music
[local_stream://moh]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 rtp-timer-name [soft]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 local-network-acl
[localnet.auto]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 manage-presence [false]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 inbound-codec-negotiation
[generous]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 nonce-ttl [60]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 auth-calls [false]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 rtp-ip [195.88.212.30]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 sip-ip [195.88.212.30]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 ext-rtp-ip [auto-nat]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 ext-sip-ip [auto-nat]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 rtp-timeout-sec [300]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 rtp-hold-timeout-sec [1800]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 tls [false]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 tls-bind-params
[transport=tls]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 tls-sip-port [5081]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 tls-cert-dir
[/usr/local/freeswitch/conf/ssl]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 tls-version [tlsv1]
2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 odbc-dsn
[COREFSdrv:adminfs:4F0s8wj]
2011-01-26 13:14:11.423588 [NOTICE] sofia_reg.c:2451 Added gateway '
example.com' to profile 'external'
2011-01-26 13:14:11.423588 [NOTICE] sofia.c:3535 Connecting ODBC Profile
external [sofia_reg_external]
2011-01-26 13:14:11.424818 [DEBUG] sofia.c:1317 Creating agent for external
2011-01-26 13:14:11.438512 [INFO] sofia_glue.c:4569 Connected ODBC DSN:
COREFSdrv
2011-01-26 13:14:11.459433 [DEBUG] sofia.c:1353 Created agent for external
2011-01-26 13:14:11.459433 [DEBUG] sofia.c:1389 Set params for external
2011-01-26 13:14:11.459433 [DEBUG] sofia.c:1412 Activated db for external
2011-01-26 13:14:11.460603 [DEBUG] sofia.c:1439 Starting thread for external

external started successfully

freeswitch at cxss01>
freeswitch at cxss01>
freeswitch at cxss01>
freeswitch at cxss01> sofia profile internal start
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 debug [0]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 sip-trace [no]
*2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 track-calls [true]*
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 log-auth-failures [true]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 context [public]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 rfc2833-pt [101]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 sip-port [5060]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 dialplan [XML]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 dtmf-duration [70]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:3279 Duration out of bounds,
using default of 2000!
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 inbound-codec-prefs
[PCMA,GSM]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 outbound-codec-prefs
[PCMA,GSM]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 rtp-timer-name [soft]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 rtp-ip [195.88.212.30]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 sip-ip [195.88.212.30]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 hold-music
[local_stream://moh]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 apply-nat-acl [nat.auto]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 apply-inbound-acl [domains]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 local-network-acl
[localnet.auto]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 dtmf-type [inband]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 record-path
[/usr/local/freeswitch/recordings]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 record-template
[${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 manage-presence [true]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 inbound-codec-negotiation
[generous]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 tls [false]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 tls-bind-params
[transport=tls]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 tls-sip-port [5061]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 tls-cert-dir
[/usr/local/freeswitch/conf/ssl]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 tls-version [tlsv1]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 odbc-dsn
[COREFSdrv:adminfs:4F0s8wj]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 inbound-late-negotiation
[true]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 nonce-ttl [60]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 auth-calls [true]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755
inbound-reg-force-matching-username [true]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 auth-all-packets [false]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 ext-rtp-ip [auto-nat]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 ext-sip-ip [auto-nat]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 rtp-timeout-sec [300]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 rtp-hold-timeout-sec [1800]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 force-register-domain
[195.88.212.31]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 force-subscription-domain
[195.88.212.31]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 force-register-db-domain
[195.88.212.31]
2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 challenge-realm [auto_from]
2011-01-26 13:14:16.440442 [NOTICE] sofia_reg.c:2451 Added gateway
'hr-sbc01' to profile 'internal'
2011-01-26 13:14:16.441444 [NOTICE] sofia_reg.c:2451 Added gateway
'hr-sbc02' to profile 'internal'
2011-01-26 13:14:16.441444 [NOTICE] sofia.c:2040 Adding Alias
[195.88.212.31] for profile [internal]
2011-01-26 13:14:16.441444 [NOTICE] sofia.c:3535 Connecting ODBC Profile
internal [sofia_reg_internal]
2011-01-26 13:14:16.441444 [DEBUG] sofia.c:1317 Creating agent for internal
2011-01-26 13:14:16.455326 [INFO] sofia_glue.c:4569 Connected ODBC DSN:
COREFSdrv
2011-01-26 13:14:16.476319 [DEBUG] sofia.c:1353 Created agent for internal
2011-01-26 13:14:16.476319 [DEBUG] sofia.c:1389 Set params for internal
2011-01-26 13:14:16.476319 [DEBUG] sofia.c:1412 Activated db for internal
2011-01-26 13:14:16.476319 [DEBUG] sofia.c:1439 Starting thread for internal

internal started successfully





i do have track-calls enabled on the sip profiles.


and yes, FS is quite outdated:

freeswitch at cxss01> version

FreeSWITCH Version 1.0.6 (svn-exported)

freeswitch at cxss01>










On Wed, Jan 26, 2011 at 1:00 PM, Avi Marcus <avi at avimarcus.net> wrote:

> The sip recovery table should have a LOT more data than that, all stuffed
> into a single column.
> Do you have track calls on in all the involved profiles? I suppose it
> really wouldn't be here otherwise.
>
> e.g. my sip_recovery table has this in the metadata field for a "simple"
> MOH call. Does yours?
>
>
> <cdr>
>   <channel_data>
>      <state>CS_EXECUTE</state>
>     <direction>inbound</direction>
>     <state_number>4</state_number>
>     <flags>0=1;1=1;3=1;35=1;41=1;51=1</flags>
>     <caps>1=1;2=1;3=1;4=1;5=1</caps>
>   </channel_data>
>   <variables>
>     <direction>inbound</direction>
>     <uuid>6cda8922-2943-11e0-b622-639ef8e43974</uuid>
>     <sip_local_network_addr>178.79.147.47</sip_local_network_addr>
>     <sip_network_ip>79.176.185.32</sip_network_ip>
>     <sip_network_port>5072</sip_network_port>
>     <sip_received_ip>79.176.yyy.32</sip_received_ip>
>     <sip_received_port>5072</sip_received_port>
>     <sip_via_protocol>udp</sip_via_protocol>
>     <sip_authorized>true</sip_authorized>
>     <sip_number_alias>1000</sip_number_alias>
>     <sip_auth_username>1000</sip_auth_username>
>     <sip_auth_realm>sip-ha.yyy.com</sip_auth_realm>
>     <number_alias>1000</number_alias>
>     <user_name>1000</user_name>
>     <domain_name>sip.yyy.com</domain_name>
>     <record_stereo>true</record_stereo>
>     <default_areacode>208</default_areacode>
>     <transfer_fallback_extension>operator</transfer_fallback_extension>
>     <accountcode>1000</accountcode>
>     <user_context>default</user_context>
>     <effective_caller_id_name>Avi%20Marcus</effective_caller_id_name>
>     <effective_caller_id_number>12013554419</effective_caller_id_number>
>     <outbound_caller_id_name>Avi%20Marcus</outbound_caller_id_name>
>     <outbound_caller_id_number>12013554419</outbound_caller_id_number>
>     <nibble_account>1000</nibble_account>
>     <sip_from_user>1000</sip_from_user>
>     <sip_from_uri>1000%40sip-ha.yyy.com</sip_from_uri>
>     <sip_from_host>sip-ha.yyy.com</sip_from_host>
>     <sip_from_user_stripped>1000</sip_from_user_stripped>
>     <sofia_profile_name>internal</sofia_profile_name>
>     <sip_Remote-Party-ID>%3Csip%3A1000%40sip-ha.yyy.com
> %3E%3Bscreen%3Dyes%3Bparty%3Dcalling</sip_Remote-Party-ID>
>     <sip_cid_type>rpid</sip_cid_type>
>
>  <sip_full_via>SIP/2.0/UDP%20192.168.1.5%3A5072%3Bbranch%3Dz9hG4bK-329da816%3Brport%3D5072%3Breceived%3D79.176.185.32</sip_full_via>
>     <sip_req_user>*9664</sip_req_user>
>     <sip_req_uri>*9664%40sip-ha.yyy.com</sip_req_uri>
>     <sip_req_host>sip-ha.yyy.com</sip_req_host>
>     <sip_to_user>*9664</sip_to_user>
>     <sip_to_uri>*9664%40sip-ha.yyy.com</sip_to_uri>
>     <sip_to_host>sip-ha.yyy.com</sip_to_host>
>     <sip_contact_user>1000</sip_contact_user>
>     <sip_contact_port>5072</sip_contact_port>
>     <sip_contact_uri>1000%4079.176.yyy.32%3A5072</sip_contact_uri>
>     <sip_contact_host>79.176.yyy.32</sip_contact_host>
>     <channel_name>sofia/internal/1000%40sip-ha.yyy.com</channel_name>
>     <sip_user_agent>Linksys/SPA2102-5.2.10</sip_user_agent>
>     <sip_via_host>192.168.1.5</sip_via_host>
>     <sip_via_port>5072</sip_via_port>
>     <sip_via_rport>5072</sip_via_rport>
>     <max_forwards>70</max_forwards>
>     <presence_id>1000%40sip-ha.yyy.com</presence_id>
>
>  <switch_r_sdp>v%3D0%0D%0Ao%3D-%2033038436%2033038436%20IN%20IP4%2079.176.185.32%0D%0As%3D-%0D%0Ac%3DIN%20IP4%2079.176.185.32%0D%0At%3D0%200%0D%0Am%3Daudio%2016422%20RTP/AVP%2018%200%202%204%208%2096%2097%2098%20100%20101%0D%0Aa%3Drtpmap%3A18%20G729/8000%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A2%20G726-32/8000%0D%0Aa%3Drtpmap%3A4%20G723/8000%0D%0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Drtpmap%3A96%20G726-40/8000%0D%0Aa%3Drtpmap%3A97%20G726-24/8000%0D%0Aa%3Drtpmap%3A98%20G726-16/8000%0D%0Aa%3Drtpmap%3A100%20NSE/8000%0D%0Aa%3Dfmtp%3A100%20192-193%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0Aa%3Dptime%3A20%0D%0A</switch_r_sdp>
>
>  <ep_codec_string>PCMU%408000h%4020i%4064000b,PCMA%408000h%4020i%4064000b</ep_codec_string>
>     <nibble_current_balance>0.000000</nibble_current_balance>
>
>  <RFC2822_DATE>Wed,%2026%20Jan%202011%2013%3A57%3A18%20%2B0200</RFC2822_DATE>
>     <zrtp_secure_media>true</zrtp_secure_media>
>     <sip_use_codec_name>PCMU</sip_use_codec_name>
>     <sip_use_codec_rate>8000</sip_use_codec_rate>
>     <sip_use_codec_ptime>20</sip_use_codec_ptime>
>     <read_codec>PCMU</read_codec>
>     <read_rate>8000</read_rate>
>     <write_codec>PCMU</write_codec>
>     <write_rate>8000</write_rate>
>     <local_media_ip>178.79.yyy.47</local_media_ip>
>     <local_media_port>19600</local_media_port>
>     <sip_use_pt>0</sip_use_pt>
>     <rtp_use_ssrc>922702377</rtp_use_ssrc>
>     <remote_media_ip>79.176.185.32</remote_media_ip>
>     <remote_media_port>16422</remote_media_port>
>
>  <sip_local_sdp_str>v%3D0%0Ao%3DFreeSWITCH%201296023438%201296023439%20IN%20IP4%20178.79.147.47%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%20178.79.147.47%0At%3D0%200%0Am%3Daudio%2019600%20RTP/AVP%200%20101%0Aa%3Drtpmap%3A0%20PCMU/8000%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A</sip_local_sdp_str>
>     <endpoint_disposition>ANSWER</endpoint_disposition>
>     <sip_to_tag>vgcH82ctg6QaD</sip_to_tag>
>     <sip_from_tag>9241a2573e4f702o0</sip_from_tag>
>     <sip_cseq>102</sip_cseq>
>     <sip_call_id>17217721-3d90de36%40192.168.1.5</sip_call_id>
>     <sip_full_from>%3Csip%3A1000%40sip-ha.yyy.com
> %3E%3Btag%3D9241a2573e4f702o0</sip_full_from>
>     <sip_full_to>%3Csip%3A*9664%40sip-ha.yyy.com
> %3E%3Btag%3DvgcH82ctg6QaD</sip_full_to>
>     <playback_seconds>2</playback_seconds>
>     <playback_ms>2000</playback_ms>
>     <playback_samples>16000</playback_samples>
>
>  <current_application_data>local_stream%3A//moh</current_application_data>
>     <current_application>playback</current_application>
>   </variables>
>   <app_log>
>     <application app_name="hash"
> app_data="insert/sip.yyy.com-spymap/1000/6cda8922-2943-11e0-b622-639ef8e43974"></application>
>     <application app_name="hash"
> app_data="insert/sip.yyy.com-last_dial/1000/*9664"></application>
>     <application app_name="hash"
> app_data="insert/sip.yyy.com-last_dial/global/6cda8922-2943-11e0-b622-639ef8e43974"></application>
>     <application app_name="set" app_data="RFC2822_DATE=Wed, 26 Jan 2011
> 13:57:18 +0200"></application>
>     <application app_name="set"
> app_data="zrtp_secure_media=true"></application>
>     <application app_name="answer" app_data=""></application>
>     <application app_name="playback"
> app_data="silence_stream://2000"></application>
>     <application app_name="execute_extension" app_data="is_zrtp_secure XML
> features"></application>
>     <application app_name="eval" app_data="not_secure"></application>
>     <application app_name="playback"
> app_data="local_stream://moh"></application>
>   </app_log>
>   <callflow dialplan="XML" profile_index="1">
>     <extension name="global" number="*9664">
>       <application app_name="hash"
> app_data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"></application>
>       <application app_name="hash"
> app_data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"></application>
>       <application app_name="hash"
> app_data="insert/${domain_name}-last_dial/global/${uuid}"></application>
>       <application app_name="set" app_data="RFC2822_DATE=${strftime(%a, %d
> %b %Y %T %z)}"></application>
>       <application app_name="set"
> app_data="zrtp_secure_media=true"></application>
>       <application app_name="answer" app_data=""></application>
>       <application app_name="playback"
> app_data="silence_stream://2000"></application>
>       <application app_name="execute_extension" app_data="is_zrtp_secure
> XML features"></application>
>       <application app_name="playback"
> app_data="local_stream://moh"></application>
>       <sub_extensions>
>         <extension name="is_zrtp_secure" number="is_zrtp_secure"
> dialplan="XML">
>           <application app_name="eval" app_data="not_secure"></application>
>         </extension>
>       </sub_extensions>
>     </extension>
>     <caller_profile>
>       <username>1000</username>
>       <dialplan>XML</dialplan>
>       <caller_id_name>1000</caller_id_name>
>       <ani>1000</ani>
>       <aniii></aniii>
>       <caller_id_number>1000</caller_id_number>
>       <network_addr>79.176.yyy.32</network_addr>
>       <rdnis></rdnis>
>       <destination_number>*9664</destination_number>
>       <uuid>6cda8922-2943-11e0-b622-639ef8e43974</uuid>
>       <source>mod_sofia</source>
>       <context>default</context>
>       <chan_name>sofia/internal/1000 at sip-ha.yyy.com</chan_name>
>     </caller_profile>
>     <times>
>       <created_time>1296043038692089</created_time>
>       <profile_created_time>1296043038692089</profile_created_time>
>       <progress_time>0</progress_time>
>       <progress_media_time>1296043038708572</progress_media_time>
>       <answered_time>1296043038708572</answered_time>
>       <hangup_time>0</hangup_time>
>       <resurrect_time>0</resurrect_time>
>       <transfer_time>0</transfer_time>
>     </times>
>   </callflow>
> </cdr>
>
>
>
>
> On Wed, Jan 26, 2011 at 12:53 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> hello Anthony,
>>
>> ya, i was afraid of that... this means moving all my modules & patches to
>> the latest git...
>>
>> radius cdr interface
>> odbc cdr interface
>>
>> within mod_xml_cdr <= i can share this as a patch... if you are
>> interested.
>>
>>
>> mod_say_hr - really bad programming .. needs big re-factoring but lack of
>> time :(
>> patches for mod_say_de & mod_say_fr - because wrong playing in some
>> scenarios
>>
>> and some small stuff i made within mod_commands... anyhow this was on the
>> road.
>>
>>
>> anyhow i was really into understanding the way the call context
>> replication works... this may be the DB connection issue as the ODBC
>> connection is reset on switchover (im switching entire resource .. floating
>> IP, & database)
>>
>>
>> so what really happens is the FS on 2nd node just getting sessions from
>> sip_recovery
>>
>> from sofia_glue_recover: i see you are selecting sip_recovery table.
>>
>>
>> select * from sip_recovery;
>>    runtime_uuid profile_name hostname uuid metadata
>> 24f77a1f-f315-4beb-904f-f779e8767c75 internal cxss01
>> 102b2c7f-466f-4b6d-a795-e2cc25630e78
>> 24f77a1f-f315-4beb-904f-f779e8767c75 external cxss01
>> 52834d99-c1be-4832-bee9-7ba05238871d
>>
>>
>>
>> so, what do you do afterward?
>>
>>
>> on the recovering node i see some portion of dialplan is executed and
>> re-INVITEs being sent ... where do you get the DSP info for the recovering
>> re-INVITE ?
>>
>>
>>
>>
>> thanks for your help,
>> Tihomir.
>>
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Nabble at slickdeals.endjunk.com  Wed Jan 26 15:22:54 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 26 Jan 2011 04:22:54 -0800 (PST)
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
	remember the BLF state)
In-Reply-To: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
Message-ID: <1296044574879-5962503.post@n2.nabble.com>



Patrick Plattes wrote:
> It's a preconception in Germany that the American people like
> (especially for the X-mas time) kitsch. Everyone here know the
> American houses with a hole bunch of blinking lights. But those
> decorated hoses are nothing against our offices! Our phones have up to
> 136 lights (BLFs). You often have to wear sunglasses at the office ;-)
I am just curious what is BLF. I could only think of a BLF is a kitsch with
Bacon, Lettuce, and Fries that every American will probably like to enjoy. 

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Persistence-BLFs-or-Dear-PBX-please-remember-the-BLF-state-tp5962196p5962503.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From tculjaga at gmail.com  Wed Jan 26 15:32:44 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 26 Jan 2011 13:32:44 +0100
Subject: [Freeswitch-users] FS HA
In-Reply-To: <AANLkTi=QkaTTBPpXfWs3bMEZqGs=9L04Mda8JU+rWpBb@mail.gmail.com>
References: <AANLkTi=_BPuE5KX4D48JDoOW9U3netsr9uDU9Lo5CgYO@mail.gmail.com>
	<FACE40F5-BB80-479A-BFFC-F07F6BA681D2@freeswitch.org>
	<350D44B3-A176-47A9-84F4-72EB62F08299@gmail.com>
	<AANLkTimJLMjPJwthN8AhDq=1_JnDbdmocbgd=-=usyTT@mail.gmail.com>
	<AANLkTi=ksT-ZkdOyXiVQkir7a9z2-58PcVirmv4xs5Kg@mail.gmail.com>
	<AANLkTikeU5FAgE-HU=0a5whf3JiD5VUCkYkCCRt4u=R0@mail.gmail.com>
	<AANLkTim3=HVjX5GZVfcVjtSCm-2kuWatyVS+YV0RpNCB@mail.gmail.com>
	<AANLkTimB-jxEjrSD7pp45DcAmCmZH_q-dycT25c1gn06@mail.gmail.com>
	<AANLkTimwD=847nJ6Lz0gDrUh=nmELW261ch3QBAhDhf-@mail.gmail.com>
	<AANLkTinDjjDMDKykSKaBJKUsKs_1A+uJo=WXgJ60ix7T@mail.gmail.com>
	<AANLkTi=QkaTTBPpXfWs3bMEZqGs=9L04Mda8JU+rWpBb@mail.gmail.com>
Message-ID: <AANLkTi=Z1ktyc1HJngDs_pmu0_zf_awfpc22s1MJfxqv@mail.gmail.com>

hmmm,


it should be working :P



void sofia_glue_tech_track(sofia_profile_t *profile, switch_core_session_t
*session)
{
        switch_event_t *event;
        private_object_t *tech_pvt = (private_object_t *)
switch_core_session_get_private(session);

        if (!sofia_test_pflag(profile, PFLAG_TRACK_CALLS) ||
sofia_test_flag(tech_pvt, TFLAG_RECOVERING)) {
                return;
        }

        if (sofia_test_flag(tech_pvt, TFLAG_TRACKED)) {
                sofia_glue_tech_untrack(profile, session, SWITCH_TRUE);
        }

        if (switch_event_create(&event, SWITCH_EVENT_CHANNEL_DATA) ==
SWITCH_STATUS_SUCCESS) {
                switch_xml_t cdr = NULL;
                char *xml_cdr_text = NULL;

                if (switch_ivr_generate_xml_cdr(session, &cdr) ==
SWITCH_STATUS_SUCCESS) {
                        xml_cdr_text = switch_xml_toxml(cdr, SWITCH_FALSE);
                        switch_xml_free(cdr);
                }

                if (xml_cdr_text) {
                        char *sql;
                        sql = switch_mprintf("insert into sip_recovery
(runtime_uuid, profile_name, hostname, uuid, metadata) values
 ('%q','%q','%q','%q','%q')",

switch_core_get_uuid(), profile->name, mod_sofia_globals.hostname,
switch_core_session_get_uuid(session), xml_cdr_text);

                        sofia_glue_execute_sql_now(profile, &sql,
SWITCH_TRUE);
                        free(xml_cdr_text);
                        sofia_set_flag(tech_pvt, TFLAG_TRACKED);
                }

        }

}


im gonna dig why xml_cdr_text is empty.





On Wed, Jan 26, 2011 at 1:20 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> dam,  your metadata is awesome ... now i see the difference.
>
> i do have track calls enabled on both profiles (internal & external) and i
> have odbc configured as well.
>
>
>
> but my version is:
>
> freeswitch at cxss01> version
>
> FreeSWITCH Version 1.0.6 (svn-exported)
>
> freeswitch at cxss01>
>
>
>
> how can i make sure it is because of the version ?
>
>
>
>
>
> freeswitch at cxss01>
> freeswitch at cxss01>
> freeswitch at cxss01> sofia profile external start
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 debug [0]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 sip-trace [no]
> *2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 track-calls [true]*
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 rfc2833-pt [101]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 sip-port [5080]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 dialplan [XML]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 context [public]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 dtmf-duration [70]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:3279 Duration out of bounds,
> using default of 2000!
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 dtmf-type [inband]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 inbound-codec-prefs
> [PCMA,GSM]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 outbound-codec-prefs
> [PCMA,GSM]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 hold-music
> [local_stream://moh]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 rtp-timer-name [soft]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 local-network-acl
> [localnet.auto]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 manage-presence [false]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 inbound-codec-negotiation
> [generous]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 nonce-ttl [60]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 auth-calls [false]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 rtp-ip [195.88.212.30]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 sip-ip [195.88.212.30]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 ext-rtp-ip [auto-nat]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 ext-sip-ip [auto-nat]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 rtp-timeout-sec [300]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 rtp-hold-timeout-sec [1800]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 tls [false]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 tls-bind-params
> [transport=tls]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 tls-sip-port [5081]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 tls-cert-dir
> [/usr/local/freeswitch/conf/ssl]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 tls-version [tlsv1]
> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 odbc-dsn
> [COREFSdrv:adminfs:4F0s8wj]
> 2011-01-26 13:14:11.423588 [NOTICE] sofia_reg.c:2451 Added gateway '
> example.com' to profile 'external'
> 2011-01-26 13:14:11.423588 [NOTICE] sofia.c:3535 Connecting ODBC Profile
> external [sofia_reg_external]
> 2011-01-26 13:14:11.424818 [DEBUG] sofia.c:1317 Creating agent for external
> 2011-01-26 13:14:11.438512 [INFO] sofia_glue.c:4569 Connected ODBC DSN:
> COREFSdrv
> 2011-01-26 13:14:11.459433 [DEBUG] sofia.c:1353 Created agent for external
> 2011-01-26 13:14:11.459433 [DEBUG] sofia.c:1389 Set params for external
> 2011-01-26 13:14:11.459433 [DEBUG] sofia.c:1412 Activated db for external
> 2011-01-26 13:14:11.460603 [DEBUG] sofia.c:1439 Starting thread for
> external
>
> external started successfully
>
> freeswitch at cxss01>
> freeswitch at cxss01>
> freeswitch at cxss01>
> freeswitch at cxss01> sofia profile internal start
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 debug [0]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 sip-trace [no]
> *2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 track-calls [true]*
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 log-auth-failures [true]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 context [public]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 rfc2833-pt [101]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 sip-port [5060]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 dialplan [XML]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 dtmf-duration [70]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:3279 Duration out of bounds,
> using default of 2000!
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 inbound-codec-prefs
> [PCMA,GSM]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 outbound-codec-prefs
> [PCMA,GSM]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 rtp-timer-name [soft]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 rtp-ip [195.88.212.30]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 sip-ip [195.88.212.30]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 hold-music
> [local_stream://moh]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 apply-nat-acl [nat.auto]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 apply-inbound-acl [domains]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 local-network-acl
> [localnet.auto]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 dtmf-type [inband]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 record-path
> [/usr/local/freeswitch/recordings]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 record-template
> [${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 manage-presence [true]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 inbound-codec-negotiation
> [generous]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 tls [false]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 tls-bind-params
> [transport=tls]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 tls-sip-port [5061]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 tls-cert-dir
> [/usr/local/freeswitch/conf/ssl]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 tls-version [tlsv1]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 odbc-dsn
> [COREFSdrv:adminfs:4F0s8wj]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 inbound-late-negotiation
> [true]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 nonce-ttl [60]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 auth-calls [true]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755
> inbound-reg-force-matching-username [true]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 auth-all-packets [false]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 ext-rtp-ip [auto-nat]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 ext-sip-ip [auto-nat]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 rtp-timeout-sec [300]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 rtp-hold-timeout-sec [1800]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 force-register-domain
> [195.88.212.31]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 force-subscription-domain
> [195.88.212.31]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 force-register-db-domain
> [195.88.212.31]
> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 challenge-realm [auto_from]
> 2011-01-26 13:14:16.440442 [NOTICE] sofia_reg.c:2451 Added gateway
> 'hr-sbc01' to profile 'internal'
> 2011-01-26 13:14:16.441444 [NOTICE] sofia_reg.c:2451 Added gateway
> 'hr-sbc02' to profile 'internal'
> 2011-01-26 13:14:16.441444 [NOTICE] sofia.c:2040 Adding Alias
> [195.88.212.31] for profile [internal]
> 2011-01-26 13:14:16.441444 [NOTICE] sofia.c:3535 Connecting ODBC Profile
> internal [sofia_reg_internal]
> 2011-01-26 13:14:16.441444 [DEBUG] sofia.c:1317 Creating agent for internal
> 2011-01-26 13:14:16.455326 [INFO] sofia_glue.c:4569 Connected ODBC DSN:
> COREFSdrv
> 2011-01-26 13:14:16.476319 [DEBUG] sofia.c:1353 Created agent for internal
> 2011-01-26 13:14:16.476319 [DEBUG] sofia.c:1389 Set params for internal
> 2011-01-26 13:14:16.476319 [DEBUG] sofia.c:1412 Activated db for internal
> 2011-01-26 13:14:16.476319 [DEBUG] sofia.c:1439 Starting thread for
> internal
>
> internal started successfully
>
>
>
>
>
> i do have track-calls enabled on the sip profiles.
>
>
> and yes, FS is quite outdated:
>
> freeswitch at cxss01> version
>
> FreeSWITCH Version 1.0.6 (svn-exported)
>
> freeswitch at cxss01>
>
>
>
>
>
>
>
>
>
>
> On Wed, Jan 26, 2011 at 1:00 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> The sip recovery table should have a LOT more data than that, all stuffed
>> into a single column.
>> Do you have track calls on in all the involved profiles? I suppose it
>> really wouldn't be here otherwise.
>>
>> e.g. my sip_recovery table has this in the metadata field for a "simple"
>> MOH call. Does yours?
>>
>>
>> <cdr>
>>   <channel_data>
>>      <state>CS_EXECUTE</state>
>>     <direction>inbound</direction>
>>     <state_number>4</state_number>
>>     <flags>0=1;1=1;3=1;35=1;41=1;51=1</flags>
>>     <caps>1=1;2=1;3=1;4=1;5=1</caps>
>>   </channel_data>
>>   <variables>
>>     <direction>inbound</direction>
>>     <uuid>6cda8922-2943-11e0-b622-639ef8e43974</uuid>
>>     <sip_local_network_addr>178.79.147.47</sip_local_network_addr>
>>     <sip_network_ip>79.176.185.32</sip_network_ip>
>>     <sip_network_port>5072</sip_network_port>
>>     <sip_received_ip>79.176.yyy.32</sip_received_ip>
>>     <sip_received_port>5072</sip_received_port>
>>     <sip_via_protocol>udp</sip_via_protocol>
>>     <sip_authorized>true</sip_authorized>
>>     <sip_number_alias>1000</sip_number_alias>
>>     <sip_auth_username>1000</sip_auth_username>
>>     <sip_auth_realm>sip-ha.yyy.com</sip_auth_realm>
>>     <number_alias>1000</number_alias>
>>     <user_name>1000</user_name>
>>     <domain_name>sip.yyy.com</domain_name>
>>     <record_stereo>true</record_stereo>
>>     <default_areacode>208</default_areacode>
>>     <transfer_fallback_extension>operator</transfer_fallback_extension>
>>     <accountcode>1000</accountcode>
>>     <user_context>default</user_context>
>>     <effective_caller_id_name>Avi%20Marcus</effective_caller_id_name>
>>     <effective_caller_id_number>12013554419</effective_caller_id_number>
>>     <outbound_caller_id_name>Avi%20Marcus</outbound_caller_id_name>
>>     <outbound_caller_id_number>12013554419</outbound_caller_id_number>
>>     <nibble_account>1000</nibble_account>
>>     <sip_from_user>1000</sip_from_user>
>>     <sip_from_uri>1000%40sip-ha.yyy.com</sip_from_uri>
>>     <sip_from_host>sip-ha.yyy.com</sip_from_host>
>>     <sip_from_user_stripped>1000</sip_from_user_stripped>
>>     <sofia_profile_name>internal</sofia_profile_name>
>>     <sip_Remote-Party-ID>%3Csip%3A1000%40sip-ha.yyy.com
>> %3E%3Bscreen%3Dyes%3Bparty%3Dcalling</sip_Remote-Party-ID>
>>     <sip_cid_type>rpid</sip_cid_type>
>>
>>  <sip_full_via>SIP/2.0/UDP%20192.168.1.5%3A5072%3Bbranch%3Dz9hG4bK-329da816%3Brport%3D5072%3Breceived%3D79.176.185.32</sip_full_via>
>>     <sip_req_user>*9664</sip_req_user>
>>     <sip_req_uri>*9664%40sip-ha.yyy.com</sip_req_uri>
>>     <sip_req_host>sip-ha.yyy.com</sip_req_host>
>>     <sip_to_user>*9664</sip_to_user>
>>     <sip_to_uri>*9664%40sip-ha.yyy.com</sip_to_uri>
>>     <sip_to_host>sip-ha.yyy.com</sip_to_host>
>>     <sip_contact_user>1000</sip_contact_user>
>>     <sip_contact_port>5072</sip_contact_port>
>>     <sip_contact_uri>1000%4079.176.yyy.32%3A5072</sip_contact_uri>
>>     <sip_contact_host>79.176.yyy.32</sip_contact_host>
>>     <channel_name>sofia/internal/1000%40sip-ha.yyy.com</channel_name>
>>     <sip_user_agent>Linksys/SPA2102-5.2.10</sip_user_agent>
>>     <sip_via_host>192.168.1.5</sip_via_host>
>>     <sip_via_port>5072</sip_via_port>
>>     <sip_via_rport>5072</sip_via_rport>
>>     <max_forwards>70</max_forwards>
>>     <presence_id>1000%40sip-ha.yyy.com</presence_id>
>>
>>  <switch_r_sdp>v%3D0%0D%0Ao%3D-%2033038436%2033038436%20IN%20IP4%2079.176.185.32%0D%0As%3D-%0D%0Ac%3DIN%20IP4%2079.176.185.32%0D%0At%3D0%200%0D%0Am%3Daudio%2016422%20RTP/AVP%2018%200%202%204%208%2096%2097%2098%20100%20101%0D%0Aa%3Drtpmap%3A18%20G729/8000%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A2%20G726-32/8000%0D%0Aa%3Drtpmap%3A4%20G723/8000%0D%0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Drtpmap%3A96%20G726-40/8000%0D%0Aa%3Drtpmap%3A97%20G726-24/8000%0D%0Aa%3Drtpmap%3A98%20G726-16/8000%0D%0Aa%3Drtpmap%3A100%20NSE/8000%0D%0Aa%3Dfmtp%3A100%20192-193%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0Aa%3Dptime%3A20%0D%0A</switch_r_sdp>
>>
>>  <ep_codec_string>PCMU%408000h%4020i%4064000b,PCMA%408000h%4020i%4064000b</ep_codec_string>
>>     <nibble_current_balance>0.000000</nibble_current_balance>
>>
>>  <RFC2822_DATE>Wed,%2026%20Jan%202011%2013%3A57%3A18%20%2B0200</RFC2822_DATE>
>>     <zrtp_secure_media>true</zrtp_secure_media>
>>     <sip_use_codec_name>PCMU</sip_use_codec_name>
>>     <sip_use_codec_rate>8000</sip_use_codec_rate>
>>     <sip_use_codec_ptime>20</sip_use_codec_ptime>
>>     <read_codec>PCMU</read_codec>
>>     <read_rate>8000</read_rate>
>>     <write_codec>PCMU</write_codec>
>>     <write_rate>8000</write_rate>
>>     <local_media_ip>178.79.yyy.47</local_media_ip>
>>     <local_media_port>19600</local_media_port>
>>     <sip_use_pt>0</sip_use_pt>
>>     <rtp_use_ssrc>922702377</rtp_use_ssrc>
>>     <remote_media_ip>79.176.185.32</remote_media_ip>
>>     <remote_media_port>16422</remote_media_port>
>>
>>  <sip_local_sdp_str>v%3D0%0Ao%3DFreeSWITCH%201296023438%201296023439%20IN%20IP4%20178.79.147.47%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%20178.79.147.47%0At%3D0%200%0Am%3Daudio%2019600%20RTP/AVP%200%20101%0Aa%3Drtpmap%3A0%20PCMU/8000%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A</sip_local_sdp_str>
>>     <endpoint_disposition>ANSWER</endpoint_disposition>
>>     <sip_to_tag>vgcH82ctg6QaD</sip_to_tag>
>>     <sip_from_tag>9241a2573e4f702o0</sip_from_tag>
>>     <sip_cseq>102</sip_cseq>
>>     <sip_call_id>17217721-3d90de36%40192.168.1.5</sip_call_id>
>>     <sip_full_from>%3Csip%3A1000%40sip-ha.yyy.com
>> %3E%3Btag%3D9241a2573e4f702o0</sip_full_from>
>>     <sip_full_to>%3Csip%3A*9664%40sip-ha.yyy.com
>> %3E%3Btag%3DvgcH82ctg6QaD</sip_full_to>
>>     <playback_seconds>2</playback_seconds>
>>     <playback_ms>2000</playback_ms>
>>     <playback_samples>16000</playback_samples>
>>
>>  <current_application_data>local_stream%3A//moh</current_application_data>
>>     <current_application>playback</current_application>
>>   </variables>
>>   <app_log>
>>     <application app_name="hash"
>> app_data="insert/sip.yyy.com-spymap/1000/6cda8922-2943-11e0-b622-639ef8e43974"></application>
>>     <application app_name="hash"
>> app_data="insert/sip.yyy.com-last_dial/1000/*9664"></application>
>>     <application app_name="hash"
>> app_data="insert/sip.yyy.com-last_dial/global/6cda8922-2943-11e0-b622-639ef8e43974"></application>
>>     <application app_name="set" app_data="RFC2822_DATE=Wed, 26 Jan 2011
>> 13:57:18 +0200"></application>
>>     <application app_name="set"
>> app_data="zrtp_secure_media=true"></application>
>>     <application app_name="answer" app_data=""></application>
>>     <application app_name="playback"
>> app_data="silence_stream://2000"></application>
>>     <application app_name="execute_extension" app_data="is_zrtp_secure XML
>> features"></application>
>>     <application app_name="eval" app_data="not_secure"></application>
>>     <application app_name="playback"
>> app_data="local_stream://moh"></application>
>>   </app_log>
>>   <callflow dialplan="XML" profile_index="1">
>>     <extension name="global" number="*9664">
>>       <application app_name="hash"
>> app_data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"></application>
>>       <application app_name="hash"
>> app_data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"></application>
>>       <application app_name="hash"
>> app_data="insert/${domain_name}-last_dial/global/${uuid}"></application>
>>       <application app_name="set" app_data="RFC2822_DATE=${strftime(%a, %d
>> %b %Y %T %z)}"></application>
>>       <application app_name="set"
>> app_data="zrtp_secure_media=true"></application>
>>       <application app_name="answer" app_data=""></application>
>>       <application app_name="playback"
>> app_data="silence_stream://2000"></application>
>>       <application app_name="execute_extension" app_data="is_zrtp_secure
>> XML features"></application>
>>       <application app_name="playback"
>> app_data="local_stream://moh"></application>
>>       <sub_extensions>
>>         <extension name="is_zrtp_secure" number="is_zrtp_secure"
>> dialplan="XML">
>>           <application app_name="eval"
>> app_data="not_secure"></application>
>>         </extension>
>>       </sub_extensions>
>>     </extension>
>>     <caller_profile>
>>       <username>1000</username>
>>       <dialplan>XML</dialplan>
>>       <caller_id_name>1000</caller_id_name>
>>       <ani>1000</ani>
>>       <aniii></aniii>
>>       <caller_id_number>1000</caller_id_number>
>>       <network_addr>79.176.yyy.32</network_addr>
>>       <rdnis></rdnis>
>>       <destination_number>*9664</destination_number>
>>       <uuid>6cda8922-2943-11e0-b622-639ef8e43974</uuid>
>>       <source>mod_sofia</source>
>>       <context>default</context>
>>       <chan_name>sofia/internal/1000 at sip-ha.yyy.com</chan_name>
>>     </caller_profile>
>>     <times>
>>       <created_time>1296043038692089</created_time>
>>       <profile_created_time>1296043038692089</profile_created_time>
>>       <progress_time>0</progress_time>
>>       <progress_media_time>1296043038708572</progress_media_time>
>>       <answered_time>1296043038708572</answered_time>
>>       <hangup_time>0</hangup_time>
>>       <resurrect_time>0</resurrect_time>
>>       <transfer_time>0</transfer_time>
>>     </times>
>>   </callflow>
>> </cdr>
>>
>>
>>
>>
>> On Wed, Jan 26, 2011 at 12:53 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>> hello Anthony,
>>>
>>> ya, i was afraid of that... this means moving all my modules & patches to
>>> the latest git...
>>>
>>> radius cdr interface
>>> odbc cdr interface
>>>
>>> within mod_xml_cdr <= i can share this as a patch... if you are
>>> interested.
>>>
>>>
>>> mod_say_hr - really bad programming .. needs big re-factoring but lack of
>>> time :(
>>> patches for mod_say_de & mod_say_fr - because wrong playing in some
>>> scenarios
>>>
>>> and some small stuff i made within mod_commands... anyhow this was on the
>>> road.
>>>
>>>
>>> anyhow i was really into understanding the way the call context
>>> replication works... this may be the DB connection issue as the ODBC
>>> connection is reset on switchover (im switching entire resource .. floating
>>> IP, & database)
>>>
>>>
>>> so what really happens is the FS on 2nd node just getting sessions from
>>> sip_recovery
>>>
>>> from sofia_glue_recover: i see you are selecting sip_recovery table.
>>>
>>>
>>> select * from sip_recovery;
>>>    runtime_uuid profile_name hostname uuid metadata
>>> 24f77a1f-f315-4beb-904f-f779e8767c75 internal cxss01
>>> 102b2c7f-466f-4b6d-a795-e2cc25630e78
>>> 24f77a1f-f315-4beb-904f-f779e8767c75 external cxss01
>>> 52834d99-c1be-4832-bee9-7ba05238871d
>>>
>>>
>>>
>>> so, what do you do afterward?
>>>
>>>
>>> on the recovering node i see some portion of dialplan is executed and
>>> re-INVITEs being sent ... where do you get the DSP info for the recovering
>>> re-INVITE ?
>>>
>>>
>>>
>>>
>>> thanks for your help,
>>> Tihomir.
>>>
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From tculjaga at gmail.com  Wed Jan 26 16:15:39 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 26 Jan 2011 14:15:39 +0100
Subject: [Freeswitch-users] FS HA
In-Reply-To: <AANLkTi=Z1ktyc1HJngDs_pmu0_zf_awfpc22s1MJfxqv@mail.gmail.com>
References: <AANLkTi=_BPuE5KX4D48JDoOW9U3netsr9uDU9Lo5CgYO@mail.gmail.com>
	<FACE40F5-BB80-479A-BFFC-F07F6BA681D2@freeswitch.org>
	<350D44B3-A176-47A9-84F4-72EB62F08299@gmail.com>
	<AANLkTimJLMjPJwthN8AhDq=1_JnDbdmocbgd=-=usyTT@mail.gmail.com>
	<AANLkTi=ksT-ZkdOyXiVQkir7a9z2-58PcVirmv4xs5Kg@mail.gmail.com>
	<AANLkTikeU5FAgE-HU=0a5whf3JiD5VUCkYkCCRt4u=R0@mail.gmail.com>
	<AANLkTim3=HVjX5GZVfcVjtSCm-2kuWatyVS+YV0RpNCB@mail.gmail.com>
	<AANLkTimB-jxEjrSD7pp45DcAmCmZH_q-dycT25c1gn06@mail.gmail.com>
	<AANLkTimwD=847nJ6Lz0gDrUh=nmELW261ch3QBAhDhf-@mail.gmail.com>
	<AANLkTinDjjDMDKykSKaBJKUsKs_1A+uJo=WXgJ60ix7T@mail.gmail.com>
	<AANLkTi=QkaTTBPpXfWs3bMEZqGs=9L04Mda8JU+rWpBb@mail.gmail.com>
	<AANLkTi=Z1ktyc1HJngDs_pmu0_zf_awfpc22s1MJfxqv@mail.gmail.com>
Message-ID: <AANLkTim6MYRAfdG-mVrQDkAvVPDCmND7aP2wzQwyr3-q@mail.gmail.com>

ya,

I added a line:

switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_NOTICE,
"sip_recovery SQL => %s\n", sql);

and found that it actually inserts the xml_cdr_text into sip_recovery table
...


its just isql command line utility that was unable to display it ... maybe
because of xml tags.


Anyhow,  metadata looks like:


----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
<cdr>


<channel_data>


<state>CS_EXECUTE</state>


<direction>inbound</direction>


<state_number>4</state_number>


<flags>0=1;1=1;35=1;41=1;51=1</flags>


<caps>1=1;2=1;3=1</caps>


</channel_data>


<variables>


<uuid>4be94b6f-1df1-45bd-b920-1c55185dc3bd</uuid>


<sip_local_network_addr>195.88.212.30</sip_local_network_addr>


<sip_network_ip>195.88.212.39</sip_network_ip>


<sip_network_port>49603</sip_network_port>


<sip_received_ip>195.88.212.39</sip_received_ip>


<sip_received_port>49603</sip_received_port>


<sip_via_protocol>udp</sip_via_protocol>


<sip_from_user>38515492122</sip_from_user>


<sip_from_uri>38515492122%40195.88.212.39</sip_from_uri>


<sip_from_host>195.88.212.39</sip_from_host>


<sip_from_user_stripped>38515492122</sip_from_user_stripped>


<sofia_profile_name>external</sofia_profile_name>


<sip_Remote-Party-ID>%3Csip%3A38515492122%40195.88.212.39%3E%3Bparty%3Dcalling%3Bscreen%3Dyes%3Bprivacy%3Doff</sip_Remote-Party-ID>


<sip_cid_type>rpid</sip_cid_type>


<sip_full_via>SIP/2.0/UDP%20195.88.212.39%3A5060%3Bbranch%3Dz9hG4bK27A84C</sip_full_via>


<sip_req_user>38518880050</sip_req_user>


<sip_req_port>5080</sip_req_port>


<sip_req_uri>38518880050%40195.88.212.30%3A5080</sip_req_uri>


<sip_req_host>195.88.212.30</sip_req_host>


<sip_to_user>38518880050</sip_to_user>


<sip_to_uri>38518880050%40195.88.212.30</sip_to_uri>


<sip_to_host>195.88.212.30</sip_to_host>


<sip_contact_user>38515492122</sip_contact_user>


<sip_contact_port>5060</sip_contact_port>


<sip_contact_uri>38515492122%40195.88.212.39%3A5060</sip_contact_uri>


<sip_contact_host>195.88.212.39</sip_contact_host>


<channel_name>sofia/external/38515492122%40195.88.212.39</channel_name>


<sip_user_agent>Cisco-SIPGateway/IOS-12.x</sip_user_agent>


<sip_via_host>195.88.212.39</sip_via_host>


<sip_via_port>5060</sip_via_port>


<switch_r_sdp>v%3D0%0D%0Ao%3DCiscoSystemsSIP-GW-UserAgent%203218%205821%20IN%20IP4%20195.88.212.39%0D%0As%3DSIP%20Call%0D%0Ac%3DIN%20IP4%20195.88.212.39%0D%0At%3D0%200%0D%0Am%3Daudio%2016464%20RTP/AVP%208%20101%0D%0Ac%3DIN%20IP4%20195.88.212.39%0D%0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-16%0D%0Aa%3Dptime%3A20%0D%0A</switch_r_sdp>


<sip_use_codec_name>PCMA</sip_use_codec_name>


<sip_use_codec_rate>8000</sip_use_codec_rate>


<sip_use_codec_ptime>20</sip_use_codec_ptime>


<read_codec>PCMA</read_codec>


<read_rate>8000</read_rate>


<write_codec>PCMA</write_codec>


<write_rate>8000</write_rate>


<outside_call>true</outside_call>


<domain_name>195.88.212.31</domain_name>


<max_forwards>69</max_forwards>


<myDNIS>38518880050</myDNIS>


<sip_local_sdp_str>v%3D0%0Ao%3DFreeSWITCH%201296021156%201296021157%20IN%20IP4%20195.88.212.30%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%20195.88.212.30%0At%3D0%200%0Am%3Daudio%2025930%20RTP/AVP%208%0Aa%3Drtpmap%3A8%20PCMA/8000%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A</sip_local_sdp_str>


<local_media_ip>195.88.212.30</local_media_ip>


<local_media_port>25930</local_media_port>


<sip_use_pt>8</sip_use_pt>


<rtp_use_ssrc>1713917945</rtp_use_ssrc>


<remote_media_ip>195.88.212.39</remote_media_ip>


<remote_media_port>16464</remote_media_port>


<endpoint_disposition>ANSWER</endpoint_disposition>


<sip_to_tag>va1j6gN6FQKtp</sip_to_tag>


<sip_from_tag>CDA3D150-264D</sip_from_tag>


<sip_cseq>101</sip_cseq>


<sip_call_id>AED34DF0-288311E0-89BBD88B-B59C2BD1%40195.88.212.39</sip_call_id>


<sip_full_from>%3Csip%3A38515492122%40195.88.212.39%3E%3Btag%3DCDA3D150-264D</sip_full_from>


<sip_full_to>%3Csip%3A38518880050%40195.88.212.30%3E%3Btag%3Dva1j6gN6FQKtp</sip_full_to>


<language>hr</language>


<CALLID>h323-conf-id%3D4be94b6f-1df1-45bd-b920-1c55185dc3bd</CALLID>


<SERVICENUM>h323-prompt-id%3D38518880050</SERVICENUM>


<TRANSACTIONID>h323-ivr-out%3DtransactionID%3A1234</TRANSACTIONID>


<auto_hunt>true</auto_hunt>


<CALLINGNUMBER>38515492122</CALLINGNUMBER>


<billing_model>h323-billing-model%3D0</billing_model>


<return_code>h323-return-code%3D0</return_code>


<ANI_AUTH_RESULT>OK</ANI_AUTH_RESULT>


<UNAME>38515492122</UNAME>


<PASSWD>1234</PASSWD>


<export_vars>myDNIS,UNAME,PASSWD</export_vars>


<sound_prefix>/usr/local/freeswitch/sounds/hr/HR/teta1</sound_prefix>


<ENTER_DEST_PR>tone_stream%3A//%25(200,300,425,425)%3B%25(700,800,425,425)%3Bloops%3D5</ENTER_DEST_PR>


<DST_TIMEOUT>1000</DST_TIMEOUT>


<playback_seconds>0</playback_seconds>


<playback_ms>0</playback_ms>


<playback_samples>0</playback_samples>


<read_result>failure</read_result>


<DN_INT>false</DN_INT>


<DN_NAT>false</DN_NAT>


<DN_LOC>false</DN_LOC>


<DST_ERR_PR>ivr/_no_dest_entered2.wav</DST_ERR_PR>


<DST_RETRIES>2</DST_RETRIES>


<bind_meta_key></bind_meta_key>


<playback_delimiter>!</playback_delimiter>


<playback_terminators>%23*0123456789</playback_terminators>


<current_application_data>6%2020%20ivr/_no_dest_entered2.wav!tone_stream%3A//%25(200,300,425,425)%3B%25(700,800,425,425)%3Bloops%3D5%20DN%201000%20*</current_application_data>


<current_application>read</current_application>


</variables>


<app_log>

    <application app_name="set"
app_data="outside_call=true"></application>

    <application app_name="set"
app_data="domain_name=195.88.212.31"></application>

    <application app_name="transfer" app_data="38518880050 XML
default"></application>

    <application app_name="export"
app_data="myDNIS=38518880050"></application>

    <application app_name="set"
app_data="playback_terminators=#*0123456789"></application>

    <application app_name="answer"
app_data=""></application>

    <application app_name="sleep"
app_data="100"></application>

    <application app_name="set"
app_data="language=hr"></application>

    <application app_name="set"
app_data="CALLID=h323-conf-id=4be94b6f-1df1-45bd-b920-1c55185dc3bd"></application>

    <application app_name="set"
app_data="SERVICENUM=h323-prompt-id=38518880050"></application>

    <application app_name="set"
app_data="TRANSACTIONID=h323-ivr-out=transactionID:1234"></application>

    <application app_name="set"
app_data="auto_hunt=true"></application>

    <application app_name="execute_extension" app_data="ANIorPIN XML
NXIVR"></application>

    <application app_name="export"
app_data="DIALED_NUMBER="></application>

    <application app_name="set"
app_data="CALLINGNUMBER=38515492122"></application>

    <application app_name="auth_function" app_data="in , in 38515492122, in
1234, out
ANI_AUTH_RESULT"></application>

    <application app_name="execute_extension" app_data="CheckANI XML
NXIVR"></application>

    <application app_name="export"
app_data="UNAME=38515492122"></application>

    <application app_name="export"
app_data="PASSWD=1234"></application>

    <application app_name="set"
app_data="sound_prefix=/usr/local/freeswitch/sounds/hr/HR/teta1"></application>

    <application app_name="set"
app_data="ENTER_DEST_PR=tone_stream://%(200,300,425,425);%(700,800,425,425);loops=5"></application>

    <application app_name="set"
app_data="DST_TIMEOUT=1000"></application>

    <application app_name="execute_extension" app_data="GetDstNum XML
NXIVR"></application>

    <application app_name="set"
app_data="DST_RETRIES=1"></application>

    <application app_name="set"
app_data="bind_meta_key=#"></application>

    <application app_name="bind_meta_app" app_data="0 a a transfer::LongDTMF
XML
NXIVR"></application>

    <application app_name="set"
app_data="playback_delimiter=!"></application>

    <application app_name="set"
app_data="playback_terminators=#*0123456789"></application>

    <application app_name="start_dtmf"
app_data=""></application>

    <application app_name="read" app_data="6 20
!tone_stream://%(200,300,425,425);%(700,800,425,425);loops=5 DN 1000
*"></application>

    <application app_name="stop_dtmf"
app_data=""></application>

    <application app_name="execute_extension" app_data="TranslateLocal XML
NXIVR"></application>

    <application app_name="set"
app_data="DN_INT=false"></application>

    <application app_name="set"
app_data="DN_NAT=false"></application>

    <application app_name="set"
app_data="DN_LOC=false"></application>

    <application app_name="export"
app_data="DIALED_NUMBER="></application>

    <application app_name="export"
app_data="DIALED_NUMBER="></application>

    <application app_name="export"
app_data="DIALED_NUMBER="></application>

    <application app_name="execute_extension" app_data="ParseDN XML
NXIVR"></application>

    <application app_name="log" app_data="INFO ################# EMPTY
DESTINATION
################

"></application>

    <application app_name="set"
app_data="DST_ERR_PR=ivr/_no_dest_entered2.wav"></application>

    <application app_name="execute_extension" app_data="DstRetries XML
NXIVR"></application>

    <application app_name="execute_extension" app_data="GetDstNum XML
NXIVR"></application>

    <application app_name="set"
app_data="DST_RETRIES=2"></application>

    <application app_name="set"
app_data="bind_meta_key=#"></application>

    <application app_name="bind_meta_app" app_data="0 a a transfer::LongDTMF
XML
NXIVR"></application>

    <application app_name="set"
app_data="playback_delimiter=!"></application>

    <application app_name="set"
app_data="playback_terminators=#*0123456789"></application>

    <application app_name="start_dtmf"
app_data=""></application>

    <application app_name="read" app_data="6 20
ivr/_no_dest_entered2.wav!tone_stream://%(200,300,425,425);%(700,800,425,425);loops=5
DN 1000
*"></application>


</app_log>

  <callflow
dialplan="XML">

    <extension name="GetDstNum" number="GetDstNum"
current_app="read">

      <application app_name="set"
app_data="bind_meta_key=#"></application>

      <application app_name="bind_meta_app" app_data="0 a a
transfer::LongDTMF XML
NXIVR"></application>

      <application app_name="set"
app_data="playback_delimiter=!"></application>

      <application app_name="set"
app_data="playback_terminators=#*0123456789"></application>

      <application app_name="start_dtmf"
app_data=""></application>

      <application last_executed="true" app_name="read"
app_data="${DST_MIN_DIG} ${DST_MAX_DIG} ${DST_ERR_PR}!${ENTER_DEST_PR} DN
${DST_TIMEOUT}
*"></application>

      <application app_name="stop_dtmf"
app_data=""></application>

      <application app_name="execute_extension" app_data="TranslateLocal XML
NXIVR"></application>

      <application app_name="execute_extension" app_data="ParseDN XML
NXIVR"></application>

      <application app_name="auth_function" app_data="in ${DIALED_NUMBER},
in ${UNAME}, in ${PASSWD}, out
AUTH_RESULT"></application>

      <application app_name="set"
app_data="CALLINGNUMBER=${caller_id_number}"></application>

      <application app_name="execute_extension" app_data="PARSE_RET_CODE XML
NXIVR"></application>

      <application app_name="execute_extension" app_data="CheckDstNum XML
NXIVR"></application>


</extension>


<caller_profile>


<username>38515492122</username>


<dialplan>XML</dialplan>


<caller_id_name>38515492122</caller_id_name>


<ani>38515492122</ani>


<aniii></aniii>


<caller_id_number>38515492122</caller_id_number>


<network_addr>195.88.212.39</network_addr>


<rdnis>38518880050</rdnis>


<destination_number>GetDstNum</destination_number>


<uuid>4be94b6f-1df1-45bd-b920-1c55185dc3bd</uuid>


<source>mod_sofia</source>


<context>NXIVR</context>

      <chan_name>sofia/external/38515492122 at 195.88.212.39</chan_name>


</caller_profile>


</callflow>

</cdr>





is we have full info of the call ... and still no joy :)

well, its time for the update i guess... a lot of pain in the next hours.





On Wed, Jan 26, 2011 at 1:32 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> hmmm,
>
>
> it should be working :P
>
>
>
> void sofia_glue_tech_track(sofia_profile_t *profile, switch_core_session_t
> *session)
> {
>         switch_event_t *event;
>         private_object_t *tech_pvt = (private_object_t *)
> switch_core_session_get_private(session);
>
>         if (!sofia_test_pflag(profile, PFLAG_TRACK_CALLS) ||
> sofia_test_flag(tech_pvt, TFLAG_RECOVERING)) {
>                 return;
>         }
>
>         if (sofia_test_flag(tech_pvt, TFLAG_TRACKED)) {
>                 sofia_glue_tech_untrack(profile, session, SWITCH_TRUE);
>         }
>
>         if (switch_event_create(&event, SWITCH_EVENT_CHANNEL_DATA) ==
> SWITCH_STATUS_SUCCESS) {
>                 switch_xml_t cdr = NULL;
>                 char *xml_cdr_text = NULL;
>
>                 if (switch_ivr_generate_xml_cdr(session, &cdr) ==
> SWITCH_STATUS_SUCCESS) {
>                         xml_cdr_text = switch_xml_toxml(cdr, SWITCH_FALSE);
>                         switch_xml_free(cdr);
>                 }
>
>                 if (xml_cdr_text) {
>                         char *sql;
>                         sql = switch_mprintf("insert into sip_recovery
> (runtime_uuid, profile_name, hostname, uuid, metadata) values
>  ('%q','%q','%q','%q','%q')",
>
> switch_core_get_uuid(), profile->name, mod_sofia_globals.hostname,
> switch_core_session_get_uuid(session), xml_cdr_text);
>
>                         sofia_glue_execute_sql_now(profile, &sql,
> SWITCH_TRUE);
>                         free(xml_cdr_text);
>                         sofia_set_flag(tech_pvt, TFLAG_TRACKED);
>                 }
>
>         }
>
> }
>
>
> im gonna dig why xml_cdr_text is empty.
>
>
>
>
>
>
> On Wed, Jan 26, 2011 at 1:20 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> dam,  your metadata is awesome ... now i see the difference.
>>
>> i do have track calls enabled on both profiles (internal & external) and i
>> have odbc configured as well.
>>
>>
>>
>> but my version is:
>>
>> freeswitch at cxss01> version
>>
>> FreeSWITCH Version 1.0.6 (svn-exported)
>>
>> freeswitch at cxss01>
>>
>>
>>
>> how can i make sure it is because of the version ?
>>
>>
>>
>>
>>
>> freeswitch at cxss01>
>> freeswitch at cxss01>
>> freeswitch at cxss01> sofia profile external start
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 debug [0]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 sip-trace [no]
>> *2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 track-calls [true]*
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 rfc2833-pt [101]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 sip-port [5080]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 dialplan [XML]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 context [public]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 dtmf-duration [70]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:3279 Duration out of bounds,
>> using default of 2000!
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 dtmf-type [inband]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 inbound-codec-prefs
>> [PCMA,GSM]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 outbound-codec-prefs
>> [PCMA,GSM]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 hold-music
>> [local_stream://moh]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 rtp-timer-name [soft]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 local-network-acl
>> [localnet.auto]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 manage-presence [false]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 inbound-codec-negotiation
>> [generous]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 nonce-ttl [60]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 auth-calls [false]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 rtp-ip [195.88.212.30]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 sip-ip [195.88.212.30]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 ext-rtp-ip [auto-nat]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 ext-sip-ip [auto-nat]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 rtp-timeout-sec [300]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 rtp-hold-timeout-sec
>> [1800]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 tls [false]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 tls-bind-params
>> [transport=tls]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 tls-sip-port [5081]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 tls-cert-dir
>> [/usr/local/freeswitch/conf/ssl]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 tls-version [tlsv1]
>> 2011-01-26 13:14:11.423588 [DEBUG] sofia.c:2755 odbc-dsn
>> [COREFSdrv:adminfs:4F0s8wj]
>> 2011-01-26 13:14:11.423588 [NOTICE] sofia_reg.c:2451 Added gateway '
>> example.com' to profile 'external'
>> 2011-01-26 13:14:11.423588 [NOTICE] sofia.c:3535 Connecting ODBC Profile
>> external [sofia_reg_external]
>> 2011-01-26 13:14:11.424818 [DEBUG] sofia.c:1317 Creating agent for
>> external
>> 2011-01-26 13:14:11.438512 [INFO] sofia_glue.c:4569 Connected ODBC DSN:
>> COREFSdrv
>> 2011-01-26 13:14:11.459433 [DEBUG] sofia.c:1353 Created agent for external
>> 2011-01-26 13:14:11.459433 [DEBUG] sofia.c:1389 Set params for external
>> 2011-01-26 13:14:11.459433 [DEBUG] sofia.c:1412 Activated db for external
>> 2011-01-26 13:14:11.460603 [DEBUG] sofia.c:1439 Starting thread for
>> external
>>
>> external started successfully
>>
>> freeswitch at cxss01>
>> freeswitch at cxss01>
>> freeswitch at cxss01>
>> freeswitch at cxss01> sofia profile internal start
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 debug [0]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 sip-trace [no]
>> *2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 track-calls [true]*
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 log-auth-failures [true]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 context [public]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 rfc2833-pt [101]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 sip-port [5060]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 dialplan [XML]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 dtmf-duration [70]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:3279 Duration out of bounds,
>> using default of 2000!
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 inbound-codec-prefs
>> [PCMA,GSM]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 outbound-codec-prefs
>> [PCMA,GSM]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 rtp-timer-name [soft]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 rtp-ip [195.88.212.30]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 sip-ip [195.88.212.30]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 hold-music
>> [local_stream://moh]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 apply-nat-acl [nat.auto]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 apply-inbound-acl
>> [domains]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 local-network-acl
>> [localnet.auto]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 dtmf-type [inband]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 record-path
>> [/usr/local/freeswitch/recordings]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 record-template
>> [${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 manage-presence [true]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 inbound-codec-negotiation
>> [generous]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 tls [false]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 tls-bind-params
>> [transport=tls]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 tls-sip-port [5061]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 tls-cert-dir
>> [/usr/local/freeswitch/conf/ssl]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 tls-version [tlsv1]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 odbc-dsn
>> [COREFSdrv:adminfs:4F0s8wj]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 inbound-late-negotiation
>> [true]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 nonce-ttl [60]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 auth-calls [true]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755
>> inbound-reg-force-matching-username [true]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 auth-all-packets [false]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 ext-rtp-ip [auto-nat]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 ext-sip-ip [auto-nat]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 rtp-timeout-sec [300]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 rtp-hold-timeout-sec
>> [1800]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 force-register-domain
>> [195.88.212.31]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 force-subscription-domain
>> [195.88.212.31]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 force-register-db-domain
>> [195.88.212.31]
>> 2011-01-26 13:14:16.440442 [DEBUG] sofia.c:2755 challenge-realm
>> [auto_from]
>> 2011-01-26 13:14:16.440442 [NOTICE] sofia_reg.c:2451 Added gateway
>> 'hr-sbc01' to profile 'internal'
>> 2011-01-26 13:14:16.441444 [NOTICE] sofia_reg.c:2451 Added gateway
>> 'hr-sbc02' to profile 'internal'
>> 2011-01-26 13:14:16.441444 [NOTICE] sofia.c:2040 Adding Alias
>> [195.88.212.31] for profile [internal]
>> 2011-01-26 13:14:16.441444 [NOTICE] sofia.c:3535 Connecting ODBC Profile
>> internal [sofia_reg_internal]
>> 2011-01-26 13:14:16.441444 [DEBUG] sofia.c:1317 Creating agent for
>> internal
>> 2011-01-26 13:14:16.455326 [INFO] sofia_glue.c:4569 Connected ODBC DSN:
>> COREFSdrv
>> 2011-01-26 13:14:16.476319 [DEBUG] sofia.c:1353 Created agent for internal
>> 2011-01-26 13:14:16.476319 [DEBUG] sofia.c:1389 Set params for internal
>> 2011-01-26 13:14:16.476319 [DEBUG] sofia.c:1412 Activated db for internal
>> 2011-01-26 13:14:16.476319 [DEBUG] sofia.c:1439 Starting thread for
>> internal
>>
>> internal started successfully
>>
>>
>>
>>
>>
>> i do have track-calls enabled on the sip profiles.
>>
>>
>> and yes, FS is quite outdated:
>>
>> freeswitch at cxss01> version
>>
>> FreeSWITCH Version 1.0.6 (svn-exported)
>>
>> freeswitch at cxss01>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> On Wed, Jan 26, 2011 at 1:00 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> The sip recovery table should have a LOT more data than that, all stuffed
>>> into a single column.
>>> Do you have track calls on in all the involved profiles? I suppose it
>>> really wouldn't be here otherwise.
>>>
>>> e.g. my sip_recovery table has this in the metadata field for a "simple"
>>> MOH call. Does yours?
>>>
>>>
>>> <cdr>
>>>   <channel_data>
>>>      <state>CS_EXECUTE</state>
>>>     <direction>inbound</direction>
>>>     <state_number>4</state_number>
>>>     <flags>0=1;1=1;3=1;35=1;41=1;51=1</flags>
>>>     <caps>1=1;2=1;3=1;4=1;5=1</caps>
>>>   </channel_data>
>>>   <variables>
>>>     <direction>inbound</direction>
>>>     <uuid>6cda8922-2943-11e0-b622-639ef8e43974</uuid>
>>>     <sip_local_network_addr>178.79.147.47</sip_local_network_addr>
>>>     <sip_network_ip>79.176.185.32</sip_network_ip>
>>>     <sip_network_port>5072</sip_network_port>
>>>     <sip_received_ip>79.176.yyy.32</sip_received_ip>
>>>     <sip_received_port>5072</sip_received_port>
>>>     <sip_via_protocol>udp</sip_via_protocol>
>>>     <sip_authorized>true</sip_authorized>
>>>     <sip_number_alias>1000</sip_number_alias>
>>>     <sip_auth_username>1000</sip_auth_username>
>>>     <sip_auth_realm>sip-ha.yyy.com</sip_auth_realm>
>>>     <number_alias>1000</number_alias>
>>>     <user_name>1000</user_name>
>>>     <domain_name>sip.yyy.com</domain_name>
>>>     <record_stereo>true</record_stereo>
>>>     <default_areacode>208</default_areacode>
>>>     <transfer_fallback_extension>operator</transfer_fallback_extension>
>>>     <accountcode>1000</accountcode>
>>>     <user_context>default</user_context>
>>>     <effective_caller_id_name>Avi%20Marcus</effective_caller_id_name>
>>>     <effective_caller_id_number>12013554419</effective_caller_id_number>
>>>     <outbound_caller_id_name>Avi%20Marcus</outbound_caller_id_name>
>>>     <outbound_caller_id_number>12013554419</outbound_caller_id_number>
>>>     <nibble_account>1000</nibble_account>
>>>     <sip_from_user>1000</sip_from_user>
>>>     <sip_from_uri>1000%40sip-ha.yyy.com</sip_from_uri>
>>>     <sip_from_host>sip-ha.yyy.com</sip_from_host>
>>>     <sip_from_user_stripped>1000</sip_from_user_stripped>
>>>     <sofia_profile_name>internal</sofia_profile_name>
>>>     <sip_Remote-Party-ID>%3Csip%3A1000%40sip-ha.yyy.com
>>> %3E%3Bscreen%3Dyes%3Bparty%3Dcalling</sip_Remote-Party-ID>
>>>     <sip_cid_type>rpid</sip_cid_type>
>>>
>>>  <sip_full_via>SIP/2.0/UDP%20192.168.1.5%3A5072%3Bbranch%3Dz9hG4bK-329da816%3Brport%3D5072%3Breceived%3D79.176.185.32</sip_full_via>
>>>     <sip_req_user>*9664</sip_req_user>
>>>     <sip_req_uri>*9664%40sip-ha.yyy.com</sip_req_uri>
>>>     <sip_req_host>sip-ha.yyy.com</sip_req_host>
>>>     <sip_to_user>*9664</sip_to_user>
>>>     <sip_to_uri>*9664%40sip-ha.yyy.com</sip_to_uri>
>>>     <sip_to_host>sip-ha.yyy.com</sip_to_host>
>>>     <sip_contact_user>1000</sip_contact_user>
>>>     <sip_contact_port>5072</sip_contact_port>
>>>     <sip_contact_uri>1000%4079.176.yyy.32%3A5072</sip_contact_uri>
>>>     <sip_contact_host>79.176.yyy.32</sip_contact_host>
>>>     <channel_name>sofia/internal/1000%40sip-ha.yyy.com</channel_name>
>>>     <sip_user_agent>Linksys/SPA2102-5.2.10</sip_user_agent>
>>>     <sip_via_host>192.168.1.5</sip_via_host>
>>>     <sip_via_port>5072</sip_via_port>
>>>     <sip_via_rport>5072</sip_via_rport>
>>>     <max_forwards>70</max_forwards>
>>>     <presence_id>1000%40sip-ha.yyy.com</presence_id>
>>>
>>>  <switch_r_sdp>v%3D0%0D%0Ao%3D-%2033038436%2033038436%20IN%20IP4%2079.176.185.32%0D%0As%3D-%0D%0Ac%3DIN%20IP4%2079.176.185.32%0D%0At%3D0%200%0D%0Am%3Daudio%2016422%20RTP/AVP%2018%200%202%204%208%2096%2097%2098%20100%20101%0D%0Aa%3Drtpmap%3A18%20G729/8000%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A2%20G726-32/8000%0D%0Aa%3Drtpmap%3A4%20G723/8000%0D%0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Drtpmap%3A96%20G726-40/8000%0D%0Aa%3Drtpmap%3A97%20G726-24/8000%0D%0Aa%3Drtpmap%3A98%20G726-16/8000%0D%0Aa%3Drtpmap%3A100%20NSE/8000%0D%0Aa%3Dfmtp%3A100%20192-193%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0Aa%3Dptime%3A20%0D%0A</switch_r_sdp>
>>>
>>>  <ep_codec_string>PCMU%408000h%4020i%4064000b,PCMA%408000h%4020i%4064000b</ep_codec_string>
>>>     <nibble_current_balance>0.000000</nibble_current_balance>
>>>
>>>  <RFC2822_DATE>Wed,%2026%20Jan%202011%2013%3A57%3A18%20%2B0200</RFC2822_DATE>
>>>     <zrtp_secure_media>true</zrtp_secure_media>
>>>     <sip_use_codec_name>PCMU</sip_use_codec_name>
>>>     <sip_use_codec_rate>8000</sip_use_codec_rate>
>>>     <sip_use_codec_ptime>20</sip_use_codec_ptime>
>>>     <read_codec>PCMU</read_codec>
>>>     <read_rate>8000</read_rate>
>>>     <write_codec>PCMU</write_codec>
>>>     <write_rate>8000</write_rate>
>>>     <local_media_ip>178.79.yyy.47</local_media_ip>
>>>     <local_media_port>19600</local_media_port>
>>>     <sip_use_pt>0</sip_use_pt>
>>>     <rtp_use_ssrc>922702377</rtp_use_ssrc>
>>>     <remote_media_ip>79.176.185.32</remote_media_ip>
>>>     <remote_media_port>16422</remote_media_port>
>>>
>>>  <sip_local_sdp_str>v%3D0%0Ao%3DFreeSWITCH%201296023438%201296023439%20IN%20IP4%20178.79.147.47%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%20178.79.147.47%0At%3D0%200%0Am%3Daudio%2019600%20RTP/AVP%200%20101%0Aa%3Drtpmap%3A0%20PCMU/8000%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A</sip_local_sdp_str>
>>>     <endpoint_disposition>ANSWER</endpoint_disposition>
>>>     <sip_to_tag>vgcH82ctg6QaD</sip_to_tag>
>>>     <sip_from_tag>9241a2573e4f702o0</sip_from_tag>
>>>     <sip_cseq>102</sip_cseq>
>>>     <sip_call_id>17217721-3d90de36%40192.168.1.5</sip_call_id>
>>>     <sip_full_from>%3Csip%3A1000%40sip-ha.yyy.com
>>> %3E%3Btag%3D9241a2573e4f702o0</sip_full_from>
>>>     <sip_full_to>%3Csip%3A*9664%40sip-ha.yyy.com
>>> %3E%3Btag%3DvgcH82ctg6QaD</sip_full_to>
>>>     <playback_seconds>2</playback_seconds>
>>>     <playback_ms>2000</playback_ms>
>>>     <playback_samples>16000</playback_samples>
>>>
>>>  <current_application_data>local_stream%3A//moh</current_application_data>
>>>     <current_application>playback</current_application>
>>>   </variables>
>>>   <app_log>
>>>     <application app_name="hash"
>>> app_data="insert/sip.yyy.com-spymap/1000/6cda8922-2943-11e0-b622-639ef8e43974"></application>
>>>     <application app_name="hash"
>>> app_data="insert/sip.yyy.com-last_dial/1000/*9664"></application>
>>>     <application app_name="hash"
>>> app_data="insert/sip.yyy.com-last_dial/global/6cda8922-2943-11e0-b622-639ef8e43974"></application>
>>>     <application app_name="set" app_data="RFC2822_DATE=Wed, 26 Jan 2011
>>> 13:57:18 +0200"></application>
>>>     <application app_name="set"
>>> app_data="zrtp_secure_media=true"></application>
>>>     <application app_name="answer" app_data=""></application>
>>>     <application app_name="playback"
>>> app_data="silence_stream://2000"></application>
>>>     <application app_name="execute_extension" app_data="is_zrtp_secure
>>> XML features"></application>
>>>     <application app_name="eval" app_data="not_secure"></application>
>>>     <application app_name="playback"
>>> app_data="local_stream://moh"></application>
>>>   </app_log>
>>>   <callflow dialplan="XML" profile_index="1">
>>>     <extension name="global" number="*9664">
>>>       <application app_name="hash"
>>> app_data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"></application>
>>>       <application app_name="hash"
>>> app_data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"></application>
>>>       <application app_name="hash"
>>> app_data="insert/${domain_name}-last_dial/global/${uuid}"></application>
>>>       <application app_name="set" app_data="RFC2822_DATE=${strftime(%a,
>>> %d %b %Y %T %z)}"></application>
>>>       <application app_name="set"
>>> app_data="zrtp_secure_media=true"></application>
>>>       <application app_name="answer" app_data=""></application>
>>>       <application app_name="playback"
>>> app_data="silence_stream://2000"></application>
>>>       <application app_name="execute_extension" app_data="is_zrtp_secure
>>> XML features"></application>
>>>       <application app_name="playback"
>>> app_data="local_stream://moh"></application>
>>>       <sub_extensions>
>>>         <extension name="is_zrtp_secure" number="is_zrtp_secure"
>>> dialplan="XML">
>>>           <application app_name="eval"
>>> app_data="not_secure"></application>
>>>         </extension>
>>>       </sub_extensions>
>>>     </extension>
>>>     <caller_profile>
>>>       <username>1000</username>
>>>       <dialplan>XML</dialplan>
>>>       <caller_id_name>1000</caller_id_name>
>>>       <ani>1000</ani>
>>>       <aniii></aniii>
>>>       <caller_id_number>1000</caller_id_number>
>>>       <network_addr>79.176.yyy.32</network_addr>
>>>       <rdnis></rdnis>
>>>       <destination_number>*9664</destination_number>
>>>       <uuid>6cda8922-2943-11e0-b622-639ef8e43974</uuid>
>>>       <source>mod_sofia</source>
>>>       <context>default</context>
>>>       <chan_name>sofia/internal/1000 at sip-ha.yyy.com</chan_name>
>>>     </caller_profile>
>>>     <times>
>>>       <created_time>1296043038692089</created_time>
>>>       <profile_created_time>1296043038692089</profile_created_time>
>>>       <progress_time>0</progress_time>
>>>       <progress_media_time>1296043038708572</progress_media_time>
>>>       <answered_time>1296043038708572</answered_time>
>>>       <hangup_time>0</hangup_time>
>>>       <resurrect_time>0</resurrect_time>
>>>       <transfer_time>0</transfer_time>
>>>     </times>
>>>   </callflow>
>>> </cdr>
>>>
>>>
>>>
>>>
>>> On Wed, Jan 26, 2011 at 12:53 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>>
>>>> hello Anthony,
>>>>
>>>> ya, i was afraid of that... this means moving all my modules & patches
>>>> to the latest git...
>>>>
>>>> radius cdr interface
>>>> odbc cdr interface
>>>>
>>>> within mod_xml_cdr <= i can share this as a patch... if you are
>>>> interested.
>>>>
>>>>
>>>> mod_say_hr - really bad programming .. needs big re-factoring but lack
>>>> of time :(
>>>> patches for mod_say_de & mod_say_fr - because wrong playing in some
>>>> scenarios
>>>>
>>>> and some small stuff i made within mod_commands... anyhow this was on
>>>> the road.
>>>>
>>>>
>>>> anyhow i was really into understanding the way the call context
>>>> replication works... this may be the DB connection issue as the ODBC
>>>> connection is reset on switchover (im switching entire resource .. floating
>>>> IP, & database)
>>>>
>>>>
>>>> so what really happens is the FS on 2nd node just getting sessions from
>>>> sip_recovery
>>>>
>>>> from sofia_glue_recover: i see you are selecting sip_recovery table.
>>>>
>>>>
>>>> select * from sip_recovery;
>>>>    runtime_uuid profile_name hostname uuid metadata
>>>> 24f77a1f-f315-4beb-904f-f779e8767c75 internal cxss01
>>>> 102b2c7f-466f-4b6d-a795-e2cc25630e78
>>>> 24f77a1f-f315-4beb-904f-f779e8767c75 external cxss01
>>>> 52834d99-c1be-4832-bee9-7ba05238871d
>>>>
>>>>
>>>>
>>>> so, what do you do afterward?
>>>>
>>>>
>>>> on the recovering node i see some portion of dialplan is executed and
>>>> re-INVITEs being sent ... where do you get the DSP info for the recovering
>>>> re-INVITE ?
>>>>
>>>>
>>>>
>>>>
>>>> thanks for your help,
>>>> Tihomir.
>>>>
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
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From steveayre at gmail.com  Wed Jan 26 16:28:29 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 26 Jan 2011 13:28:29 +0000
Subject: [Freeswitch-users] FS HA
In-Reply-To: <AANLkTimwD=847nJ6Lz0gDrUh=nmELW261ch3QBAhDhf-@mail.gmail.com>
References: <AANLkTi=_BPuE5KX4D48JDoOW9U3netsr9uDU9Lo5CgYO@mail.gmail.com>
	<FACE40F5-BB80-479A-BFFC-F07F6BA681D2@freeswitch.org>
	<350D44B3-A176-47A9-84F4-72EB62F08299@gmail.com>
	<AANLkTimJLMjPJwthN8AhDq=1_JnDbdmocbgd=-=usyTT@mail.gmail.com>
	<AANLkTi=ksT-ZkdOyXiVQkir7a9z2-58PcVirmv4xs5Kg@mail.gmail.com>
	<AANLkTikeU5FAgE-HU=0a5whf3JiD5VUCkYkCCRt4u=R0@mail.gmail.com>
	<AANLkTim3=HVjX5GZVfcVjtSCm-2kuWatyVS+YV0RpNCB@mail.gmail.com>
	<AANLkTimB-jxEjrSD7pp45DcAmCmZH_q-dycT25c1gn06@mail.gmail.com>
	<AANLkTimwD=847nJ6Lz0gDrUh=nmELW261ch3QBAhDhf-@mail.gmail.com>
Message-ID: <AANLkTindsD6E7g=fqr0jr6sCtJrJg6M0Hb1P0LGKh6LR@mail.gmail.com>

It shouldn't be hard to update them. Most interfaces haven't changed, and
those that have won't have changed much. Any changes will show up as
compile-time errors. If you wrote them in the first place then you'll be
easily capable of updating them for the latest version.

If you have any patches that you think would be beneficial for the entire
community you can share them on http://jira.freeswitch.org, then they'll get
considered for adding to the trunk - what's the patch for?

Patches to trunk modules will always be easier for you to maintain if you
can get them into trunk.

-Steve


On 26 January 2011 10:53, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> hello Anthony,
>
> ya, i was afraid of that... this means moving all my modules & patches to
> the latest git...
>
> radius cdr interface
> odbc cdr interface
>

> within mod_xml_cdr <= i can share this as a patch... if you are interested.
>
>
> mod_say_hr - really bad programming .. needs big re-factoring but lack of
> time :(
> patches for mod_say_de & mod_say_fr - because wrong playing in some
> scenarios
>
> and some small stuff i made within mod_commands... anyhow this was on the
> road.
>

>
> anyhow i was really into understanding the way the call context replication
> works... this may be the DB connection issue as the ODBC connection is reset
> on switchover (im switching entire resource .. floating IP, & database)
>
>
> so what really happens is the FS on 2nd node just getting sessions from
> sip_recovery
>
> from sofia_glue_recover: i see you are selecting sip_recovery table.
>
>
> select * from sip_recovery;
>    runtime_uuid profile_name hostname uuid metadata
> 24f77a1f-f315-4beb-904f-f779e8767c75 internal cxss01
> 102b2c7f-466f-4b6d-a795-e2cc25630e78
> 24f77a1f-f315-4beb-904f-f779e8767c75 external cxss01
> 52834d99-c1be-4832-bee9-7ba05238871d
>
>
>
> so, what do you do afterward?
>
>
> on the recovering node i see some portion of dialplan is executed and
> re-INVITEs being sent ... where do you get the DSP info for the recovering
> re-INVITE ?
>
>
>
>
> thanks for your help,
> Tihomir.
>
>
>
>
>
> On Wed, Jan 26, 2011 at 1:12 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> There is an xml_cdr snapshot in sip_recovery sql table that contains
>> the data used for the recovery.
>> Also based on your logs, you are not using latest git.  I would try
>> that first before anything else.
>>
>>
>>
>> On Tue, Jan 25, 2011 at 7:24 AM, Tihomir Culjaga <tculjaga at gmail.com>
>> wrote:
>> > yap, both using the same config
>> >
>> >
>> > can you advice where is FS getting the SDP info for the re-INVITE ?
>> >
>> >
>> > On Tue, Jan 25, 2011 at 2:03 PM, Avi Marcus <avi at avimarcus.net> wrote:
>> >>
>> >> Are you using two machines for the HA? do both have the same configs?
>> >> -Avi
>> >>
>> >> On Tue, Jan 25, 2011 at 11:41 AM, Tihomir Culjaga <tculjaga at gmail.com>
>> >> wrote:
>> >> > Here is the debug: http://pastebin.freeswitch.org/15133
>> >> >
>> >> > i have set verbose_sdp=true in vars.xml as.
>> >> >   <X-PRE-PROCESS cmd="set" data="verbose_sdp=true"/>
>> >> >
>> >> > but not much to be seen of the verbose thing in the debug...
>> >> >
>> >> > Still, FS is sending a re-INVITE with wrong SDP. The call to be
>> >> > recovered is
>> >> > using ALAW... and ULAW is not supported.
>> >> > FS sends a re-INVITE with ULAW and CN in SDP. The re-INVITE is
>> rejected
>> >> > due
>> >> > to incompatible SDP.
>> >> >
>> >> > Where does FS get the information for the SDP in re-INVITE message?
>> >> >
>> >> >
>> >> > please advice,
>> >> > T.
>> >> >
>> >> >
>> >> >
>> >> > On Tue, Jan 25, 2011 at 12:01 AM, Tihomir Culjaga <
>> tculjaga at gmail.com>
>> >> > wrote:
>> >> >>
>> >> >> yap, i do have PCMA ... and the debug shows it correctly :=)
>> >> >>
>> >> >> i will try to see what it does with verbose. Post new debug
>> tomorrow.
>> >> >>
>> >> >> ty.
>> >> >>
>> >> >>
>> >> >> On Mon, Jan 24, 2011 at 11:29 PM, Steven Ayre <steveayre at gmail.com>
>> >> >> wrote:
>> >> >>>
>> >> >>> Brian, it has PCMU in the sdp, but Sofia thinks it's setting PCMA?
>> >> >>>
>> >> >>> Steve on iPhone
>> >> >>>
>> >> >>> On 24 Jan 2011, at 20:17, Brian West <brian at freeswitch.org> wrote:
>> >> >>>
>> >> >>> > What makes you think that fails?  It has ULAW and CN in the codec
>> >> >>> > list!
>> >> >>> >  Sounds like you need the verbose sdp... set the global variable
>> >> >>> > "verbose_sdp=true"
>> >> >>> >
>> >> >>> > /b
>> >> >>> >
>> >> >>> > On Jan 24, 2011, at 1:50 PM, Tihomir Culjaga wrote:
>> >> >>> >
>> >> >>> >>
>> >> >>> >>
>> >> >>> >> i configured FS HA and looks like its trying to recover the call
>> ..
>> >> >>> >> but the re-INVITE fails due to "wrong/missed" codec capability.
>> >> >>> >
>> >> >>> >
>> >> >>> > _______________________________________________
>> >> >>> > FreeSWITCH-users mailing list
>> >> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>> >
>> >> >>> >
>> >> >>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> > http://www.freeswitch.org
>> >> >>>
>> >> >>> _______________________________________________
>> >> >>> FreeSWITCH-users mailing list
>> >> >>> FreeSWITCH-users at lists.freeswitch.org
>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>
>> >> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> http://www.freeswitch.org
>> >> >>
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kond at nstel.ru  Wed Jan 26 17:11:27 2011
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Wed, 26 Jan 2011 17:11:27 +0300
Subject: [Freeswitch-users] freetdm (sangoma A101): isdn-sip display name
	interworking
Message-ID: <20110126141127.4FB30121A3@mail.nstel.ru>

Hi all,
can anybody please clarify if mod_freetdm and sangoma a101 pri card support isdn <-> sip display name interworking?
I mean translating the name in the isdn display information element into >From header and vice versa?
Thanks in advance, 
Nikolay.
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From dyatsin at sangoma.com  Wed Jan 26 17:34:49 2011
From: dyatsin at sangoma.com (David Yat Sin)
Date: Wed, 26 Jan 2011 09:34:49 -0500
Subject: [Freeswitch-users] freetdm (sangoma A101): isdn-sip display
 name interworking
In-Reply-To: <20110126141127.4FB30121A3@mail.nstel.ru>
References: <20110126141127.4FB30121A3@mail.nstel.ru>
Message-ID: <4D403109.30902@sangoma.com>

Hi Nikolay,
Yes.

The Display IE is only transmited from Network to User, so  if you are 
on the CPE side, on an incoming call, the Caller Name from the Display 
IE will be forwarded to the SIP header. And if you are on the Network 
side, on an outgoing call, the Caller Name will be transmitted in the 
Display IE.

David

*David Yat Sin, BEng*
*/Senior Software Engineer/*
Sangoma Technologies
100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada

	



t.   +1 800 388 2475 x119
t.   +1 905 474 1990 x119
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On 1/26/2011 9:11 AM, Nikolay Kondratyev wrote:
> Hi all,
> can anybody please clarify if mod_freetdm and sangoma a101 pri card 
> support isdn <-> sip display name interworking?
> I mean translating the name in the isdn display information element 
> into From header and vice versa?
> Thanks in advance,
> Nikolay.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From michal.bielicki at seventhsignal.de  Wed Jan 26 17:53:30 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Wed, 26 Jan 2011 15:53:30 +0100
Subject: [Freeswitch-users] FS HA
In-Reply-To: <AANLkTimwD=847nJ6Lz0gDrUh=nmELW261ch3QBAhDhf-@mail.gmail.com>
References: <AANLkTi=_BPuE5KX4D48JDoOW9U3netsr9uDU9Lo5CgYO@mail.gmail.com>
	<FACE40F5-BB80-479A-BFFC-F07F6BA681D2@freeswitch.org>
	<350D44B3-A176-47A9-84F4-72EB62F08299@gmail.com>
	<AANLkTimJLMjPJwthN8AhDq=1_JnDbdmocbgd=-=usyTT@mail.gmail.com>
	<AANLkTi=ksT-ZkdOyXiVQkir7a9z2-58PcVirmv4xs5Kg@mail.gmail.com>
	<AANLkTikeU5FAgE-HU=0a5whf3JiD5VUCkYkCCRt4u=R0@mail.gmail.com>
	<AANLkTim3=HVjX5GZVfcVjtSCm-2kuWatyVS+YV0RpNCB@mail.gmail.com>
	<AANLkTimB-jxEjrSD7pp45DcAmCmZH_q-dycT25c1gn06@mail.gmail.com>
	<AANLkTimwD=847nJ6Lz0gDrUh=nmELW261ch3QBAhDhf-@mail.gmail.com>
Message-ID: <70D5725B-8662-4250-BD06-9BA06DF8CEF5@seventhsignal.de>


Am 26.01.2011 um 11:53 schrieb Tihomir Culjaga:

> hello Anthony, 
> 
> ya, i was afraid of that... this means moving all my modules & patches to the latest git...
> 
> radius cdr interface

What about the one thats in tree ?

> odbc cdr interface
> 
> within mod_xml_cdr <= i can share this as a patch... if you are interested.
> 
> 
> mod_say_hr - really bad programming .. needs big re-factoring but lack of time :(
> patches for mod_say_de & mod_say_fr - because wrong playing in some scenarios

sent me those I will work them in.
> 
> and some small stuff i made within mod_commands... anyhow this was on the road.
> 
> 
> anyhow i was really into understanding the way the call context replication works... this may be the DB connection issue as the ODBC connection is reset on switchover (im switching entire resource .. floating IP, & database)
> 
> 
> so what really happens is the FS on 2nd node just getting sessions from sip_recovery 
> 
> from sofia_glue_recover: i see you are selecting sip_recovery table.
> 
> 
> select * from sip_recovery;
> runtime_uuid	profile_name	hostname	uuid	metadata
> 24f77a1f-f315-4beb-904f-f779e8767c75	internal	cxss01	102b2c7f-466f-4b6d-a795-e2cc25630e78	 
> 24f77a1f-f315-4beb-904f-f779e8767c75	external	cxss01	52834d99-c1be-4832-bee9-7ba05238871d	 
> 
> 
> 
> 
> so, what do you do afterward?
> 
> 
> on the recovering node i see some portion of dialplan is executed and re-INVITEs being sent ... where do you get the DSP info for the recovering re-INVITE ?
> 
> 
> 
> 
> thanks for your help,
> Tihomir.
> 
> 
> 
> 
> On Wed, Jan 26, 2011 at 1:12 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> There is an xml_cdr snapshot in sip_recovery sql table that contains
> the data used for the recovery.
> Also based on your logs, you are not using latest git.  I would try
> that first before anything else.
> 
> 
> 
> On Tue, Jan 25, 2011 at 7:24 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
> > yap, both using the same config
> >
> >
> > can you advice where is FS getting the SDP info for the re-INVITE ?
> >
> >
> > On Tue, Jan 25, 2011 at 2:03 PM, Avi Marcus <avi at avimarcus.net> wrote:
> >>
> >> Are you using two machines for the HA? do both have the same configs?
> >> -Avi
> >>
> >> On Tue, Jan 25, 2011 at 11:41 AM, Tihomir Culjaga <tculjaga at gmail.com>
> >> wrote:
> >> > Here is the debug: http://pastebin.freeswitch.org/15133
> >> >
> >> > i have set verbose_sdp=true in vars.xml as.
> >> >   <X-PRE-PROCESS cmd="set" data="verbose_sdp=true"/>
> >> >
> >> > but not much to be seen of the verbose thing in the debug...
> >> >
> >> > Still, FS is sending a re-INVITE with wrong SDP. The call to be
> >> > recovered is
> >> > using ALAW... and ULAW is not supported.
> >> > FS sends a re-INVITE with ULAW and CN in SDP. The re-INVITE is rejected
> >> > due
> >> > to incompatible SDP.
> >> >
> >> > Where does FS get the information for the SDP in re-INVITE message?
> >> >
> >> >
> >> > please advice,
> >> > T.
> >> >
> >> >
> >> >
> >> > On Tue, Jan 25, 2011 at 12:01 AM, Tihomir Culjaga <tculjaga at gmail.com>
> >> > wrote:
> >> >>
> >> >> yap, i do have PCMA ... and the debug shows it correctly :=)
> >> >>
> >> >> i will try to see what it does with verbose. Post new debug tomorrow.
> >> >>
> >> >> ty.
> >> >>
> >> >>
> >> >> On Mon, Jan 24, 2011 at 11:29 PM, Steven Ayre <steveayre at gmail.com>
> >> >> wrote:
> >> >>>
> >> >>> Brian, it has PCMU in the sdp, but Sofia thinks it's setting PCMA?
> >> >>>
> >> >>> Steve on iPhone
> >> >>>
> >> >>> On 24 Jan 2011, at 20:17, Brian West <brian at freeswitch.org> wrote:
> >> >>>
> >> >>> > What makes you think that fails?  It has ULAW and CN in the codec
> >> >>> > list!
> >> >>> >  Sounds like you need the verbose sdp... set the global variable
> >> >>> > "verbose_sdp=true"
> >> >>> >
> >> >>> > /b
> >> >>> >
> >> >>> > On Jan 24, 2011, at 1:50 PM, Tihomir Culjaga wrote:
> >> >>> >
> >> >>> >>
> >> >>> >>
> >> >>> >> i configured FS HA and looks like its trying to recover the call ..
> >> >>> >> but the re-INVITE fails due to "wrong/missed" codec capability.
> >> >>> >
> >> >>> >
> >> >>> > _______________________________________________
> >> >>> > FreeSWITCH-users mailing list
> >> >>> > FreeSWITCH-users at lists.freeswitch.org
> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >
> >> >>> >
> >> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> > http://www.freeswitch.org
> >> >>>
> >> >>> _______________________________________________
> >> >>> FreeSWITCH-users mailing list
> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>
> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> http://www.freeswitch.org
> >> >>
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> 
> 
> --
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de

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From mkopacki at gmail.com  Wed Jan 26 15:38:25 2011
From: mkopacki at gmail.com (Michal Kopacki)
Date: Wed, 26 Jan 2011 13:38:25 +0100
Subject: [Freeswitch-users] network scenario doubts
In-Reply-To: <4D3FD3B0.4080309@gmail.com>
References: <4D3FD3B0.4080309@gmail.com>
Message-ID: <AANLkTikre6NZjg9KZgckxYEJO49s+B59w5W0aKt32ROT@mail.gmail.com>

? ? ?Hello,

?This is my first post to this list, so hello everyone.

?I'm at the very beginning of freeswitch journey and i have a problem
with fit FS to my network scenario.

OS: fedora 13 (x86_64)
FS: 1.0.7 (compiled from sources), default config

Desired scenario:

softphone -> mydomain.com (with ip 193.59.72.xx) -> my isp network ->
10.25.48.xx on eth1 -> FS -> 192.168.0.1 on eth0 -> softphone

And now, I'm able to connect to FS from my local network, but I'm not
able to connect from outside (neither domain nor ip). In internal.xml
I set internal ip of server and i wanted to set external ip in
external.xml, but there is a question: which one ? ?In case of
193.59.72.xx external profile didnt' start and with 10.25.48.xx
outside softphone didn't register. ?I checked with netstat and
realized that port 5060 is bind to internal nic only and 5080 to
external nic (10.25.48.xx) and I have no idea what next.

Is it even possible to work with such network scenario ? I would be
grateful for pointing me to right direction or maybe propose different
approach.

--
Best regards,
Michal


From nemrod at reaper.pl  Wed Jan 26 16:20:06 2011
From: nemrod at reaper.pl (Michal Kopacki)
Date: Wed, 26 Jan 2011 14:20:06 +0100
Subject: [Freeswitch-users] network scenario doubts
Message-ID: <4D401F86.7060002@reaper.pl>

       Hello,

   This is my first post to this list, so hello everyone.

   I'm at the very beginning of freeswitch journey and i have a problem 
with fit FS to my network scenario.

OS: fedora 13 (x86_64)
FS: 1.0.7 (compiled from sources), default config

Desired scenario:

softphone -> mydomain.com (with ip 193.59.72.xx) -> my isp network -> 
10.25.48.xx on eth1 -> FS <- 192.168.0.1 on eth0 <- softphone

And now, I'm able to connect to FS from my local network, but I'm not 
able to connect from outside (neither domain nor ip). In internal.xml I 
set internal ip of server and i wanted to set external ip in 
external.xml, but there is a question: which one ?  In case of 
193.59.72.xx external profile didnt' start and with 10.25.48.xx outside 
softphone didn't register.  I checked with netstat and realized that 
port 5060 is bind to internal nic only and 5080 to external nic 
(10.25.48.xx) and I have no idea what next.

Is it even possible to work with such network scenario ? I would be 
grateful for pointing me to right direction or maybe propose different 
approach.

-- 
Best regards,
Michal

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From patrick.plattes at niemann-frey.info  Wed Jan 26 15:50:57 2011
From: patrick.plattes at niemann-frey.info (Patrick Plattes)
Date: Wed, 26 Jan 2011 13:50:57 +0100
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <1296044574879-5962503.post@n2.nabble.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<1296044574879-5962503.post@n2.nabble.com>
Message-ID: <AANLkTi=K5VMyz0ebcwnJQc5E5cSUMe_T_WKu9kyMetqr@mail.gmail.com>

2011/1/26 mazilo <Nabble at slickdeals.endjunk.com>:
> I am just curious what is BLF. I could only think of a BLF is a kitsch with
> Bacon, Lettuce, and Fries that every American will probably like to enjoy.

Busy Lamp Field (BLF) is a light on an IP phone which tells you
whether another extension connected to the same PBX is busy or not.
The snom 37 has 12 leds on the right side
(http://blog.tmcnet.com/beyond-voip/snom%20370.jpg)


From steveayre at gmail.com  Wed Jan 26 18:09:19 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 26 Jan 2011 15:09:19 +0000
Subject: [Freeswitch-users] network scenario doubts
In-Reply-To: <4D401F86.7060002@reaper.pl>
References: <4D401F86.7060002@reaper.pl>
Message-ID: <A4459666-7A4C-46F0-BB89-29CD028275F6@gmail.com>

Please only send to the list once. You might not get an immediate response, but someone will eventually reply. If you need a quicker answer you can try joining #freeswitch on freenode IRC.

Steve on iPhone

On 26 Jan 2011, at 13:20, Michal Kopacki <nemrod at reaper.pl> wrote:

>       Hello, 
> 
>   This is my first post to this list, so hello everyone. 
> 
>   I'm at the very beginning of freeswitch journey and i have a problem with fit FS to my network scenario. 
> 
> OS: fedora 13 (x86_64) 
> FS: 1.0.7 (compiled from sources), default config 
> 
> Desired scenario: 
> 
> softphone -> mydomain.com (with ip 193.59.72.xx) -> my isp network -> 10.25.48.xx on eth1 -> FS <- 192.168.0.1 on eth0 <- softphone 
> 
> And now, I'm able to connect to FS from my local network, but I'm not able to connect from outside (neither domain nor ip). In internal.xml I set internal ip of server and i wanted to set external ip in external.xml, but there is a question: which one ?  In case of 193.59.72.xx external profile didnt' start and with 10.25.48.xx outside softphone didn't register.  I checked with netstat and realized that port 5060 is bind to internal nic only and 5080 to external nic (10.25.48.xx) and I have no idea what next. 
> 
> Is it even possible to work with such network scenario ? I would be grateful for pointing me to right direction or maybe propose different approach. 
> 
> -- 
> Best regards, 
> Michal 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From Nabble at slickdeals.endjunk.com  Wed Jan 26 18:17:46 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 26 Jan 2011 07:17:46 -0800 (PST)
Subject: [Freeswitch-users] network scenario doubts
In-Reply-To: <4D401F86.7060002@reaper.pl>
References: <4D401F86.7060002@reaper.pl>
Message-ID: <1296055066658-5963004.post@n2.nabble.com>



Michal Kopacki-2 wrote:
> softphone -> mydomain.com (with ip 193.59.72.xx) -> my isp network -> 
> 10.25.48.xx on eth1 -> FS <- 192.168.0.1 on eth0 <- softphone
R U saying your ISP assigned your a 10.x.x.x IP space? If so, then you are
out of luck trying to register your device/softphone to your FS from outside
because a 10.x.x.x is a private subnet class A.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/network-scenario-doubts-tp5962941p5963004.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From wstephen80 at gmail.com  Wed Jan 26 18:31:39 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Wed, 26 Jan 2011 16:31:39 +0100
Subject: [Freeswitch-users] How to check call state in lua script?
Message-ID: <AANLkTimU9_4NUpdh2tsO48PZpno6BX3prDEcm_ZjWESD@mail.gmail.com>

Hi,
I have difficult to check the state of an originated session in lua script.

I want to know when the originated session is ringing back or progressing or
answered.

I have tried with session:getState() but returns always CS_SOFT_EXECUTE.

My lua test script is called from dialplan with a:

        <action application="lua" data="test1.lua"/>


And the "test1.lua" script is:

    session2 = freeswitch.Session("sofia/external/xxxxxx at a.b.c.d");

    while (session:ready() and session2:ready()) do
          state2 = session2:getVariable("channel_call_state");
          freeswitch.consoleLog("warning", "State2 = " .. state2 .. "\n");
          session:execute("sleep", "500");
    end

    if (session:ready()) then
          session:hangup();
    end

    if (session2:ready()) then
          session2:hangup();
    end

This script fails because the session:getVariable("channel_call_state")
returns nil.

What is the correct way to know the state of a call?

Stephen
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From steveayre at gmail.com  Wed Jan 26 18:34:54 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 26 Jan 2011 15:34:54 +0000
Subject: [Freeswitch-users] FS HA
In-Reply-To: <AANLkTindsD6E7g=fqr0jr6sCtJrJg6M0Hb1P0LGKh6LR@mail.gmail.com>
References: <AANLkTi=_BPuE5KX4D48JDoOW9U3netsr9uDU9Lo5CgYO@mail.gmail.com>
	<FACE40F5-BB80-479A-BFFC-F07F6BA681D2@freeswitch.org>
	<350D44B3-A176-47A9-84F4-72EB62F08299@gmail.com>
	<AANLkTimJLMjPJwthN8AhDq=1_JnDbdmocbgd=-=usyTT@mail.gmail.com>
	<AANLkTi=ksT-ZkdOyXiVQkir7a9z2-58PcVirmv4xs5Kg@mail.gmail.com>
	<AANLkTikeU5FAgE-HU=0a5whf3JiD5VUCkYkCCRt4u=R0@mail.gmail.com>
	<AANLkTim3=HVjX5GZVfcVjtSCm-2kuWatyVS+YV0RpNCB@mail.gmail.com>
	<AANLkTimB-jxEjrSD7pp45DcAmCmZH_q-dycT25c1gn06@mail.gmail.com>
	<AANLkTimwD=847nJ6Lz0gDrUh=nmELW261ch3QBAhDhf-@mail.gmail.com>
	<AANLkTindsD6E7g=fqr0jr6sCtJrJg6M0Hb1P0LGKh6LR@mail.gmail.com>
Message-ID: <AANLkTikjBm=XT4FrY1N5n_RDQ-TvjANuy149+f4Ai8sH@mail.gmail.com>

Tihomir,

Are you saying all of those are implemented in mod_xml_cdr?

If so, that's probably not the best place to implement them. They'd be
better off in their own separate modules.

- added radius accounting interface
>

There's already a module for this I believe?
http://wiki.freeswitch.org/wiki/Mod_radius_cdr

- added odbc cdr interface
>

This would be better in its own module.

- not the best place but
>   added a function to build up my dial-string... basically the same
> functionality of LCR module but much more simpler :=)
>

Indeed it's not, I don't see how CDRs and LCR is related.
This would be better as something like mod_custom_lcr. mod_lcr may well do
what you need though with custom SQL.

Warm Regards,
-Steve





On 26 January 2011 14:03, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> patch for mod_xml_cdr
>
> - added radius accounting interface
> - added odbc cdr interface
>
> - not the best place but
>   added a function to build up my dial-string... basically the same
> functionality of LCR module but much more simpler :=)
>




On 26 January 2011 13:28, Steven Ayre <steveayre at gmail.com> wrote:

> It shouldn't be hard to update them. Most interfaces haven't changed, and
> those that have won't have changed much. Any changes will show up as
> compile-time errors. If you wrote them in the first place then you'll be
> easily capable of updating them for the latest version.
>
> If you have any patches that you think would be beneficial for the entire
> community you can share them on http://jira.freeswitch.org, then they'll
> get considered for adding to the trunk - what's the patch for?
>
> Patches to trunk modules will always be easier for you to maintain if you
> can get them into trunk.
>
> -Steve
>
>
>
> On 26 January 2011 10:53, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>
>> hello Anthony,
>>
>> ya, i was afraid of that... this means moving all my modules & patches to
>> the latest git...
>>
>> radius cdr interface
>> odbc cdr interface
>>
>
>> within mod_xml_cdr <= i can share this as a patch... if you are
>> interested.
>>
>>
>> mod_say_hr - really bad programming .. needs big re-factoring but lack of
>> time :(
>> patches for mod_say_de & mod_say_fr - because wrong playing in some
>> scenarios
>>
>> and some small stuff i made within mod_commands... anyhow this was on the
>> road.
>>
>
>>
>> anyhow i was really into understanding the way the call context
>> replication works... this may be the DB connection issue as the ODBC
>> connection is reset on switchover (im switching entire resource .. floating
>> IP, & database)
>>
>>
>> so what really happens is the FS on 2nd node just getting sessions from
>> sip_recovery
>>
>> from sofia_glue_recover: i see you are selecting sip_recovery table.
>>
>>
>> select * from sip_recovery;
>>    runtime_uuid profile_name hostname uuid metadata
>> 24f77a1f-f315-4beb-904f-f779e8767c75 internal cxss01
>> 102b2c7f-466f-4b6d-a795-e2cc25630e78
>> 24f77a1f-f315-4beb-904f-f779e8767c75 external cxss01
>> 52834d99-c1be-4832-bee9-7ba05238871d
>>
>>
>>
>> so, what do you do afterward?
>>
>>
>> on the recovering node i see some portion of dialplan is executed and
>> re-INVITEs being sent ... where do you get the DSP info for the recovering
>> re-INVITE ?
>>
>>
>>
>>
>> thanks for your help,
>> Tihomir.
>>
>>
>>
>>
>>
>> On Wed, Jan 26, 2011 at 1:12 AM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> There is an xml_cdr snapshot in sip_recovery sql table that contains
>>> the data used for the recovery.
>>> Also based on your logs, you are not using latest git.  I would try
>>> that first before anything else.
>>>
>>>
>>>
>>> On Tue, Jan 25, 2011 at 7:24 AM, Tihomir Culjaga <tculjaga at gmail.com>
>>> wrote:
>>> > yap, both using the same config
>>> >
>>> >
>>> > can you advice where is FS getting the SDP info for the re-INVITE ?
>>> >
>>> >
>>> > On Tue, Jan 25, 2011 at 2:03 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>> >>
>>> >> Are you using two machines for the HA? do both have the same configs?
>>> >> -Avi
>>> >>
>>> >> On Tue, Jan 25, 2011 at 11:41 AM, Tihomir Culjaga <tculjaga at gmail.com
>>> >
>>> >> wrote:
>>> >> > Here is the debug: http://pastebin.freeswitch.org/15133
>>> >> >
>>> >> > i have set verbose_sdp=true in vars.xml as.
>>> >> >   <X-PRE-PROCESS cmd="set" data="verbose_sdp=true"/>
>>> >> >
>>> >> > but not much to be seen of the verbose thing in the debug...
>>> >> >
>>> >> > Still, FS is sending a re-INVITE with wrong SDP. The call to be
>>> >> > recovered is
>>> >> > using ALAW... and ULAW is not supported.
>>> >> > FS sends a re-INVITE with ULAW and CN in SDP. The re-INVITE is
>>> rejected
>>> >> > due
>>> >> > to incompatible SDP.
>>> >> >
>>> >> > Where does FS get the information for the SDP in re-INVITE message?
>>> >> >
>>> >> >
>>> >> > please advice,
>>> >> > T.
>>> >> >
>>> >> >
>>> >> >
>>> >> > On Tue, Jan 25, 2011 at 12:01 AM, Tihomir Culjaga <
>>> tculjaga at gmail.com>
>>> >> > wrote:
>>> >> >>
>>> >> >> yap, i do have PCMA ... and the debug shows it correctly :=)
>>> >> >>
>>> >> >> i will try to see what it does with verbose. Post new debug
>>> tomorrow.
>>> >> >>
>>> >> >> ty.
>>> >> >>
>>> >> >>
>>> >> >> On Mon, Jan 24, 2011 at 11:29 PM, Steven Ayre <steveayre at gmail.com
>>> >
>>> >> >> wrote:
>>> >> >>>
>>> >> >>> Brian, it has PCMU in the sdp, but Sofia thinks it's setting PCMA?
>>> >> >>>
>>> >> >>> Steve on iPhone
>>> >> >>>
>>> >> >>> On 24 Jan 2011, at 20:17, Brian West <brian at freeswitch.org>
>>> wrote:
>>> >> >>>
>>> >> >>> > What makes you think that fails?  It has ULAW and CN in the
>>> codec
>>> >> >>> > list!
>>> >> >>> >  Sounds like you need the verbose sdp... set the global variable
>>> >> >>> > "verbose_sdp=true"
>>> >> >>> >
>>> >> >>> > /b
>>> >> >>> >
>>> >> >>> > On Jan 24, 2011, at 1:50 PM, Tihomir Culjaga wrote:
>>> >> >>> >
>>> >> >>> >>
>>> >> >>> >>
>>> >> >>> >> i configured FS HA and looks like its trying to recover the
>>> call ..
>>> >> >>> >> but the re-INVITE fails due to "wrong/missed" codec capability.
>>> >> >>> >
>>> >> >>> >
>>> >> >>> > _______________________________________________
>>> >> >>> > FreeSWITCH-users mailing list
>>> >> >>> > FreeSWITCH-users at lists.freeswitch.org
>>> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >>> >
>>> >> >>> >
>>> >> >>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >>> > http://www.freeswitch.org
>>> >> >>>
>>> >> >>> _______________________________________________
>>> >> >>> FreeSWITCH-users mailing list
>>> >> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >>>
>>> >> >>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >>> http://www.freeswitch.org
>>> >> >>
>>> >> >
>>> >> >
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> > http://www.freeswitch.org
>>> >> >
>>> >> >
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From vik_kom at mail.ru  Wed Jan 26 18:14:29 2011
From: vik_kom at mail.ru (vik_kom)
Date: Wed, 26 Jan 2011 07:14:29 -0800 (PST)
Subject: [Freeswitch-users] Sangoma A104D
Message-ID: <1296054869082-5962992.post@n2.nabble.com>


Hello ALL, 
I installed FWS 1.0.7 + Sangoma A104D
CentOS 5.5 
wanpipe-3.5.18 
libsng_isdn-7.0.0.x86_64.
When I try call from outside, I see such messages

freeswitch at internal> 2011-01-26 17:59:44.501244 [INFO]
ftmod_sangoma_isdn_stack_rcv.c:75 [s1c1][1:1] Received SETUP (suId:1
suInstId:0 spInstId:35)                                                                                                         
2011-01-26 17:59:44.501244 [INFO] ftmod_sangoma_isdn_trace.c:223 [SNGISDN
Q931] wp1 FRAME INCOMING:  Prot Disc:Q.931/I.451 (0x08)                                                                                                                           
  Call Ref:1136 (Originating side)                                                                                            
  Type:SETUP (0x5)                                                                                                            
  Bearer Capability:Coding:ITU-T(0) TransferCap:Speech(0) TransferRate:64
Kbit/s(16) L1Prot:G.711 A-Law(3)                    
  Channel Id:No:1 Type:B-chans(3)  Preferred/Implicit                                                                         
  Calling Party Number:495XXXXXXX(l:10) plan:isdn(1)
type:national(2)scr:network, provided(3) pres:allowed(0)                 
  Called Party Number:XXXXXXX(l:7) plan:isdn(1) type:national(2)                                                              
  Sending complete:                                                                                                           
  [ 08 02 11 36 05 04 03 80 90 a3 18 03 a1 83 81 6c 0c 21 83 34 39 35 37 38
39 33 35 39 31 70 08 a1                           
    32 38 37 38 37 38 36 a1 00 ]                                                                                              
                                                                                                                              
2011-01-26 17:59:44.501244 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:57
[s1c1][1:1] Processing SETUP (suId:1 suInstId:0 spInstId:35)                                                                                                                          
2011-01-26 17:59:44.501244 [WARNING] ftdm_io.c:1789 [s1c1][1:1] Cannot open
channel when is alarmed                           
2011-01-26 17:59:44.501244 [INFO] ftmod_sangoma_isdn_stack_hndl.c:64
[s1c1][1:1] Received SETUP but channel is in USE, saving call for later
processing                                                                                                     
2011-01-26 17:59:48.501214 [INFO] ftmod_sangoma_isdn_trace.c:223 [SNGISDN
Q931] wp1 FRAME INCOMING:  Prot Disc:Q.931/I.451 (0x08)                                                                                                                           
  Call Ref:1136 (Originating side)                                                                                            
  Type:SETUP (0x5)                                                                                                            
  Bearer Capability:Coding:ITU-T(0) TransferCap:Speech(0) TransferRate:64
Kbit/s(16) L1Prot:G.711 A-Law(3)                    
  Channel Id:No:1 Type:B-chans(3)  Preferred/Implicit                                                                         
  Calling Party Number:495XXXXXXX(l:10) plan:isdn(1)
type:national(2)scr:network, provided(3) pres:allowed(0)                 
  Called Party Number:XXXXXXX(l:7) plan:isdn(1) type:national(2)                                                              
  Sending complete:                                                                                                           
  [ 08 02 11 36 05 04 03 80 90 a3 18 03 a1 83 81 6c 0c 21 83 34 39 35 37 38
39 33 35 39 31 70 08 a1                           
    32 38 37 38 37 38 36 a1 00 ]                                                                                              
                                                                                                                              
2011-01-26 17:59:48.501214 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:760
[SNGISDN Q931] s1: Protocol: Unknown Event Code(2): Incomp Msg(276)                                                                                                                 
2011-01-26 17:59:52.547185 [INFO] ftmod_sangoma_isdn_stack_rcv.c:245
[s1c1][1:1] Received RELEASE/RELEASE COMPLETE (suId:1 suInstId:0
spInstId:35)                                                                                                          
2011-01-26 17:59:52.547185 [INFO] ftmod_sangoma_isdn_trace.c:223 [SNGISDN
Q931] wp1 FRAME INCOMING:  Prot Disc:Q.931/I.451 (0x08)                                                                                                                           
  Call Ref:1136 (Originating side)                                                                                            
  Type:RELEASE COMPLETE (0x5a)                                                                                                
  Cause:coding:ITU-T(0) location:User(0) val:Recovery on timer expired(102)                                                   
  Timer T                                                                                                                     
  [ 08 02 11 36 5a 08 02 80 e6 00 ]                                                                                           
                                                                                                                              
2011-01-26 17:59:52.547185 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:578
[s1c1][1:1] Processing RELEASE/RELEASE COMPLETE (suId:1 suInstId:0
spInstId:35)                                                                                                      
2011-01-26 17:59:52.547185 [DEBUG] ftmod_sangoma_isdn_support.c:73
[s1c1][1:1] Clearing glare data (suId:1 suInstId:0 spInstId:35
actv-suInstId:0  actv-spInstId:0)  

I check freetdm.conf and freetdm.conf.xml it seem like correct,
but 
freeswitch at internal> ftdm list
+OK
span: 1 (wp1)
type: Sangoma (ISDN)
physical_status: alarmed
signaling_status: UP
chan_count: 31
dialplan: XML
context: public
dial_regex: 
fail_dial_regex: 
hold_music: 
analog_options none

Maybe anybody know about such problem?
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Sangoma-A104D-tp5962992p5962992.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From singhujjwal at gmail.com  Wed Jan 26 19:04:04 2011
From: singhujjwal at gmail.com (Ujjwal SIngh)
Date: Wed, 26 Jan 2011 21:34:04 +0530
Subject: [Freeswitch-users] SRTP offer from FreeSWITCH
Message-ID: <AANLkTikS=KvRmbaqwuKhaMbaON1qKxsEresGs4LrpUx=@mail.gmail.com>

Hi,

Can we initiate Best Effort SRTP offer from FreeSWITCH in the same way as
Polycom phones do,
i.e sending two "m=" lines for the same media, the first line having a SAVP
profile, while the second one having the normal AVP profile.

Kindly please let me know if this can be done.

Thanks in advance.
Regards,
Ujjwal
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From infos at madovsky.org  Wed Jan 26 19:44:04 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 26 Jan 2011 11:44:04 -0500
Subject: [Freeswitch-users] php esl with outbound socket connection
References: <4D3FFFE4.7030404@xpirio.com>
Message-ID: <EF5621D3C076439CAA873DCB3889547C@e1705>

I'm testing ESL perl and php since two weeks now with
average results.
My challenge is to only create a stream that receives events
so I tried ivrd with ivrd-demo.php. it connects well, the script
receive the CHANNEL_DATA after a "connect\n\n" but if
I send a "event plain all", nothing in return.
maybe I misunderstood the ESL concept, my challenge is
to create a persistent socket that receive events I selected first,
but unitl now no success. (the PHP examples on wiki makes a T_CLASS error so 
I can't try it)



----- Original Message ----- 
From: "Christian L?schenkohl" <christian.loeschenkohl at xpirio.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, January 26, 2011 6:05 AM
Subject: [Freeswitch-users] php esl with outbound socket connection


hello list

has somebody a working example of an php esl outbound socket connection?
the example at http://wiki.freeswitch.org/wiki/PHP_ESL shows only sending 
manual
commands.
i have written my own php class here (not based on the esl module) that is 
in use for
a year now, but i wonder if it's possible with the esl module too (in 
combination with ivrd).
the advantage would be to communicate more directly, like in mod_perl 
(setting and getting
switch variables within the current connection and so on).

the comeback of mod_php would be my dream here, but it doesn't seem to be 
possible (high costs and
no maintainer).

please share your thoughts on this topic

br

-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 5 77 11 - 1000
F  +43 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 



From steveayre at gmail.com  Wed Jan 26 20:19:26 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 26 Jan 2011 17:19:26 +0000
Subject: [Freeswitch-users] make phpmod and test.php
In-Reply-To: <70DF9C14348A4760BCBB0944910D9E22@e1705>
References: <70DF9C14348A4760BCBB0944910D9E22@e1705>
Message-ID: <AANLkTi=RaFr6Rf3q+y-1Y2HU5xD9R3fL1RBtJ27rpJyE@mail.gmail.com>

Yes, it uses the class abstraction which was introduced in PHP5 (
http://php.net/manual/en/language.oop5.abstract.php)

Try upgrading to PHP5.

-Steve


On 26 January 2011 05:26, Madovsky <infos at madovsky.org> wrote:

>  Seems that ESL.php is for PHP5 only ?
>
> ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Wednesday, January 26, 2011 12:18 AM
> *Subject:* make phpmod and test.php
>
> after removed warn as errors and compile ESL.so
> and did on bash command
>
> php test.php
>
> Parse error: syntax error, unexpected T_CLASS in
> /home/src/freeswitch/libs/esl/php/ESL.php on line 30
>
>
> I just want to try to listen ESL events.... ;)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Wed Jan 26 20:19:53 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 26 Jan 2011 17:19:53 +0000
Subject: [Freeswitch-users] php esl with outbound socket connection
In-Reply-To: <EF5621D3C076439CAA873DCB3889547C@e1705>
References: <4D3FFFE4.7030404@xpirio.com>
	<EF5621D3C076439CAA873DCB3889547C@e1705>
Message-ID: <AANLkTinyvbe35=BXvdZ0_KBGzy1NkCrLOOJ=89a4eJBQ@mail.gmail.com>

>
> (the PHP examples on wiki makes a T_CLASS error so I can't try it)
>

That's because it requires PHP5.

-Steve




On 26 January 2011 16:44, Madovsky <infos at madovsky.org> wrote:

> I'm testing ESL perl and php since two weeks now with
> average results.
> My challenge is to only create a stream that receives events
> so I tried ivrd with ivrd-demo.php. it connects well, the script
> receive the CHANNEL_DATA after a "connect\n\n" but if
> I send a "event plain all", nothing in return.
> maybe I misunderstood the ESL concept, my challenge is
> to create a persistent socket that receive events I selected first,
> but unitl now no success. (the PHP examples on wiki makes a T_CLASS error
> so
> I can't try it)
>
>
>
> ----- Original Message -----
> From: "Christian L?schenkohl" <christian.loeschenkohl at xpirio.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, January 26, 2011 6:05 AM
> Subject: [Freeswitch-users] php esl with outbound socket connection
>
>
> hello list
>
> has somebody a working example of an php esl outbound socket connection?
> the example at http://wiki.freeswitch.org/wiki/PHP_ESL shows only sending
> manual
> commands.
> i have written my own php class here (not based on the esl module) that is
> in use for
> a year now, but i wonder if it's possible with the esl module too (in
> combination with ivrd).
> the advantage would be to communicate more directly, like in mod_perl
> (setting and getting
> switch variables within the current connection and so on).
>
> the comeback of mod_php would be my dream here, but it doesn't seem to be
> possible (high costs and
> no maintainer).
>
> please share your thoughts on this topic
>
> br
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 5 77 11 - 1000
> F  +43 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Wed Jan 26 20:31:07 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 26 Jan 2011 12:31:07 -0500
Subject: [Freeswitch-users] make phpmod and test.php
References: <70DF9C14348A4760BCBB0944910D9E22@e1705>
	<AANLkTi=RaFr6Rf3q+y-1Y2HU5xD9R3fL1RBtJ27rpJyE@mail.gmail.com>
Message-ID: <42B6B4DDC3AD4CE280C8B7DD04EAE657@e1705>

> Try upgrading to PHP5.
LOL! do you think I'm in PHP4 yet for pleasure ? ;)
I manage 100 app I developed in PHP4 so it will take maybe another 1 year
to upgrade and test all. No sorry, I won't upgrade it
  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, January 26, 2011 12:19 PM
  Subject: Re: [Freeswitch-users] make phpmod and test.php


  Yes, it uses the class abstraction which was introduced in PHP5 (http://php.net/manual/en/language.oop5.abstract.php)

  Try upgrading to PHP5.

  -Steve



  On 26 January 2011 05:26, Madovsky <infos at madovsky.org> wrote:

    Seems that ESL.php is for PHP5 only ?
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Wednesday, January 26, 2011 12:18 AM
      Subject: make phpmod and test.php


      after removed warn as errors and compile ESL.so
      and did on bash command

      php test.php

      Parse error: syntax error, unexpected T_CLASS in /home/src/freeswitch/libs/esl/php/ESL.php on line 30


      I just want to try to listen ESL events.... ;)

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org






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From marcdecorny at gmail.com  Wed Jan 26 20:38:43 2011
From: marcdecorny at gmail.com (Marc de Corny)
Date: Wed, 26 Jan 2011 17:38:43 +0000
Subject: [Freeswitch-users] Caller ID using Fifo
In-Reply-To: <201101251040421253879@asiainfo-linkage.com>
References: <AANLkTik6pdgXgND6kmzZA4qe8vDZQ0_7e47+ciVWPfub@mail.gmail.com>
	<AANLkTikKDO3qru-9=nuPEQcW+iCN7ccizyJAh9sqUkCN@mail.gmail.com>
	<AANLkTi=p7_CsUDWS62+weZv50+PBZ8RLop0Q04y-c4hB@mail.gmail.com>
	<AANLkTi=m_hW9vQV5nbC2jwcos0bKy2rCf7YAr6aA23oz@mail.gmail.com>
	<AANLkTinX7fZOO9edgHs739RxhcBJwuh70E9f07mM0mzL@mail.gmail.com>
	<201101251040421253879@asiainfo-linkage.com>
Message-ID: <AANLkTinOzCVm4rFqm8MGp-_mgc6Ub3t4BZ+c9NLeG=oQ@mail.gmail.com>

I can see that a number of us are interested in this.
I have tried to set that outbound_name and fifo_outbound_name before send
ing the call the queue but they made no difference.

and when I do  fifo list, I cannot see any variables with that name
available to set. Am I not looking in the right place.


I'm looking into <action application="set" data="fifo_orbit_dialplan=XML"/>
as a potential way out, whereby as the call enters the FIFO I could record
the outbound_name requested and then if I can control the call on the way
out I can set it again. How does this command above allow me to send the
calls out a certain way and treat them. what are the options for that
fifo_orbit_dialplan

thanks
Marc
On Tue, Jan 25, 2011 at 2:40 AM, liuyp2 <liuyp2 at asiainfo-linkage.com> wrote:

>  mod_fifo can't transfer sip header  message which defined by
> myself(sip_h_X-xxx) to b-leg also.
>
> Is there any solution in latest version?
>
>  ------------------------------
>  liuyp2
> 2011-01-25
>  ------------------------------
>
> *????* Anthony Minessale
> *?????* 2011-01-25 09:28:25
> *????* FreeSWITCH Users Help
> *???*
> *???* Re: [Freeswitch-users] Caller ID using Fifo
>
>
> You should all confer to make sure you are all using fs latest git because
> that is the version I am talking about.  Fifo has some major new features in
> latest that do not exist in older versions including showing the customers
> cid when it calls agents.  The dilemma jm describes used to be true but is
> no longer the case with the default ringall strategy on latest git.
>
> The customers cid is sent to the agent and if the fifo xml defines
> outbound_name param that will be included as well.
>
> If you want to override it you must do what you quoted in the wiki in the
> dialstring contained in the member tag of the xml for that membership not in
> the dialplan.
>
> On Jan 14, 2011 10:36 AM, "Marc de Corny" <marcdecorny at gmail.com> wrote:
> >
> > Just to follow up on this subject.
> >
> > I have done a lot of testing on the fifo trying to get the caller_id_name
> changed on the outbound call to the agent and to be honest I cannot
> understand the explanation.
> >
> > If mod_fifo does not know which call it will connect until the agent
> answers, how come it displays the CLI correctly, jsut won;t let me change
> it.
> >
> > Still seems strange. I am looking into the Mod_callcentre to check if it
> sends caller_id information. but the same logic if valid could apply
> >
> > Also maybe someone should change the Wiki ( I would but do not have
> enough expertise on the subject) because the following is a bit misleading
> >
> >  "Note: If you wish to specify the caller ID presented when a fifo calls
> an agent, set the origination_caller_id_name and origination_caller_id_num
> variables to the values desired. These could be set within the {} of the
> dialstring, or they could be set using the set application in the dialplan
> which places the caller into the fifo (before the 'fifo in' executed on the
> caller). "
> > thanks
> > Marc
> > On Thu, Jan 13, 2011 at 10:47 PM, Joao Leme <joaocarlosleme at gmail.com>
> wrote:
> >>
> >> What about showing the Caller ID after it is answered? Any way to do
> that?
> >>
> >> 2011/1/12 Jo?o Mesquita <jmesquita at freeswitch.org>
> >>
> >>> Jo?o Leme,
> >>>
> >>> The caller id is not passed when the phone is ringing because mod_fifo
> does not know which call is going to be sent to that channel once it is
> answered until it is really answered. I don't know if mod_callcenter does
> show anything but you should consider looking at the documentation if you
> really need this feature.
> >>>
> >>> Regards,
> >>> Jo?o Mesquita
> >>>
> >>>
> >>> On Wed, Jan 12, 2011 at 9:15 PM, Joao Leme <joaocarlosleme at gmail.com>
> wrote:
> >>>>
> >>>> Hi there,
> >>>> I would like to know if there is a way to see the caller ID on my Sip
> Client (X-Lite for example) of the caller that I answear from a Fifo queue?
> >>>> Thanks,
> >>>> John
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Wed Jan 26 20:42:23 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 26 Jan 2011 18:42:23 +0100
Subject: [Freeswitch-users] make phpmod and test.php
In-Reply-To: <42B6B4DDC3AD4CE280C8B7DD04EAE657@e1705>
References: <70DF9C14348A4760BCBB0944910D9E22@e1705>
	<AANLkTi=RaFr6Rf3q+y-1Y2HU5xD9R3fL1RBtJ27rpJyE@mail.gmail.com>
	<42B6B4DDC3AD4CE280C8B7DD04EAE657@e1705>
Message-ID: <58CD61C9-3987-4D0C-BAC0-D478714B1BA4@ipeva.fr>

Well I think actually, we are sorry for you.
There is a way to have 2 versions of PHP running at the same time, AFAIR.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 26/01/2011 ? 18:31, Madovsky a ?crit :

> > Try upgrading to PHP5.
> LOL! do you think I'm in PHP4 yet for pleasure ? ;)
> I manage 100 app I developed in PHP4 so it will take maybe another 1 year
> to upgrade and test all. No sorry, I won't upgrade it
> ----- Original Message -----
> From: Steven Ayre
> To: FreeSWITCH Users Help
> Sent: Wednesday, January 26, 2011 12:19 PM
> Subject: Re: [Freeswitch-users] make phpmod and test.php
> 
> Yes, it uses the class abstraction which was introduced in PHP5 (http://php.net/manual/en/language.oop5.abstract.php)
> 
> Try upgrading to PHP5.
> 
> -Steve
> 
> 
> On 26 January 2011 05:26, Madovsky <infos at madovsky.org> wrote:
> Seems that ESL.php is for PHP5 only ?
> ----- Original Message -----
> From: Madovsky
> To: freeswitch-users at lists.freeswitch.org
> Sent: Wednesday, January 26, 2011 12:18 AM
> Subject: make phpmod and test.php
> 
> after removed warn as errors and compile ESL.so
> and did on bash command
>  
> php test.php
>  
> Parse error: syntax error, unexpected T_CLASS in /home/src/freeswitch/libs/esl/php/ESL.php on line 30
>  
>  
> I just want to try to listen ESL events.... ;)
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Wed Jan 26 21:03:20 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 26 Jan 2011 13:03:20 -0500
Subject: [Freeswitch-users] make phpmod and test.php
References: <70DF9C14348A4760BCBB0944910D9E22@e1705><AANLkTi=RaFr6Rf3q+y-1Y2HU5xD9R3fL1RBtJ27rpJyE@mail.gmail.com><42B6B4DDC3AD4CE280C8B7DD04EAE657@e1705>
	<58CD61C9-3987-4D0C-BAC0-D478714B1BA4@ipeva.fr>
Message-ID: <4B6BB650E411440B9244B73DFE4D2CB9@e1705>

did you try already to make php4 and php5 ?
I did already years ago, I don't want to restart it,
a big mess
  ----- Original Message ----- 
  From: David Ponzone 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, January 26, 2011 12:42 PM
  Subject: Re: [Freeswitch-users] make phpmod and test.php


  Well I think actually, we are sorry for you.
  There is a way to have 2 versions of PHP running at the same time, AFAIR.


  David Ponzone  Direction Technique
  email: david.ponzone at ipeva.fr
  tel:      01 74 03 18 97
  gsm:   06 66 98 76 34


  Service Client IPeva
  tel:      0811 46 26 26
  www.ipeva.fr  -   www.ipeva-studio.com


  Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.







  Le 26/01/2011 ? 18:31, Madovsky a ?crit :


    > Try upgrading to PHP5.
    LOL! do you think I'm in PHP4 yet for pleasure ? ;)
    I manage 100 app I developed in PHP4 so it will take maybe another 1 year
    to upgrade and test all. No sorry, I won't upgrade it
      ----- Original Message -----
      From: Steven Ayre
      To: FreeSWITCH Users Help
      Sent: Wednesday, January 26, 2011 12:19 PM
      Subject: Re: [Freeswitch-users] make phpmod and test.php


      Yes, it uses the class abstraction which was introduced in PHP5 (http://php.net/manual/en/language.oop5.abstract.php)

      Try upgrading to PHP5.

      -Steve



      On 26 January 2011 05:26, Madovsky <infos at madovsky.org> wrote:

        Seems that ESL.php is for PHP5 only ?
          ----- Original Message -----
          From: Madovsky
          To: freeswitch-users at lists.freeswitch.org
          Sent: Wednesday, January 26, 2011 12:18 AM
          Subject: make phpmod and test.php


          after removed warn as errors and compile ESL.so
          and did on bash command

          php test.php

          Parse error: syntax error, unexpected T_CLASS in /home/src/freeswitch/libs/esl/php/ESL.php on line 30


          I just want to try to listen ESL events.... ;)

        _______________________________________________
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        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From msc at freeswitch.org  Wed Jan 26 21:07:27 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Jan 2011 10:07:27 -0800
Subject: [Freeswitch-users] *FreeSWITCH Conf Call Now! Hear About New Vestec
	Speech Rec Engine!
Message-ID: <AANLkTim2jWDkwXOGvD7Au1f1S+wvobVAtSMb175NDOTv@mail.gmail.com>


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From tculjaga at gmail.com  Wed Jan 26 21:12:48 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 26 Jan 2011 19:12:48 +0100
Subject: [Freeswitch-users] FS HA
In-Reply-To: <AANLkTikjBm=XT4FrY1N5n_RDQ-TvjANuy149+f4Ai8sH@mail.gmail.com>
References: <AANLkTi=_BPuE5KX4D48JDoOW9U3netsr9uDU9Lo5CgYO@mail.gmail.com>
	<FACE40F5-BB80-479A-BFFC-F07F6BA681D2@freeswitch.org>
	<350D44B3-A176-47A9-84F4-72EB62F08299@gmail.com>
	<AANLkTimJLMjPJwthN8AhDq=1_JnDbdmocbgd=-=usyTT@mail.gmail.com>
	<AANLkTi=ksT-ZkdOyXiVQkir7a9z2-58PcVirmv4xs5Kg@mail.gmail.com>
	<AANLkTikeU5FAgE-HU=0a5whf3JiD5VUCkYkCCRt4u=R0@mail.gmail.com>
	<AANLkTim3=HVjX5GZVfcVjtSCm-2kuWatyVS+YV0RpNCB@mail.gmail.com>
	<AANLkTimB-jxEjrSD7pp45DcAmCmZH_q-dycT25c1gn06@mail.gmail.com>
	<AANLkTimwD=847nJ6Lz0gDrUh=nmELW261ch3QBAhDhf-@mail.gmail.com>
	<AANLkTindsD6E7g=fqr0jr6sCtJrJg6M0Hb1P0LGKh6LR@mail.gmail.com>
	<AANLkTikjBm=XT4FrY1N5n_RDQ-TvjANuy149+f4Ai8sH@mail.gmail.com>
Message-ID: <AANLkTin0Dxzz2wM5Vm2AKDAb2VetTiczq3xAfPhDabfo@mail.gmail.com>

On Wed, Jan 26, 2011 at 4:34 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Tihomir,
>
> Are you saying all of those are implemented in mod_xml_cdr?
>
> If so, that's probably not the best place to implement them. They'd be
> better off in their own separate modules.
>
>
well, i have tested it with the original module ... don't wary about that
:=) ... I always use default modules if i have issues to see the behaviour.



> - added radius accounting interface
>>
>
> There's already a module for this I believe?
> http://wiki.freeswitch.org/wiki/Mod_radius_cdr
>

sorry to say that and pls don't get me wrong, but its not good ... and
doesn't support adding  custom VSAs  (something as in mod_rad_auth).


>
> - added odbc cdr interface
>>
>
> This would be better in its own module.
>
> - not the best place but
>>
>
I don't agree with you here, mod_xml_cdr has already two interfaces (curl
and fs) .. why not add radius and odbc as well ?


Of course i can move all of that in separate modules ... but why.. i look at
that as additional interfaces where i can send the CDR.


LCR stuff is on the roadmap already... and i agree its has no place here.


>   added a function to build up my dial-string... basically the same
>> functionality of LCR module but much more simpler :=)
>>
>
> Indeed it's not, I don't see how CDRs and LCR is related.
> This would be better as something like mod_custom_lcr. mod_lcr may well do
> what you need though with custom SQL.
>
> Warm Regards,
> -Steve
>
>
>
>
>
> On 26 January 2011 14:03, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>
>> patch for mod_xml_cdr
>>
>> - added radius accounting interface
>> - added odbc cdr interface
>>
>> - not the best place but
>>   added a function to build up my dial-string... basically the same
>> functionality of LCR module but much more simpler :=)
>>
>
>
>
>
> On 26 January 2011 13:28, Steven Ayre <steveayre at gmail.com> wrote:
>
>> It shouldn't be hard to update them. Most interfaces haven't changed, and
>> those that have won't have changed much. Any changes will show up as
>> compile-time errors. If you wrote them in the first place then you'll be
>> easily capable of updating them for the latest version.
>>
>> If you have any patches that you think would be beneficial for the entire
>> community you can share them on http://jira.freeswitch.org, then they'll
>> get considered for adding to the trunk - what's the patch for?
>>
>> Patches to trunk modules will always be easier for you to maintain if you
>> can get them into trunk.
>>
>> -Steve
>>
>>
>>
>> On 26 January 2011 10:53, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>>
>>> hello Anthony,
>>>
>>> ya, i was afraid of that... this means moving all my modules & patches to
>>> the latest git...
>>>
>>> radius cdr interface
>>> odbc cdr interface
>>>
>>
>>> within mod_xml_cdr <= i can share this as a patch... if you are
>>> interested.
>>>
>>>
>>> mod_say_hr - really bad programming .. needs big re-factoring but lack of
>>> time :(
>>> patches for mod_say_de & mod_say_fr - because wrong playing in some
>>> scenarios
>>>
>>> and some small stuff i made within mod_commands... anyhow this was on the
>>> road.
>>>
>>
>>>
>>> anyhow i was really into understanding the way the call context
>>> replication works... this may be the DB connection issue as the ODBC
>>> connection is reset on switchover (im switching entire resource .. floating
>>> IP, & database)
>>>
>>>
>>> so what really happens is the FS on 2nd node just getting sessions from
>>> sip_recovery
>>>
>>> from sofia_glue_recover: i see you are selecting sip_recovery table.
>>>
>>>
>>> select * from sip_recovery;
>>>    runtime_uuid profile_name hostname uuid metadata
>>> 24f77a1f-f315-4beb-904f-f779e8767c75 internal cxss01
>>> 102b2c7f-466f-4b6d-a795-e2cc25630e78
>>> 24f77a1f-f315-4beb-904f-f779e8767c75 external cxss01
>>> 52834d99-c1be-4832-bee9-7ba05238871d
>>>
>>>
>>>
>>> so, what do you do afterward?
>>>
>>>
>>> on the recovering node i see some portion of dialplan is executed and
>>> re-INVITEs being sent ... where do you get the DSP info for the recovering
>>> re-INVITE ?
>>>
>>>
>>>
>>>
>>> thanks for your help,
>>> Tihomir.
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Jan 26, 2011 at 1:12 AM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> There is an xml_cdr snapshot in sip_recovery sql table that contains
>>>> the data used for the recovery.
>>>> Also based on your logs, you are not using latest git.  I would try
>>>> that first before anything else.
>>>>
>>>>
>>>>
>>>> On Tue, Jan 25, 2011 at 7:24 AM, Tihomir Culjaga <tculjaga at gmail.com>
>>>> wrote:
>>>> > yap, both using the same config
>>>> >
>>>> >
>>>> > can you advice where is FS getting the SDP info for the re-INVITE ?
>>>> >
>>>> >
>>>> > On Tue, Jan 25, 2011 at 2:03 PM, Avi Marcus <avi at avimarcus.net>
>>>> wrote:
>>>> >>
>>>> >> Are you using two machines for the HA? do both have the same configs?
>>>> >> -Avi
>>>> >>
>>>> >> On Tue, Jan 25, 2011 at 11:41 AM, Tihomir Culjaga <
>>>> tculjaga at gmail.com>
>>>> >> wrote:
>>>> >> > Here is the debug: http://pastebin.freeswitch.org/15133
>>>> >> >
>>>> >> > i have set verbose_sdp=true in vars.xml as.
>>>> >> >   <X-PRE-PROCESS cmd="set" data="verbose_sdp=true"/>
>>>> >> >
>>>> >> > but not much to be seen of the verbose thing in the debug...
>>>> >> >
>>>> >> > Still, FS is sending a re-INVITE with wrong SDP. The call to be
>>>> >> > recovered is
>>>> >> > using ALAW... and ULAW is not supported.
>>>> >> > FS sends a re-INVITE with ULAW and CN in SDP. The re-INVITE is
>>>> rejected
>>>> >> > due
>>>> >> > to incompatible SDP.
>>>> >> >
>>>> >> > Where does FS get the information for the SDP in re-INVITE message?
>>>> >> >
>>>> >> >
>>>> >> > please advice,
>>>> >> > T.
>>>> >> >
>>>> >> >
>>>> >> >
>>>> >> > On Tue, Jan 25, 2011 at 12:01 AM, Tihomir Culjaga <
>>>> tculjaga at gmail.com>
>>>> >> > wrote:
>>>> >> >>
>>>> >> >> yap, i do have PCMA ... and the debug shows it correctly :=)
>>>> >> >>
>>>> >> >> i will try to see what it does with verbose. Post new debug
>>>> tomorrow.
>>>> >> >>
>>>> >> >> ty.
>>>> >> >>
>>>> >> >>
>>>> >> >> On Mon, Jan 24, 2011 at 11:29 PM, Steven Ayre <
>>>> steveayre at gmail.com>
>>>> >> >> wrote:
>>>> >> >>>
>>>> >> >>> Brian, it has PCMU in the sdp, but Sofia thinks it's setting
>>>> PCMA?
>>>> >> >>>
>>>> >> >>> Steve on iPhone
>>>> >> >>>
>>>> >> >>> On 24 Jan 2011, at 20:17, Brian West <brian at freeswitch.org>
>>>> wrote:
>>>> >> >>>
>>>> >> >>> > What makes you think that fails?  It has ULAW and CN in the
>>>> codec
>>>> >> >>> > list!
>>>> >> >>> >  Sounds like you need the verbose sdp... set the global
>>>> variable
>>>> >> >>> > "verbose_sdp=true"
>>>> >> >>> >
>>>> >> >>> > /b
>>>> >> >>> >
>>>> >> >>> > On Jan 24, 2011, at 1:50 PM, Tihomir Culjaga wrote:
>>>> >> >>> >
>>>> >> >>> >>
>>>> >> >>> >>
>>>> >> >>> >> i configured FS HA and looks like its trying to recover the
>>>> call ..
>>>> >> >>> >> but the re-INVITE fails due to "wrong/missed" codec
>>>> capability.
>>>> >> >>> >
>>>> >> >>> >
>>>> >> >>> > _______________________________________________
>>>> >> >>> > FreeSWITCH-users mailing list
>>>> >> >>> > FreeSWITCH-users at lists.freeswitch.org
>>>> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> >>> >
>>>> >> >>> >
>>>> >> >>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> >>> > http://www.freeswitch.org
>>>> >> >>>
>>>> >> >>> _______________________________________________
>>>> >> >>> FreeSWITCH-users mailing list
>>>> >> >>> FreeSWITCH-users at lists.freeswitch.org
>>>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> >>>
>>>> >> >>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> >>> http://www.freeswitch.org
>>>> >> >>
>>>> >> >
>>>> >> >
>>>> >> > _______________________________________________
>>>> >> > FreeSWITCH-users mailing list
>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> > http://www.freeswitch.org
>>>> >> >
>>>> >> >
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Wed Jan 26 21:41:53 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 26 Jan 2011 13:41:53 -0500
Subject: [Freeswitch-users] make phpmod and test.php
Message-ID: <5724A8BEAE3348E58AF9D5A8B6A5D7FA@e1705>

I'm also blocked with PHP4 because some PHP projects
use extension not available anymore on PHP5.
  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, January 26, 2011 1:03 PM
  Subject: Re: [Freeswitch-users] make phpmod and test.php


  did you try already to make php4 and php5 ?
  I did already years ago, I don't want to restart it,
  a big mess
    ----- Original Message ----- 
    From: David Ponzone 
    To: FreeSWITCH Users Help 
    Sent: Wednesday, January 26, 2011 12:42 PM
    Subject: Re: [Freeswitch-users] make phpmod and test.php


    Well I think actually, we are sorry for you. 
    There is a way to have 2 versions of PHP running at the same time, AFAIR.


    David Ponzone  Direction Technique
    email: david.ponzone at ipeva.fr
    tel:      01 74 03 18 97
    gsm:   06 66 98 76 34


    Service Client IPeva
    tel:      0811 46 26 26
    www.ipeva.fr  -   www.ipeva-studio.com


    Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.







    Le 26/01/2011 ? 18:31, Madovsky a ?crit :


      > Try upgrading to PHP5.
      LOL! do you think I'm in PHP4 yet for pleasure ? ;)
      I manage 100 app I developed in PHP4 so it will take maybe another 1 year
      to upgrade and test all. No sorry, I won't upgrade it
        ----- Original Message -----
        From: Steven Ayre
        To: FreeSWITCH Users Help
        Sent: Wednesday, January 26, 2011 12:19 PM
        Subject: Re: [Freeswitch-users] make phpmod and test.php


        Yes, it uses the class abstraction which was introduced in PHP5 (http://php.net/manual/en/language.oop5.abstract.php)

        Try upgrading to PHP5.

        -Steve



        On 26 January 2011 05:26, Madovsky <infos at madovsky.org> wrote:

          Seems that ESL.php is for PHP5 only ?
            ----- Original Message -----
            From: Madovsky
            To: freeswitch-users at lists.freeswitch.org
            Sent: Wednesday, January 26, 2011 12:18 AM
            Subject: make phpmod and test.php


            after removed warn as errors and compile ESL.so
            and did on bash command

            php test.php

            Parse error: syntax error, unexpected T_CLASS in /home/src/freeswitch/libs/esl/php/ESL.php on line 30


            I just want to try to listen ESL events.... ;)

          _______________________________________________
          FreeSWITCH-users mailing list
          FreeSWITCH-users at lists.freeswitch.org
          http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
          http://www.freeswitch.org







------------------------------------------------------------------------



        _______________________________________________
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        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org

      _______________________________________________
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      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org





----------------------------------------------------------------------------


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From gchen00 at insightbb.com  Wed Jan 26 22:18:06 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Wed, 26 Jan 2011 14:18:06 -0500
Subject: [Freeswitch-users] question about acl and context
Message-ID: <fc76a4804bce.4d402d1e@insightbb.com>

Installed freeswitch (FreeSWITCH Version 1.0.7 (hacked-20110126T141401Z)) with default configuration.
I registered two sip phones with user id 1007 and 1001. I then dial 9196 for echo test from sip phone 1001 and it worked. Then I make change to acl.conf.xml:
from??? ?<list name="domains" default="deny">
to??? ?<list name="domains" default="allow">

After this change, dialing 9196 stopped working. By looking at console, I can see that before the change
the call first hit default context. and then after the change the call first hit public context.

Does anybody know why?
I'd like to setup freeswitch to allow both inbound and outbound calls using port 5060. That is why I made above change on acl.conf.xml.

Gary?


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From steveayre at gmail.com  Wed Jan 26 22:20:14 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 26 Jan 2011 19:20:14 +0000
Subject: [Freeswitch-users] FS HA
In-Reply-To: <AANLkTin0Dxzz2wM5Vm2AKDAb2VetTiczq3xAfPhDabfo@mail.gmail.com>
References: <AANLkTi=_BPuE5KX4D48JDoOW9U3netsr9uDU9Lo5CgYO@mail.gmail.com>
	<FACE40F5-BB80-479A-BFFC-F07F6BA681D2@freeswitch.org>
	<350D44B3-A176-47A9-84F4-72EB62F08299@gmail.com>
	<AANLkTimJLMjPJwthN8AhDq=1_JnDbdmocbgd=-=usyTT@mail.gmail.com>
	<AANLkTi=ksT-ZkdOyXiVQkir7a9z2-58PcVirmv4xs5Kg@mail.gmail.com>
	<AANLkTikeU5FAgE-HU=0a5whf3JiD5VUCkYkCCRt4u=R0@mail.gmail.com>
	<AANLkTim3=HVjX5GZVfcVjtSCm-2kuWatyVS+YV0RpNCB@mail.gmail.com>
	<AANLkTimB-jxEjrSD7pp45DcAmCmZH_q-dycT25c1gn06@mail.gmail.com>
	<AANLkTimwD=847nJ6Lz0gDrUh=nmELW261ch3QBAhDhf-@mail.gmail.com>
	<AANLkTindsD6E7g=fqr0jr6sCtJrJg6M0Hb1P0LGKh6LR@mail.gmail.com>
	<AANLkTikjBm=XT4FrY1N5n_RDQ-TvjANuy149+f4Ai8sH@mail.gmail.com>
	<AANLkTin0Dxzz2wM5Vm2AKDAb2VetTiczq3xAfPhDabfo@mail.gmail.com>
Message-ID: <AANLkTikoNS+seph5zuKWVom_5d=Xt1m7m0zw5CiiBaDH@mail.gmail.com>

>
> Tihomir,
>>
>> Are you saying all of those are implemented in mod_xml_cdr?
>>
>> If so, that's probably not the best place to implement them. They'd be
>> better off in their own separate modules.
>>
>>
> well, i have tested it with the original module ... don't wary about that
> :=) ... I always use default modules if i have issues to see the behaviour.
>

There's no reason it should work in a default module if it wouldn't work in
its own.


> - added radius accounting interface
>>>
>>
>> There's already a module for this I believe?
>> http://wiki.freeswitch.org/wiki/Mod_radius_cdr
>>
>
> sorry to say that and pls don't get me wrong, but its not good ... and
> doesn't support adding  custom VSAs  (something as in mod_rad_auth).


Fair enough. If you think it can be improved post jira tickets requesting
new features and possibly supplying patches to improve it.



> - not the best place but
>>
>
> I don't agree with you here, mod_xml_cdr has already two interfaces (curl
> and fs) .. why not add radius and odbc as well ?
>

curl and fs are writing the same generated XML cdr content. The fs support
is there partly to give the ability to store CDRs when the HTTP request
fails so they can be resent later.
Radius and ODBC on the otherhand aren't XML, so it's an entirely different
type of CDR. (Well, ODBC could be but it's probably not the best way to
store it).

-Steve


On 26 January 2011 18:12, Tihomir Culjaga <tculjaga at gmail.com> wrote:

>
>
> On Wed, Jan 26, 2011 at 4:34 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Tihomir,
>>
>> Are you saying all of those are implemented in mod_xml_cdr?
>>
>> If so, that's probably not the best place to implement them. They'd be
>> better off in their own separate modules.
>>
>>
> well, i have tested it with the original module ... don't wary about that
> :=) ... I always use default modules if i have issues to see the behaviour.
>
>
>
>>  - added radius accounting interface
>>>
>>
>> There's already a module for this I believe?
>> http://wiki.freeswitch.org/wiki/Mod_radius_cdr
>>
>
> sorry to say that and pls don't get me wrong, but its not good ... and
> doesn't support adding  custom VSAs  (something as in mod_rad_auth).
>
>
>>
>> - added odbc cdr interface
>>>
>>
>> This would be better in its own module.
>>
>> - not the best place but
>>>
>>
> I don't agree with you here, mod_xml_cdr has already two interfaces (curl
> and fs) .. why not add radius and odbc as well ?
>
>
> Of course i can move all of that in separate modules ... but why.. i look
> at that as additional interfaces where i can send the CDR.
>
>
> LCR stuff is on the roadmap already... and i agree its has no place here.
>
>
>>   added a function to build up my dial-string... basically the same
>>> functionality of LCR module but much more simpler :=)
>>>
>>
>> Indeed it's not, I don't see how CDRs and LCR is related.
>> This would be better as something like mod_custom_lcr. mod_lcr may well do
>> what you need though with custom SQL.
>>
>> Warm Regards,
>> -Steve
>>
>>
>>
>>
>>
>> On 26 January 2011 14:03, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>>
>>> patch for mod_xml_cdr
>>>
>>> - added radius accounting interface
>>> - added odbc cdr interface
>>>
>>> - not the best place but
>>>   added a function to build up my dial-string... basically the same
>>> functionality of LCR module but much more simpler :=)
>>>
>>
>>
>>
>>
>> On 26 January 2011 13:28, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> It shouldn't be hard to update them. Most interfaces haven't changed, and
>>> those that have won't have changed much. Any changes will show up as
>>> compile-time errors. If you wrote them in the first place then you'll be
>>> easily capable of updating them for the latest version.
>>>
>>> If you have any patches that you think would be beneficial for the entire
>>> community you can share them on http://jira.freeswitch.org, then they'll
>>> get considered for adding to the trunk - what's the patch for?
>>>
>>> Patches to trunk modules will always be easier for you to maintain if you
>>> can get them into trunk.
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 26 January 2011 10:53, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>>>
>>>> hello Anthony,
>>>>
>>>> ya, i was afraid of that... this means moving all my modules & patches
>>>> to the latest git...
>>>>
>>>> radius cdr interface
>>>> odbc cdr interface
>>>>
>>>
>>>> within mod_xml_cdr <= i can share this as a patch... if you are
>>>> interested.
>>>>
>>>>
>>>> mod_say_hr - really bad programming .. needs big re-factoring but lack
>>>> of time :(
>>>> patches for mod_say_de & mod_say_fr - because wrong playing in some
>>>> scenarios
>>>>
>>>> and some small stuff i made within mod_commands... anyhow this was on
>>>> the road.
>>>>
>>>
>>>>
>>>> anyhow i was really into understanding the way the call context
>>>> replication works... this may be the DB connection issue as the ODBC
>>>> connection is reset on switchover (im switching entire resource .. floating
>>>> IP, & database)
>>>>
>>>>
>>>> so what really happens is the FS on 2nd node just getting sessions from
>>>> sip_recovery
>>>>
>>>> from sofia_glue_recover: i see you are selecting sip_recovery table.
>>>>
>>>>
>>>> select * from sip_recovery;
>>>>    runtime_uuid profile_name hostname uuid metadata
>>>> 24f77a1f-f315-4beb-904f-f779e8767c75 internal cxss01
>>>> 102b2c7f-466f-4b6d-a795-e2cc25630e78
>>>> 24f77a1f-f315-4beb-904f-f779e8767c75 external cxss01
>>>> 52834d99-c1be-4832-bee9-7ba05238871d
>>>>
>>>>
>>>>
>>>> so, what do you do afterward?
>>>>
>>>>
>>>> on the recovering node i see some portion of dialplan is executed and
>>>> re-INVITEs being sent ... where do you get the DSP info for the recovering
>>>> re-INVITE ?
>>>>
>>>>
>>>>
>>>>
>>>> thanks for your help,
>>>> Tihomir.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Wed, Jan 26, 2011 at 1:12 AM, Anthony Minessale <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>>> There is an xml_cdr snapshot in sip_recovery sql table that contains
>>>>> the data used for the recovery.
>>>>> Also based on your logs, you are not using latest git.  I would try
>>>>> that first before anything else.
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Jan 25, 2011 at 7:24 AM, Tihomir Culjaga <tculjaga at gmail.com>
>>>>> wrote:
>>>>> > yap, both using the same config
>>>>> >
>>>>> >
>>>>> > can you advice where is FS getting the SDP info for the re-INVITE ?
>>>>> >
>>>>> >
>>>>> > On Tue, Jan 25, 2011 at 2:03 PM, Avi Marcus <avi at avimarcus.net>
>>>>> wrote:
>>>>> >>
>>>>> >> Are you using two machines for the HA? do both have the same
>>>>> configs?
>>>>> >> -Avi
>>>>> >>
>>>>> >> On Tue, Jan 25, 2011 at 11:41 AM, Tihomir Culjaga <
>>>>> tculjaga at gmail.com>
>>>>> >> wrote:
>>>>> >> > Here is the debug: http://pastebin.freeswitch.org/15133
>>>>> >> >
>>>>> >> > i have set verbose_sdp=true in vars.xml as.
>>>>> >> >   <X-PRE-PROCESS cmd="set" data="verbose_sdp=true"/>
>>>>> >> >
>>>>> >> > but not much to be seen of the verbose thing in the debug...
>>>>> >> >
>>>>> >> > Still, FS is sending a re-INVITE with wrong SDP. The call to be
>>>>> >> > recovered is
>>>>> >> > using ALAW... and ULAW is not supported.
>>>>> >> > FS sends a re-INVITE with ULAW and CN in SDP. The re-INVITE is
>>>>> rejected
>>>>> >> > due
>>>>> >> > to incompatible SDP.
>>>>> >> >
>>>>> >> > Where does FS get the information for the SDP in re-INVITE
>>>>> message?
>>>>> >> >
>>>>> >> >
>>>>> >> > please advice,
>>>>> >> > T.
>>>>> >> >
>>>>> >> >
>>>>> >> >
>>>>> >> > On Tue, Jan 25, 2011 at 12:01 AM, Tihomir Culjaga <
>>>>> tculjaga at gmail.com>
>>>>> >> > wrote:
>>>>> >> >>
>>>>> >> >> yap, i do have PCMA ... and the debug shows it correctly :=)
>>>>> >> >>
>>>>> >> >> i will try to see what it does with verbose. Post new debug
>>>>> tomorrow.
>>>>> >> >>
>>>>> >> >> ty.
>>>>> >> >>
>>>>> >> >>
>>>>> >> >> On Mon, Jan 24, 2011 at 11:29 PM, Steven Ayre <
>>>>> steveayre at gmail.com>
>>>>> >> >> wrote:
>>>>> >> >>>
>>>>> >> >>> Brian, it has PCMU in the sdp, but Sofia thinks it's setting
>>>>> PCMA?
>>>>> >> >>>
>>>>> >> >>> Steve on iPhone
>>>>> >> >>>
>>>>> >> >>> On 24 Jan 2011, at 20:17, Brian West <brian at freeswitch.org>
>>>>> wrote:
>>>>> >> >>>
>>>>> >> >>> > What makes you think that fails?  It has ULAW and CN in the
>>>>> codec
>>>>> >> >>> > list!
>>>>> >> >>> >  Sounds like you need the verbose sdp... set the global
>>>>> variable
>>>>> >> >>> > "verbose_sdp=true"
>>>>> >> >>> >
>>>>> >> >>> > /b
>>>>> >> >>> >
>>>>> >> >>> > On Jan 24, 2011, at 1:50 PM, Tihomir Culjaga wrote:
>>>>> >> >>> >
>>>>> >> >>> >>
>>>>> >> >>> >>
>>>>> >> >>> >> i configured FS HA and looks like its trying to recover the
>>>>> call ..
>>>>> >> >>> >> but the re-INVITE fails due to "wrong/missed" codec
>>>>> capability.
>>>>> >> >>> >
>>>>> >> >>> >
>>>>> >> >>> > _______________________________________________
>>>>> >> >>> > FreeSWITCH-users mailing list
>>>>> >> >>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> >>> >
>>>>> >> >>> >
>>>>> >> >>> > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> >>> > http://www.freeswitch.org
>>>>> >> >>>
>>>>> >> >>> _______________________________________________
>>>>> >> >>> FreeSWITCH-users mailing list
>>>>> >> >>> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> >>>
>>>>> >> >>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> >>> http://www.freeswitch.org
>>>>> >> >>
>>>>> >> >
>>>>> >> >
>>>>> >> > _______________________________________________
>>>>> >> > FreeSWITCH-users mailing list
>>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> > http://www.freeswitch.org
>>>>> >> >
>>>>> >> >
>>>>> >>
>>>>> >> _______________________________________________
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> http://www.freeswitch.org
>>>>> >
>>>>> >
>>>>> > _______________________________________________
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>> >
>>>>> >
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
> _______________________________________________
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From infos at madovsky.org  Wed Jan 26 22:56:31 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 26 Jan 2011 14:56:31 -0500
Subject: [Freeswitch-users] conference with outbound socket
Message-ID: <1D0C3FB175264DE086391F4CD1A0A6A6@e1705>

I'm trying to create a conference with ESL outbound socket


echo "sendmsg\n";
echo "call-command: execute\n";
echo "execute-app-name: conference\n";
echo "execute-app-arg: myconf at default\n\n"


but I get

Content-Type: command/reply
Reply-Text: -ERR command not found

Any idea ?

Thanks


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From tculjaga at gmail.com  Wed Jan 26 22:57:13 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 26 Jan 2011 20:57:13 +0100
Subject: [Freeswitch-users] FS HA
In-Reply-To: <AANLkTikoNS+seph5zuKWVom_5d=Xt1m7m0zw5CiiBaDH@mail.gmail.com>
References: <AANLkTi=_BPuE5KX4D48JDoOW9U3netsr9uDU9Lo5CgYO@mail.gmail.com>
	<FACE40F5-BB80-479A-BFFC-F07F6BA681D2@freeswitch.org>
	<350D44B3-A176-47A9-84F4-72EB62F08299@gmail.com>
	<AANLkTimJLMjPJwthN8AhDq=1_JnDbdmocbgd=-=usyTT@mail.gmail.com>
	<AANLkTi=ksT-ZkdOyXiVQkir7a9z2-58PcVirmv4xs5Kg@mail.gmail.com>
	<AANLkTikeU5FAgE-HU=0a5whf3JiD5VUCkYkCCRt4u=R0@mail.gmail.com>
	<AANLkTim3=HVjX5GZVfcVjtSCm-2kuWatyVS+YV0RpNCB@mail.gmail.com>
	<AANLkTimB-jxEjrSD7pp45DcAmCmZH_q-dycT25c1gn06@mail.gmail.com>
	<AANLkTimwD=847nJ6Lz0gDrUh=nmELW261ch3QBAhDhf-@mail.gmail.com>
	<AANLkTindsD6E7g=fqr0jr6sCtJrJg6M0Hb1P0LGKh6LR@mail.gmail.com>
	<AANLkTikjBm=XT4FrY1N5n_RDQ-TvjANuy149+f4Ai8sH@mail.gmail.com>
	<AANLkTin0Dxzz2wM5Vm2AKDAb2VetTiczq3xAfPhDabfo@mail.gmail.com>
	<AANLkTikoNS+seph5zuKWVom_5d=Xt1m7m0zw5CiiBaDH@mail.gmail.com>
Message-ID: <AANLkTin-sGbVsCT_GNiu3p_gDT_d7sBPLMTs=wGHJ515@mail.gmail.com>

points taken :=)


just updated to latest git, included the mod_xml_cdr modifications and i can
recover the calls now. Thanks for your help.

anyhow, im going to move the functionalities to appropriate modules instead.



On Wed, Jan 26, 2011 at 8:20 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Tihomir,
>>>
>>> Are you saying all of those are implemented in mod_xml_cdr?
>>>
>>> If so, that's probably not the best place to implement them. They'd be
>>> better off in their own separate modules.
>>>
>>>
>> well, i have tested it with the original module ... don't wary about that
>> :=) ... I always use default modules if i have issues to see the behaviour.
>>
>
> There's no reason it should work in a default module if it wouldn't work in
> its own.
>
>
>>  - added radius accounting interface
>>>>
>>>
>>> There's already a module for this I believe?
>>> http://wiki.freeswitch.org/wiki/Mod_radius_cdr
>>>
>>
>> sorry to say that and pls don't get me wrong, but its not good ... and
>> doesn't support adding  custom VSAs  (something as in mod_rad_auth).
>
>
> Fair enough. If you think it can be improved post jira tickets requesting
> new features and possibly supplying patches to improve it.
>
>
>
>> - not the best place but
>>>
>>
>> I don't agree with you here, mod_xml_cdr has already two interfaces (curl
>> and fs) .. why not add radius and odbc as well ?
>>
>
> curl and fs are writing the same generated XML cdr content. The fs support
> is there partly to give the ability to store CDRs when the HTTP request
> fails so they can be resent later.
> Radius and ODBC on the otherhand aren't XML, so it's an entirely different
> type of CDR. (Well, ODBC could be but it's probably not the best way to
> store it).
>
> -Steve
>
>
>
> On 26 January 2011 18:12, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>
>>
>>
>> On Wed, Jan 26, 2011 at 4:34 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> Tihomir,
>>>
>>> Are you saying all of those are implemented in mod_xml_cdr?
>>>
>>> If so, that's probably not the best place to implement them. They'd be
>>> better off in their own separate modules.
>>>
>>>
>> well, i have tested it with the original module ... don't wary about that
>> :=) ... I always use default modules if i have issues to see the behaviour.
>>
>>
>>
>>>  - added radius accounting interface
>>>>
>>>
>>> There's already a module for this I believe?
>>> http://wiki.freeswitch.org/wiki/Mod_radius_cdr
>>>
>>
>> sorry to say that and pls don't get me wrong, but its not good ... and
>> doesn't support adding  custom VSAs  (something as in mod_rad_auth).
>>
>>
>>>
>>> - added odbc cdr interface
>>>>
>>>
>>> This would be better in its own module.
>>>
>>> - not the best place but
>>>>
>>>
>> I don't agree with you here, mod_xml_cdr has already two interfaces (curl
>> and fs) .. why not add radius and odbc as well ?
>>
>>
>> Of course i can move all of that in separate modules ... but why.. i look
>> at that as additional interfaces where i can send the CDR.
>>
>>
>> LCR stuff is on the roadmap already... and i agree its has no place here.
>>
>>
>>>   added a function to build up my dial-string... basically the same
>>>> functionality of LCR module but much more simpler :=)
>>>>
>>>
>>> Indeed it's not, I don't see how CDRs and LCR is related.
>>> This would be better as something like mod_custom_lcr. mod_lcr may well
>>> do what you need though with custom SQL.
>>>
>>> Warm Regards,
>>> -Steve
>>>
>>>
>>>
>>>
>>>
>>> On 26 January 2011 14:03, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>>>
>>>> patch for mod_xml_cdr
>>>>
>>>> - added radius accounting interface
>>>> - added odbc cdr interface
>>>>
>>>> - not the best place but
>>>>   added a function to build up my dial-string... basically the same
>>>> functionality of LCR module but much more simpler :=)
>>>>
>>>
>>>
>>>
>>>
>>> On 26 January 2011 13:28, Steven Ayre <steveayre at gmail.com> wrote:
>>>
>>>> It shouldn't be hard to update them. Most interfaces haven't changed,
>>>> and those that have won't have changed much. Any changes will show up as
>>>> compile-time errors. If you wrote them in the first place then you'll be
>>>> easily capable of updating them for the latest version.
>>>>
>>>> If you have any patches that you think would be beneficial for the
>>>> entire community you can share them on http://jira.freeswitch.org, then
>>>> they'll get considered for adding to the trunk - what's the patch for?
>>>>
>>>> Patches to trunk modules will always be easier for you to maintain if
>>>> you can get them into trunk.
>>>>
>>>> -Steve
>>>>
>>>>
>>>>
>>>> On 26 January 2011 10:53, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>>>>
>>>>> hello Anthony,
>>>>>
>>>>> ya, i was afraid of that... this means moving all my modules & patches
>>>>> to the latest git...
>>>>>
>>>>> radius cdr interface
>>>>> odbc cdr interface
>>>>>
>>>>
>>>>> within mod_xml_cdr <= i can share this as a patch... if you are
>>>>> interested.
>>>>>
>>>>>
>>>>> mod_say_hr - really bad programming .. needs big re-factoring but lack
>>>>> of time :(
>>>>> patches for mod_say_de & mod_say_fr - because wrong playing in some
>>>>> scenarios
>>>>>
>>>>> and some small stuff i made within mod_commands... anyhow this was on
>>>>> the road.
>>>>>
>>>>
>>>>>
>>>>> anyhow i was really into understanding the way the call context
>>>>> replication works... this may be the DB connection issue as the ODBC
>>>>> connection is reset on switchover (im switching entire resource .. floating
>>>>> IP, & database)
>>>>>
>>>>>
>>>>> so what really happens is the FS on 2nd node just getting sessions from
>>>>> sip_recovery
>>>>>
>>>>> from sofia_glue_recover: i see you are selecting sip_recovery table.
>>>>>
>>>>>
>>>>> select * from sip_recovery;
>>>>>    runtime_uuid profile_name hostname uuid metadata
>>>>> 24f77a1f-f315-4beb-904f-f779e8767c75 internal cxss01
>>>>> 102b2c7f-466f-4b6d-a795-e2cc25630e78
>>>>> 24f77a1f-f315-4beb-904f-f779e8767c75 external cxss01
>>>>> 52834d99-c1be-4832-bee9-7ba05238871d
>>>>>
>>>>>
>>>>>
>>>>> so, what do you do afterward?
>>>>>
>>>>>
>>>>> on the recovering node i see some portion of dialplan is executed and
>>>>> re-INVITEs being sent ... where do you get the DSP info for the recovering
>>>>> re-INVITE ?
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> thanks for your help,
>>>>> Tihomir.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Wed, Jan 26, 2011 at 1:12 AM, Anthony Minessale <
>>>>> anthony.minessale at gmail.com> wrote:
>>>>>
>>>>>> There is an xml_cdr snapshot in sip_recovery sql table that contains
>>>>>> the data used for the recovery.
>>>>>> Also based on your logs, you are not using latest git.  I would try
>>>>>> that first before anything else.
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Jan 25, 2011 at 7:24 AM, Tihomir Culjaga <tculjaga at gmail.com>
>>>>>> wrote:
>>>>>> > yap, both using the same config
>>>>>> >
>>>>>> >
>>>>>> > can you advice where is FS getting the SDP info for the re-INVITE ?
>>>>>> >
>>>>>> >
>>>>>> > On Tue, Jan 25, 2011 at 2:03 PM, Avi Marcus <avi at avimarcus.net>
>>>>>> wrote:
>>>>>> >>
>>>>>> >> Are you using two machines for the HA? do both have the same
>>>>>> configs?
>>>>>> >> -Avi
>>>>>> >>
>>>>>> >> On Tue, Jan 25, 2011 at 11:41 AM, Tihomir Culjaga <
>>>>>> tculjaga at gmail.com>
>>>>>> >> wrote:
>>>>>> >> > Here is the debug: http://pastebin.freeswitch.org/15133
>>>>>> >> >
>>>>>> >> > i have set verbose_sdp=true in vars.xml as.
>>>>>> >> >   <X-PRE-PROCESS cmd="set" data="verbose_sdp=true"/>
>>>>>> >> >
>>>>>> >> > but not much to be seen of the verbose thing in the debug...
>>>>>> >> >
>>>>>> >> > Still, FS is sending a re-INVITE with wrong SDP. The call to be
>>>>>> >> > recovered is
>>>>>> >> > using ALAW... and ULAW is not supported.
>>>>>> >> > FS sends a re-INVITE with ULAW and CN in SDP. The re-INVITE is
>>>>>> rejected
>>>>>> >> > due
>>>>>> >> > to incompatible SDP.
>>>>>> >> >
>>>>>> >> > Where does FS get the information for the SDP in re-INVITE
>>>>>> message?
>>>>>> >> >
>>>>>> >> >
>>>>>> >> > please advice,
>>>>>> >> > T.
>>>>>> >> >
>>>>>> >> >
>>>>>> >> >
>>>>>> >> > On Tue, Jan 25, 2011 at 12:01 AM, Tihomir Culjaga <
>>>>>> tculjaga at gmail.com>
>>>>>> >> > wrote:
>>>>>> >> >>
>>>>>> >> >> yap, i do have PCMA ... and the debug shows it correctly :=)
>>>>>> >> >>
>>>>>> >> >> i will try to see what it does with verbose. Post new debug
>>>>>> tomorrow.
>>>>>> >> >>
>>>>>> >> >> ty.
>>>>>> >> >>
>>>>>> >> >>
>>>>>> >> >> On Mon, Jan 24, 2011 at 11:29 PM, Steven Ayre <
>>>>>> steveayre at gmail.com>
>>>>>> >> >> wrote:
>>>>>> >> >>>
>>>>>> >> >>> Brian, it has PCMU in the sdp, but Sofia thinks it's setting
>>>>>> PCMA?
>>>>>> >> >>>
>>>>>> >> >>> Steve on iPhone
>>>>>> >> >>>
>>>>>> >> >>> On 24 Jan 2011, at 20:17, Brian West <brian at freeswitch.org>
>>>>>> wrote:
>>>>>> >> >>>
>>>>>> >> >>> > What makes you think that fails?  It has ULAW and CN in the
>>>>>> codec
>>>>>> >> >>> > list!
>>>>>> >> >>> >  Sounds like you need the verbose sdp... set the global
>>>>>> variable
>>>>>> >> >>> > "verbose_sdp=true"
>>>>>> >> >>> >
>>>>>> >> >>> > /b
>>>>>> >> >>> >
>>>>>> >> >>> > On Jan 24, 2011, at 1:50 PM, Tihomir Culjaga wrote:
>>>>>> >> >>> >
>>>>>> >> >>> >>
>>>>>> >> >>> >>
>>>>>> >> >>> >> i configured FS HA and looks like its trying to recover the
>>>>>> call ..
>>>>>> >> >>> >> but the re-INVITE fails due to "wrong/missed" codec
>>>>>> capability.
>>>>>> >> >>> >
>>>>>> >> >>> >
>>>>>> >> >>> > _______________________________________________
>>>>>> >> >>> > FreeSWITCH-users mailing list
>>>>>> >> >>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>> >> >>> >
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >> >>> >
>>>>>> >> >>> >
>>>>>> >> >>> > UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> >> >>> > http://www.freeswitch.org
>>>>>> >> >>>
>>>>>> >> >>> _______________________________________________
>>>>>> >> >>> FreeSWITCH-users mailing list
>>>>>> >> >>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >> >>>
>>>>>> >> >>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> >> >>> http://www.freeswitch.org
>>>>>> >> >>
>>>>>> >> >
>>>>>> >> >
>>>>>> >> > _______________________________________________
>>>>>> >> > FreeSWITCH-users mailing list
>>>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >> > UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> >> > http://www.freeswitch.org
>>>>>> >> >
>>>>>> >> >
>>>>>> >>
>>>>>> >> _______________________________________________
>>>>>> >> FreeSWITCH-users mailing list
>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> >> http://www.freeswitch.org
>>>>>> >
>>>>>> >
>>>>>> > _______________________________________________
>>>>>> > FreeSWITCH-users mailing list
>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> > UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> > http://www.freeswitch.org
>>>>>> >
>>>>>> >
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II
>>>>>>
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>
>>>>>> AIM: anthm
>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>
>>>>>> FreeSWITCH Developer Conference
>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>> pstn:+19193869900
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From dyatsin at sangoma.com  Wed Jan 26 23:33:10 2011
From: dyatsin at sangoma.com (David Yat Sin)
Date: Wed, 26 Jan 2011 15:33:10 -0500
Subject: [Freeswitch-users] Sangoma A104D
In-Reply-To: <1296054869082-5962992.post@n2.nabble.com>
References: <1296054869082-5962992.post@n2.nabble.com>
Message-ID: <4D408506.207@sangoma.com>

Hi vik_kom,
We are looking into this and will get back to you.

David,

*David Yat Sin, BEng*
*/Senior Software Engineer/*
Sangoma Technologies
100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada

	



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On 1/26/2011 10:14 AM, vik_kom wrote:
> Hello ALL,
> I installed FWS 1.0.7 + Sangoma A104D
> CentOS 5.5
> wanpipe-3.5.18
> libsng_isdn-7.0.0.x86_64.
> When I try call from outside, I see such messages
>
> freeswitch at internal>  2011-01-26 17:59:44.501244 [INFO]
> ftmod_sangoma_isdn_stack_rcv.c:75 [s1c1][1:1] Received SETUP (suId:1
> suInstId:0 spInstId:35)
> 2011-01-26 17:59:44.501244 [INFO] ftmod_sangoma_isdn_trace.c:223 [SNGISDN
> Q931] wp1 FRAME INCOMING:  Prot Disc:Q.931/I.451 (0x08)
>    Call Ref:1136 (Originating side)
>    Type:SETUP (0x5)
>    Bearer Capability:Coding:ITU-T(0) TransferCap:Speech(0) TransferRate:64
> Kbit/s(16) L1Prot:G.711 A-Law(3)
>    Channel Id:No:1 Type:B-chans(3)  Preferred/Implicit
>    Calling Party Number:495XXXXXXX(l:10) plan:isdn(1)
> type:national(2)scr:network, provided(3) pres:allowed(0)
>    Called Party Number:XXXXXXX(l:7) plan:isdn(1) type:national(2)
>    Sending complete:
>    [ 08 02 11 36 05 04 03 80 90 a3 18 03 a1 83 81 6c 0c 21 83 34 39 35 37 38
> 39 33 35 39 31 70 08 a1
>      32 38 37 38 37 38 36 a1 00 ]
>
> 2011-01-26 17:59:44.501244 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:57
> [s1c1][1:1] Processing SETUP (suId:1 suInstId:0 spInstId:35)
> 2011-01-26 17:59:44.501244 [WARNING] ftdm_io.c:1789 [s1c1][1:1] Cannot open
> channel when is alarmed
> 2011-01-26 17:59:44.501244 [INFO] ftmod_sangoma_isdn_stack_hndl.c:64
> [s1c1][1:1] Received SETUP but channel is in USE, saving call for later
> processing
> 2011-01-26 17:59:48.501214 [INFO] ftmod_sangoma_isdn_trace.c:223 [SNGISDN
> Q931] wp1 FRAME INCOMING:  Prot Disc:Q.931/I.451 (0x08)
>    Call Ref:1136 (Originating side)
>    Type:SETUP (0x5)
>    Bearer Capability:Coding:ITU-T(0) TransferCap:Speech(0) TransferRate:64
> Kbit/s(16) L1Prot:G.711 A-Law(3)
>    Channel Id:No:1 Type:B-chans(3)  Preferred/Implicit
>    Calling Party Number:495XXXXXXX(l:10) plan:isdn(1)
> type:national(2)scr:network, provided(3) pres:allowed(0)
>    Called Party Number:XXXXXXX(l:7) plan:isdn(1) type:national(2)
>    Sending complete:
>    [ 08 02 11 36 05 04 03 80 90 a3 18 03 a1 83 81 6c 0c 21 83 34 39 35 37 38
> 39 33 35 39 31 70 08 a1
>      32 38 37 38 37 38 36 a1 00 ]
>
> 2011-01-26 17:59:48.501214 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:760
> [SNGISDN Q931] s1: Protocol: Unknown Event Code(2): Incomp Msg(276)
> 2011-01-26 17:59:52.547185 [INFO] ftmod_sangoma_isdn_stack_rcv.c:245
> [s1c1][1:1] Received RELEASE/RELEASE COMPLETE (suId:1 suInstId:0
> spInstId:35)
> 2011-01-26 17:59:52.547185 [INFO] ftmod_sangoma_isdn_trace.c:223 [SNGISDN
> Q931] wp1 FRAME INCOMING:  Prot Disc:Q.931/I.451 (0x08)
>    Call Ref:1136 (Originating side)
>    Type:RELEASE COMPLETE (0x5a)
>    Cause:coding:ITU-T(0) location:User(0) val:Recovery on timer expired(102)
>    Timer T
>    [ 08 02 11 36 5a 08 02 80 e6 00 ]
>
> 2011-01-26 17:59:52.547185 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:578
> [s1c1][1:1] Processing RELEASE/RELEASE COMPLETE (suId:1 suInstId:0
> spInstId:35)
> 2011-01-26 17:59:52.547185 [DEBUG] ftmod_sangoma_isdn_support.c:73
> [s1c1][1:1] Clearing glare data (suId:1 suInstId:0 spInstId:35
> actv-suInstId:0  actv-spInstId:0)
>
> I check freetdm.conf and freetdm.conf.xml it seem like correct,
> but
> freeswitch at internal>  ftdm list
> +OK
> span: 1 (wp1)
> type: Sangoma (ISDN)
> physical_status: alarmed
> signaling_status: UP
> chan_count: 31
> dialplan: XML
> context: public
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options none
>
> Maybe anybody know about such problem?
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From steveayre at gmail.com  Wed Jan 26 23:52:54 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 26 Jan 2011 20:52:54 +0000
Subject: [Freeswitch-users] conference with outbound socket
In-Reply-To: <1D0C3FB175264DE086391F4CD1A0A6A6@e1705>
References: <1D0C3FB175264DE086391F4CD1A0A6A6@e1705>
Message-ID: <B62AFB53-7344-425C-9E78-8F6F0C18F141@gmail.com>

Firstly, is mod_conference loaded?

Second, try SendMsg. Wiki seems to show both lowercase and capitalisedin the examples.

It also shows uuid after SendMsg, but that might not apply on an outbound socket.

Steve on iPhone

On 26 Jan 2011, at 19:56, "Madovsky" <infos at madovsky.org> wrote:

> I'm trying to create a conference with ESL outbound socket
>  
>  
> echo "sendmsg\n";
> echo "call-command: execute\n";
> echo "execute-app-name: conference\n";
> echo "execute-app-arg: myconf at default\n\n"
> 
>  
> but I get
>  
> Content-Type: command/reply
> Reply-Text: -ERR command not found
>  
> Any idea ?
>  
> Thanks
> 
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From infos at madovsky.org  Thu Jan 27 00:13:56 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 26 Jan 2011 16:13:56 -0500
Subject: [Freeswitch-users] conference with outbound socket
References: <1D0C3FB175264DE086391F4CD1A0A6A6@e1705>
	<B62AFB53-7344-425C-9E78-8F6F0C18F141@gmail.com>
Message-ID: <0C84532889CD473EAECEE8643DF41141@e1705>

> Firstly, is mod_conference loaded?
yes, I can use it in as app in dialplan

> Second, try SendMsg. Wiki seems to show both lowercase and capitalisedin the examples.
ok I'll check it out

thanks
  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, January 26, 2011 3:52 PM
  Subject: Re: [Freeswitch-users] conference with outbound socket


  Firstly, is mod_conference loaded?


  Second, try SendMsg. Wiki seems to show both lowercase and capitalisedin the examples.


  It also shows uuid after SendMsg, but that might not apply on an outbound socket.

  Steve on iPhone

  On 26 Jan 2011, at 19:56, "Madovsky" <infos at madovsky.org> wrote:


    I'm trying to create a conference with ESL outbound socket


    echo "sendmsg\n";
    echo "call-command: execute\n";
    echo "execute-app-name: conference\n";
    echo "execute-app-arg: myconf at default\n\n"

     
    but I get

    Content-Type: command/reply
    Reply-Text: -ERR command not found

    Any idea ?

    Thanks



    _______________________________________________
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    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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------------------------------------------------------------------------------


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Thu Jan 27 01:13:01 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 26 Jan 2011 17:13:01 -0500
Subject: [Freeswitch-users] conference with outbound socket
References: <1D0C3FB175264DE086391F4CD1A0A6A6@e1705>
	<B62AFB53-7344-425C-9E78-8F6F0C18F141@gmail.com>
Message-ID: <0DF77BD7B7D449D8AE68381C9BB1E3E3@e1705>

ok got it,

the msg was not sent entirely to the socket.

for  those who are interested to use an ESL server in PHP (4 and 5)
that use a low cpu (about 10% for 9000 connections)
let me know I will post on wiki.

I finally killed the Chimere ESL ;)
  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, January 26, 2011 3:52 PM
  Subject: Re: [Freeswitch-users] conference with outbound socket


  Firstly, is mod_conference loaded?


  Second, try SendMsg. Wiki seems to show both lowercase and capitalisedin the examples.


  It also shows uuid after SendMsg, but that might not apply on an outbound socket.

  Steve on iPhone

  On 26 Jan 2011, at 19:56, "Madovsky" <infos at madovsky.org> wrote:


    I'm trying to create a conference with ESL outbound socket


    echo "sendmsg\n";
    echo "call-command: execute\n";
    echo "execute-app-name: conference\n";
    echo "execute-app-arg: myconf at default\n\n"

     
    but I get

    Content-Type: command/reply
    Reply-Text: -ERR command not found

    Any idea ?

    Thanks



    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org



------------------------------------------------------------------------------


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From miguel.mirandag at gmail.com  Thu Jan 27 01:10:50 2011
From: miguel.mirandag at gmail.com (Miguel Miranda)
Date: Wed, 26 Jan 2011 16:10:50 -0600
Subject: [Freeswitch-users] conference with did example
Message-ID: <AANLkTik-wYp99Mbw6==EnT21U5RGk_MLPCt+g-wA+nFs@mail.gmail.com>

Hi, i have searched the arvhices and could not find a simple example on how
to configure a DID for  mod_conference, i mean you dial an DID, an ivr
responds "please enter you conference number and press puond key, etc", and
if the conference room was configured with pin it asks "please enter ?pin
number, etc".
as i undestend, in this example from wiki:

<extension name="test_bridging_conference">
   <condition field="destination_number" expression="^(3000)$">
      <action application="answer"/>
      <action application="playback" data="connecting_your_call.wav"/>
      <action application="set" data="ringback=${us-ring}"/>
      <action application="conference"
data="bridge:$1-${domain_name}@default:user/1000@${domain_name}"/>
   </condition>
</extension>


i need a gateway and if you receive the DID 3000 you join the conference
right?
What i want is a single did for all the conferences.
regards
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From steveayre at gmail.com  Thu Jan 27 02:57:11 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 26 Jan 2011 23:57:11 +0000
Subject: [Freeswitch-users] conference with did example
In-Reply-To: <AANLkTik-wYp99Mbw6==EnT21U5RGk_MLPCt+g-wA+nFs@mail.gmail.com>
References: <AANLkTik-wYp99Mbw6==EnT21U5RGk_MLPCt+g-wA+nFs@mail.gmail.com>
Message-ID: <AANLkTinsDpRV6p5uFp4WGiNeOupcL6TAfteoJ31K8Qf4@mail.gmail.com>

Something like this:

<extension name="conference">
   <condition field="destination_number" expression="^12345$">
      <action application="answer"/>
      <action application="play_and_get_digits" data="4 4 3 7000 #
$${base_dir}/sounds/conf-number.wav /invalid.wav confnumber \d+"/>
      <action application="playback" data="connecting_your_call.wav"/>
      <action application="set" data="ringback=${us-ring}"/>
      <action application="conference" data="${confnumber}@${domain_name}"/>
   </condition>
</extension>




On 26 January 2011 22:10, Miguel Miranda <miguel.mirandag at gmail.com> wrote:

> Hi, i have searched the arvhices and could not find a simple example on how
> to configure a DID for  mod_conference, i mean you dial an DID, an ivr
> responds "please enter you conference number and press puond key, etc", and
> if the conference room was configured with pin it asks "please enter ?pin
> number, etc".
> as i undestend, in this example from wiki:
>
> <extension name="test_bridging_conference">
>    <condition field="destination_number" expression="^(3000)$">
>       <action application="answer"/>
>
>       <action application="playback" data="connecting_your_call.wav"/>
>       <action application="set" data="ringback=${us-ring}"/>
>       <action application="conference" data="bridge:$1-${domain_name}@default:user/1000@${domain_name}"/>
>
>    </condition>
> </extension>
>
>
> i need a gateway and if you receive the DID 3000 you join the conference
> right?
> What i want is a single did for all the conferences.
> regards
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Jan 27 03:39:37 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Jan 2011 16:39:37 -0800
Subject: [Freeswitch-users] network scenario doubts
In-Reply-To: <4D3FD3B0.4080309@gmail.com>
References: <4D3FD3B0.4080309@gmail.com>
Message-ID: <AANLkTinBd00iKLysi1O97hOupQpYdzBchBN5t8oZOJre@mail.gmail.com>

yes, you should be able to handle this scenario. You probably just need to
set your external profile to use port 5060 instead of 5080. Most people only
have a single NIC so their sofia profiles need to be on different ports. (A
sofia profile is a SIP user agent (UA) that can listen/respond to SIP
messages on an IP address and port.)

Look in external.xml for these lines:
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>

If you have a static IP address (which it appears you do) then put that ip
addr in there in place of the variable. Then look at the end of vars.xml and
you'll see where the external profile's port is set to 5080/5081. Change it
to 5060/5061 and then restart FS.

Let us know how it goes.

-MC

On Tue, Jan 25, 2011 at 11:56 PM, Michal Kopacki <mkopacki at gmail.com> wrote:

>       Hello,
>
>   This is my first post to this list, so hello everyone.
>
>   I'm at the very beginning of freeswitch journey and i have a problem
> with fit FS to my network scenario.
>
> OS: fedora 13 (x86_64)
> FS: 1.0.7 (compiled from sources), default config
>
> Desired scenario:
>
> softphone -> mydomain.com (with ip 193.59.72.xx) -> my isp network ->
> 10.25.48.xx on eth1 -> FS -> 192.168.0.1 on eth0 -> softphone
>
> And now, I'm able to connect to FS from my local network, but I'm not
> able to connect from outside (neither domain nor ip). In internal.xml I
> set internal ip of server and i wanted to set external ip in
> external.xml, but there is a question: which one ?  In case of
> 193.59.72.xx external profile didnt' start and with 10.25.48.xx outside
> softphone didn't register.  I checked with netstat and realized that
> port 5060 is bind to internal nic only and 5080 to external nic
> (10.25.48.xx) and I have no idea what next.
>
> Is it even possible to work with such network scenario ? I would be
> grateful for pointing me to right direction or maybe propose different
> approach.
>
> --
> Best regards,
> Michal
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Jan 27 03:55:49 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Jan 2011 16:55:49 -0800
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
Message-ID: <AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>

Could you humor us and get sip traces of the blf working on a reboot w/
asterisk and not working with freeswitch? Put them in
pastebin.freeswitch.org and drop the link in this thread.

Thanks,
MC

On Wed, Jan 26, 2011 at 12:33 AM, Patrick Plattes <
patrick.plattes at niemann-frey.info> wrote:

> Hi List :-),
>
> I'm currently switching from Asterisk to FreeSWITCH. It's really hard
> work for an Asterisk user, but using Asterisk becomes more and more
> painful even for small installations (less than 100 sip users). I know
> FreeSWITCH is not a drop-in replacement for the Asterisk PBX, but I
> don't want to change the behaviour of the PBX for the users.
>
> It's a preconception in Germany that the American people like
> (especially for the X-mas time) kitsch. Everyone here know the
> American houses with a hole bunch of blinking lights. But those
> decorated hoses are nothing against our offices! Our phones have up to
> 136 lights (BLFs). You often have to wear sunglasses at the office ;-)
>
> A typical usage of BLFs is to check if an agent is a member of the
> queue. I've build a simple extension to add and delete a member. I can
> user "presence in" and "presence out" to enable or disable the BLF,
> but there is one big issue. After a reboot of the phone the the user
> is still a member of the queue, but the BLF is off. We are using hints
> at Asterisk to show the user if he is a member and it works even after
> a reboot - with "presence" at FreeSWITCH it works (of cause) not. Does
> anyone have an idea how to implement it?
>
> My current extension is just for testing and so I use mod_fifo. It
> shouldn't be a problem to use mod_callcenter. The phone calls
> "queue-the_name_of_the_queue-the_name_of_the_user at pbxdomain" eg.
> "queue-sales_de-1000 at freeswitch.cust"
>
> So how implement BLF persistence?
>
> Thanks,
>  Patrick
>
>
> Colourized version: http://pastebin.com/ufJ7U930
>
>
>
> <extension name="Queue Management">
>
>  <condition field="destination_number"
> expression="^queue-(\w+)-(\d+)$" break="on-false">
>   <action application="set"
> data="queue_name=${regex(${destination_number}|^queue-(\w+)-(\d+)$|$1)}"
> inline="true"/>
>   <action application="set"
> data="queue_user=${regex(${destination_number}|^queue-(\w+)-(\d+)$|$2)}"
> inline="true"/>
>  </condition>
>
>  <condition field="${fifo(list ${queue_name}@${domain_name})}"
> expression="${queue_user}">
>   <action application="answer"/>
>   <action application="set" data="result=${fifo_member(del
> ${queue_name}@${domain_name}
> {fifo_member_wait=nowait}user/${queue_user} )}"/>
>   <action application="log" data="INFO Delete FIFO agent result:
> ${result}"/>
>   <action application="log" data="INFO User Logout: ${user_name}"/>
>   <action application="sleep" data="600"/>
>   <action application="playback"
> data="ivr/ivr-you_are_now_logged_out.wav"/>
>   <action application="set" data="result=${presence(out
> queue-${queue_name}-${queue_user}@${domain_name} inuse inuse)}"/>
>   <action application="log" data="INFO Presence OUT ${result}"/>
>   <action application="hangup"/>
>
>   <anti-action application="answer"/>
>   <anti-action application="set" data="result=${fifo_member(add
> ${queue_name}@${domain_name}
> {fifo_member_wait=nowait}user/${queue_user} )}"/>
>   <anti-action application="log" data="INFO Add FIFO agent result:
> ${result}"/>
>   <anti-action application="log" data="INFO User Login: ${user_name}"/>
>   <anti-action application="sleep" data="600"/>
>   <anti-action application="playback"
> data="ivr/ivr-you_are_now_logged_in.wav"/>
>   <anti-action application="set" data="result=${presence(in
> queue-${queue_name}-${queue_user}@${domain_name} inuse inuse)}"/>
>   <anti-action application="log" data="INFO Presence IN ${result}"/>
>   <anti-action application="hangup"/>
>  </condition>
> </extension>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Jan 27 04:05:26 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Jan 2011 17:05:26 -0800
Subject: [Freeswitch-users] How to check call state in lua script?
In-Reply-To: <AANLkTimU9_4NUpdh2tsO48PZpno6BX3prDEcm_ZjWESD@mail.gmail.com>
References: <AANLkTimU9_4NUpdh2tsO48PZpno6BX3prDEcm_ZjWESD@mail.gmail.com>
Message-ID: <AANLkTi=0dY+s9Niq5ss9qdrqSNPzXukDS5ry471E230H@mail.gmail.com>

Use the eval API:

api = freeswitch.API()
my_uuid = session:getVariable('uuid')
...

state2 = api:executeString('eval uuid:' .. my_uuid .. '
${Channel-Call-State}')
...


The 'eval' API is useful for getting those "magic" variables that show up
when you do the info app or uuid_dump xxx.

-MC

On Wed, Jan 26, 2011 at 7:31 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> Hi,
> I have difficult to check the state of an originated session in lua script.
>
> I want to know when the originated session is ringing back or progressing
> or answered.
>
> I have tried with session:getState() but returns always CS_SOFT_EXECUTE.
>
> My lua test script is called from dialplan with a:
>
>         <action application="lua" data="test1.lua"/>
>
>
> And the "test1.lua" script is:
>
>     session2 = freeswitch.Session("sofia/external/xxxxxx at a.b.c.d");
>
>     while (session:ready() and session2:ready()) do
>           state2 = session2:getVariable("channel_call_state");
>           freeswitch.consoleLog("warning", "State2 = " .. state2 .. "\n");
>           session:execute("sleep", "500");
>     end
>
>     if (session:ready()) then
>           session:hangup();
>     end
>
>     if (session2:ready()) then
>           session2:hangup();
>     end
>
> This script fails because the session:getVariable("channel_call_state")
> returns nil.
>
> What is the correct way to know the state of a call?
>
> Stephen
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Jan 27 04:07:59 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Jan 2011 17:07:59 -0800
Subject: [Freeswitch-users] question about acl and context
In-Reply-To: <fc76a4804bce.4d402d1e@insightbb.com>
References: <fc76a4804bce.4d402d1e@insightbb.com>
Message-ID: <AANLkTinddK_bafLVGHTBRcYPJkMLTC7-eqweibVHZ6-n@mail.gmail.com>

Do you want to open your switch to the whole wide world? Or do you just want
to allow any calls from certain IP addresses?

-MC

On Wed, Jan 26, 2011 at 11:18 AM, Gary Chen <gchen00 at insightbb.com> wrote:

> Installed freeswitch (FreeSWITCH Version 1.0.7 (hacked-20110126T141401Z))
> with default configuration.
> I registered two sip phones with user id 1007 and 1001. I then dial 9196
> for echo test from sip phone 1001 and it worked. Then I make change to
> acl.conf.xml:
> from     <list name="domains" default="deny">
> to     <list name="domains" default="allow">
>
> After this change, dialing 9196 stopped working. By looking at console, I
> can see that before the change
> the call first hit default context. and then after the change the call
> first hit public context.
>
> Does anybody know why?
> I'd like to setup freeswitch to allow both inbound and outbound calls using
> port 5060. That is why I made above change on acl.conf.xml.
>
> Gary
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Nabble at slickdeals.endjunk.com  Thu Jan 27 04:26:29 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 26 Jan 2011 17:26:29 -0800 (PST)
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
	remember the BLF state)
In-Reply-To: <AANLkTi=K5VMyz0ebcwnJQc5E5cSUMe_T_WKu9kyMetqr@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<1296044574879-5962503.post@n2.nabble.com>
	<AANLkTi=K5VMyz0ebcwnJQc5E5cSUMe_T_WKu9kyMetqr@mail.gmail.com>
Message-ID: <1296091589137-5964863.post@n2.nabble.com>



Patrick Plattes wrote:
> 
> 2011/1/26 mazilo <Nabble at slickdeals.endjunk.com>:
>> I am just curious what is BLF. I could only think of a BLF is a kitsch
>> with
>> Bacon, Lettuce, and Fries that every American will probably like to
>> enjoy.
> 
> Busy Lamp Field (BLF) is a light on an IP phone which tells you
> whether another extension connected to the same PBX is busy or not.
> The snom 37 has 12 leds on the right side
> (http://blog.tmcnet.com/beyond-voip/snom%20370.jpg)
Thanks and I appreciate that. I haven't got a budget to get me an IP Phone,
yet.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Persistence-BLFs-or-Dear-PBX-please-remember-the-BLF-state-tp5962196p5964863.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Thu Jan 27 04:27:12 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Jan 2011 17:27:12 -0800
Subject: [Freeswitch-users] conference with did example
In-Reply-To: <AANLkTinsDpRV6p5uFp4WGiNeOupcL6TAfteoJ31K8Qf4@mail.gmail.com>
References: <AANLkTik-wYp99Mbw6==EnT21U5RGk_MLPCt+g-wA+nFs@mail.gmail.com>
	<AANLkTinsDpRV6p5uFp4WGiNeOupcL6TAfteoJ31K8Qf4@mail.gmail.com>
Message-ID: <AANLkTik7mXLbTx2gGXDwx3EaKJaJTRofCqY+Bnyo6rnY@mail.gmail.com>

I don't believe this example will work because ${confnumber} won't be
populated at the time the dialplan is parsed. However a trivial modification
would make it work:

  <extension name="ask user for conf number">
    <condition field="destination_number" expression="^(9903)$">
      <action application="answer"/>
      <action application="play_and_get_digits"
              data="4 4 3 7000 # conference/conf-enter_conf_number.wav
ivr/ivr-that_was_an_invalid_entry.wav confnumber \d+"/>
      <action application="playback" data="ivr/ivr-one_moment_please.wav"/>
      <action application="set" data="ringback=${us-ring}"/>
      <action application="transfer" data="USER_DIALED_${confnumber} XML
default"/>
    </condition>
  </extension>

  <extension name="put user into conf">
    <condition field="destination_number" expression="^USER_DIALED_(\d+)$">
      <action application="conference" data="$1@${domain}"/>
    </condition>
  </extension>

Note that I tested this with real sound files on my system instead of the
pretend ones that were there. I also used dest num of "9903" - use a value
that works for you.

-MC

On Wed, Jan 26, 2011 at 3:57 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Something like this:
>
> <extension name="conference">
>    <condition field="destination_number" expression="^12345$">
>       <action application="answer"/>
>       <action application="play_and_get_digits" data="4 4 3 7000 #
> $${base_dir}/sounds/conf-number.wav /invalid.wav confnumber \d+"/>
>
>       <action application="playback" data="connecting_your_call.wav"/>
>       <action application="set" data="ringback=${us-ring}"/>
>       <action application="conference" data="${confnumber}@
> ${domain_name}"/>
>    </condition>
> </extension>
>
>
>
>
> On 26 January 2011 22:10, Miguel Miranda <miguel.mirandag at gmail.com>wrote:
>
>> Hi, i have searched the arvhices and could not find a simple example on
>> how to configure a DID for  mod_conference, i mean you dial an DID, an ivr
>> responds "please enter you conference number and press puond key, etc", and
>> if the conference room was configured with pin it asks "please enter ?pin
>> number, etc".
>> as i undestend, in this example from wiki:
>>
>> <extension name="test_bridging_conference">
>>    <condition field="destination_number" expression="^(3000)$">
>>       <action application="answer"/>
>>
>>
>>       <action application="playback" data="connecting_your_call.wav"/>
>>       <action application="set" data="ringback=${us-ring}"/>
>>       <action application="conference" data="bridge:$1-${domain_name}@default:user/1000@${domain_name}"/>
>>
>>
>>    </condition>
>> </extension>
>>
>>
>> i need a gateway and if you receive the DID 3000 you join the conference
>> right?
>> What i want is a single did for all the conferences.
>> regards
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Thu Jan 27 04:29:40 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Jan 2011 19:29:40 -0600
Subject: [Freeswitch-users] Need park a call with out pre-answer the
 call(not reply 183)
In-Reply-To: <235E565A-8835-4991-BF59-70D64CCB4466@visionutveckling.se>
References: <235E565A-8835-4991-BF59-70D64CCB4466@visionutveckling.se>
Message-ID: <AANLkTi=R+Ln4J5CfpCZ0Rp0WntYgKi2jUkq+b_tRxx8X@mail.gmail.com>

its  a newer addition so you must be on an older revision of the code.


On Wed, Jan 26, 2011 at 5:22 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> I think this should work already, have you tried? Park does not require media.
>
> Peter
>
>
> ----- Reply message -----
> Fr?n: "???" <lrobot.qq at gmail.com>
> Datum: ons, jan 26, 2011 17:16
> Rubrik: [Freeswitch-users] Need park a call with out pre-answer the call(not reply 183)
> Till: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
>
> Hi all,
>
> ?I am try use freeswitch. I need run freeswitch in "bypass-media" mode. So when execute park application. I want the application will not reply 183 message to per answer it.
> ?How could I configure or modify code to get this?
>
> ?Thanks
>
> Best Regards
>
> -Qingquan
> !DSPAM:4d3ff40d32767902010757!
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Thu Jan 27 04:50:33 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Jan 2011 19:50:33 -0600
Subject: [Freeswitch-users] Caller ID using Fifo
In-Reply-To: <AANLkTinOzCVm4rFqm8MGp-_mgc6Ub3t4BZ+c9NLeG=oQ@mail.gmail.com>
References: <AANLkTik6pdgXgND6kmzZA4qe8vDZQ0_7e47+ciVWPfub@mail.gmail.com>
	<AANLkTikKDO3qru-9=nuPEQcW+iCN7ccizyJAh9sqUkCN@mail.gmail.com>
	<AANLkTi=p7_CsUDWS62+weZv50+PBZ8RLop0Q04y-c4hB@mail.gmail.com>
	<AANLkTi=m_hW9vQV5nbC2jwcos0bKy2rCf7YAr6aA23oz@mail.gmail.com>
	<AANLkTinX7fZOO9edgHs739RxhcBJwuh70E9f07mM0mzL@mail.gmail.com>
	<201101251040421253879@asiainfo-linkage.com>
	<AANLkTinOzCVm4rFqm8MGp-_mgc6Ub3t4BZ+c9NLeG=oQ@mail.gmail.com>
Message-ID: <AANLkTin0BZUeaqbWNrErfpB7_wCoJaYANhAADM2QisDy@mail.gmail.com>

If they made no difference you are not on latest GIT HEAD


On Wed, Jan 26, 2011 at 11:38 AM, Marc de Corny <marcdecorny at gmail.com> wrote:
> I can see that a number of us are interested in this.
> I have tried to set that outbound_name and fifo_outbound_name before send
> ing the call the queue but they made no difference.
>
> and when I do? fifo list, I cannot see any variables with that name
> available to set. Am I not looking in the right place.
>
> I'm looking into <action application="set" data="fifo_orbit_dialplan=XML"/>
> as a potential way out, whereby as the call enters the FIFO I could record
> the outbound_name requested and then if I can control the call on the way
> out I can set it again. How does this command above allow me to send the
> calls out a certain way and treat them. what are the options for that
> fifo_orbit_dialplan
>
> thanks
> Marc
> On Tue, Jan 25, 2011 at 2:40 AM, liuyp2 <liuyp2 at asiainfo-linkage.com> wrote:
>>
>> mod_fifo can't transfer sip header? message which defined by
>> myself(sip_h_X-xxx) to b-leg also.
>>
>> Is there any solution in latest version?
>>
>> ________________________________
>> liuyp2
>> 2011-01-25
>> ________________________________
>>
>> ???? Anthony Minessale
>> ????? 2011-01-25?09:28:25
>> ???? FreeSWITCH Users Help
>> ???
>> ??? Re: [Freeswitch-users] Caller ID using Fifo
>>
>>
>> You should all confer to make sure you are all using fs latest git because
>> that is the version I am talking about.? Fifo has some major new features in
>> latest that do not exist in older versions including showing the customers
>> cid when it calls agents.? The dilemma jm describes used to be true but is
>> no longer the case with the default ringall strategy on latest git.
>>
>> The customers cid is sent to the agent and if the fifo xml defines
>> outbound_name param that will be included as well.
>>
>> If you want to override it you must do what you quoted in the wiki in the
>> dialstring contained in the member tag of the xml for that membership not in
>> the dialplan.
>>
>> On Jan 14, 2011 10:36 AM, "Marc de Corny" <marcdecorny at gmail.com> wrote:
>> >
>> > Just to follow up on this subject.
>> >
>> > I have done a lot of testing on the fifo trying to get the
>> > caller_id_name changed on the outbound call to the agent and to be honest I
>> > cannot understand the explanation.
>> >
>> > If mod_fifo does not know which call it will connect until the agent
>> > answers, how come it displays the CLI correctly, jsut won;t let me change
>> > it.
>> >
>> > Still seems strange. I am looking into the Mod_callcentre to check if it
>> > sends caller_id information. but the same logic if valid could apply
>> >
>> > Also maybe someone should change the Wiki ( I would but do not have
>> > enough expertise on the subject) because the following?is a bit misleading
>> >
>> > ?"Note: If you wish to specify the caller ID presented when a fifo calls
>> > an agent, set the origination_caller_id_name and origination_caller_id_num
>> > variables to the values desired. These could be set within the {} of the
>> > dialstring, or they could be set using the set application in the dialplan
>> > which places the caller into the fifo (before the 'fifo in' executed on the
>> > caller). "
>> > thanks
>> > Marc
>> > On Thu, Jan 13, 2011 at 10:47 PM, Joao Leme <joaocarlosleme at gmail.com>
>> > wrote:
>> >>
>> >> What about showing the Caller ID after it is?answered? Any way to do
>> >> that?
>> >>
>> >> 2011/1/12 Jo?o Mesquita <jmesquita at freeswitch.org>
>> >>
>> >>> Jo?o Leme,
>> >>>
>> >>> The caller id is not passed when the phone is ringing because mod_fifo
>> >>> does not know which call is going to be sent to that channel once it is
>> >>> answered until it is really answered. I don't know if mod_callcenter does
>> >>> show anything but you should consider looking at the documentation if you
>> >>> really need this feature.
>> >>>
>> >>> Regards,
>> >>> Jo?o Mesquita
>> >>>
>> >>>
>> >>> On Wed, Jan 12, 2011 at 9:15 PM, Joao Leme <joaocarlosleme at gmail.com>
>> >>> wrote:
>> >>>>
>> >>>> Hi there,
>> >>>> I would like to know if there is a way to see the caller ID on my Sip
>> >>>> Client (X-Lite for example) of the caller that I answear from a Fifo queue?
>> >>>> Thanks,
>> >>>> John
>> >>>>
>> >>>> _______________________________________________
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>
>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>>
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From jerry.richards at teotech.com  Wed Jan 26 00:27:17 2011
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 25 Jan 2011 13:27:17 -0800
Subject: [Freeswitch-users] Difference Between "realm" and "challenge-realm"
	in sip_profiles
Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D6960660335F9C@VA3DIAXVS351.RED001.local>

Hello All,

What is the difference between the parameters "realm" and "challenge-realm" in the conf/sip_profiles tree?

Thanks,
Jerry
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From daniel_wells at byu.edu  Thu Jan 27 06:12:53 2011
From: daniel_wells at byu.edu (Daniel Wells)
Date: Wed, 26 Jan 2011 20:12:53 -0700
Subject: [Freeswitch-users] mod_dingaling and the weekly windows build
Message-ID: <8C68232BC9314C40BBCDDAA480F7B01AEA3B493B34@harrow.exch.ad.byu.edu>

I was wanting to try google voice integration with freeswitch and downloaded the weekly build for windows.  It ran great but I was unable to load the mod_dingaling module.  I received an error saying that the dll could not be found.  I did verify that the dll was not in the mod directory.  Where can I get the dll?

Thanks in advance.


-          Daniel
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From arun.chinnachamy at cognizant.com  Thu Jan 27 06:40:24 2011
From: arun.chinnachamy at cognizant.com (Arun Chinnachamy)
Date: Wed, 26 Jan 2011 22:40:24 -0500
Subject: [Freeswitch-users] Issue when Transferring Outgoing call.
In-Reply-To: <C96517E4.3707%arun.chinnachamy@cognizant.com>
Message-ID: <C9665358.374D%arun.chinnachamy@cognizant.com>

I have uploaded the logs related to the scenario mentioned below.

http://pastebin.freeswitch.org/15153

--Arun


On 1/26/11 12:14 AM, "Arun Chinnachamy" <arun.chinnachamy at cognizant.com>
wrote:

> Thanks Brian. Please find more information below.
> 
> I have two clients 1000 and 1001 registered to FS under domain 192.168.1.10.
> When I received a call from Gateway (say extension 102 and IP Address
> 192.168.1.50) via FS, I can receive the call at 1000 and I am successful at
> transferring (Refer) the call from 1000 to 1001. Everything works fine.
> 
> But when I call the gateway 102 from the client 1000, the call is established
> fine. When I tried to transfer the call from 1000 to 1001, the transfer
> failed. I checked the logs and also captured the packets using wire shark.
> 
> FS is receiving the Refer packet. Instead of sending the Invite to
> 1001 at 192.168.1.10, the invite was sent to 1001 at 192.168.1.50 for which the
> Gateway is replying with User not registered. I found in this link (
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-June/058984.html)
> that It is an issue fixed using dialplan but even after using the latest dial
> plan I am getting the same behavior from FS.
> 
> Please suggest.
> 
> P.S: the Signature is being appended by my server which I do not have control
> over. It annoys me but can not help it.
> 
> 
> On 1/25/11 10:31 PM, "Brian West" <brian at freeswitch.org> wrote:
> 
>> Can you elaborate on this logs or something would help?  Also remove that
>> annoying unenforceable signature when emailing a public mailing list please.
>> 
>> /b
>> 
>> On Jan 25, 2011, at 9:15 PM, Arun Chinnachamy wrote:
>> 
>>> I am facing an issue in FS. I am able to transfer an incoming call from
>>> Gateway to local registered user but i am not able to do the same for
>>> outgoing call to gateway. FS is trying to call the local user through
>>> gateway. any suggestions?
>>> 
>>> ____
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org


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From jonyoung111 at gmail.com  Thu Jan 27 09:08:06 2011
From: jonyoung111 at gmail.com (Jon Young)
Date: Wed, 26 Jan 2011 23:08:06 -0700
Subject: [Freeswitch-users] mod_dingaling answering prematurely
Message-ID: <AANLkTi=3nR1ne6Un1XTrLUkfVxh1HwMjJCJbTaZb3XEE@mail.gmail.com>

I just got mod_dingaling working for outbound and inbound calls.
However, on inbound calls I have to answer it quickly on my SIP phone.
 When I do I catch the call screening announcement from google to
press a "1" to answer.  I put the "execute_on_answer=send_dtmf 1" as
recommended in the Wiki which eliminates the need to press one.  It
seems that FS is answering the call and google starts playing the
screening.  If I wait too long to answer I miss the announcement.

I am routing the call to a specific extension for testing.  The
dialplan is using the extension definitions provided in the default
configuration.  Where should I start looking to allow the phones to
ring longer and not prematurely answer the inbound call?

Thanks,

Jon


From marcdecorny at gmail.com  Thu Jan 27 09:55:39 2011
From: marcdecorny at gmail.com (Marc de Corny)
Date: Thu, 27 Jan 2011 06:55:39 +0000
Subject: [Freeswitch-users] Caller ID using Fifo
In-Reply-To: <AANLkTin0BZUeaqbWNrErfpB7_wCoJaYANhAADM2QisDy@mail.gmail.com>
References: <AANLkTik6pdgXgND6kmzZA4qe8vDZQ0_7e47+ciVWPfub@mail.gmail.com>
	<AANLkTikKDO3qru-9=nuPEQcW+iCN7ccizyJAh9sqUkCN@mail.gmail.com>
	<AANLkTi=p7_CsUDWS62+weZv50+PBZ8RLop0Q04y-c4hB@mail.gmail.com>
	<AANLkTi=m_hW9vQV5nbC2jwcos0bKy2rCf7YAr6aA23oz@mail.gmail.com>
	<AANLkTinX7fZOO9edgHs739RxhcBJwuh70E9f07mM0mzL@mail.gmail.com>
	<201101251040421253879@asiainfo-linkage.com>
	<AANLkTinOzCVm4rFqm8MGp-_mgc6Ub3t4BZ+c9NLeG=oQ@mail.gmail.com>
	<AANLkTin0BZUeaqbWNrErfpB7_wCoJaYANhAADM2QisDy@mail.gmail.com>
Message-ID: <AANLkTi=yhRKUpnyApJF4KGqnfECATSTkdURBeevq=iXX@mail.gmail.com>

Hi Anthony,

I updated it the other day.

This is my version :
freeswitch at internal> version
FreeSWITCH Version 1.0.head (git-6faa4c9 2010-12-02 17-11-04 -0600)

Having said that I am using remote SIP endpoints on another SIP platform,
maybe that is why. I will take a closer look at the signalling and see.

thanks
Marc

On Thu, Jan 27, 2011 at 1:50 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> If they made no difference you are not on latest GIT HEAD
>
>
> On Wed, Jan 26, 2011 at 11:38 AM, Marc de Corny <marcdecorny at gmail.com>
> wrote:
> > I can see that a number of us are interested in this.
> > I have tried to set that outbound_name and fifo_outbound_name before send
> > ing the call the queue but they made no difference.
> >
> > and when I do  fifo list, I cannot see any variables with that name
> > available to set. Am I not looking in the right place.
> >
> > I'm looking into <action application="set"
> data="fifo_orbit_dialplan=XML"/>
> > as a potential way out, whereby as the call enters the FIFO I could
> record
> > the outbound_name requested and then if I can control the call on the way
> > out I can set it again. How does this command above allow me to send the
> > calls out a certain way and treat them. what are the options for that
> > fifo_orbit_dialplan
> >
> > thanks
> > Marc
> > On Tue, Jan 25, 2011 at 2:40 AM, liuyp2 <liuyp2 at asiainfo-linkage.com>
> wrote:
> >>
> >> mod_fifo can't transfer sip header  message which defined by
> >> myself(sip_h_X-xxx) to b-leg also.
> >>
> >> Is there any solution in latest version?
> >>
> >> ________________________________
> >> liuyp2
> >> 2011-01-25
> >> ________________________________
> >>
> >> ???? Anthony Minessale
> >> ????? 2011-01-25 09:28:25
> >> ???? FreeSWITCH Users Help
> >> ???
> >> ??? Re: [Freeswitch-users] Caller ID using Fifo
> >>
> >>
> >> You should all confer to make sure you are all using fs latest git
> because
> >> that is the version I am talking about.  Fifo has some major new
> features in
> >> latest that do not exist in older versions including showing the
> customers
> >> cid when it calls agents.  The dilemma jm describes used to be true but
> is
> >> no longer the case with the default ringall strategy on latest git.
> >>
> >> The customers cid is sent to the agent and if the fifo xml defines
> >> outbound_name param that will be included as well.
> >>
> >> If you want to override it you must do what you quoted in the wiki in
> the
> >> dialstring contained in the member tag of the xml for that membership
> not in
> >> the dialplan.
> >>
> >> On Jan 14, 2011 10:36 AM, "Marc de Corny" <marcdecorny at gmail.com>
> wrote:
> >> >
> >> > Just to follow up on this subject.
> >> >
> >> > I have done a lot of testing on the fifo trying to get the
> >> > caller_id_name changed on the outbound call to the agent and to be
> honest I
> >> > cannot understand the explanation.
> >> >
> >> > If mod_fifo does not know which call it will connect until the agent
> >> > answers, how come it displays the CLI correctly, jsut won;t let me
> change
> >> > it.
> >> >
> >> > Still seems strange. I am looking into the Mod_callcentre to check if
> it
> >> > sends caller_id information. but the same logic if valid could apply
> >> >
> >> > Also maybe someone should change the Wiki ( I would but do not have
> >> > enough expertise on the subject) because the following is a bit
> misleading
> >> >
> >> >  "Note: If you wish to specify the caller ID presented when a fifo
> calls
> >> > an agent, set the origination_caller_id_name and
> origination_caller_id_num
> >> > variables to the values desired. These could be set within the {} of
> the
> >> > dialstring, or they could be set using the set application in the
> dialplan
> >> > which places the caller into the fifo (before the 'fifo in' executed
> on the
> >> > caller). "
> >> > thanks
> >> > Marc
> >> > On Thu, Jan 13, 2011 at 10:47 PM, Joao Leme <joaocarlosleme at gmail.com
> >
> >> > wrote:
> >> >>
> >> >> What about showing the Caller ID after it is answered? Any way to do
> >> >> that?
> >> >>
> >> >> 2011/1/12 Jo?o Mesquita <jmesquita at freeswitch.org>
> >> >>
> >> >>> Jo?o Leme,
> >> >>>
> >> >>> The caller id is not passed when the phone is ringing because
> mod_fifo
> >> >>> does not know which call is going to be sent to that channel once it
> is
> >> >>> answered until it is really answered. I don't know if mod_callcenter
> does
> >> >>> show anything but you should consider looking at the documentation
> if you
> >> >>> really need this feature.
> >> >>>
> >> >>> Regards,
> >> >>> Jo?o Mesquita
> >> >>>
> >> >>>
> >> >>> On Wed, Jan 12, 2011 at 9:15 PM, Joao Leme <
> joaocarlosleme at gmail.com>
> >> >>> wrote:
> >> >>>>
> >> >>>> Hi there,
> >> >>>> I would like to know if there is a way to see the caller ID on my
> Sip
> >> >>>> Client (X-Lite for example) of the caller that I answear from a
> Fifo queue?
> >> >>>> Thanks,
> >> >>>> John
> >> >>>>
> >> >>>> _______________________________________________
> >> >>>> FreeSWITCH-users mailing list
> >> >>>> FreeSWITCH-users at lists.freeswitch.org
> >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>>
> >> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>>> http://www.freeswitch.org
> >> >>>>
> >> >>>
> >> >>>
> >> >>> _______________________________________________
> >> >>> FreeSWITCH-users mailing list
> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>
> >> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> http://www.freeswitch.org
> >> >>>
> >> >>
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From u2nsam at gmail.com  Thu Jan 27 10:40:29 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 27 Jan 2011 13:10:29 +0530
Subject: [Freeswitch-users] deflect
In-Reply-To: <AANLkTimfCVsJFCVDv1Az-8ERh8UZRqx1cKG1hY+GfKwd@mail.gmail.com>
References: <AANLkTiksYLOeki2_XDJC5YwMW6hjUxvdpT7-Mvg7vmKZ@mail.gmail.com>
	<AANLkTi=EMn55D1xGBrWuGgVE_dW_5itoJ4WpzByyagTE@mail.gmail.com>
	<AANLkTi=F5Lhsk1tjcG5z5LTUjZb4e-cQdNXQjK3OsVD7@mail.gmail.com>
	<AANLkTinV=xZ_ahs=8DH7n8hADGL4hTOR05hNCWCasA85@mail.gmail.com>
	<AANLkTimfCVsJFCVDv1Az-8ERh8UZRqx1cKG1hY+GfKwd@mail.gmail.com>
Message-ID: <AANLkTi=OnWrFDS_JbijJnpQ+_EmPm6r9=yM_aYVNjLVA@mail.gmail.com>

Hi Michael,

Here is it. http://pastebin.freeswitch.org/15156

Regds
Sam

On Wed, Jan 26, 2011 at 12:23 AM, Michael Collins <msc at freeswitch.org>wrote:

> I strongly recommend that you capture the debug output and drop it into a
> pastebin at pastebin.freeswitch.org. You may also wish to capture the sip
> traffic as well. If you are using fs_cli then you already see the debug
> level console output. To get the sip traffic inline with the debug output
> just do "sofia global siptrace on".
>
> -MC
>
>
> On Tue, Jan 25, 2011 at 8:24 AM, Sam <u2nsam at gmail.com> wrote:
>
>> Hi,
>>
>> Is it possible in this scenario,
>>
>> I have a call (leg a) to an IVR on FS1 , after the ivr the below statement
>> is executed,
>> <action application="bridge" data="{api_on_answer='uuid_transfer  ${uuid}
>> both'}sofia/external/12127773456 at 192.168.2.130 <9999 at 192.168.2.130>"/>
>>
>> As the FS1 sends invite to 192.168.2.130 and the call is connected to the
>> moviephone IVR,
>> but here what happens is the call is getting disconnected from leg a and
>> the movie phone ivr 12127773456.
>>
>>
>>
>> Regds
>>
>> Sam
>>
>>
>>
>> On Mon, Jan 24, 2011 at 3:12 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> You could try uuid_simplify with the api_on_answer hook
>>>
>>> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_simplify
>>> http://wiki.freeswitch.org/wiki/Variable_api_on_answer
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 24 January 2011 09:05, Sam <u2nsam at gmail.com> wrote:
>>>
>>>>  Hi,
>>>>
>>>> Is it possible by having b2bua in between , would the leg A be deflected
>>>> to the another FS server from first server ?
>>>>
>>>> Regds
>>>> Sam
>>>>
>>>>
>>>> On Wed, Jan 12, 2011 at 11:42 AM, Sam <u2nsam at gmail.com> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> When call comes on 1 server and plays an application and after
>>>>> execution of the
>>>>> application the call is bridge to the other server ,but here after
>>>>> bridging the call
>>>>> should refer/deflect to other server, how this can be done ?
>>>>>
>>>>> Here just using the deflect variable is not recommended as there is
>>>>> proxy in between,
>>>>> so once the call is bridge the next step would be deflect the leg
>>>>> totally to another server via proxy.
>>>>>
>>>>> Regards
>>>>> Sam
>>>>>
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kond at nstel.ru  Thu Jan 27 12:15:09 2011
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Thu, 27 Jan 2011 12:15:09 +0300
Subject: [Freeswitch-users] freetdm (sangoma A101): isdn-sip display
	name interworking
In-Reply-To: <4D403109.30902@sangoma.com>
Message-ID: <20110127091509.592AF119FB@mail.nstel.ru>

David, 
thank you for the info.
 
Currently my A101 is configured to be isdn network side. And with outgoing call i can see the caller name on the callee phone.
But i can't see any mention about "sip isdn name interworking" in the fs log.
Is it possible? currently i have <param name="debug" value="5"/> in freetdm.conf.xml. Should i set higher value for debug? What value would be appropriate?
 
And one more question: when i use just ascii symbols in caller name i can see it on the callee phone, but when i use UTF8 (russian) i can't see it.
Is "name interworking" supported with UTF8 names?

Thanks again,
Nikolay.

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Yat Sin
Sent: Wednesday, January 26, 2011 5:35 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] freetdm (sangoma A101): isdn-sip display name interworking


Hi Nikolay,
Yes.

The Display IE is only transmited from Network to User, so  if you are on the CPE side, on an incoming call, the Caller Name from the Display IE will be forwarded to the SIP header. And if you are on the Network side, on an outgoing call, the Caller Name will be transmitted in the Display IE.

David



David Yat Sin, BEng
Senior Software Engineer
Sangoma Technologies
100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada



t.   +1 800 388 2475 x119  
t.   +1 905 474 1990 x119  
f.   +1 905 474 9223

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On 1/26/2011 9:11 AM, Nikolay Kondratyev wrote: 

Hi all,
can anybody please clarify if mod_freetdm and sangoma a101 pri card support isdn <-> sip display name interworking?
I mean translating the name in the isdn display information element into >From header and vice versa?
Thanks in advance, 
Nikolay.


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From wstephen80 at gmail.com  Thu Jan 27 13:11:50 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Thu, 27 Jan 2011 11:11:50 +0100
Subject: [Freeswitch-users] How to check call state in lua script?
In-Reply-To: <AANLkTi=0dY+s9Niq5ss9qdrqSNPzXukDS5ry471E230H@mail.gmail.com>
References: <AANLkTimU9_4NUpdh2tsO48PZpno6BX3prDEcm_ZjWESD@mail.gmail.com>
	<AANLkTi=0dY+s9Niq5ss9qdrqSNPzXukDS5ry471E230H@mail.gmail.com>
Message-ID: <AANLkTimkKUyU-qhwdZtAN=SEU0W2ANZqEkXoAOy6Ee__@mail.gmail.com>

Thank you Michael, it works!

Stephen

On Thu, Jan 27, 2011 at 2:05 AM, Michael Collins <msc at freeswitch.org> wrote:

> Use the eval API:
>
> api = freeswitch.API()
> my_uuid = session:getVariable('uuid')
> ...
>
> state2 = api:executeString('eval uuid:' .. my_uuid .. '
> ${Channel-Call-State}')
> ...
>
>
> The 'eval' API is useful for getting those "magic" variables that show up
> when you do the info app or uuid_dump xxx.
>
> -MC
>
> On Wed, Jan 26, 2011 at 7:31 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> Hi,
>> I have difficult to check the state of an originated session in lua
>> script.
>>
>> I want to know when the originated session is ringing back or progressing
>> or answered.
>>
>> I have tried with session:getState() but returns always CS_SOFT_EXECUTE.
>>
>> My lua test script is called from dialplan with a:
>>
>>         <action application="lua" data="test1.lua"/>
>>
>>
>> And the "test1.lua" script is:
>>
>>     session2 = freeswitch.Session("sofia/external/xxxxxx at a.b.c.d");
>>
>>     while (session:ready() and session2:ready()) do
>>           state2 = session2:getVariable("channel_call_state");
>>           freeswitch.consoleLog("warning", "State2 = " .. state2 .. "\n");
>>           session:execute("sleep", "500");
>>     end
>>
>>     if (session:ready()) then
>>           session:hangup();
>>     end
>>
>>     if (session2:ready()) then
>>           session2:hangup();
>>     end
>>
>> This script fails because the session:getVariable("channel_call_state")
>> returns nil.
>>
>> What is the correct way to know the state of a call?
>>
>> Stephen
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gchen00 at insightbb.com  Thu Jan 27 16:09:40 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Thu, 27 Jan 2011 08:09:40 -0500
Subject: [Freeswitch-users] question about acl and context
Message-ID: <fc7f86531200.4d412844@insightbb.com>

I'd like to only allow the calls from certain IP's. I tried to added IP to the acl configuration file and could not make it work. And there is not much information on how to use acl.conf.xml.

Let's say that I have?<list name="domains" default="deny"> and I only want outside incoming calls from certain ?IP's., how can I do that? Can anybody give me an example?


Gary Chen

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Wednesday, January 26, 2011 8:08 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] question about acl and context

?

Do you want to open your switch to the whole wide world? Or do you just want to allow any calls from certain IP addresses??

?

-MC

On Wed, Jan 26, 2011 at 11:18 AM, Gary Chen <gchen00 at insightbb.com> wrote:

Installed freeswitch (FreeSWITCH Version 1.0.7 (hacked-20110126T141401Z)) with default configuration.

I registered two sip phones with user id 1007 and 1001. I then dial 9196 for echo test from sip phone 1001 and it worked. Then I make change to acl.conf.xml:

from??? ?<list name="domains" default="deny">

to??? ?<list name="domains" default="allow">

?

After this change, dialing 9196 stopped working. By looking at console, I can see that before the change

the call first hit default context. and then after the change the call first hit public context.

?

Does anybody know why?

I'd like to setup freeswitch to allow both inbound and outbound calls using port 5060. That is why I made above change on acl.conf.xml.

?

Gary?

?
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From david.ponzone at ipeva.fr  Thu Jan 27 16:17:47 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 27 Jan 2011 14:17:47 +0100
Subject: [Freeswitch-users] question about acl and context
In-Reply-To: <fc7f86531200.4d412844@insightbb.com>
References: <fc7f86531200.4d412844@insightbb.com>
Message-ID: <14AB4990-8A34-4827-AD00-CCEB5E594A88@ipeva.fr>

For instance, in the <sip_profile>.xml file, configure:
<param name="apply-inbound-acl" value="external"/>

and in the acl.conf.xml:

    <list name="external" default="deny">
      <node type="allow" cidr="8.8.8.8/32"/>
    </list>

and complete it as required

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 27/01/2011 ? 14:09, Gary Chen a ?crit :

> I'd like to only allow the calls from certain IP's. I tried to added IP to the acl configuration file and could not make it work. And there is not much information on how to use acl.conf.xml.
> 
> Let's say that I have <list name="domains" default="deny"> and I only want outside incoming calls from certain  IP's., how can I do that? Can anybody give me an example?
> 
> 
> 
> Gary Chen
> 
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
> Sent: Wednesday, January 26, 2011 8:08 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] question about acl and context
> 
>  
> 
> Do you want to open your switch to the whole wide world? Or do you just want to allow any calls from certain IP addresses? 
> 
>  
> 
> -MC
> 
> On Wed, Jan 26, 2011 at 11:18 AM, Gary Chen <gchen00 at insightbb.com> wrote:
> 
> Installed freeswitch (FreeSWITCH Version 1.0.7 (hacked-20110126T141401Z)) with default configuration.
> 
> I registered two sip phones with user id 1007 and 1001. I then dial 9196 for echo test from sip phone 1001 and it worked. Then I make change to acl.conf.xml:
> 
> from     <list name="domains" default="deny">
> 
> to     <list name="domains" default="allow">
> 
>  
> 
> After this change, dialing 9196 stopped working. By looking at console, I can see that before the change
> 
> the call first hit default context. and then after the change the call first hit public context.
> 
>  
> 
> Does anybody know why?
> 
> I'd like to setup freeswitch to allow both inbound and outbound calls using port 5060. That is why I made above change on acl.conf.xml.
> 
>  
> 
> Gary 
> 
>  
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From gchen00 at insightbb.com  Thu Jan 27 16:39:43 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Thu, 27 Jan 2011 08:39:43 -0500
Subject: [Freeswitch-users] question about acl and context
Message-ID: <fc84c47613ba.4d412f4f@insightbb.com>

Thanks. I got it.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Ponzone
Sent: Thursday, January 27, 2011 8:18 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] question about acl and context

?

For instance, in the <sip_profile>.xml file, configure:

<param name="apply-inbound-acl" value="external"/>

?

and in the acl.conf.xml:

?

?? ?<list name="external" default="deny">

?? ? ?<node type="allow" cidr="8.8.8.8/32"/>

?? ?</list>

?

and complete it as required



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From anthony.minessale at gmail.com  Thu Jan 27 17:35:29 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Jan 2011 08:35:29 -0600
Subject: [Freeswitch-users] Caller ID using Fifo
In-Reply-To: <AANLkTi=yhRKUpnyApJF4KGqnfECATSTkdURBeevq=iXX@mail.gmail.com>
References: <AANLkTik6pdgXgND6kmzZA4qe8vDZQ0_7e47+ciVWPfub@mail.gmail.com>
	<AANLkTikKDO3qru-9=nuPEQcW+iCN7ccizyJAh9sqUkCN@mail.gmail.com>
	<AANLkTi=p7_CsUDWS62+weZv50+PBZ8RLop0Q04y-c4hB@mail.gmail.com>
	<AANLkTi=m_hW9vQV5nbC2jwcos0bKy2rCf7YAr6aA23oz@mail.gmail.com>
	<AANLkTinX7fZOO9edgHs739RxhcBJwuh70E9f07mM0mzL@mail.gmail.com>
	<201101251040421253879@asiainfo-linkage.com>
	<AANLkTinOzCVm4rFqm8MGp-_mgc6Ub3t4BZ+c9NLeG=oQ@mail.gmail.com>
	<AANLkTin0BZUeaqbWNrErfpB7_wCoJaYANhAADM2QisDy@mail.gmail.com>
	<AANLkTi=yhRKUpnyApJF4KGqnfECATSTkdURBeevq=iXX@mail.gmail.com>
Message-ID: <AANLkTinHypD7aCBQwzZxF=DgppbZ_4j8f9x6W6JM=fe+@mail.gmail.com>

not the dialplan,
the fifo config file:

autoload_configs/fifo.conf.xml


configuration name="fifo.conf" description="FIFO Configuration">
  <settings>
    <param name="delete-all-outbound-member-on-startup" value="false"/>
  </settings>
  <fifos>
    <fifo name="test_fifo@$${domain}" importance="0" outbound_name="sales">
      <member timeout="60" simo="1"
lag="20">{origination_caller_id_name=fred,origination_caller_id_number=1234}user/1005@$${domain}</member>
    </fifo>
  </fifos>
</configuration>




On Thu, Jan 27, 2011 at 12:55 AM, Marc de Corny <marcdecorny at gmail.com> wrote:
> Hi Anthony,
>
> I updated it the other day.
>
> This is my version :
> freeswitch at internal> version
> FreeSWITCH Version 1.0.head (git-6faa4c9 2010-12-02 17-11-04 -0600)
>
> Having said that I am using remote SIP endpoints on another SIP platform,
> maybe that is why. I will take a closer look at the signalling and see.
>
> thanks
> Marc
>
> On Thu, Jan 27, 2011 at 1:50 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> If they made no difference you are not on latest GIT HEAD
>>
>>
>> On Wed, Jan 26, 2011 at 11:38 AM, Marc de Corny <marcdecorny at gmail.com>
>> wrote:
>> > I can see that a number of us are interested in this.
>> > I have tried to set that outbound_name and fifo_outbound_name before
>> > send
>> > ing the call the queue but they made no difference.
>> >
>> > and when I do? fifo list, I cannot see any variables with that name
>> > available to set. Am I not looking in the right place.
>> >
>> > I'm looking into <action application="set"
>> > data="fifo_orbit_dialplan=XML"/>
>> > as a potential way out, whereby as the call enters the FIFO I could
>> > record
>> > the outbound_name requested and then if I can control the call on the
>> > way
>> > out I can set it again. How does this command above allow me to send the
>> > calls out a certain way and treat them. what are the options for that
>> > fifo_orbit_dialplan
>> >
>> > thanks
>> > Marc
>> > On Tue, Jan 25, 2011 at 2:40 AM, liuyp2 <liuyp2 at asiainfo-linkage.com>
>> > wrote:
>> >>
>> >> mod_fifo can't transfer sip header? message which defined by
>> >> myself(sip_h_X-xxx) to b-leg also.
>> >>
>> >> Is there any solution in latest version?
>> >>
>> >> ________________________________
>> >> liuyp2
>> >> 2011-01-25
>> >> ________________________________
>> >>
>> >> ???? Anthony Minessale
>> >> ????? 2011-01-25?09:28:25
>> >> ???? FreeSWITCH Users Help
>> >> ???
>> >> ??? Re: [Freeswitch-users] Caller ID using Fifo
>> >>
>> >>
>> >> You should all confer to make sure you are all using fs latest git
>> >> because
>> >> that is the version I am talking about.? Fifo has some major new
>> >> features in
>> >> latest that do not exist in older versions including showing the
>> >> customers
>> >> cid when it calls agents.? The dilemma jm describes used to be true but
>> >> is
>> >> no longer the case with the default ringall strategy on latest git.
>> >>
>> >> The customers cid is sent to the agent and if the fifo xml defines
>> >> outbound_name param that will be included as well.
>> >>
>> >> If you want to override it you must do what you quoted in the wiki in
>> >> the
>> >> dialstring contained in the member tag of the xml for that membership
>> >> not in
>> >> the dialplan.
>> >>
>> >> On Jan 14, 2011 10:36 AM, "Marc de Corny" <marcdecorny at gmail.com>
>> >> wrote:
>> >> >
>> >> > Just to follow up on this subject.
>> >> >
>> >> > I have done a lot of testing on the fifo trying to get the
>> >> > caller_id_name changed on the outbound call to the agent and to be
>> >> > honest I
>> >> > cannot understand the explanation.
>> >> >
>> >> > If mod_fifo does not know which call it will connect until the agent
>> >> > answers, how come it displays the CLI correctly, jsut won;t let me
>> >> > change
>> >> > it.
>> >> >
>> >> > Still seems strange. I am looking into the Mod_callcentre to check if
>> >> > it
>> >> > sends caller_id information. but the same logic if valid could apply
>> >> >
>> >> > Also maybe someone should change the Wiki ( I would but do not have
>> >> > enough expertise on the subject) because the following?is a bit
>> >> > misleading
>> >> >
>> >> > ?"Note: If you wish to specify the caller ID presented when a fifo
>> >> > calls
>> >> > an agent, set the origination_caller_id_name and
>> >> > origination_caller_id_num
>> >> > variables to the values desired. These could be set within the {} of
>> >> > the
>> >> > dialstring, or they could be set using the set application in the
>> >> > dialplan
>> >> > which places the caller into the fifo (before the 'fifo in' executed
>> >> > on the
>> >> > caller). "
>> >> > thanks
>> >> > Marc
>> >> > On Thu, Jan 13, 2011 at 10:47 PM, Joao Leme
>> >> > <joaocarlosleme at gmail.com>
>> >> > wrote:
>> >> >>
>> >> >> What about showing the Caller ID after it is?answered? Any way to do
>> >> >> that?
>> >> >>
>> >> >> 2011/1/12 Jo?o Mesquita <jmesquita at freeswitch.org>
>> >> >>
>> >> >>> Jo?o Leme,
>> >> >>>
>> >> >>> The caller id is not passed when the phone is ringing because
>> >> >>> mod_fifo
>> >> >>> does not know which call is going to be sent to that channel once
>> >> >>> it is
>> >> >>> answered until it is really answered. I don't know if
>> >> >>> mod_callcenter does
>> >> >>> show anything but you should consider looking at the documentation
>> >> >>> if you
>> >> >>> really need this feature.
>> >> >>>
>> >> >>> Regards,
>> >> >>> Jo?o Mesquita
>> >> >>>
>> >> >>>
>> >> >>> On Wed, Jan 12, 2011 at 9:15 PM, Joao Leme
>> >> >>> <joaocarlosleme at gmail.com>
>> >> >>> wrote:
>> >> >>>>
>> >> >>>> Hi there,
>> >> >>>> I would like to know if there is a way to see the caller ID on my
>> >> >>>> Sip
>> >> >>>> Client (X-Lite for example) of the caller that I answear from a
>> >> >>>> Fifo queue?
>> >> >>>> Thanks,
>> >> >>>> John
>> >> >>>>
>> >> >>>> _______________________________________________
>> >> >>>> FreeSWITCH-users mailing list
>> >> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>>
>> >> >>>>
>> >> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>>> http://www.freeswitch.org
>> >> >>>>
>> >> >>>
>> >> >>>
>> >> >>> _______________________________________________
>> >> >>> FreeSWITCH-users mailing list
>> >> >>> FreeSWITCH-users at lists.freeswitch.org
>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>
>> >> >>>
>> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> http://www.freeswitch.org
>> >> >>>
>> >> >>
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >>
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From 28adam28 at gmail.com  Thu Jan 27 13:16:11 2011
From: 28adam28 at gmail.com (aaaa bbbb)
Date: Thu, 27 Jan 2011 11:16:11 +0100
Subject: [Freeswitch-users] change codec on the bypass point
Message-ID: <AANLkTikiLKcsb+oCLmdrxa_BqKPVX7mzRi3tQxvtqj0q@mail.gmail.com>

Hi,

I am using FS and when i make calls from "A" to "C" through "B". On the "B"
i am using bypass media. "B" takes the codec list from A and send it to C as
outgoing codec list. I would like to change the outgoing codec on "B" but I
change anything on "B", it only using the codec list what get from "A".

How can i change the codec list on "B"?

Regards,
Adam
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From patrick.plattes at niemann-frey.info  Thu Jan 27 15:08:25 2011
From: patrick.plattes at niemann-frey.info (Patrick Plattes)
Date: Thu, 27 Jan 2011 13:08:25 +0100
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
Message-ID: <AANLkTimaOaKfADikHuEQ8uXXtV4xJUJo=4oxErJr2Wsf@mail.gmail.com>

Hi Michael :-),

would it be okay to you to get a pcap file? I think the sip traces
from the phone are not very useful

Thanks,
 Patrick


2011/1/27 Michael Collins <msc at freeswitch.org>:
> Could you humor us and get sip traces of the blf working on a reboot w/
> asterisk and not working with freeswitch? Put them in
> pastebin.freeswitch.org and drop the link in this thread.
> Thanks,
> MC


From mkopacki at gmail.com  Thu Jan 27 17:25:00 2011
From: mkopacki at gmail.com (Michal Kopacki)
Date: Thu, 27 Jan 2011 15:25:00 +0100
Subject: [Freeswitch-users] network scenario doubts
In-Reply-To: <AANLkTinBd00iKLysi1O97hOupQpYdzBchBN5t8oZOJre@mail.gmail.com>
References: <4D3FD3B0.4080309@gmail.com>
	<AANLkTinBd00iKLysi1O97hOupQpYdzBchBN5t8oZOJre@mail.gmail.com>
Message-ID: <4D41803C.1050601@gmail.com>

     Everything is set as you wrote and I'm able to login (using domain 
name). Now I'm facing a media problem during outside call. I'm logged to 
FS from ip phone (outside lan) and trying to dial 5000 (default ivr). 
Call is connected but i'm not able to hear anything.

    There is iptable based firewall with opened udp ports 16384:32768, 
but it seems nothing even hit those ports:

Chain INPUT (policy DROP 0 packets, 0 bytes)
  pkts bytes target     prot opt in     out     source               
destination
     0     0 ACCEPT     udp  --  eth0   any     anywhere             
anywhere            udp dpts:connected:filenet-tms state NEW

   Any thoughts ?
-- 
Best regards,
Michal

On 01/27/2011 01:39 AM, Michael Collins wrote:
> yes, you should be able to handle this scenario. You probably just 
> need to set your external profile to use port 5060 instead of 5080. 
> Most people only have a single NIC so their sofia profiles need to be 
> on different ports. (A sofia profile is a SIP user agent (UA) that can 
> listen/respond to SIP messages on an IP address and port.)
>
> Look in external.xml for these lines:
> <param name="rtp-ip" value="$${local_ip_v4}"/>
> <param name="sip-ip" value="$${local_ip_v4}"/>
>
> If you have a static IP address (which it appears you do) then put 
> that ip addr in there in place of the variable. Then look at the end 
> of vars.xml and you'll see where the external profile's port is set to 
> 5080/5081. Change it to 5060/5061 and then restart FS.
>
> Let us know how it goes.
>
> -MC
>
> On Tue, Jan 25, 2011 at 11:56 PM, Michal Kopacki <mkopacki at gmail.com 
> <mailto:mkopacki at gmail.com>> wrote:
>
>           Hello,
>
>       This is my first post to this list, so hello everyone.
>
>       I'm at the very beginning of freeswitch journey and i have a problem
>     with fit FS to my network scenario.
>
>     OS: fedora 13 (x86_64)
>     FS: 1.0.7 (compiled from sources), default config
>
>     Desired scenario:
>
>     softphone -> mydomain.com <http://mydomain.com> (with ip
>     193.59.72.xx) -> my isp network ->
>     10.25.48.xx on eth1 -> FS -> 192.168.0.1 on eth0 -> softphone
>
>     And now, I'm able to connect to FS from my local network, but I'm not
>     able to connect from outside (neither domain nor ip). In
>     internal.xml I
>     set internal ip of server and i wanted to set external ip in
>     external.xml, but there is a question: which one ?  In case of
>     193.59.72.xx external profile didnt' start and with 10.25.48.xx
>     outside
>     softphone didn't register.  I checked with netstat and realized that
>     port 5060 is bind to internal nic only and 5080 to external nic
>     (10.25.48.xx) and I have no idea what next.
>
>     Is it even possible to work with such network scenario ? I would be
>     grateful for pointing me to right direction or maybe propose different
>     approach.
>
>     --
>     Best regards,
>     Michal
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From Michal.Muszalski at telekomunikacja.pl  Thu Jan 27 17:26:41 2011
From: Michal.Muszalski at telekomunikacja.pl (=?iso-8859-2?Q?Muszalski_Micha=B3_-_Korpo_TP?=)
Date: Thu, 27 Jan 2011 15:26:41 +0100
Subject: [Freeswitch-users] PD: SIP -> XMPP calls
Message-ID: <964C8735B9EC444A99C4EC298B1BA0AB0207BF59@OPEXCN07.tp.gk.corp.tepenet>


> Hello,
> 
> We are trying to make some kind of a gateway between SIP and XMPP domain.
> We have an environment with FreeSWITCH and OpenFire (FreeSWITCH is
> registered as a component in OF). Calls from XMPP to SIP are working fine.
> The problem is with calls from SIP to XMPP. The caller (SIP) has a ringing
> tone, a the callee (XMPP) has a 'connecting...' message after answer the
> call.
> Do you have any experience in making calls between SIP and XMPP users? Is
> it possible to establish the video calls also?
> 
> Best Regards,
> 
> Micha? Muszalski
> Senior Specialist
> 
> Orange Labs Poland
> Obrze?na 7, 02-691 Warsaw
> 
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From helmut.kuper at ewetel.de  Thu Jan 27 18:31:10 2011
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Thu, 27 Jan 2011 16:31:10 +0100
Subject: [Freeswitch-users] Problems with t38 and tone detection
Message-ID: <4D418FBE.5040909@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

I'm using mod_spandsp and t38gateway function for FAX transport. Some
months ago I get it up and running and documented it in FS's wiki. What
I forgot is, that I had to modify mod_spanddsp.c in the following way:

replace this:
switch_ivr_tone_detect_session(session, "t38", "1100.0", "rw", timeout,
1, data, NULL, t38_gateway_start);

with this:
switch_ivr_tone_detect_session(session, "t38", "2100.0", "rw", timeout,
1, data, NULL, t38_gateway_start);


I'm not sure whether this is a dirty hack which works just for my
environment, or whether this is a general bug. This difference is the
frequency to which the gateways is listening.

If I use 1100 the gateway never detects the fax tone which is send by
the internal fax device.



regards
Helmut
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.10 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/

iEYEARECAAYFAk1Bj74ACgkQ4tZeNddg3dzmugCgkLtnqsaBnf5Yhqz/BtDBzh35
gz0AniwyCAtR5gKQZHc0GSmF3nxyhNqR
=gI21
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From rajesh.npnr at yahoo.com  Thu Jan 27 18:38:14 2011
From: rajesh.npnr at yahoo.com (rex.alex)
Date: Thu, 27 Jan 2011 07:38:14 -0800 (PST)
Subject: [Freeswitch-users] Send email on successful fax sending.
In-Reply-To: <AANLkTin6BdJLZ6O2KydDY768U1tOU-xqxSACSFmL94bq@mail.gmail.com>
References: <AANLkTinJ3N+zwg+ZK2RmdAZECEkLgcvOSKT0hf_mbpix@mail.gmail.com>
	<AANLkTi=5g-rOZtS9Dy4wq_J6OQU2roLzBDwhzhsuhQSG@mail.gmail.com>
	<4D3976FD.4090805@chaschperli.ch>
	<AANLkTimo0n3UP2c09F3AmgQdkRdU+DVAMxjYrw7cP9gm@mail.gmail.com>
	<AANLkTi=zBgX-trHwVUUgfGNZh3SmcJfru1tD=5-aE=MY@mail.gmail.com>
	<AANLkTimKgZDpuDUgNaqT1=61mqwOoJ66u4m-u4LBjxXM@mail.gmail.com>
	<1295627914599-5948386.post@n2.nabble.com>
	<AANLkTi==DN40Ebdi0A0bqX95Ct7Ra_Px+uSX8i4acf0i@mail.gmail.com>
	<AANLkTin6BdJLZ6O2KydDY768U1tOU-xqxSACSFmL94bq@mail.gmail.com>
Message-ID: <1296142694375-5966720.post@n2.nabble.com>


Hello,

I am using api_hangup_hook and calling a lua function with
env:serialize(),env:getHeader("fax_success") to fetch the delivery status.
But I would like to use javascript since I am familiar with javascript odbc
to store the status in db, so what is the javascript equivalent to these
functions as I tried env.serialize(); and event.serialize(); nothing seems
to be working.

Please assist how to call these methods
http://wiki.freeswitch.org/wiki/Javascript_Event#Methods

Thanks,
Rex
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Send-email-on-successful-fax-sending-tp5881415p5966720.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Thu Jan 27 18:47:17 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Jan 2011 09:47:17 -0600
Subject: [Freeswitch-users] Problems with t38 and tone detection
In-Reply-To: <4D418FBE.5040909@ewetel.de>
References: <4D418FBE.5040909@ewetel.de>
Message-ID: <AANLkTi=Sk5xnUpuy377zuzs-gQ7am0WmuTctT2Xf+pd-@mail.gmail.com>

are you using 16k audio by any chance?
The tone is supposed to be 1100 if 21 is working it's possible you are
transcoding it and doubling the frequency?


On Thu, Jan 27, 2011 at 9:31 AM, Helmut Kuper <helmut.kuper at ewetel.de> wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> I'm using mod_spandsp and t38gateway function for FAX transport. Some
> months ago I get it up and running and documented it in FS's wiki. What
> I forgot is, that I had to modify mod_spanddsp.c in the following way:
>
> replace this:
> switch_ivr_tone_detect_session(session, "t38", "1100.0", "rw", timeout,
> 1, data, NULL, t38_gateway_start);
>
> with this:
> switch_ivr_tone_detect_session(session, "t38", "2100.0", "rw", timeout,
> 1, data, NULL, t38_gateway_start);
>
>
> I'm not sure whether this is a dirty hack which works just for my
> environment, or whether this is a general bug. This difference is the
> frequency to which the gateways is listening.
>
> If I use 1100 the gateway never detects the fax tone which is send by
> the internal fax device.
>
>
>
> regards
> Helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.10 (MingW32)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/
>
> iEYEARECAAYFAk1Bj74ACgkQ4tZeNddg3dzmugCgkLtnqsaBnf5Yhqz/BtDBzh35
> gz0AniwyCAtR5gKQZHc0GSmF3nxyhNqR
> =gI21
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From Nabble at slickdeals.endjunk.com  Thu Jan 27 19:25:12 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 27 Jan 2011 08:25:12 -0800 (PST)
Subject: [Freeswitch-users] PD: SIP -> XMPP calls
In-Reply-To: <964C8735B9EC444A99C4EC298B1BA0AB0207BF59@OPEXCN07.tp.gk.corp.tepenet>
References: <964C8735B9EC444A99C4EC298B1BA0AB0207BF59@OPEXCN07.tp.gk.corp.tepenet>
Message-ID: <1296145512923-5966876.post@n2.nabble.com>



Muszalski Micha? - Korpo TP wrote:
> 
> 
>> Hello,
>> 
>> We are trying to make some kind of a gateway between SIP and XMPP domain.
>> We have an environment with FreeSWITCH and OpenFire (FreeSWITCH is
>> registered as a component in OF). Calls from XMPP to SIP are working
>> fine.
>> The problem is with calls from SIP to XMPP. The caller (SIP) has a
>> ringing
>> tone, a the callee (XMPP) has a 'connecting...' message after answer the
>> call.
I see this problem with a self-built FS git as of 01/26/2011, but not on a
FreeSWITCH Version 1.0.head (git-cf253c3 2011-01-11 15-31-40 -0600).

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/PD-SIP-XMPP-calls-tp5966636p5966876.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From miguel.mirandag at gmail.com  Thu Jan 27 19:37:04 2011
From: miguel.mirandag at gmail.com (Miguel Miranda)
Date: Thu, 27 Jan 2011 10:37:04 -0600
Subject: [Freeswitch-users] conference with did example
In-Reply-To: <AANLkTik7mXLbTx2gGXDwx3EaKJaJTRofCqY+Bnyo6rnY@mail.gmail.com>
References: <AANLkTik-wYp99Mbw6==EnT21U5RGk_MLPCt+g-wA+nFs@mail.gmail.com>
	<AANLkTinsDpRV6p5uFp4WGiNeOupcL6TAfteoJ31K8Qf4@mail.gmail.com>
	<AANLkTik7mXLbTx2gGXDwx3EaKJaJTRofCqY+Bnyo6rnY@mail.gmail.com>
Message-ID: <AANLkTinJP0irvvw_stiX=7GGxYsAt-R1kmY8=wMCLr7M@mail.gmail.com>

Great, a few questions:

1) ${domain} is the  conference's profile name right?
2)what are the digits 4437000 ? the conference number?
3) the sound files should be any regular.wav file, even one that i can
record using my laptop's mic, no special requeriments?

thanks in advance

On Wed, Jan 26, 2011 at 7:27 PM, Michael Collins <msc at freeswitch.org> wrote:

> I don't believe this example will work because ${confnumber} won't be
> populated at the time the dialplan is parsed. However a trivial modification
> would make it work:
>
>   <extension name="ask user for conf number">
>     <condition field="destination_number" expression="^(9903)$">
>       <action application="answer"/>
>       <action application="play_and_get_digits"
>               data="4 4 3 7000 # conference/conf-enter_conf_number.wav
> ivr/ivr-that_was_an_invalid_entry.wav confnumber \d+"/>
>       <action application="playback" data="ivr/ivr-one_moment_please.wav"/>
>       <action application="set" data="ringback=${us-ring}"/>
>       <action application="transfer" data="USER_DIALED_${confnumber} XML
> default"/>
>     </condition>
>   </extension>
>
>   <extension name="put user into conf">
>     <condition field="destination_number" expression="^USER_DIALED_(\d+)$">
>       <action application="conference" data="$1@${domain}"/>
>     </condition>
>   </extension>
>
> Note that I tested this with real sound files on my system instead of the
> pretend ones that were there. I also used dest num of "9903" - use a value
> that works for you.
>
> -MC
>
> On Wed, Jan 26, 2011 at 3:57 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Something like this:
>>
>> <extension name="conference">
>>    <condition field="destination_number" expression="^12345$">
>>       <action application="answer"/>
>>       <action application="play_and_get_digits" data="4 4 3 7000 #
>> $${base_dir}/sounds/conf-number.wav /invalid.wav confnumber \d+"/>
>>
>>       <action application="playback" data="connecting_your_call.wav"/>
>>       <action application="set" data="ringback=${us-ring}"/>
>>       <action application="conference" data="${confnumber}@
>> ${domain_name}"/>
>>    </condition>
>> </extension>
>>
>>
>>
>>
>> On 26 January 2011 22:10, Miguel Miranda <miguel.mirandag at gmail.com>wrote:
>>
>>> Hi, i have searched the arvhices and could not find a simple example on
>>> how to configure a DID for  mod_conference, i mean you dial an DID, an ivr
>>> responds "please enter you conference number and press puond key, etc", and
>>> if the conference room was configured with pin it asks "please enter ?pin
>>> number, etc".
>>> as i undestend, in this example from wiki:
>>>
>>> <extension name="test_bridging_conference">
>>>    <condition field="destination_number" expression="^(3000)$">
>>>       <action application="answer"/>
>>>
>>>
>>>
>>>       <action application="playback" data="connecting_your_call.wav"/>
>>>       <action application="set" data="ringback=${us-ring}"/>
>>>       <action application="conference" data="bridge:$1-${domain_name}@default:user/1000@${domain_name}"/>
>>>
>>>
>>>
>>>    </condition>
>>> </extension>
>>>
>>>
>>> i need a gateway and if you receive the DID 3000 you join the conference
>>> right?
>>> What i want is a single did for all the conferences.
>>> regards
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From helmut.kuper at ewetel.de  Thu Jan 27 19:39:54 2011
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Thu, 27 Jan 2011 17:39:54 +0100
Subject: [Freeswitch-users] Problems with t38 and tone detection
In-Reply-To: <AANLkTi=Sk5xnUpuy377zuzs-gQ7am0WmuTctT2Xf+pd-@mail.gmail.com>
References: <4D418FBE.5040909@ewetel.de>
	<AANLkTi=Sk5xnUpuy377zuzs-gQ7am0WmuTctT2Xf+pd-@mail.gmail.com>
Message-ID: <4D419FDA.8040809@ewetel.de>

erm, afaik no. The used codec is G711-A on both sides.

How can I make sure it is 8k?


Am 27.01.2011 16:47, schrieb Anthony Minessale:
> are you using 16k audio by any chance?
> The tone is supposed to be 1100 if 21 is working it's possible you are
> transcoding it and doubling the frequency?



From david.ponzone at ipeva.fr  Thu Jan 27 19:44:27 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 27 Jan 2011 17:44:27 +0100
Subject: [Freeswitch-users] conference with did example
In-Reply-To: <AANLkTinJP0irvvw_stiX=7GGxYsAt-R1kmY8=wMCLr7M@mail.gmail.com>
References: <AANLkTik-wYp99Mbw6==EnT21U5RGk_MLPCt+g-wA+nFs@mail.gmail.com>
	<AANLkTinsDpRV6p5uFp4WGiNeOupcL6TAfteoJ31K8Qf4@mail.gmail.com>
	<AANLkTik7mXLbTx2gGXDwx3EaKJaJTRofCqY+Bnyo6rnY@mail.gmail.com>
	<AANLkTinJP0irvvw_stiX=7GGxYsAt-R1kmY8=wMCLr7M@mail.gmail.com>
Message-ID: <64B53B21-223A-4E0D-BFBB-F940E22A848F@ipeva.fr>

Miguel,

I think you should be able to find most answers in the wiki.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 27/01/2011 ? 17:37, Miguel Miranda a ?crit :

> Great, a few questions:
> 
> 1) ${domain} is the  conference's profile name right?
> 2)what are the digits 4437000 ? the conference number?
> 3) the sound files should be any regular.wav file, even one that i can record using my laptop's mic, no special requeriments?
> 
> thanks in advance
> 
> On Wed, Jan 26, 2011 at 7:27 PM, Michael Collins <msc at freeswitch.org> wrote:
> I don't believe this example will work because ${confnumber} won't be populated at the time the dialplan is parsed. However a trivial modification would make it work:
> 
>   <extension name="ask user for conf number">
>     <condition field="destination_number" expression="^(9903)$">
>       <action application="answer"/>
>       <action application="play_and_get_digits"
>               data="4 4 3 7000 # conference/conf-enter_conf_number.wav ivr/ivr-that_was_an_invalid_entry.wav confnumber \d+"/>
>       <action application="playback" data="ivr/ivr-one_moment_please.wav"/>
>       <action application="set" data="ringback=${us-ring}"/>
>       <action application="transfer" data="USER_DIALED_${confnumber} XML default"/>
>     </condition>
>   </extension>
> 
>   <extension name="put user into conf">
>     <condition field="destination_number" expression="^USER_DIALED_(\d+)$">
>       <action application="conference" data="$1@${domain}"/>
>     </condition>
>   </extension>
> 
> Note that I tested this with real sound files on my system instead of the pretend ones that were there. I also used dest num of "9903" - use a value that works for you.
> 
> -MC
> 
> On Wed, Jan 26, 2011 at 3:57 PM, Steven Ayre <steveayre at gmail.com> wrote:
> Something like this:
> 
> <extension name="conference">
>    <condition field="destination_number" expression="^12345$">
>       <action application="answer"/>
>       <action application="play_and_get_digits" data="4 4 3 7000 # $${base_dir}/sounds/conf-number.wav /invalid.wav confnumber \d+"/>
> 
>       <action application="playback" data="connecting_your_call.wav"/>
>       <action application="set" data="ringback=${us-ring}"/>
>       <action application="conference" data="${confnumber}@${domain_name}"/>
>    </condition>
> </extension>
> 
> 
> 
> 
> On 26 January 2011 22:10, Miguel Miranda <miguel.mirandag at gmail.com> wrote:
> Hi, i have searched the arvhices and could not find a simple example on how to configure a DID for  mod_conference, i mean you dial an DID, an ivr responds "please enter you conference number and press puond key, etc", and if the conference room was configured with pin it asks "please enter ?pin number, etc".
> as i undestend, in this example from wiki:
> 
> <extension name="test_bridging_conference">
>    <condition field="destination_number" expression="^(3000)$">
>       <action application="answer"/>
> 
> 
> 
> 
>       <action application="playback" data="connecting_your_call.wav"/>
>       <action application="set" data="ringback=${us-ring}"/>
>       <action application="conference" data="bridge:$1-${domain_name}@default:user/1000@${domain_name}"/>
> 
> 
> 
> 
>    </condition>
> </extension>
> 
> i need a gateway and if you receive the DID 3000 you join the conference right?
> What i want is a single did for all the conferences.
> regards
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From infos at madovsky.org  Thu Jan 27 19:54:26 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 27 Jan 2011 11:54:26 -0500
Subject: [Freeswitch-users] conference with did example
References: <AANLkTik-wYp99Mbw6==EnT21U5RGk_MLPCt+g-wA+nFs@mail.gmail.com><AANLkTinsDpRV6p5uFp4WGiNeOupcL6TAfteoJ31K8Qf4@mail.gmail.com><AANLkTik7mXLbTx2gGXDwx3EaKJaJTRofCqY+Bnyo6rnY@mail.gmail.com><AANLkTinJP0irvvw_stiX=7GGxYsAt-R1kmY8=wMCLr7M@mail.gmail.com>
	<64B53B21-223A-4E0D-BFBB-F940E22A848F@ipeva.fr>
Message-ID: <F5DB2244E0CF44569D2CD44A562742E2@e1705>

I recommand to all new user to buy the FS book,
you'll find all your answer easily with good examples.
  ----- Original Message ----- 
  From: David Ponzone 
  To: FreeSWITCH Users Help 
  Sent: Thursday, January 27, 2011 11:44 AM
  Subject: Re: [Freeswitch-users] conference with did example


  Miguel,


  I think you should be able to find most answers in the wiki.


  David Ponzone  Direction Technique
  email: david.ponzone at ipeva.fr
  tel:      01 74 03 18 97
  gsm:   06 66 98 76 34


  Service Client IPeva
  tel:      0811 46 26 26
  www.ipeva.fr  -   www.ipeva-studio.com


  Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.







  Le 27/01/2011 ? 17:37, Miguel Miranda a ?crit :


    Great, a few questions:

    1) ${domain} is the  conference's profile name right?
    2)what are the digits 4437000 ? the conference number?
    3) the sound files should be any regular.wav file, even one that i can record using my laptop's mic, no special requeriments?

    thanks in advance


    On Wed, Jan 26, 2011 at 7:27 PM, Michael Collins <msc at freeswitch.org> wrote:

      I don't believe this example will work because ${confnumber} won't be populated at the time the dialplan is parsed. However a trivial modification would make it work:


        <extension name="ask user for conf number">
          <condition field="destination_number" expression="^(9903)$">
            <action application="answer"/>
            <action application="play_and_get_digits"
                    data="4 4 3 7000 # conference/conf-enter_conf_number.wav ivr/ivr-that_was_an_invalid_entry.wav confnumber \d+"/>
            <action application="playback" data="ivr/ivr-one_moment_please.wav"/>
            <action application="set" data="ringback=${us-ring}"/>
            <action application="transfer" data="USER_DIALED_${confnumber} XML default"/>
          </condition>
        </extension>


        <extension name="put user into conf">
          <condition field="destination_number" expression="^USER_DIALED_(\d+)$">
            <action application="conference" data="$1@${domain}"/>
          </condition>
        </extension>


      Note that I tested this with real sound files on my system instead of the pretend ones that were there. I also used dest num of "9903" - use a value that works for you.


      -MC


      On Wed, Jan 26, 2011 at 3:57 PM, Steven Ayre <steveayre at gmail.com> wrote:

        Something like this:

        <extension name="conference">
           <condition field="destination_number" expression="^12345$">
              <action application="answer"/>
              <action application="play_and_get_digits" data="4 4 3 7000 # $${base_dir}/sounds/conf-number.wav /invalid.wav confnumber \d+"/>

              <action application="playback" data="connecting_your_call.wav"/>
              <action application="set" data="ringback=${us-ring}"/>

              <action application="conference" data="${confnumber}@${domain_name}"/>
           </condition>
        </extension>





        On 26 January 2011 22:10, Miguel Miranda <miguel.mirandag at gmail.com> wrote:

          Hi, i have searched the arvhices and could not find a simple example on how to configure a DID for  mod_conference, i mean you dial an DID, an ivr responds "please enter you conference number and press puond key, etc", and if the conference room was configured with pin it asks "please enter ?pin number, etc".
          as i undestend, in this example from wiki:


<extension name="test_bridging_conference">   <condition field="destination_number" expression="^(3000)$">      <action application="answer"/>



      <action application="playback" data="connecting_your_call.wav"/>      <action application="set" data="ringback=${us-ring}"/>      <action application="conference" data="bridge:$1-${domain_name}@default:user/1000@${domain_name}"/>



   </condition></extension>
          i need a gateway and if you receive the DID 3000 you join the conference right?
          What i want is a single did for all the conferences.
          regards



          _______________________________________________
          FreeSWITCH-users mailing list
          FreeSWITCH-users at lists.freeswitch.org
          http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
          http://www.freeswitch.org




        _______________________________________________
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        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org





      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org



    _______________________________________________
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    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





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From infos at madovsky.org  Thu Jan 27 19:56:01 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 27 Jan 2011 11:56:01 -0500
Subject: [Freeswitch-users] Problems with t38 and tone detection
References: <4D418FBE.5040909@ewetel.de><AANLkTi=Sk5xnUpuy377zuzs-gQ7am0WmuTctT2Xf+pd-@mail.gmail.com>
	<4D419FDA.8040809@ewetel.de>
Message-ID: <4EEFDA45BF4846F38271456CFBDFB3A6@e1705>

there's a channel var codec_rate maybe it can help you

----- Original Message ----- 
From: "Helmut Kuper" <helmut.kuper at ewetel.de>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, January 27, 2011 11:39 AM
Subject: Re: [Freeswitch-users] Problems with t38 and tone detection


> erm, afaik no. The used codec is G711-A on both sides.
> 
> How can I make sure it is 8k?
> 
> 
> Am 27.01.2011 16:47, schrieb Anthony Minessale:
>> are you using 16k audio by any chance?
>> The tone is supposed to be 1100 if 21 is working it's possible you are
>> transcoding it and doubling the frequency?
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From frank at telonium.com  Thu Jan 27 20:38:12 2011
From: frank at telonium.com (Frank Park)
Date: Thu, 27 Jan 2011 12:38:12 -0500
Subject: [Freeswitch-users] Choppy VM
Message-ID: <AANLkTimBXKiC2frKN3UBOC_c-gdfgk8N--k1xX2F7fqR@mail.gmail.com>

Hey guys,

I've been trying to troubleshoot this, but haven't been able to figure it
out...
I am currently experiencing choppy voicemail recordings on my freeswitch.
The calls are coming in using SIP trunk and the call quality using the same
trunk has no problem, but when I listen back the vm, that can be a different
story.
This is running on a dedicate server and not only a virtual instance. I am
currently not using mod_shout, so this is being saved as a wav file.
I've monitored CPU and memory usage while this happens, and because they are
so low, I don't think resource exhaustion is a factor.

Any direction as to what I should try to troubleshoot what's wrong?
I appreciate any input.

Thank you,
Frank


-- 

----=======================----
Frank Park
Telonium Communications, LLC
frank at telonium.com
http://www.telonium.com
Follow Us on Twitter: @GetTelonium
404-566-8888 x1001 Office
404-939-4242 Cell
----=======================----
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From anthony.minessale at gmail.com  Thu Jan 27 20:41:13 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Jan 2011 11:41:13 -0600
Subject: [Freeswitch-users] Choppy VM
In-Reply-To: <AANLkTimBXKiC2frKN3UBOC_c-gdfgk8N--k1xX2F7fqR@mail.gmail.com>
References: <AANLkTimBXKiC2frKN3UBOC_c-gdfgk8N--k1xX2F7fqR@mail.gmail.com>
Message-ID: <AANLkTinU6GsT3NMKJNQxr4_Etis=v3LaZhTmUZtaMsNa@mail.gmail.com>

have you tried the latest version of the code?
Are you using mod_loopback in your VM calls?


On Thu, Jan 27, 2011 at 11:38 AM, Frank Park <frank at telonium.com> wrote:
> Hey guys,
> I've been trying to troubleshoot this, but haven't been able to figure it
> out...
> I am currently experiencing choppy voicemail recordings on my freeswitch.
> The calls are coming in using SIP trunk and the call quality using the same
> trunk has no problem, but when I listen back the vm, that can be a different
> story.
> This is running on a dedicate server and not only a virtual instance. I am
> currently not using mod_shout, so this is being saved as a wav file.
> I've monitored CPU and memory usage while this happens, and because they are
> so low, I don't think resource?exhaustion?is a factor.
> Any direction as to what I should try to troubleshoot what's wrong?
> I appreciate any input.
> Thank you,
> Frank
>
> --
>
> ----=======================----
> Frank Park
> Telonium Communications, LLC
> frank at telonium.com
> http://www.telonium.com
> Follow Us on Twitter: @GetTelonium
> 404-566-8888 x1001 Office
> 404-939-4242 Cell
> ----=======================----
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From jeff at jefflenk.com  Thu Jan 27 20:43:04 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 27 Jan 2011 09:43:04 -0800 (PST)
Subject: [Freeswitch-users] mod_dingaling and the weekly windows build
In-Reply-To: <8C68232BC9314C40BBCDDAA480F7B01AEA3B493B34@harrow.exch.ad.byu.edu>
References: <8C68232BC9314C40BBCDDAA480F7B01AEA3B493B34@harrow.exch.ad.byu.edu>
Message-ID: <1296150184550-5967186.post@n2.nabble.com>


That dll is not part of the weekly dist.
If you want to experiment with that you will have to build the code
yourself. You will also have to obtain and integrate GnuTLS into the build
for that to work(Google Voice). There are some introductory docs on the Wiki
that outline what may be required for doing this but keep in mind that this
functionality is only experimental.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-dingaling-and-the-weekly-windows-build-tp5965119p5967186.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From david.ponzone at ipeva.fr  Thu Jan 27 20:44:26 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 27 Jan 2011 18:44:26 +0100
Subject: [Freeswitch-users] Choppy VM
In-Reply-To: <AANLkTimBXKiC2frKN3UBOC_c-gdfgk8N--k1xX2F7fqR@mail.gmail.com>
References: <AANLkTimBXKiC2frKN3UBOC_c-gdfgk8N--k1xX2F7fqR@mail.gmail.com>
Message-ID: <61CA99D2-1FD0-41E1-92A3-5540CB3097E6@ipeva.fr>

Is your kernel compiled with 1000hz ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 27/01/2011 ? 18:38, Frank Park a ?crit :

> Hey guys,
> 
> I've been trying to troubleshoot this, but haven't been able to figure it out...
> I am currently experiencing choppy voicemail recordings on my freeswitch. The calls are coming in using SIP trunk and the call quality using the same trunk has no problem, but when I listen back the vm, that can be a different story.
> This is running on a dedicate server and not only a virtual instance. I am currently not using mod_shout, so this is being saved as a wav file.
> I've monitored CPU and memory usage while this happens, and because they are so low, I don't think resource exhaustion is a factor.
> 
> Any direction as to what I should try to troubleshoot what's wrong?
> I appreciate any input.
> 
> Thank you,
> Frank
> 
> 
> -- 
> 
> ----=======================----
> Frank Park
> Telonium Communications, LLC
> frank at telonium.com
> http://www.telonium.com
> Follow Us on Twitter: @GetTelonium
> 404-566-8888 x1001 Office
> 404-939-4242 Cell
> ----=======================----
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From covici at ccs.covici.com  Thu Jan 27 20:54:50 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 27 Jan 2011 12:54:50 -0500
Subject: [Freeswitch-users] Choppy VM
In-Reply-To: <AANLkTinU6GsT3NMKJNQxr4_Etis=v3LaZhTmUZtaMsNa@mail.gmail.com>
References: <AANLkTimBXKiC2frKN3UBOC_c-gdfgk8N--k1xX2F7fqR@mail.gmail.com>
	<AANLkTinU6GsT3NMKJNQxr4_Etis=v3LaZhTmUZtaMsNa@mail.gmail.com>
Message-ID: <20718.1296150890@ccs.covici.com>

I have experienced the choppy vm sometimes -- is there an alternative to
using loopback and why should it make a difference?

Anthony Minessale <anthony.minessale at gmail.com> wrote:

> have you tried the latest version of the code?
> Are you using mod_loopback in your VM calls?
> 
> 
> On Thu, Jan 27, 2011 at 11:38 AM, Frank Park <frank at telonium.com> wrote:
> > Hey guys,
> > I've been trying to troubleshoot this, but haven't been able to figure it
> > out...
> > I am currently experiencing choppy voicemail recordings on my freeswitch.
> > The calls are coming in using SIP trunk and the call quality using the same
> > trunk has no problem, but when I listen back the vm, that can be a different
> > story.
> > This is running on a dedicate server and not only a virtual instance. I am
> > currently not using mod_shout, so this is being saved as a wav file.
> > I've monitored CPU and memory usage while this happens, and because they are
> > so low, I don't think resource?exhaustion?is a factor.
> > Any direction as to what I should try to troubleshoot what's wrong?
> > I appreciate any input.
> > Thank you,
> > Frank
> >
> > --
> >
> > ----=======================----
> > Frank Park
> > Telonium Communications, LLC
> > frank at telonium.com
> > http://www.telonium.com
> > Follow Us on Twitter: @GetTelonium
> > 404-566-8888 x1001 Office
> > 404-939-4242 Cell
> > ----=======================----
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From steveayre at gmail.com  Thu Jan 27 20:56:46 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 27 Jan 2011 17:56:46 +0000
Subject: [Freeswitch-users] conference with did example
In-Reply-To: <AANLkTinJP0irvvw_stiX=7GGxYsAt-R1kmY8=wMCLr7M@mail.gmail.com>
References: <AANLkTik-wYp99Mbw6==EnT21U5RGk_MLPCt+g-wA+nFs@mail.gmail.com>
	<AANLkTinsDpRV6p5uFp4WGiNeOupcL6TAfteoJ31K8Qf4@mail.gmail.com>
	<AANLkTik7mXLbTx2gGXDwx3EaKJaJTRofCqY+Bnyo6rnY@mail.gmail.com>
	<AANLkTinJP0irvvw_stiX=7GGxYsAt-R1kmY8=wMCLr7M@mail.gmail.com>
Message-ID: <AANLkTinq8y7nA3=U=xDetosgfzUvvnSaRwZt38u8+ksg@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits

On 27 January 2011 16:37, Miguel Miranda <miguel.mirandag at gmail.com> wrote:

> Great, a few questions:
>
> 1) ${domain} is the  conference's profile name right?
>

You can replace that with whatever the profile is named


> 2)what are the digits 4437000 ? the conference number?
>

Read the play_and_get_digits documentation on the wiki, that explains the
syntax:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits

Basically, they specify the number of digits, number of retries to allow and
the timeout waiting for the number to be entered.

3) the sound files should be any regular.wav file, even one that i can
> record using my laptop's mic, no special requeriments?
>
> thanks in advance


Almost any sound format supported by sndfile will probably work. WAV files
will use more space but less CPU (it's raw audio so FS won't need to
decompress it).


>
>
> On Wed, Jan 26, 2011 at 7:27 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> I don't believe this example will work because ${confnumber} won't be
>> populated at the time the dialplan is parsed. However a trivial modification
>> would make it work:
>>
>>   <extension name="ask user for conf number">
>>     <condition field="destination_number" expression="^(9903)$">
>>        <action application="answer"/>
>>       <action application="play_and_get_digits"
>>               data="4 4 3 7000 # conference/conf-enter_conf_number.wav
>> ivr/ivr-that_was_an_invalid_entry.wav confnumber \d+"/>
>>       <action application="playback"
>> data="ivr/ivr-one_moment_please.wav"/>
>>       <action application="set" data="ringback=${us-ring}"/>
>>       <action application="transfer" data="USER_DIALED_${confnumber} XML
>> default"/>
>>     </condition>
>>   </extension>
>>
>>   <extension name="put user into conf">
>>     <condition field="destination_number"
>> expression="^USER_DIALED_(\d+)$">
>>       <action application="conference" data="$1@${domain}"/>
>>     </condition>
>>   </extension>
>>
>> Note that I tested this with real sound files on my system instead of the
>> pretend ones that were there. I also used dest num of "9903" - use a value
>> that works for you.
>>
>> -MC
>>
>> On Wed, Jan 26, 2011 at 3:57 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> Something like this:
>>>
>>> <extension name="conference">
>>>    <condition field="destination_number" expression="^12345$">
>>>       <action application="answer"/>
>>>       <action application="play_and_get_digits" data="4 4 3 7000 #
>>> $${base_dir}/sounds/conf-number.wav /invalid.wav confnumber \d+"/>
>>>
>>>       <action application="playback" data="connecting_your_call.wav"/>
>>>       <action application="set" data="ringback=${us-ring}"/>
>>>       <action application="conference" data="${confnumber}@
>>> ${domain_name}"/>
>>>    </condition>
>>> </extension>
>>>
>>>
>>>
>>>
>>> On 26 January 2011 22:10, Miguel Miranda <miguel.mirandag at gmail.com>wrote:
>>>
>>>> Hi, i have searched the arvhices and could not find a simple example on
>>>> how to configure a DID for  mod_conference, i mean you dial an DID, an ivr
>>>> responds "please enter you conference number and press puond key, etc", and
>>>> if the conference room was configured with pin it asks "please enter ?pin
>>>> number, etc".
>>>> as i undestend, in this example from wiki:
>>>>
>>>> <extension name="test_bridging_conference">
>>>>    <condition field="destination_number" expression="^(3000)$">
>>>>       <action application="answer"/>
>>>>
>>>>
>>>>
>>>>
>>>>       <action application="playback" data="connecting_your_call.wav"/>
>>>>       <action application="set" data="ringback=${us-ring}"/>
>>>>       <action application="conference" data="bridge:$1-${domain_name}@default:user/1000@${domain_name}"/>
>>>>
>>>>
>>>>
>>>>
>>>>    </condition>
>>>> </extension>
>>>>
>>>>
>>>> i need a gateway and if you receive the DID 3000 you join the conference
>>>> right?
>>>> What i want is a single did for all the conferences.
>>>> regards
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Jan 27 21:17:10 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Jan 2011 10:17:10 -0800
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTimaOaKfADikHuEQ8uXXtV4xJUJo=4oxErJr2Wsf@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
	<AANLkTimaOaKfADikHuEQ8uXXtV4xJUJo=4oxErJr2Wsf@mail.gmail.com>
Message-ID: <AANLkTi=ZGXEXmF_MSaKJ6J1TYJcObu4vnVQ3ApKpBeL=@mail.gmail.com>

pcaps are marvelous! Can you put those out on the web somewhere, like in a
drop box?
-MC

On Thu, Jan 27, 2011 at 4:08 AM, Patrick Plattes <
patrick.plattes at niemann-frey.info> wrote:

> Hi Michael :-),
>
> would it be okay to you to get a pcap file? I think the sip traces
> from the phone are not very useful
>
> Thanks,
>  Patrick
>
>
> 2011/1/27 Michael Collins <msc at freeswitch.org>:
> > Could you humor us and get sip traces of the blf working on a reboot w/
> > asterisk and not working with freeswitch? Put them in
> > pastebin.freeswitch.org and drop the link in this thread.
> > Thanks,
> > MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From patrick.plattes at niemann-frey.info  Thu Jan 27 21:32:35 2011
From: patrick.plattes at niemann-frey.info (Patrick Plattes)
Date: Thu, 27 Jan 2011 19:32:35 +0100
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
Message-ID: <AANLkTi=in3CLd-n0mT3SRKYA0kxLMGVWk9KeBx3_jhY6@mail.gmail.com>

Hi,

if've pcaps and videos to show you our problem.
http://kwixo.de/fs

Thank you,
 Patrick.

2011/1/27 Michael Collins <msc at freeswitch.org>:
> Could you humor us and get sip traces of the blf working on a reboot w/
> asterisk and not working with freeswitch? Put them in
> pastebin.freeswitch.org and drop the link in this thread.
> Thanks,
> MC
>
> On Wed, Jan 26, 2011 at 12:33 AM, Patrick Plattes
> <patrick.plattes at niemann-frey.info> wrote:
>>
>> Hi List :-),
>>
>> I'm currently switching from Asterisk to FreeSWITCH. It's really hard
>> work for an Asterisk user, but using Asterisk becomes more and more
>> painful even for small installations (less than 100 sip users). I know
>> FreeSWITCH is not a drop-in replacement for the Asterisk PBX, but I
>> don't want to change the behaviour of the PBX for the users.
>>
>> It's a preconception in Germany that the American people like
>> (especially for the X-mas time) kitsch. Everyone here know the
>> American houses with a hole bunch of blinking lights. But those
>> decorated hoses are nothing against our offices! Our phones have up to
>> 136 lights (BLFs). You often have to wear sunglasses at the office ;-)
>>
>> A typical usage of BLFs is to check if an agent is a member of the
>> queue. I've build a simple extension to add and delete a member. I can
>> user "presence in" and "presence out" to enable or disable the BLF,
>> but there is one big issue. After a reboot of the phone the the user
>> is still a member of the queue, but the BLF is off. We are using hints
>> at Asterisk to show the user if he is a member and it works even after
>> a reboot - with "presence" at FreeSWITCH it works (of cause) not. Does
>> anyone have an idea how to implement it?
>>
>> My current extension is just for testing and so I use mod_fifo. It
>> shouldn't be a problem to use mod_callcenter. The phone calls
>> "queue-the_name_of_the_queue-the_name_of_the_user at pbxdomain" eg.
>> "queue-sales_de-1000 at freeswitch.cust"
>>
>> So how implement BLF persistence?
>>
>> Thanks,
>> ?Patrick
>>
>>
>> Colourized version: http://pastebin.com/ufJ7U930
>>
>>
>>
>> <extension name="Queue Management">
>>
>> ?<condition field="destination_number"
>> expression="^queue-(\w+)-(\d+)$" break="on-false">
>> ? <action application="set"
>> data="queue_name=${regex(${destination_number}|^queue-(\w+)-(\d+)$|$1)}"
>> inline="true"/>
>> ? <action application="set"
>> data="queue_user=${regex(${destination_number}|^queue-(\w+)-(\d+)$|$2)}"
>> inline="true"/>
>> ?</condition>
>>
>> ?<condition field="${fifo(list ${queue_name}@${domain_name})}"
>> expression="${queue_user}">
>> ? <action application="answer"/>
>> ? <action application="set" data="result=${fifo_member(del
>> ${queue_name}@${domain_name}
>> {fifo_member_wait=nowait}user/${queue_user} )}"/>
>> ? <action application="log" data="INFO Delete FIFO agent result:
>> ${result}"/>
>> ? <action application="log" data="INFO User Logout: ${user_name}"/>
>> ? <action application="sleep" data="600"/>
>> ? <action application="playback"
>> data="ivr/ivr-you_are_now_logged_out.wav"/>
>> ? <action application="set" data="result=${presence(out
>> queue-${queue_name}-${queue_user}@${domain_name} inuse inuse)}"/>
>> ? <action application="log" data="INFO Presence OUT ${result}"/>
>> ? <action application="hangup"/>
>>
>> ? <anti-action application="answer"/>
>> ? <anti-action application="set" data="result=${fifo_member(add
>> ${queue_name}@${domain_name}
>> {fifo_member_wait=nowait}user/${queue_user} )}"/>
>> ? <anti-action application="log" data="INFO Add FIFO agent result:
>> ${result}"/>
>> ? <anti-action application="log" data="INFO User Login: ${user_name}"/>
>> ? <anti-action application="sleep" data="600"/>
>> ? <anti-action application="playback"
>> data="ivr/ivr-you_are_now_logged_in.wav"/>
>> ? <anti-action application="set" data="result=${presence(in
>> queue-${queue_name}-${queue_user}@${domain_name} inuse inuse)}"/>
>> ? <anti-action application="log" data="INFO Presence IN ${result}"/>
>> ? <anti-action application="hangup"/>
>> ?</condition>
>> </extension>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Patrick Plattes
IT - Projektleiter

Niemann + Frey GmbH
Adolf-Dembach-Str. 24
47829 Krefeld

Tel.: +49/2151 - 5554-263
Fax : +49/2151 - 5554-123
patrick.plattes at niemann-frey.info

Gesch?ftsf?hrer: Gerd Frey
Sitz und Registergericht: Krefeld HRB 10851

www.niemann-frey.de


From daniel_wells at byu.edu  Thu Jan 27 21:37:09 2011
From: daniel_wells at byu.edu (Daniel Wells)
Date: Thu, 27 Jan 2011 11:37:09 -0700
Subject: [Freeswitch-users] mod_dingaling and the weekly windows build
In-Reply-To: <1296150184550-5967186.post@n2.nabble.com>
References: <8C68232BC9314C40BBCDDAA480F7B01AEA3B493B34@harrow.exch.ad.byu.edu>
	<1296150184550-5967186.post@n2.nabble.com>
Message-ID: <8C68232BC9314C40BBCDDAA480F7B01AEA55E1F8B6@harrow.exch.ad.byu.edu>

So will mod_dingaling ever be included in the build provided by freeswitch (a final release for example) or is it always going to be something we can only add with a custom build?

Thanks for the response. 

    - Daniel Wells
 
 
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk
Sent: Thursday, January 27, 2011 10:43 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_dingaling and the weekly windows build


That dll is not part of the weekly dist.
If you want to experiment with that you will have to build the code yourself. You will also have to obtain and integrate GnuTLS into the build for that to work(Google Voice). There are some introductory docs on the Wiki that outline what may be required for doing this but keep in mind that this functionality is only experimental.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-dingaling-and-the-weekly-windows-build-tp5965119p5967186.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From miguel.mirandag at gmail.com  Thu Jan 27 21:44:05 2011
From: miguel.mirandag at gmail.com (Miguel Miranda)
Date: Thu, 27 Jan 2011 12:44:05 -0600
Subject: [Freeswitch-users] conference with did example
In-Reply-To: <AANLkTinq8y7nA3=U=xDetosgfzUvvnSaRwZt38u8+ksg@mail.gmail.com>
References: <AANLkTik-wYp99Mbw6==EnT21U5RGk_MLPCt+g-wA+nFs@mail.gmail.com>
	<AANLkTinsDpRV6p5uFp4WGiNeOupcL6TAfteoJ31K8Qf4@mail.gmail.com>
	<AANLkTik7mXLbTx2gGXDwx3EaKJaJTRofCqY+Bnyo6rnY@mail.gmail.com>
	<AANLkTinJP0irvvw_stiX=7GGxYsAt-R1kmY8=wMCLr7M@mail.gmail.com>
	<AANLkTinq8y7nA3=U=xDetosgfzUvvnSaRwZt38u8+ksg@mail.gmail.com>
Message-ID: <AANLkTinKFxZm+=VD785OVzFNdX=k8S=wsNjjDSE_yyiQ@mail.gmail.com>

Thanks Steve and all, that is exactly what i was looking for, beacuse of
that link all is clear now.

regards,

On Thu, Jan 27, 2011 at 11:56 AM, Steven Ayre <steveayre at gmail.com> wrote:

> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits
>
> On 27 January 2011 16:37, Miguel Miranda <miguel.mirandag at gmail.com>wrote:
>
>> Great, a few questions:
>>
>> 1) ${domain} is the  conference's profile name right?
>>
>
> You can replace that with whatever the profile is named
>
>
>> 2)what are the digits 4437000 ? the conference number?
>>
>
> Read the play_and_get_digits documentation on the wiki, that explains the
> syntax:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits
>
> Basically, they specify the number of digits, number of retries to allow
> and the timeout waiting for the number to be entered.
>
> 3) the sound files should be any regular.wav file, even one that i can
>> record using my laptop's mic, no special requeriments?
>>
>> thanks in advance
>
>
> Almost any sound format supported by sndfile will probably work. WAV files
> will use more space but less CPU (it's raw audio so FS won't need to
> decompress it).
>
>
>>
>>
>> On Wed, Jan 26, 2011 at 7:27 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> I don't believe this example will work because ${confnumber} won't be
>>> populated at the time the dialplan is parsed. However a trivial modification
>>> would make it work:
>>>
>>>   <extension name="ask user for conf number">
>>>     <condition field="destination_number" expression="^(9903)$">
>>>        <action application="answer"/>
>>>       <action application="play_and_get_digits"
>>>               data="4 4 3 7000 # conference/conf-enter_conf_number.wav
>>> ivr/ivr-that_was_an_invalid_entry.wav confnumber \d+"/>
>>>       <action application="playback"
>>> data="ivr/ivr-one_moment_please.wav"/>
>>>       <action application="set" data="ringback=${us-ring}"/>
>>>       <action application="transfer" data="USER_DIALED_${confnumber} XML
>>> default"/>
>>>     </condition>
>>>   </extension>
>>>
>>>   <extension name="put user into conf">
>>>     <condition field="destination_number"
>>> expression="^USER_DIALED_(\d+)$">
>>>       <action application="conference" data="$1@${domain}"/>
>>>     </condition>
>>>   </extension>
>>>
>>> Note that I tested this with real sound files on my system instead of the
>>> pretend ones that were there. I also used dest num of "9903" - use a value
>>> that works for you.
>>>
>>> -MC
>>>
>>> On Wed, Jan 26, 2011 at 3:57 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>>
>>>> Something like this:
>>>>
>>>> <extension name="conference">
>>>>    <condition field="destination_number" expression="^12345$">
>>>>       <action application="answer"/>
>>>>       <action application="play_and_get_digits" data="4 4 3 7000 #
>>>> $${base_dir}/sounds/conf-number.wav /invalid.wav confnumber \d+"/>
>>>>
>>>>       <action application="playback" data="connecting_your_call.wav"/>
>>>>       <action application="set" data="ringback=${us-ring}"/>
>>>>       <action application="conference" data="${confnumber}@
>>>> ${domain_name}"/>
>>>>    </condition>
>>>> </extension>
>>>>
>>>>
>>>>
>>>>
>>>> On 26 January 2011 22:10, Miguel Miranda <miguel.mirandag at gmail.com>wrote:
>>>>
>>>>> Hi, i have searched the arvhices and could not find a simple example on
>>>>> how to configure a DID for  mod_conference, i mean you dial an DID, an ivr
>>>>> responds "please enter you conference number and press puond key, etc", and
>>>>> if the conference room was configured with pin it asks "please enter ?pin
>>>>> number, etc".
>>>>> as i undestend, in this example from wiki:
>>>>>
>>>>> <extension name="test_bridging_conference">
>>>>>    <condition field="destination_number" expression="^(3000)$">
>>>>>       <action application="answer"/>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>       <action application="playback" data="connecting_your_call.wav"/>
>>>>>       <action application="set" data="ringback=${us-ring}"/>
>>>>>       <action application="conference" data="bridge:$1-${domain_name}@default:user/1000@${domain_name}"/>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>    </condition>
>>>>> </extension>
>>>>>
>>>>>
>>>>> i need a gateway and if you receive the DID 3000 you join the
>>>>> conference right?
>>>>> What i want is a single did for all the conferences.
>>>>> regards
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From frank at telonium.com  Thu Jan 27 21:53:14 2011
From: frank at telonium.com (Frank Park)
Date: Thu, 27 Jan 2011 13:53:14 -0500
Subject: [Freeswitch-users] Choppy VM
In-Reply-To: <20718.1296150890@ccs.covici.com>
References: <AANLkTimBXKiC2frKN3UBOC_c-gdfgk8N--k1xX2F7fqR@mail.gmail.com>
	<AANLkTinU6GsT3NMKJNQxr4_Etis=v3LaZhTmUZtaMsNa@mail.gmail.com>
	<20718.1296150890@ccs.covici.com>
Message-ID: <AANLkTikZhW1Q=bGx=nfCcP7xuvgjef_broOX6mVDe4jt@mail.gmail.com>

I'm using loopback.
Our production server is currently using a git from November and I should
play with another instance with new git-head. Was there any changes done
with voicemail or loopback mod?
How do I check if my kernel was compiled with 1000hz?

Sorry for somewhat of basic questions.

Frank


On Thu, Jan 27, 2011 at 12:54 PM, <covici at ccs.covici.com> wrote:

> I have experienced the choppy vm sometimes -- is there an alternative to
> using loopback and why should it make a difference?
>
> Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
> > have you tried the latest version of the code?
> > Are you using mod_loopback in your VM calls?
> >
> >
> > On Thu, Jan 27, 2011 at 11:38 AM, Frank Park <frank at telonium.com> wrote:
> > > Hey guys,
> > > I've been trying to troubleshoot this, but haven't been able to figure
> it
> > > out...
> > > I am currently experiencing choppy voicemail recordings on my
> freeswitch.
> > > The calls are coming in using SIP trunk and the call quality using the
> same
> > > trunk has no problem, but when I listen back the vm, that can be a
> different
> > > story.
> > > This is running on a dedicate server and not only a virtual instance. I
> am
> > > currently not using mod_shout, so this is being saved as a wav file.
> > > I've monitored CPU and memory usage while this happens, and because
> they are
> > > so low, I don't think resource exhaustion is a factor.
> > > Any direction as to what I should try to troubleshoot what's wrong?
> > > I appreciate any input.
> > > Thank you,
> > > Frank
> > >
> > > --
> > >
> > > ----=======================----
> > > Frank Park
> > > Telonium Communications, LLC
> > > frank at telonium.com
> > > http://www.telonium.com
> > > Follow Us on Twitter: @GetTelonium
> > > 404-566-8888 x1001 Office
> > > 404-939-4242 Cell
> > > ----=======================----
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

----=======================----
Frank Park
Telonium Communications, LLC
frank at telonium.com
http://www.telonium.com
Follow Us on Twitter: @GetTelonium
404-566-8888 x1001 Office
404-939-4242 Cell
----=======================----
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From david.ponzone at ipeva.fr  Thu Jan 27 22:07:21 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 27 Jan 2011 20:07:21 +0100
Subject: [Freeswitch-users] Choppy VM
In-Reply-To: <AANLkTikZhW1Q=bGx=nfCcP7xuvgjef_broOX6mVDe4jt@mail.gmail.com>
References: <AANLkTimBXKiC2frKN3UBOC_c-gdfgk8N--k1xX2F7fqR@mail.gmail.com>
	<AANLkTinU6GsT3NMKJNQxr4_Etis=v3LaZhTmUZtaMsNa@mail.gmail.com>
	<20718.1296150890@ccs.covici.com>
	<AANLkTikZhW1Q=bGx=nfCcP7xuvgjef_broOX6mVDe4jt@mail.gmail.com>
Message-ID: <934BB9A5-56CE-4EEB-85DC-27F84773D1C4@ipeva.fr>

get the kernel config file and do a grep 1000 in it :)

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 27/01/2011 ? 19:53, Frank Park a ?crit :

> I'm using loopback.
> Our production server is currently using a git from November and I should play with another instance with new git-head. Was there any changes done with voicemail or loopback mod?
> How do I check if my kernel was compiled with 1000hz?
> 
> Sorry for somewhat of basic questions.
> 
> Frank
> 
> 
> On Thu, Jan 27, 2011 at 12:54 PM, <covici at ccs.covici.com> wrote:
> I have experienced the choppy vm sometimes -- is there an alternative to
> using loopback and why should it make a difference?
> 
> Anthony Minessale <anthony.minessale at gmail.com> wrote:
> 
> > have you tried the latest version of the code?
> > Are you using mod_loopback in your VM calls?
> >
> >
> > On Thu, Jan 27, 2011 at 11:38 AM, Frank Park <frank at telonium.com> wrote:
> > > Hey guys,
> > > I've been trying to troubleshoot this, but haven't been able to figure it
> > > out...
> > > I am currently experiencing choppy voicemail recordings on my freeswitch.
> > > The calls are coming in using SIP trunk and the call quality using the same
> > > trunk has no problem, but when I listen back the vm, that can be a different
> > > story.
> > > This is running on a dedicate server and not only a virtual instance. I am
> > > currently not using mod_shout, so this is being saved as a wav file.
> > > I've monitored CPU and memory usage while this happens, and because they are
> > > so low, I don't think resource exhaustion is a factor.
> > > Any direction as to what I should try to troubleshoot what's wrong?
> > > I appreciate any input.
> > > Thank you,
> > > Frank
> > >
> > > --
> > >
> > > ----=======================----
> > > Frank Park
> > > Telonium Communications, LLC
> > > frank at telonium.com
> > > http://www.telonium.com
> > > Follow Us on Twitter: @GetTelonium
> > > 404-566-8888 x1001 Office
> > > 404-939-4242 Cell
> > > ----=======================----
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> > googletalk:conf+888 at conference.freeswitch.org
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>         John Covici
>         covici at ccs.covici.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> -- 
> 
> ----=======================----
> Frank Park
> Telonium Communications, LLC
> frank at telonium.com
> http://www.telonium.com
> Follow Us on Twitter: @GetTelonium
> 404-566-8888 x1001 Office
> 404-939-4242 Cell
> ----=======================----
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
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From anthony.minessale at gmail.com  Thu Jan 27 22:33:55 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Jan 2011 13:33:55 -0600
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTi=in3CLd-n0mT3SRKYA0kxLMGVWk9KeBx3_jhY6@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
	<AANLkTi=in3CLd-n0mT3SRKYA0kxLMGVWk9KeBx3_jhY6@mail.gmail.com>
Message-ID: <AANLkTin3Bu+8EEr_v_GaRGNPzOjfcnfzsV2Av2cgnWgO@mail.gmail.com>

try setting the variable presence_id to the blf entity you are
subscribing to (include the domain)


On Thu, Jan 27, 2011 at 12:32 PM, Patrick Plattes
<patrick.plattes at niemann-frey.info> wrote:
> Hi,
>
> if've pcaps and videos to show you our problem.
> http://kwixo.de/fs
>
> Thank you,
> ?Patrick.
>
> 2011/1/27 Michael Collins <msc at freeswitch.org>:
>> Could you humor us and get sip traces of the blf working on a reboot w/
>> asterisk and not working with freeswitch? Put them in
>> pastebin.freeswitch.org and drop the link in this thread.
>> Thanks,
>> MC
>>
>> On Wed, Jan 26, 2011 at 12:33 AM, Patrick Plattes
>> <patrick.plattes at niemann-frey.info> wrote:
>>>
>>> Hi List :-),
>>>
>>> I'm currently switching from Asterisk to FreeSWITCH. It's really hard
>>> work for an Asterisk user, but using Asterisk becomes more and more
>>> painful even for small installations (less than 100 sip users). I know
>>> FreeSWITCH is not a drop-in replacement for the Asterisk PBX, but I
>>> don't want to change the behaviour of the PBX for the users.
>>>
>>> It's a preconception in Germany that the American people like
>>> (especially for the X-mas time) kitsch. Everyone here know the
>>> American houses with a hole bunch of blinking lights. But those
>>> decorated hoses are nothing against our offices! Our phones have up to
>>> 136 lights (BLFs). You often have to wear sunglasses at the office ;-)
>>>
>>> A typical usage of BLFs is to check if an agent is a member of the
>>> queue. I've build a simple extension to add and delete a member. I can
>>> user "presence in" and "presence out" to enable or disable the BLF,
>>> but there is one big issue. After a reboot of the phone the the user
>>> is still a member of the queue, but the BLF is off. We are using hints
>>> at Asterisk to show the user if he is a member and it works even after
>>> a reboot - with "presence" at FreeSWITCH it works (of cause) not. Does
>>> anyone have an idea how to implement it?
>>>
>>> My current extension is just for testing and so I use mod_fifo. It
>>> shouldn't be a problem to use mod_callcenter. The phone calls
>>> "queue-the_name_of_the_queue-the_name_of_the_user at pbxdomain" eg.
>>> "queue-sales_de-1000 at freeswitch.cust"
>>>
>>> So how implement BLF persistence?
>>>
>>> Thanks,
>>> ?Patrick
>>>
>>>
>>> Colourized version: http://pastebin.com/ufJ7U930
>>>
>>>
>>>
>>> <extension name="Queue Management">
>>>
>>> ?<condition field="destination_number"
>>> expression="^queue-(\w+)-(\d+)$" break="on-false">
>>> ? <action application="set"
>>> data="queue_name=${regex(${destination_number}|^queue-(\w+)-(\d+)$|$1)}"
>>> inline="true"/>
>>> ? <action application="set"
>>> data="queue_user=${regex(${destination_number}|^queue-(\w+)-(\d+)$|$2)}"
>>> inline="true"/>
>>> ?</condition>
>>>
>>> ?<condition field="${fifo(list ${queue_name}@${domain_name})}"
>>> expression="${queue_user}">
>>> ? <action application="answer"/>
>>> ? <action application="set" data="result=${fifo_member(del
>>> ${queue_name}@${domain_name}
>>> {fifo_member_wait=nowait}user/${queue_user} )}"/>
>>> ? <action application="log" data="INFO Delete FIFO agent result:
>>> ${result}"/>
>>> ? <action application="log" data="INFO User Logout: ${user_name}"/>
>>> ? <action application="sleep" data="600"/>
>>> ? <action application="playback"
>>> data="ivr/ivr-you_are_now_logged_out.wav"/>
>>> ? <action application="set" data="result=${presence(out
>>> queue-${queue_name}-${queue_user}@${domain_name} inuse inuse)}"/>
>>> ? <action application="log" data="INFO Presence OUT ${result}"/>
>>> ? <action application="hangup"/>
>>>
>>> ? <anti-action application="answer"/>
>>> ? <anti-action application="set" data="result=${fifo_member(add
>>> ${queue_name}@${domain_name}
>>> {fifo_member_wait=nowait}user/${queue_user} )}"/>
>>> ? <anti-action application="log" data="INFO Add FIFO agent result:
>>> ${result}"/>
>>> ? <anti-action application="log" data="INFO User Login: ${user_name}"/>
>>> ? <anti-action application="sleep" data="600"/>
>>> ? <anti-action application="playback"
>>> data="ivr/ivr-you_are_now_logged_in.wav"/>
>>> ? <anti-action application="set" data="result=${presence(in
>>> queue-${queue_name}-${queue_user}@${domain_name} inuse inuse)}"/>
>>> ? <anti-action application="log" data="INFO Presence IN ${result}"/>
>>> ? <anti-action application="hangup"/>
>>> ?</condition>
>>> </extension>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Patrick Plattes
> IT - Projektleiter
>
> Niemann + Frey GmbH
> Adolf-Dembach-Str. 24
> 47829 Krefeld
>
> Tel.: +49/2151 - 5554-263
> Fax : +49/2151 - 5554-123
> patrick.plattes at niemann-frey.info
>
> Gesch?ftsf?hrer: Gerd Frey
> Sitz und Registergericht: Krefeld HRB 10851
>
> www.niemann-frey.de
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Thu Jan 27 22:22:17 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Jan 2011 13:22:17 -0600
Subject: [Freeswitch-users] mod_dingaling answering prematurely
In-Reply-To: <AANLkTi=3nR1ne6Un1XTrLUkfVxh1HwMjJCJbTaZb3XEE@mail.gmail.com>
References: <AANLkTi=3nR1ne6Un1XTrLUkfVxh1HwMjJCJbTaZb3XEE@mail.gmail.com>
Message-ID: <AANLkTintUoqhCD24ceDZMWhG4Wnew_=x04Yt+V9TSFgz@mail.gmail.com>

That is because there is no concept of answer or starting media etc.
we must answer right away or they will give up very quickly when we
don't respond.
Jingle is not a telephone protocol so there is no 180 ringing
equivalent otherwise we could do it.



On Thu, Jan 27, 2011 at 12:08 AM, Jon Young <jonyoung111 at gmail.com> wrote:
> I just got mod_dingaling working for outbound and inbound calls.
> However, on inbound calls I have to answer it quickly on my SIP phone.
> ?When I do I catch the call screening announcement from google to
> press a "1" to answer. ?I put the "execute_on_answer=send_dtmf 1" as
> recommended in the Wiki which eliminates the need to press one. ?It
> seems that FS is answering the call and google starts playing the
> screening. ?If I wait too long to answer I miss the announcement.
>
> I am routing the call to a specific extension for testing. ?The
> dialplan is using the extension definitions provided in the default
> configuration. ?Where should I start looking to allow the phones to
> ring longer and not prematurely answer the inbound call?
>
> Thanks,
>
> Jon
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Thu Jan 27 22:17:10 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Jan 2011 13:17:10 -0600
Subject: [Freeswitch-users] Choppy VM
In-Reply-To: <934BB9A5-56CE-4EEB-85DC-27F84773D1C4@ipeva.fr>
References: <AANLkTimBXKiC2frKN3UBOC_c-gdfgk8N--k1xX2F7fqR@mail.gmail.com>
	<AANLkTinU6GsT3NMKJNQxr4_Etis=v3LaZhTmUZtaMsNa@mail.gmail.com>
	<20718.1296150890@ccs.covici.com>
	<AANLkTikZhW1Q=bGx=nfCcP7xuvgjef_broOX6mVDe4jt@mail.gmail.com>
	<934BB9A5-56CE-4EEB-85DC-27F84773D1C4@ipeva.fr>
Message-ID: <AANLkTikEgEw6Voj-ZatmaUR0vT4P-E2ekNrwdGdr09ev@mail.gmail.com>

I did a patch this week that improved audio quality when using
loopback talking to an app.
I would try that first.



On Thu, Jan 27, 2011 at 1:07 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
> get the kernel config file and do a grep 1000 in it :)
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 27/01/2011 ? 19:53, Frank Park a ?crit :
>
> I'm using loopback.
> Our production server is currently using a git from November and I should
> play with another instance with new git-head. Was there any changes done
> with voicemail or loopback mod?
> How do I check if my kernel was compiled with 1000hz?
> Sorry for somewhat of basic questions.
> Frank
>
> On Thu, Jan 27, 2011 at 12:54 PM, <covici at ccs.covici.com> wrote:
>>
>> I have experienced the choppy vm sometimes -- is there an alternative to
>> using loopback and why should it make a difference?
>>
>> Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>
>> > have you tried the latest version of the code?
>> > Are you using mod_loopback in your VM calls?
>> >
>> >
>> > On Thu, Jan 27, 2011 at 11:38 AM, Frank Park <frank at telonium.com> wrote:
>> > > Hey guys,
>> > > I've been trying to troubleshoot this, but haven't been able to figure
>> > > it
>> > > out...
>> > > I am currently experiencing choppy voicemail recordings on my
>> > > freeswitch.
>> > > The calls are coming in using SIP trunk and the call quality using the
>> > > same
>> > > trunk has no problem, but when I listen back the vm, that can be a
>> > > different
>> > > story.
>> > > This is running on a dedicate server and not only a virtual instance.
>> > > I am
>> > > currently not using mod_shout, so this is being saved as a wav file.
>> > > I've monitored CPU and memory usage while this happens, and because
>> > > they are
>> > > so low, I don't think resource?exhaustion?is a factor.
>> > > Any direction as to what I should try to troubleshoot what's wrong?
>> > > I appreciate any input.
>> > > Thank you,
>> > > Frank
>> > >
>> > > --
>> > >
>> > > ----=======================----
>> > > Frank Park
>> > > Telonium Communications, LLC
>> > > frank at telonium.com
>> > > http://www.telonium.com
>> > > Follow Us on Twitter: @GetTelonium
>> > > 404-566-8888 x1001 Office
>> > > 404-939-4242 Cell
>> > > ----=======================----
>> > >
>> > >
>> > > _______________________________________________
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > >
>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> > >
>> > >
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> --
>> Your life is like a penny. ?You're going to lose it. ?The question is:
>> How do
>> you spend it?
>>
>> ? ? ? ? John Covici
>> ? ? ? ? covici at ccs.covici.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
>
> ----=======================----
> Frank Park
> Telonium Communications, LLC
> frank at telonium.com
> http://www.telonium.com
> Follow Us on Twitter: @GetTelonium
> 404-566-8888 x1001 Office
> 404-939-4242 Cell
> ----=======================----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From jonyoung111 at gmail.com  Thu Jan 27 22:43:53 2011
From: jonyoung111 at gmail.com (Jon Young)
Date: Thu, 27 Jan 2011 12:43:53 -0700
Subject: [Freeswitch-users] mod_dingaling answering prematurely
In-Reply-To: <AANLkTintUoqhCD24ceDZMWhG4Wnew_=x04Yt+V9TSFgz@mail.gmail.com>
References: <AANLkTi=3nR1ne6Un1XTrLUkfVxh1HwMjJCJbTaZb3XEE@mail.gmail.com>
	<AANLkTintUoqhCD24ceDZMWhG4Wnew_=x04Yt+V9TSFgz@mail.gmail.com>
Message-ID: <AANLkTi=SJeXp0DszimNBcRdnkCMvVFZ3tbi50CFnXqMS@mail.gmail.com>

I see, thanks.  So my best option would be to send the incoming call
to an IVR/Auto Attendant and send the DTMF "1"?

Or is there a way to trigger sending the "1" without waiting for me to
pick up a handset?

Jon

On Thu, Jan 27, 2011 at 12:22 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> That is because there is no concept of answer or starting media etc.
> we must answer right away or they will give up very quickly when we
> don't respond.
> Jingle is not a telephone protocol so there is no 180 ringing
> equivalent otherwise we could do it.
>
>
>
> On Thu, Jan 27, 2011 at 12:08 AM, Jon Young <jonyoung111 at gmail.com> wrote:
>> I just got mod_dingaling working for outbound and inbound calls.
>> However, on inbound calls I have to answer it quickly on my SIP phone.
>> ?When I do I catch the call screening announcement from google to
>> press a "1" to answer. ?I put the "execute_on_answer=send_dtmf 1" as
>> recommended in the Wiki which eliminates the need to press one. ?It
>> seems that FS is answering the call and google starts playing the
>> screening. ?If I wait too long to answer I miss the announcement.
>>
>> I am routing the call to a specific extension for testing. ?The
>> dialplan is using the extension definitions provided in the default
>> configuration. ?Where should I start looking to allow the phones to
>> ring longer and not prematurely answer the inbound call?
>>
>> Thanks,
>>
>> Jon
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From frank at telonium.com  Thu Jan 27 22:56:15 2011
From: frank at telonium.com (Frank Park)
Date: Thu, 27 Jan 2011 14:56:15 -0500
Subject: [Freeswitch-users] Choppy VM
In-Reply-To: <AANLkTikEgEw6Voj-ZatmaUR0vT4P-E2ekNrwdGdr09ev@mail.gmail.com>
References: <AANLkTimBXKiC2frKN3UBOC_c-gdfgk8N--k1xX2F7fqR@mail.gmail.com>
	<AANLkTinU6GsT3NMKJNQxr4_Etis=v3LaZhTmUZtaMsNa@mail.gmail.com>
	<20718.1296150890@ccs.covici.com>
	<AANLkTikZhW1Q=bGx=nfCcP7xuvgjef_broOX6mVDe4jt@mail.gmail.com>
	<934BB9A5-56CE-4EEB-85DC-27F84773D1C4@ipeva.fr>
	<AANLkTikEgEw6Voj-ZatmaUR0vT4P-E2ekNrwdGdr09ev@mail.gmail.com>
Message-ID: <AANLkTikBD-MAGT8S-msd+njSoOFWLGfYTmy1k3P81ykC@mail.gmail.com>

David & Anthony,

Thanks, I will look at the config file and see if I can update the loopback
mod to the latest one to see if that helps.
I'll keep you guys posted on the result in few days.

Frank


On Thu, Jan 27, 2011 at 2:17 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I did a patch this week that improved audio quality when using
> loopback talking to an app.
> I would try that first.
>
>
>
> On Thu, Jan 27, 2011 at 1:07 PM, David Ponzone <david.ponzone at ipeva.fr>
> wrote:
> > get the kernel config file and do a grep 1000 in it :)
> > David Ponzone  Direction Technique
> > email: david.ponzone at ipeva.fr
> > tel:      01 74 03 18 97
> > gsm:   06 66 98 76 34
> > Service Client IPeva
> > tel:      0811 46 26 26
> > www.ipeva.fr  -   www.ipeva-studio.com
> > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> > l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion
> > non autoris?e est interdite. Tout message ?lectronique est susceptible
> > d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il
> > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> >
> >
> >
> > Le 27/01/2011 ? 19:53, Frank Park a ?crit :
> >
> > I'm using loopback.
> > Our production server is currently using a git from November and I should
> > play with another instance with new git-head. Was there any changes done
> > with voicemail or loopback mod?
> > How do I check if my kernel was compiled with 1000hz?
> > Sorry for somewhat of basic questions.
> > Frank
> >
> > On Thu, Jan 27, 2011 at 12:54 PM, <covici at ccs.covici.com> wrote:
> >>
> >> I have experienced the choppy vm sometimes -- is there an alternative to
> >> using loopback and why should it make a difference?
> >>
> >> Anthony Minessale <anthony.minessale at gmail.com> wrote:
> >>
> >> > have you tried the latest version of the code?
> >> > Are you using mod_loopback in your VM calls?
> >> >
> >> >
> >> > On Thu, Jan 27, 2011 at 11:38 AM, Frank Park <frank at telonium.com>
> wrote:
> >> > > Hey guys,
> >> > > I've been trying to troubleshoot this, but haven't been able to
> figure
> >> > > it
> >> > > out...
> >> > > I am currently experiencing choppy voicemail recordings on my
> >> > > freeswitch.
> >> > > The calls are coming in using SIP trunk and the call quality using
> the
> >> > > same
> >> > > trunk has no problem, but when I listen back the vm, that can be a
> >> > > different
> >> > > story.
> >> > > This is running on a dedicate server and not only a virtual
> instance.
> >> > > I am
> >> > > currently not using mod_shout, so this is being saved as a wav file.
> >> > > I've monitored CPU and memory usage while this happens, and because
> >> > > they are
> >> > > so low, I don't think resource exhaustion is a factor.
> >> > > Any direction as to what I should try to troubleshoot what's wrong?
> >> > > I appreciate any input.
> >> > > Thank you,
> >> > > Frank
> >> > >
> >> > > --
> >> > >
> >> > > ----=======================----
> >> > > Frank Park
> >> > > Telonium Communications, LLC
> >> > > frank at telonium.com
> >> > > http://www.telonium.com
> >> > > Follow Us on Twitter: @GetTelonium
> >> > > 404-566-8888 x1001 Office
> >> > > 404-939-4242 Cell
> >> > > ----=======================----
> >> > >
> >> > >
> >> > > _______________________________________________
> >> > > FreeSWITCH-users mailing list
> >> > > FreeSWITCH-users at lists.freeswitch.org
> >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > >
> >> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > > http://www.freeswitch.org
> >> > >
> >> > >
> >> >
> >> >
> >> >
> >> > --
> >> > Anthony Minessale II
> >> >
> >> > FreeSWITCH http://www.freeswitch.org/
> >> > ClueCon http://www.cluecon.com/
> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >
> >> > AIM: anthm
> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> > pstn:+19193869900
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >>
> >> --
> >> Your life is like a penny.  You're going to lose it.  The question is:
> >> How do
> >> you spend it?
> >>
> >>         John Covici
> >>         covici at ccs.covici.com
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> >
> > ----=======================----
> > Frank Park
> > Telonium Communications, LLC
> > frank at telonium.com
> > http://www.telonium.com
> > Follow Us on Twitter: @GetTelonium
> > 404-566-8888 x1001 Office
> > 404-939-4242 Cell
> > ----=======================----
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

----=======================----
Frank Park
Telonium Communications, LLC
frank at telonium.com
http://www.telonium.com
Follow Us on Twitter: @GetTelonium
404-566-8888 x1001 Office
404-939-4242 Cell
----=======================----
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From hesser4900 at gmail.com  Fri Jan 28 03:16:25 2011
From: hesser4900 at gmail.com (Holger Esser)
Date: Thu, 27 Jan 2011 18:16:25 -0600
Subject: [Freeswitch-users] TBCT 2bchannel transfer
Message-ID: <AANLkTinaZCrohy_RLoxFifSvRbzbXUfrhJcCRv61dsV5@mail.gmail.com>

Hi, does anybody have a detailed description of how to implement the TBCT
transfer method with Sangoma cards?
I was looking at the att_xfer but I am not sure if that would cover TBCT.

Any suggestions would be greatly appreciated.

Holger
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From hesser4900 at gmail.com  Fri Jan 28 03:19:01 2011
From: hesser4900 at gmail.com (Holger Esser)
Date: Thu, 27 Jan 2011 18:19:01 -0600
Subject: [Freeswitch-users] supported say languages
Message-ID: <AANLkTimgf5N2K5yTa4T0pZV+J0GCB2_pwrMOwjfZ3noy@mail.gmail.com>

Hi, is there a list of all the supported say (mod_say_xx) modules? I am
looking at my build and so far I am seeing Russian and English. Are there
more?

Kind regards,
Holger
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From dujinfang at gmail.com  Fri Jan 28 03:51:25 2011
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 28 Jan 2011 08:51:25 +0800
Subject: [Freeswitch-users] question about outbound socket async
Message-ID: <AANLkTinMmHpqbk2zF87XBiEjNQ+pg2ADmZLFjzVFz2dd@mail.gmail.com>

Hi all,

I'm using mod_erlang_event and I think it's equivalent to event_socket
async full.

When I run sendmsg playback wav1 and then playback wav2, wav2 will
stop wav1 and when wav2 is done, wav1 continues by default. Is there a
way to automatically break wav1 when play wav2 ?

I could send break to wav1 but because everything is async, sometimes
break also breaks wav2 even with event-lock = true which made async
mode a sort of pain.

It seems like there is a var can control this but I cannot recall. Any
help is appreciated.

Thanks.

-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From cmrienzo at gmail.com  Fri Jan 28 03:55:27 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Thu, 27 Jan 2011 19:55:27 -0500
Subject: [Freeswitch-users] supported say languages
In-Reply-To: <AANLkTimgf5N2K5yTa4T0pZV+J0GCB2_pwrMOwjfZ3noy@mail.gmail.com>
References: <AANLkTimgf5N2K5yTa4T0pZV+J0GCB2_pwrMOwjfZ3noy@mail.gmail.com>
Message-ID: <AANLkTi=S1cdsB7EiyV-ZirO1KpzQWPt7Lthn-+MBf3J0@mail.gmail.com>

$ ls src/mod/say
mod_say_de  mod_say_en  mod_say_es  mod_say_fr  mod_say_hr  mod_say_hu
mod_say_it  mod_say_ja  mod_say_nl  mod_say_pt  mod_say_ru  mod_say_th
mod_say_zh



On Thu, Jan 27, 2011 at 7:19 PM, Holger Esser <hesser4900 at gmail.com> wrote:

> Hi, is there a list of all the supported say (mod_say_xx) modules? I am
> looking at my build and so far I am seeing Russian and English. Are there
> more?
>
> Kind regards,
> Holger
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From djbinter at gmail.com  Fri Jan 28 04:13:08 2011
From: djbinter at gmail.com (DJB International)
Date: Thu, 27 Jan 2011 17:13:08 -0800
Subject: [Freeswitch-users] Question about IVR Menu
Message-ID: <AANLkTikr_hoJuZb-QS2PQomJefYc94Vio+z5oyLJ-bD=@mail.gmail.com>

In IVR menu, I have: <entry action="menu-play-sound" digits="2"
param="voicemail/company_directory.wav"/> to play company directory when the
caller press 2,
but the caller have to wait until the wave file finished playing to the end
of the file before he can enter any digits .

Is there a way to enter digits (extension number) at any time without
waiting for the wave file to finish playing?

Thank you.
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From msc at freeswitch.org  Fri Jan 28 04:38:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Jan 2011 17:38:36 -0800
Subject: [Freeswitch-users] network scenario doubts
In-Reply-To: <4D41803C.1050601@gmail.com>
References: <4D3FD3B0.4080309@gmail.com>
	<AANLkTinBd00iKLysi1O97hOupQpYdzBchBN5t8oZOJre@mail.gmail.com>
	<4D41803C.1050601@gmail.com>
Message-ID: <AANLkTi=d+7Yxd6CKD-NyuW6ionAAxbgmBy2StMJO=1K7@mail.gmail.com>

For a test, stop iptables and make a call. If audio still isn't flowing then
you know something other than iptables is interfering. I would get a capture
of the SIP traffic and see what it looks like. Make sure that there aren't
any SIP ALGs in the path that might be re-writing port numbers.

-MC

On Thu, Jan 27, 2011 at 6:25 AM, Michal Kopacki <mkopacki at gmail.com> wrote:

>      Everything is set as you wrote and I'm able to login (using domain
> name). Now I'm facing a media problem during outside call. I'm logged to FS
> from ip phone (outside lan) and trying to dial 5000 (default ivr). Call is
> connected but i'm not able to hear anything.
>
>    There is iptable based firewall with opened udp ports 16384:32768, but
> it seems nothing even hit those ports:
>
> Chain INPUT (policy DROP 0 packets, 0 bytes)
>  pkts bytes target     prot opt in     out     source
> destination
>     0     0 ACCEPT     udp  --  eth0   any     anywhere
> anywhere            udp dpts:connected:filenet-tms state NEW
>
>   Any thoughts ?
> --
> Best regards,
> Michal
>
>
> On 01/27/2011 01:39 AM, Michael Collins wrote:
>
> yes, you should be able to handle this scenario. You probably just need to
> set your external profile to use port 5060 instead of 5080. Most people only
> have a single NIC so their sofia profiles need to be on different ports. (A
> sofia profile is a SIP user agent (UA) that can listen/respond to SIP
> messages on an IP address and port.)
>
>  Look in external.xml for these lines:
>      <param name="rtp-ip" value="$${local_ip_v4}"/>
>     <param name="sip-ip" value="$${local_ip_v4}"/>
>
>  If you have a static IP address (which it appears you do) then put that
> ip addr in there in place of the variable. Then look at the end of vars.xml
> and you'll see where the external profile's port is set to 5080/5081. Change
> it to 5060/5061 and then restart FS.
>
>  Let us know how it goes.
>
>  -MC
>
> On Tue, Jan 25, 2011 at 11:56 PM, Michal Kopacki <mkopacki at gmail.com>wrote:
>
>>       Hello,
>>
>>   This is my first post to this list, so hello everyone.
>>
>>   I'm at the very beginning of freeswitch journey and i have a problem
>> with fit FS to my network scenario.
>>
>> OS: fedora 13 (x86_64)
>> FS: 1.0.7 (compiled from sources), default config
>>
>> Desired scenario:
>>
>> softphone -> mydomain.com (with ip 193.59.72.xx) -> my isp network ->
>> 10.25.48.xx on eth1 -> FS -> 192.168.0.1 on eth0 -> softphone
>>
>> And now, I'm able to connect to FS from my local network, but I'm not
>> able to connect from outside (neither domain nor ip). In internal.xml I
>> set internal ip of server and i wanted to set external ip in
>> external.xml, but there is a question: which one ?  In case of
>> 193.59.72.xx external profile didnt' start and with 10.25.48.xx outside
>> softphone didn't register.  I checked with netstat and realized that
>> port 5060 is bind to internal nic only and 5080 to external nic
>> (10.25.48.xx) and I have no idea what next.
>>
>> Is it even possible to work with such network scenario ? I would be
>> grateful for pointing me to right direction or maybe propose different
>> approach.
>>
>> --
>> Best regards,
>> Michal
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri Jan 28 04:41:17 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Jan 2011 17:41:17 -0800
Subject: [Freeswitch-users] supported say languages
In-Reply-To: <AANLkTi=S1cdsB7EiyV-ZirO1KpzQWPt7Lthn-+MBf3J0@mail.gmail.com>
References: <AANLkTimgf5N2K5yTa4T0pZV+J0GCB2_pwrMOwjfZ3noy@mail.gmail.com>
	<AANLkTi=S1cdsB7EiyV-ZirO1KpzQWPt7Lthn-+MBf3J0@mail.gmail.com>
Message-ID: <AANLkTims8XYAnNbqL0vTt9NbZ_5RpB2UZT7EK-4PnfO5@mail.gmail.com>

That being said, we only have sound files in our repo for en and ru. If you
want fr then talk to our buddy Moc. He's got Canadian fr. I hear rumors that
there are also sound file sets for de and nl. We are working on getting some
es sounds.

If anyone has a sound set that they'd like to have us host let me know and
we'll see about getting your sound files up on files.freeswitch.org.

Thanks,
MC

On Thu, Jan 27, 2011 at 4:55 PM, Christopher Rienzo <cmrienzo at gmail.com>wrote:

> $ ls src/mod/say
> mod_say_de  mod_say_en  mod_say_es  mod_say_fr  mod_say_hr  mod_say_hu
> mod_say_it  mod_say_ja  mod_say_nl  mod_say_pt  mod_say_ru  mod_say_th
> mod_say_zh
>
>
>
> On Thu, Jan 27, 2011 at 7:19 PM, Holger Esser <hesser4900 at gmail.com>wrote:
>
>> Hi, is there a list of all the supported say (mod_say_xx) modules? I am
>> looking at my build and so far I am seeing Russian and English. Are there
>> more?
>>
>> Kind regards,
>> Holger
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri Jan 28 04:42:52 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Jan 2011 17:42:52 -0800
Subject: [Freeswitch-users] mod_dingaling answering prematurely
In-Reply-To: <AANLkTi=SJeXp0DszimNBcRdnkCMvVFZ3tbi50CFnXqMS@mail.gmail.com>
References: <AANLkTi=3nR1ne6Un1XTrLUkfVxh1HwMjJCJbTaZb3XEE@mail.gmail.com>
	<AANLkTintUoqhCD24ceDZMWhG4Wnew_=x04Yt+V9TSFgz@mail.gmail.com>
	<AANLkTi=SJeXp0DszimNBcRdnkCMvVFZ3tbi50CFnXqMS@mail.gmail.com>
Message-ID: <AANLkTinXjdxZfimpAsrNrCNJS+Bk9fsf7Z5Um8ADgsDT@mail.gmail.com>

You can answer the call, send the one, then bridge it to your extension.
It's wonky, but then again, so is GV.

-MC

On Thu, Jan 27, 2011 at 11:43 AM, Jon Young <jonyoung111 at gmail.com> wrote:

> I see, thanks.  So my best option would be to send the incoming call
> to an IVR/Auto Attendant and send the DTMF "1"?
>
> Or is there a way to trigger sending the "1" without waiting for me to
> pick up a handset?
>
> Jon
>
> On Thu, Jan 27, 2011 at 12:22 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > That is because there is no concept of answer or starting media etc.
> > we must answer right away or they will give up very quickly when we
> > don't respond.
> > Jingle is not a telephone protocol so there is no 180 ringing
> > equivalent otherwise we could do it.
> >
> >
> >
> > On Thu, Jan 27, 2011 at 12:08 AM, Jon Young <jonyoung111 at gmail.com>
> wrote:
> >> I just got mod_dingaling working for outbound and inbound calls.
> >> However, on inbound calls I have to answer it quickly on my SIP phone.
> >>  When I do I catch the call screening announcement from google to
> >> press a "1" to answer.  I put the "execute_on_answer=send_dtmf 1" as
> >> recommended in the Wiki which eliminates the need to press one.  It
> >> seems that FS is answering the call and google starts playing the
> >> screening.  If I wait too long to answer I miss the announcement.
> >>
> >> I am routing the call to a specific extension for testing.  The
> >> dialplan is using the extension definitions provided in the default
> >> configuration.  Where should I start looking to allow the phones to
> >> ring longer and not prematurely answer the inbound call?
> >>
> >> Thanks,
> >>
> >> Jon
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Fri Jan 28 04:48:43 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Jan 2011 17:48:43 -0800
Subject: [Freeswitch-users] Question about IVR Menu
In-Reply-To: <AANLkTikr_hoJuZb-QS2PQomJefYc94Vio+z5oyLJ-bD=@mail.gmail.com>
References: <AANLkTikr_hoJuZb-QS2PQomJefYc94Vio+z5oyLJ-bD=@mail.gmail.com>
Message-ID: <AANLkTimQAVOLk7vE7uNJniGWKjJC3h9boOq6ZQzv52mv@mail.gmail.com>

Have the action be "menu-sub" and then create a sub menu whose greeting is
the company_directory.wav file. Then you can define a menu-exec-app with
digits="/^(10\d\d)$/" or whatever your extension numbers are and then have
param="transfer $1 XML features". Just be sure to modify features.xml to
have the correct regex in the please_hold extension!

-MC

On Thu, Jan 27, 2011 at 5:13 PM, DJB International <djbinter at gmail.com>wrote:

> In IVR menu, I have: <entry action="menu-play-sound" digits="2"
> param="voicemail/company_directory.wav"/> to play company directory when the
> caller press 2,
> but the caller have to wait until the wave file finished playing to the end
> of the file before he can enter any digits .
>
> Is there a way to enter digits (extension number) at any time without
> waiting for the wave file to finish playing?
>
> Thank you.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From hesser4900 at gmail.com  Fri Jan 28 04:55:05 2011
From: hesser4900 at gmail.com (Holger Esser)
Date: Thu, 27 Jan 2011 19:55:05 -0600
Subject: [Freeswitch-users] supported say languages
In-Reply-To: <AANLkTims8XYAnNbqL0vTt9NbZ_5RpB2UZT7EK-4PnfO5@mail.gmail.com>
References: <AANLkTimgf5N2K5yTa4T0pZV+J0GCB2_pwrMOwjfZ3noy@mail.gmail.com>
	<AANLkTi=S1cdsB7EiyV-ZirO1KpzQWPt7Lthn-+MBf3J0@mail.gmail.com>
	<AANLkTims8XYAnNbqL0vTt9NbZ_5RpB2UZT7EK-4PnfO5@mail.gmail.com>
Message-ID: <AANLkTimu4jQYek6ayUZh647DnjiERJRK8wa9vQ6ZG0Xu@mail.gmail.com>

Many thanks guys! As always....

On Thu, Jan 27, 2011 at 7:41 PM, Michael Collins <msc at freeswitch.org> wrote:

> That being said, we only have sound files in our repo for en and ru. If you
> want fr then talk to our buddy Moc. He's got Canadian fr. I hear rumors that
> there are also sound file sets for de and nl. We are working on getting some
> es sounds.
>
> If anyone has a sound set that they'd like to have us host let me know and
> we'll see about getting your sound files up on files.freeswitch.org.
>
> Thanks,
> MC
>
> On Thu, Jan 27, 2011 at 4:55 PM, Christopher Rienzo <cmrienzo at gmail.com>wrote:
>
>> $ ls src/mod/say
>> mod_say_de  mod_say_en  mod_say_es  mod_say_fr  mod_say_hr  mod_say_hu
>> mod_say_it  mod_say_ja  mod_say_nl  mod_say_pt  mod_say_ru  mod_say_th
>> mod_say_zh
>>
>>
>>
>> On Thu, Jan 27, 2011 at 7:19 PM, Holger Esser <hesser4900 at gmail.com>wrote:
>>
>>> Hi, is there a list of all the supported say (mod_say_xx) modules? I am
>>> looking at my build and so far I am seeing Russian and English. Are there
>>> more?
>>>
>>> Kind regards,
>>> Holger
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peter.olsson at visionutveckling.se  Fri Jan 28 05:08:50 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 28 Jan 2011 03:08:50 +0100
Subject: [Freeswitch-users] question about outbound socket async
Message-ID: <CE77FAA7-A13A-40BF-AD6E-7E0E10807AE3@visionutveckling.se>

Are you using sendmsg break when issuing a break? That forces it to be queued in the execute queue the same way as playback, and it should be impossible to break the second file when not supposed to (since that should be the next one in queue).

/Peter


----- Reply message -----
Fr?n: "Seven Du" <dujinfang at gmail.com>
Datum: fre, jan 28, 2011 07:59
Rubrik: [Freeswitch-users] question about outbound socket async
Till: "freeswitch-users" <freeswitch-users at lists.freeswitch.org>

Hi all,

I'm using mod_erlang_event and I think it's equivalent to event_socket
async full.

When I run sendmsg playback wav1 and then playback wav2, wav2 will
stop wav1 and when wav2 is done, wav1 continues by default. Is there a
way to automatically break wav1 when play wav2 ?

I could send break to wav1 but because everything is async, sometimes
break also breaks wav2 even with event-lock = true which made async
mode a sort of pain.

It seems like there is a var can control this but I cannot recall. Any
help is appreciated.

Thanks.

--
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4d42149232767334012844!


From djbinter at gmail.com  Fri Jan 28 05:32:59 2011
From: djbinter at gmail.com (DJB International)
Date: Thu, 27 Jan 2011 18:32:59 -0800
Subject: [Freeswitch-users] Question about IVR Menu
In-Reply-To: <AANLkTimQAVOLk7vE7uNJniGWKjJC3h9boOq6ZQzv52mv@mail.gmail.com>
References: <AANLkTikr_hoJuZb-QS2PQomJefYc94Vio+z5oyLJ-bD=@mail.gmail.com>
	<AANLkTimQAVOLk7vE7uNJniGWKjJC3h9boOq6ZQzv52mv@mail.gmail.com>
Message-ID: <AANLkTims8xKn4vyBzBvguKiTVtD2G0ppYL5Xzv19ONT8@mail.gmail.com>

Michael,

Thank you very much.  Your method is working great.

-djbinter

On Thu, Jan 27, 2011 at 5:48 PM, Michael Collins <msc at freeswitch.org> wrote:

> Have the action be "menu-sub" and then create a sub menu whose greeting is
> the company_directory.wav file. Then you can define a menu-exec-app with
> digits="/^(10\d\d)$/" or whatever your extension numbers are and then have
> param="transfer $1 XML features". Just be sure to modify features.xml to
> have the correct regex in the please_hold extension!
>
> -MC
>
> On Thu, Jan 27, 2011 at 5:13 PM, DJB International <djbinter at gmail.com>wrote:
>
>> In IVR menu, I have: <entry action="menu-play-sound" digits="2"
>> param="voicemail/company_directory.wav"/> to play company directory when the
>> caller press 2,
>> but the caller have to wait until the wave file finished playing to the
>> end of the file before he can enter any digits .
>>
>> Is there a way to enter digits (extension number) at any time without
>> waiting for the wave file to finish playing?
>>
>> Thank you.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Fri Jan 28 05:37:43 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 27 Jan 2011 23:37:43 -0300
Subject: [Freeswitch-users] supported say languages
In-Reply-To: <AANLkTimu4jQYek6ayUZh647DnjiERJRK8wa9vQ6ZG0Xu@mail.gmail.com>
References: <AANLkTimgf5N2K5yTa4T0pZV+J0GCB2_pwrMOwjfZ3noy@mail.gmail.com>
	<AANLkTi=S1cdsB7EiyV-ZirO1KpzQWPt7Lthn-+MBf3J0@mail.gmail.com>
	<AANLkTims8XYAnNbqL0vTt9NbZ_5RpB2UZT7EK-4PnfO5@mail.gmail.com>
	<AANLkTimu4jQYek6ayUZh647DnjiERJRK8wa9vQ6ZG0Xu@mail.gmail.com>
Message-ID: <AANLkTim_O0J25jT_tdpdiQkj4ocrKrmAQm7GM7Cgrmdg@mail.gmail.com>

I have a company that might be interested in sponsoring the recording of es
sounds... I am still negotiating with them, but there might be some
surprises soon.

Regards,
Jo?o Mesquita


On Thu, Jan 27, 2011 at 10:55 PM, Holger Esser <hesser4900 at gmail.com> wrote:

> Many thanks guys! As always....
>
>
> On Thu, Jan 27, 2011 at 7:41 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> That being said, we only have sound files in our repo for en and ru. If
>> you want fr then talk to our buddy Moc. He's got Canadian fr. I hear rumors
>> that there are also sound file sets for de and nl. We are working on getting
>> some es sounds.
>>
>> If anyone has a sound set that they'd like to have us host let me know and
>> we'll see about getting your sound files up on files.freeswitch.org.
>>
>> Thanks,
>> MC
>>
>> On Thu, Jan 27, 2011 at 4:55 PM, Christopher Rienzo <cmrienzo at gmail.com>wrote:
>>
>>> $ ls src/mod/say
>>> mod_say_de  mod_say_en  mod_say_es  mod_say_fr  mod_say_hr  mod_say_hu
>>> mod_say_it  mod_say_ja  mod_say_nl  mod_say_pt  mod_say_ru  mod_say_th
>>> mod_say_zh
>>>
>>>
>>>
>>>  On Thu, Jan 27, 2011 at 7:19 PM, Holger Esser <hesser4900 at gmail.com>wrote:
>>>
>>>> Hi, is there a list of all the supported say (mod_say_xx) modules? I am
>>>> looking at my build and so far I am seeing Russian and English. Are there
>>>> more?
>>>>
>>>> Kind regards,
>>>> Holger
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bwibowo at gmail.com  Fri Jan 28 05:45:28 2011
From: bwibowo at gmail.com (Budi wibowo)
Date: Fri, 28 Jan 2011 02:45:28 +0000
Subject: [Freeswitch-users] Number mapping (DID like)
Message-ID: <1806109314-1296182728-cardhu_decombobulator_blackberry.rim.net-210966170-@b25.c2.bise3.blackberry>

Hi 
I want to do B# (called) rewriting, for example 
>From user 1000 call to 1001, and freeswitch rewrite 1001 to other number (pstn). 
Is it doable with freeswitch?

Regards
Budi


From Nabble at slickdeals.endjunk.com  Fri Jan 28 06:04:11 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 27 Jan 2011 19:04:11 -0800 (PST)
Subject: [Freeswitch-users] execute_on_answer=send_dtmf 1
Message-ID: <1296183851075-5968605.post@n2.nabble.com>


I have the following simple google_in dialplan
(conf/dialplan/public/094_gtalk_in.xml) meant to handle incoming calls for
my Google Voice DID by mod_dingaling, but it doesn't seem like it ever gets
executed. As such, I always have to manually press 1 to accept the incoming
call on my GV DID. If I answer the call after the 1st ring, then I hear no
GV call presentation announcer asking me to press 1 to accept the call (see
the 
http://freeswitch-users.2379917.n2.nabble.com/mod-dingaling-answering-prematurely-tp5965311p5967533.html
remarks  by Anthony Menessale if you want to know why). By the time I
realize this, the incoming call has been intercepted by the GV voicemail and
pressing 1 will have no use. I am hoping anyone here who has managed to
configure his/her FS which automatically sends dtmf 1 as outlined in 
http://wiki.freeswitch.org/wiki/Google_Voice here  will be able to help me
out.
<extension name="google_in" continue="true">
  <condition field="caller_id_name" expression="^(Google Voice)$">
    <action application="log" data="Google Voice Call Incoming" />
    <action application="set" data="execute_on_answer=send_dtmf 1"/>
    <action application="transfer" data="1000 XML default"/>
  </condition>
</extension>

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/execute-on-answer-send-dtmf-1-tp5968605p5968605.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From phone.bytes at gmail.com  Fri Jan 28 07:02:13 2011
From: phone.bytes at gmail.com (phone.bytes)
Date: Thu, 27 Jan 2011 21:02:13 -0700
Subject: [Freeswitch-users] TBCT 2bchannel transfer
In-Reply-To: <AANLkTinaZCrohy_RLoxFifSvRbzbXUfrhJcCRv61dsV5@mail.gmail.com>
References: <AANLkTinaZCrohy_RLoxFifSvRbzbXUfrhJcCRv61dsV5@mail.gmail.com>
Message-ID: <4D423FC5.3010007@gmail.com>

I talked to Sangoma about this several months ago.  They said it was on 
their road map, but, not at the top.  This is a sweet feature to have 
available on a PRI.  We wrote code to do this on a Dialogic system 
several years ago, still works great.  It was not too difficult to do.  
You need the Call Reference Value of the two calls, then you invoke a 
facility command, and it is done.  The calls have to be answered or in 
the alerting stage to succeed.  We may make a stab at this for 
Freeswitch when we get a better grip on it.  I have a spec I can try to 
dig up for you if you would like.

Maybe it is making some progress moving up on the Sangoma list.

On 1/27/2011 5:16 PM, Holger Esser wrote:
> Hi, does anybody have a detailed description of how to implement the 
> TBCT transfer method with Sangoma cards?
> I was looking at the att_xfer but I am not sure if that would cover TBCT.
>
> Any suggestions would be greatly appreciated.
>
> Holger
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From infos at madovsky.org  Fri Jan 28 07:05:13 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 27 Jan 2011 23:05:13 -0500
Subject: [Freeswitch-users] nibblebill and ODBC
Message-ID: <1F6390AB77694D38A2FFE90C1A125486@e1705>

I don't know yet if it's related to ODBC,
but if I use nibblebill pause, (works first time) hangup and call
the same again so the dialplan is held at nibblebill pause
infinetly. I have to restart FS  (on console I can see nibblebill is waiting some
reference or process)

?

thanks
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From u2nsam at gmail.com  Fri Jan 28 07:28:23 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 28 Jan 2011 09:58:23 +0530
Subject: [Freeswitch-users] deflect
In-Reply-To: <AANLkTinV=xZ_ahs=8DH7n8hADGL4hTOR05hNCWCasA85@mail.gmail.com>
References: <AANLkTiksYLOeki2_XDJC5YwMW6hjUxvdpT7-Mvg7vmKZ@mail.gmail.com>
	<AANLkTi=EMn55D1xGBrWuGgVE_dW_5itoJ4WpzByyagTE@mail.gmail.com>
	<AANLkTi=F5Lhsk1tjcG5z5LTUjZb4e-cQdNXQjK3OsVD7@mail.gmail.com>
	<AANLkTinV=xZ_ahs=8DH7n8hADGL4hTOR05hNCWCasA85@mail.gmail.com>
Message-ID: <AANLkTikGxmoG+zqPxHQ7RbOWby-KwhbSJBCEBJ_hFdWa@mail.gmail.com>

Here,

The Leg a should get transferred to 12127773456 right !

Regards
Sam

On Tue, Jan 25, 2011 at 9:54 PM, Sam <u2nsam at gmail.com> wrote:

> Hi,
>
> Is it possible in this scenario,
>
> I have a call (leg a) to an IVR on FS1 , after the ivr the below statement
> is executed,
> <action application="bridge" data="{api_on_answer='uuid_transfer  ${uuid}
> both'}sofia/external/12127773456 at 192.168.2.130 <9999 at 192.168.2.130>"/>
>
> As the FS1 sends invite to 192.168.2.130 and the call is connected to the
> moviephone IVR,
> but here what happens is the call is getting disconnected from leg a and
> the movie phone ivr 12127773456.
>
>
>
> Regds
> Sam
>
>
>
> On Mon, Jan 24, 2011 at 3:12 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> You could try uuid_simplify with the api_on_answer hook
>>
>> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_simplify
>> http://wiki.freeswitch.org/wiki/Variable_api_on_answer
>>
>> -Steve
>>
>>
>>
>> On 24 January 2011 09:05, Sam <u2nsam at gmail.com> wrote:
>>
>>>  Hi,
>>>
>>> Is it possible by having b2bua in between , would the leg A be deflected
>>> to the another FS server from first server ?
>>>
>>> Regds
>>> Sam
>>>
>>>
>>> On Wed, Jan 12, 2011 at 11:42 AM, Sam <u2nsam at gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> When call comes on 1 server and plays an application and after execution
>>>> of the
>>>> application the call is bridge to the other server ,but here after
>>>> bridging the call
>>>> should refer/deflect to other server, how this can be done ?
>>>>
>>>> Here just using the deflect variable is not recommended as there is
>>>> proxy in between,
>>>> so once the call is bridge the next step would be deflect the leg
>>>> totally to another server via proxy.
>>>>
>>>> Regards
>>>> Sam
>>>>
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From u2nsam at gmail.com  Fri Jan 28 09:41:42 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 28 Jan 2011 12:11:42 +0530
Subject: [Freeswitch-users] blocking 183 w/o sdp
Message-ID: <AANLkTinH1s+Uy-EPvX9oV-x+aXYkyzUEBSYm+hQ+-+K-@mail.gmail.com>

 Hi,

how can i ignore 183 without sdp,
what happens is the provider sends 183 without sdp and by applying "<action
application="set" data="sip_ignore_183nosdp=true"/>"  the FS sends 180 to
the leg a.
Here i want to block the 183 with SDP just like router as b2bua and send
nothing to leg a, and when actual 183 with sdp comes it should send .

Its because, providers are sending false signaling by sending 183 without
sdp,and it hampers while @ production,
Although by cisco sbc i have done this but i want to do it by FS,
Take a scenario, when call is send 183 without sdp for 10 secs and then
followed by 183 with sdp ( actual signal),
but when some one dials invalid number it rings for 10 secs and then gives
SIP cause 404, which is bad from the providers.
So this is the reason i want to block it.

Most of the providers do this, the way out is blocking.

I have got an advice from Tihomir  to do "execute_on_ring and parse your 180
/ 183 messages in search of SDP,
once you get 183 without SDP do not send it back to leg a and send signal
only when you got 183 with sdp or 180 "
And this could be valid call flow.

This happens in many cases where the provider is having nextone as a sbc and
that too tier 1 !

Regards
Sam
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From steveayre at gmail.com  Fri Jan 28 11:32:36 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 28 Jan 2011 08:32:36 +0000
Subject: [Freeswitch-users] Send email on successful fax sending.
In-Reply-To: <AANLkTimKgZDpuDUgNaqT1=61mqwOoJ66u4m-u4LBjxXM@mail.gmail.com>
References: <AANLkTinJ3N+zwg+ZK2RmdAZECEkLgcvOSKT0hf_mbpix@mail.gmail.com>
	<AANLkTi=5g-rOZtS9Dy4wq_J6OQU2roLzBDwhzhsuhQSG@mail.gmail.com>
	<4D3976FD.4090805@chaschperli.ch>
	<AANLkTimo0n3UP2c09F3AmgQdkRdU+DVAMxjYrw7cP9gm@mail.gmail.com>
	<AANLkTi=zBgX-trHwVUUgfGNZh3SmcJfru1tD=5-aE=MY@mail.gmail.com>
	<AANLkTimKgZDpuDUgNaqT1=61mqwOoJ66u4m-u4LBjxXM@mail.gmail.com>
Message-ID: <AANLkTin7M1H_Rxeet8QC-JcqkupLmVYhONAijXkuQ0-u@mail.gmail.com>

Hi Marc,

Brian's committed this patch to the trunk 27/1/2011 in 314a2a1e.

You'd now be able to connect via ESL, do originate and then wait for the
spandsp::txfaxresult event, then send the success/failure email.

-Steve


On 21 January 2011 13:10, Steven Ayre <steveayre at gmail.com> wrote:

> I've just put a patch together for mod_spandsp that should add the required
> event.
>
> It's on Jira, FS-3004.
> http://jira.freeswitch.org/browse/FS-3004
>
> I've checked it compiles, but I have no T38 capable gateways so someone
> else will need to test it out. Does anyone fancy trying it out?
>
> -Steve
>
>
>
>
> On 21 January 2011 12:40, Marc de Corny <marcdecorny at gmail.com> wrote:
>
>> Thanks, all I have not investigated the ESL options yet, I will have a
>> look and see,
>>
>>
>> many thanks for the responses,
>>
>> Marc
>>
>> On Fri, Jan 21, 2011 at 12:33 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>
>>> But since the channel will hangup, you'd need to wait for and check the
>>> CDR to see that variable.
>>>
>>> mod_spandsp_fax.c:412 is "TODO Fire events"
>>>
>>> It looks like that it's planned for mod_spandsp to fire an event that'll
>>> indicate that the fax has finished sending and would contain the status.
>>>
>>> If that was implemented, you could connect via ESL, do the originate and
>>> wait for the event to tell you whether it worked or not.
>>>
>>> -Steve
>>>
>>>
>>>
>>>   On 21 January 2011 12:07, Thomas Mueller <thomas at chaschperli.ch>wrote:
>>>
>>>>   On 21.01.2011 09:14, Marc de Corny wrote:
>>>> > Hi There,
>>>> > Has anybody had any ideas on this ? I imagine you must all have the
>>>> > same requirement in the Email to Fax scenario ?
>>>>
>>>>
>>>> reading the sourcecode the channel var "fax_success" gets set to 1 for
>>>> OK, to 0 for fail.
>>>>
>>>> - Thomas
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From avi at avimarcus.net  Fri Jan 28 11:41:01 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 28 Jan 2011 10:41:01 +0200
Subject: [Freeswitch-users] Number mapping (DID like)
In-Reply-To: <AANLkTinujT2YjspkUMn9fEnkk0w=vCH5wCWoZnQ07LLZ@mail.gmail.com>
References: <1806109314-1296182728-cardhu_decombobulator_blackberry.rim.net-210966170-@b25.c2.bise3.blackberry>
	<AANLkTinujT2YjspkUMn9fEnkk0w=vCH5wCWoZnQ07LLZ@mail.gmail.com>
Message-ID: <AANLkTikf2gS+=3NMUb=HW_QasRxiDZ3Ny+0a5=0d5VGK@mail.gmail.com>

Propaby, but that could mean many things. Do you mean rewrite the number in
the CDRs? Allow the owner of the extension to set a call forwarding / follow
me? Or basically to use 1001 as a speeddial for the one user?
I use mod_xml_curl extensively which allow me to query a database before I
do just about anything with te normal dialplan.
First, any 1-3 digit numbers it checks a extension-specific and then global
speeddial table.
Soon, I intend to have it query FusionPBX's follow-me settings so that I can
implement that with poper billing.
Some of tis might help you?
-Avi Marcus

> On Jan 28, 2011 4:46 AM, "Budi wibowo" <bwibowo at gmail.com> wrote:
> > Hi
> > I want to do B# (called) rewriting, for example
> >>From user 1000 call to 1001, and freeswitch rewrite 1001 to other number
(pstn).
> > Is it doable with freeswitch?
> >
> > Regards
> > Budi
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
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From louis.huppenbauer at gmail.com  Fri Jan 28 13:08:38 2011
From: louis.huppenbauer at gmail.com (Louis Huppenbauer)
Date: Fri, 28 Jan 2011 11:08:38 +0100
Subject: [Freeswitch-users] time_test on Centos 5.5
Message-ID: <AANLkTiksCg70iEdZiN1+CnfyBW87mg7T4jDHjeq7JW5Y@mail.gmail.com>

Hi,

I am testing FreeSWITCH on a Centos 5.5. The hardware is a 32 bit AMD
Opteron 275.

On the system I recognize a avg of 2000 if I run the time_test 1000
application.

Currently I have no audio problems but I want to make sure that the system
works well.

There are 4 possible options:
- Try to update the kernel to a newer one
- Try to use debian.
- Try to use the kernel-rt (realtime kernel from Centos)
- Set some nice kernel options. Which?

What do you think is the better solution?

Thanks in advance.

Kind regards,
Louis
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From hesser4900 at gmail.com  Fri Jan 28 14:33:32 2011
From: hesser4900 at gmail.com (Holger Esser)
Date: Fri, 28 Jan 2011 05:33:32 -0600
Subject: [Freeswitch-users] TBCT 2bchannel transfer
In-Reply-To: <4D423FC5.3010007@gmail.com>
References: <AANLkTinaZCrohy_RLoxFifSvRbzbXUfrhJcCRv61dsV5@mail.gmail.com>
	<4D423FC5.3010007@gmail.com>
Message-ID: <AANLkTi=eMBCiKUiw6pKom4yNT9oJ7ALfYpJAWuyt0CD6@mail.gmail.com>

Hi phone.bytes,

Thx for your suggestions. We did the same for Dialogic as well. Maybe
Sangoma made some progress. Take care.

Holger

On Thu, Jan 27, 2011 at 10:02 PM, phone.bytes <phone.bytes at gmail.com> wrote:

>  I talked to Sangoma about this several months ago.  They said it was on
> their road map, but, not at the top.  This is a sweet feature to have
> available on a PRI.  We wrote code to do this on a Dialogic system several
> years ago, still works great.  It was not too difficult to do.  You need the
> Call Reference Value of the two calls, then you invoke a facility command,
> and it is done.  The calls have to be answered or in the alerting stage to
> succeed.  We may make a stab at this for Freeswitch when we get a better
> grip on it.  I have a spec I can try to dig up for you if you would like.
>
> Maybe it is making some progress moving up on the Sangoma list.
>
> On 1/27/2011 5:16 PM, Holger Esser wrote:
>
> Hi, does anybody have a detailed description of how to implement the TBCT
> transfer method with Sangoma cards?
> I was looking at the att_xfer but I am not sure if that would cover TBCT.
>
>  Any suggestions would be greatly appreciated.
>
>  Holger
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Fri Jan 28 14:35:02 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 28 Jan 2011 12:35:02 +0100
Subject: [Freeswitch-users] time_test on Centos 5.5
In-Reply-To: <AANLkTiksCg70iEdZiN1+CnfyBW87mg7T4jDHjeq7JW5Y@mail.gmail.com>
References: <AANLkTiksCg70iEdZiN1+CnfyBW87mg7T4jDHjeq7JW5Y@mail.gmail.com>
Message-ID: <0E4A540B-3E61-4940-8246-6FBF67CF91D8@ipeva.fr>

Louis,

check that your kernel is compiled with the 1000hz timer (it should with Centos).
You have no audio issues as long as what you do does not require mixing audio.
Try to call a conference room, and from the CLI, play an audio file to the conf room.
With a bad timer, I had issues that easily.

Stay away from Debian, Centos is the right choice.
You could eventually try to fallback to centos 5.3 or 5.4.

Are you sure this processor is 32 bits ?
All info I find says it's 64bits, so you coul try a 64 bits kernel.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 28/01/2011 ? 11:08, Louis Huppenbauer a ?crit :

> Hi,
> 
> I am testing FreeSWITCH on a Centos 5.5. The hardware is a 32 bit AMD Opteron 275. 
> 
> On the system I recognize a avg of 2000 if I run the time_test 1000 application. 
> 
> Currently I have no audio problems but I want to make sure that the system works well.
> 
> There are 4 possible options:
> - Try to update the kernel to a newer one
> - Try to use debian.
> - Try to use the kernel-rt (realtime kernel from Centos)
> - Set some nice kernel options. Which?
> 
> What do you think is the better solution? 
> 
> Thanks in advance.
> 
> Kind regards,
> Louis
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mkopacki at gmail.com  Fri Jan 28 14:45:22 2011
From: mkopacki at gmail.com (Michal Kopacki)
Date: Fri, 28 Jan 2011 12:45:22 +0100
Subject: [Freeswitch-users] network scenario doubts
In-Reply-To: <AANLkTi=d+7Yxd6CKD-NyuW6ionAAxbgmBy2StMJO=1K7@mail.gmail.com>
References: <4D3FD3B0.4080309@gmail.com>	<AANLkTinBd00iKLysi1O97hOupQpYdzBchBN5t8oZOJre@mail.gmail.com>	<4D41803C.1050601@gmail.com>
	<AANLkTi=d+7Yxd6CKD-NyuW6ionAAxbgmBy2StMJO=1K7@mail.gmail.com>
Message-ID: <4D42AC52.7080301@gmail.com>

     Disabling iptables didn't change anything, but thanks to STUN it's 
working well now.

-- 
Michal

On 01/28/2011 02:38 AM, Michael Collins wrote:
> For a test, stop iptables and make a call. If audio still isn't 
> flowing then you know something other than iptables is interfering. I 
> would get a capture of the SIP traffic and see what it looks like. 
> Make sure that there aren't any SIP ALGs in the path that might be 
> re-writing port numbers.
>
> -MC
>
> On Thu, Jan 27, 2011 at 6:25 AM, Michal Kopacki <mkopacki at gmail.com 
> <mailto:mkopacki at gmail.com>> wrote:
>
>         Everything is set as you wrote and I'm able to login (using
>     domain name). Now I'm facing a media problem during outside call.
>     I'm logged to FS from ip phone (outside lan) and trying to dial
>     5000 (default ivr). Call is connected but i'm not able to hear
>     anything.
>
>        There is iptable based firewall with opened udp ports
>     16384:32768, but it seems nothing even hit those ports:
>
>     Chain INPUT (policy DROP 0 packets, 0 bytes)
>      pkts bytes target     prot opt in     out    
>     source               destination
>         0     0 ACCEPT     udp  --  eth0   any    
>     anywhere             anywhere            udp
>     dpts:connected:filenet-tms state NEW
>
>       Any thoughts ?
>     -- 
>     Best regards,
>     Michal
>
>
>     On 01/27/2011 01:39 AM, Michael Collins wrote:
>>     yes, you should be able to handle this scenario. You probably
>>     just need to set your external profile to use port 5060 instead
>>     of 5080. Most people only have a single NIC so their sofia
>>     profiles need to be on different ports. (A sofia profile is a SIP
>>     user agent (UA) that can listen/respond to SIP messages on an IP
>>     address and port.)
>>
>>     Look in external.xml for these lines:
>>     <param name="rtp-ip" value="$${local_ip_v4}"/>
>>     <param name="sip-ip" value="$${local_ip_v4}"/>
>>
>>     If you have a static IP address (which it appears you do) then
>>     put that ip addr in there in place of the variable. Then look at
>>     the end of vars.xml and you'll see where the external profile's
>>     port is set to 5080/5081. Change it to 5060/5061 and then restart FS.
>>
>>     Let us know how it goes.
>>
>>     -MC
>>
>>     On Tue, Jan 25, 2011 at 11:56 PM, Michal Kopacki
>>     <mkopacki at gmail.com <mailto:mkopacki at gmail.com>> wrote:
>>
>>               Hello,
>>
>>           This is my first post to this list, so hello everyone.
>>
>>           I'm at the very beginning of freeswitch journey and i have
>>         a problem
>>         with fit FS to my network scenario.
>>
>>         OS: fedora 13 (x86_64)
>>         FS: 1.0.7 (compiled from sources), default config
>>
>>         Desired scenario:
>>
>>         softphone -> mydomain.com <http://mydomain.com> (with ip
>>         193.59.72.xx) -> my isp network ->
>>         10.25.48.xx on eth1 -> FS -> 192.168.0.1 on eth0 -> softphone
>>
>>         And now, I'm able to connect to FS from my local network, but
>>         I'm not
>>         able to connect from outside (neither domain nor ip). In
>>         internal.xml I
>>         set internal ip of server and i wanted to set external ip in
>>         external.xml, but there is a question: which one ?  In case of
>>         193.59.72.xx external profile didnt' start and with
>>         10.25.48.xx outside
>>         softphone didn't register.  I checked with netstat and
>>         realized that
>>         port 5060 is bind to internal nic only and 5080 to external nic
>>         (10.25.48.xx) and I have no idea what next.
>>
>>         Is it even possible to work with such network scenario ? I
>>         would be
>>         grateful for pointing me to right direction or maybe propose
>>         different
>>         approach.
>>
>>         --
>>         Best regards,
>>         Michal
>>
>>
>>         _______________________________________________
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From covici at ccs.covici.com  Fri Jan 28 14:46:50 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 28 Jan 2011 06:46:50 -0500
Subject: [Freeswitch-users] time_test on Centos 5.5
In-Reply-To: <AANLkTiksCg70iEdZiN1+CnfyBW87mg7T4jDHjeq7JW5Y@mail.gmail.com>
References: <AANLkTiksCg70iEdZiN1+CnfyBW87mg7T4jDHjeq7JW5Y@mail.gmail.com>
Message-ID: <4908.1296215210@ccs.covici.com>

I don't even have time_test -- is this supposed to be a freeswitch
command?

Louis Huppenbauer <louis.huppenbauer at gmail.com> wrote:

> Hi,
> 
> I am testing FreeSWITCH on a Centos 5.5. The hardware is a 32 bit AMD
> Opteron 275.
> 
> On the system I recognize a avg of 2000 if I run the time_test 1000
> application.
> 
> Currently I have no audio problems but I want to make sure that the system
> works well.
> 
> There are 4 possible options:
> - Try to update the kernel to a newer one
> - Try to use debian.
> - Try to use the kernel-rt (realtime kernel from Centos)
> - Set some nice kernel options. Which?
> 
> What do you think is the better solution?
> 
> Thanks in advance.
> 
> Kind regards,
> Louis
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From fdelawarde at wirelessmundi.com  Fri Jan 28 14:47:45 2011
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Fri, 28 Jan 2011 12:47:45 +0100
Subject: [Freeswitch-users] execute_on_ring executing on answer
Message-ID: <1296215265.8986.147.camel@luna.tc.commsmundi.com>

Hi,

Doing some testing with this morning's git (Fri Jan 28) I just found out
that the execute_on_ring application runs when the destination answers
instead of when it rings.

So far, I can't seem to find out the reason. Could it be some
configuration issue?


Here a call log showing the phenomenon with a simple bridge:

<action application="bridge" data="[execute_on_ring=info,absolute_codec_string=^^:PCMA:G722]sofia/192.168.10.1/sip:2103 at 192.168.10.22:5060"/>

http://pastebin.freeswitch.org/15168


Thanks,
Fran?ois.



From vizentini at hotmail.com  Fri Jan 28 14:51:05 2011
From: vizentini at hotmail.com (Paulo Vicentini)
Date: Fri, 28 Jan 2011 11:51:05 +0000
Subject: [Freeswitch-users] Portaudio Improvements
In-Reply-To: <AANLkTi=GRGJbQwE8yF=y_yUWJobUMnhdYfYpnGge5qf-@mail.gmail.com>
References: <AANLkTim9AmtEGZoK0NyG1yKVsmJ_MSwXAppi-6AcCqnz@mail.gmail.com>,
	<AANLkTimm69f6adgkgDz03GUOq8xJw_oaVvrkAGxd1yjG@mail.gmail.com>,
	<AANLkTikk==DKhXft67HWcj0ytya030bS=4vtt2o7+tD8@mail.gmail.com>,
	<AANLkTi=T6a9nU54toi5+fMA7kq_xkmZrRnRT3Fa8Pao5@mail.gmail.com>,
	<AANLkTimbULX-_ne4ZyirMFxcOcRBaEZwxTfude6=L8F_@mail.gmail.com>,
	<AANLkTin_ue-i-Mw5CTQiZo3CTWi3Kcvt_4pdGYTS1jk7@mail.gmail.com>,
	<AANLkTi=GRGJbQwE8yF=y_yUWJobUMnhdYfYpnGge5qf-@mail.gmail.com>
Message-ID: <SNT124-W43801D7DD8A4B86A2A0B8EBBE10@phx.gbl>


HiYou can try PortMixer for volume control that is intended to work along with PortAudio 
I am developing/testing "mod_portaudio2" that does not rely on pablio, maybe things will be better with this approach  Paulo
From: mitch.capper at gmail.com
Date: Sat, 22 Jan 2011 21:05:22 -0800
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Portaudio Improvements

Ah well viewing volume levels had not been something I had looked into.  It wouldn't be too hard but the question is how do you constantly report that (FS event every second? every half second? port audio commands to return that value frequently?).  If your client is windows however again the native windows volume handler for apps does also keep track of their current output.   



~Mitch

On Sat, Jan 22, 2011 at 5:26 PM, Seven Du <dujinfang at gmail.com> wrote:


Thanks Mitch, I actually had tried the equivalent uuid_audio and looks

like it's the only way to do it for now. We had wanted to get energy

levels dynamically to show a live indication but we given up that

feature.





On Sun, Jan 23, 2011 at 2:52 AM, Mitch Capper <mitch.capper at gmail.com> wrote:

> It is true PA lacks volume control, it had been something I was debating.

> Adding gain is pretty simple as you are for the most part just multiplying

> up the samples I believe so would not be a major item to add to portaudio, I

> however decided not to go this route for two reasons.  1) In Windows each

> application has its own volume setting (Vista and higher),  and apps are

> starting to tie into this than internal volume controls.  The downside is it

> doesn't distort audio/have a gain option.   This is when I thought about

> allowing for a fixed gain amount added to portaudio.  2) Freeswitch actually

> has its own built in volume control settings with set_audio_level.

> set_audio_level allows for 9 settings (4 lower 4 higher) and seems to do a

> good job at digital volume control.

> Avoiding modifying the audio stream itself could also result in less

> distortion when using actual volume controls (headsets, etc).

>

> So if you want volume control I would say take a look at set_audio_level as

> it may be the simplest method.   Adding it to portaudio would work but one

> of the nice advantages of set_audio_level is the fact you can also do it per

> channel rather than globally:)

>

> ~Mitch

>

> On Fri, Jan 21, 2011 at 5:44 PM, Seven Du <dujinfang at gmail.com> wrote:

>>

>> In addition to AEC. I also found PA lacks the ability to change the

>> sound device volume. I tried to control sound devices in QT but

>> haven't find a way. Ideas on this?

>

>

> _______________________________________________

> FreeSWITCH-users mailing list

> FreeSWITCH-users at lists.freeswitch.org

> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

> http://www.freeswitch.org

>

>







--

About: http://about.me/dujinfang

Blog: http://www.dujinfang.com

Proj:  http://www.freeswitch.org.cn



_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org




_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From louis.huppenbauer at gmail.com  Fri Jan 28 15:20:22 2011
From: louis.huppenbauer at gmail.com (Louis Huppenbauer)
Date: Fri, 28 Jan 2011 13:20:22 +0100
Subject: [Freeswitch-users] time_test on Centos 5.5
In-Reply-To: <4908.1296215210@ccs.covici.com>
References: <AANLkTiksCg70iEdZiN1+CnfyBW87mg7T4jDHjeq7JW5Y@mail.gmail.com>
	<4908.1296215210@ccs.covici.com>
Message-ID: <AANLkTi=5MrJ7C5sozFvm3CEV2Oh7MiuLAtm_dtkhHZW2@mail.gmail.com>

Hi,

yeah, your right. It's a 64 bit processor.

Is it possible to "downgrade" to Centos 5.3? I found only the DVD from
Centos 5.3 but the machine is within a server farm and therefore a
netinstall is much easier.

Did someone tried the kernel-rt package?

Kind regards,
Louis

2011/1/28 <covici at ccs.covici.com>

> I don't even have time_test -- is this supposed to be a freeswitch
> command?
>
> Louis Huppenbauer <louis.huppenbauer at gmail.com> wrote:
>
> > Hi,
> >
> > I am testing FreeSWITCH on a Centos 5.5. The hardware is a 32 bit AMD
> > Opteron 275.
> >
> > On the system I recognize a avg of 2000 if I run the time_test 1000
> > application.
> >
> > Currently I have no audio problems but I want to make sure that the
> system
> > works well.
> >
> > There are 4 possible options:
> > - Try to update the kernel to a newer one
> > - Try to use debian.
> > - Try to use the kernel-rt (realtime kernel from Centos)
> > - Set some nice kernel options. Which?
> >
> > What do you think is the better solution?
> >
> > Thanks in advance.
> >
> > Kind regards,
> > Louis
> >
> > ----------------------------------------------------
> > Alternatives:
> >
> > ----------------------------------------------------
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sos at sokhapkin.dyndns.org  Fri Jan 28 15:30:17 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 28 Jan 2011 07:30:17 -0500
Subject: [Freeswitch-users] 200 OK without SDP
Message-ID: <201101280730.17483.sos@sokhapkin.dyndns.org>

What are possible reasons for FS to send 200 OK without SDP in response to SST 
reinvite? bypass_media=true, initial INVITE and initial 200 OK response have 
more than one codec (in different order) if this matters.


From steveayre at gmail.com  Fri Jan 28 15:34:44 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 28 Jan 2011 12:34:44 +0000
Subject: [Freeswitch-users] 200 OK without SDP
In-Reply-To: <201101280730.17483.sos@sokhapkin.dyndns.org>
References: <201101280730.17483.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTinCypuBiNnVNfrYF6wn_YOqPnbM0Fjuau8tzBNA@mail.gmail.com>

Do you have a siptrace?

On 28 January 2011 12:30, Sergey Okhapkin <sos at sokhapkin.dyndns.org> wrote:

> What are possible reasons for FS to send 200 OK without SDP in response to
> SST
> reinvite? bypass_media=true, initial INVITE and initial 200 OK response
> have
> more than one codec (in different order) if this matters.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sos at sokhapkin.dyndns.org  Fri Jan 28 15:53:32 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 28 Jan 2011 07:53:32 -0500
Subject: [Freeswitch-users] 200 OK without SDP
In-Reply-To: <AANLkTinCypuBiNnVNfrYF6wn_YOqPnbM0Fjuau8tzBNA@mail.gmail.com>
References: <201101280730.17483.sos@sokhapkin.dyndns.org>
	<AANLkTinCypuBiNnVNfrYF6wn_YOqPnbM0Fjuau8tzBNA@mail.gmail.com>
Message-ID: <201101280753.32148.sos@sokhapkin.dyndns.org>

http://pastebin.freeswitch.org/15169 , 200 OK in question at line 439.

On Friday 28 January 2011, Steven Ayre wrote:
> Do you have a siptrace?
> 
> On 28 January 2011 12:30, Sergey Okhapkin <sos at sokhapkin.dyndns.org> wrote:
> > What are possible reasons for FS to send 200 OK without SDP in response
> > to SST
> > reinvite? bypass_media=true, initial INVITE and initial 200 OK response
> > have
> > more than one codec (in different order) if this matters.
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org



From megahohol at gmail.com  Fri Jan 28 14:52:02 2011
From: megahohol at gmail.com (Grygoriy Dobrovolskyy)
Date: Fri, 28 Jan 2011 12:52:02 +0100
Subject: [Freeswitch-users] Virtualisation and freeswitch
Message-ID: <AANLkTimZTwZfUYMjC1ddAQMWHHWSLH=BLajv0cgSux4B@mail.gmail.com>

Hello list, le me introduce myself first, I am Grigoriy Dobrovolskyy
currently working as voip administrator and call centeres integrator. I
worked a lot with asterisk, i think Fs is a great software but not easy to
learn.

I have googled my question and did not found a clear answer.

With what technology i can virtualise freeswitch without timer problems ?

I know that i can use OpenVZ but it's more 'jail' then virtualisation. I
heard that Amazon EC with High CPU instance works (confirm?)

Can someone give me info of a succesfull installations with Conferences, Moh
 / mixing ? And a technology used ?

Thank you.
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From jalsot at gmail.com  Fri Jan 28 15:54:10 2011
From: jalsot at gmail.com (Tamas Jalsovszky)
Date: Fri, 28 Jan 2011 13:54:10 +0100
Subject: [Freeswitch-users] Portaudio Improvements
In-Reply-To: <SNT124-W43801D7DD8A4B86A2A0B8EBBE10@phx.gbl>
References: <AANLkTim9AmtEGZoK0NyG1yKVsmJ_MSwXAppi-6AcCqnz@mail.gmail.com>
	<AANLkTimm69f6adgkgDz03GUOq8xJw_oaVvrkAGxd1yjG@mail.gmail.com>
	<AANLkTikk==DKhXft67HWcj0ytya030bS=4vtt2o7+tD8@mail.gmail.com>
	<AANLkTi=T6a9nU54toi5+fMA7kq_xkmZrRnRT3Fa8Pao5@mail.gmail.com>
	<AANLkTimbULX-_ne4ZyirMFxcOcRBaEZwxTfude6=L8F_@mail.gmail.com>
	<AANLkTin_ue-i-Mw5CTQiZo3CTWi3Kcvt_4pdGYTS1jk7@mail.gmail.com>
	<AANLkTi=GRGJbQwE8yF=y_yUWJobUMnhdYfYpnGge5qf-@mail.gmail.com>
	<SNT124-W43801D7DD8A4B86A2A0B8EBBE10@phx.gbl>
Message-ID: <AANLkTi=DQTDrFLv8HmMSOaL+PWKjg0B0htq+a=16QAUa@mail.gmail.com>

Wow, I didn't even know that somebody is working on PA2.
Paulo, could you share some more information about that module? Why would
somebody use it instead of the current one? Advantages/disadvantages?
Is the source code available somewhere? Do you have a roadmap for the
module? :)

Regards,
  Tamas

On Fri, Jan 28, 2011 at 12:51 PM, Paulo Vicentini <vizentini at hotmail.com>wrote:

>  Hi
> You can try PortMixer for volume control that is intended to work along
> with PortAudio
>
> I am developing/testing "mod_portaudio2" that does not rely on pablio,
> maybe things will be better with this approach
>
> Paulo
>
> ------------------------------
> From: mitch.capper at gmail.com
> Date: Sat, 22 Jan 2011 21:05:22 -0800
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Portaudio Improvements
>
>
> Ah well viewing volume levels had not been something I had looked into.  It
> wouldn't be too hard but the question is how do you constantly report that
> (FS event every second? every half second? port audio commands to return
> that value frequently?).  If your client is windows however again the native
> windows volume handler for apps does also keep track of their current
> output.
>
> ~Mitch
>
> On Sat, Jan 22, 2011 at 5:26 PM, Seven Du <dujinfang at gmail.com> wrote:
>
> Thanks Mitch, I actually had tried the equivalent uuid_audio and looks
> like it's the only way to do it for now. We had wanted to get energy
> levels dynamically to show a live indication but we given up that
> feature.
>
>
> On Sun, Jan 23, 2011 at 2:52 AM, Mitch Capper <mitch.capper at gmail.com>
> wrote:
> > It is true PA lacks volume control, it had been something I was debating.
> > Adding gain is pretty simple as you are for the most part just
> multiplying
> > up the samples I believe so would not be a major item to add to
> portaudio, I
> > however decided not to go this route for two reasons.  1) In Windows each
> > application has its own volume setting (Vista and higher),  and apps are
> > starting to tie into this than internal volume controls.  The downside is
> it
> > doesn't distort audio/have a gain option.   This is when I thought about
> > allowing for a fixed gain amount added to portaudio.  2) Freeswitch
> actually
> > has its own built in volume control settings with set_audio_level.
> > set_audio_level allows for 9 settings (4 lower 4 higher) and seems to do
> a
> > good job at digital volume control.
> > Avoiding modifying the audio stream itself could also result in less
> > distortion when using actual volume controls (headsets, etc).
> >
> > So if you want volume control I would say take a look at set_audio_level
> as
> > it may be the simplest method.   Adding it to portaudio would work but
> one
> > of the nice advantages of set_audio_level is the fact you can also do it
> per
> > channel rather than globally:)
> >
> > ~Mitch
> >
> > On Fri, Jan 21, 2011 at 5:44 PM, Seven Du <dujinfang at gmail.com> wrote:
> >>
> >> In addition to AEC. I also found PA lacks the ability to change the
> >> sound device volume. I tried to control sound devices in QT but
> >> haven't find a way. Ideas on this?
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From oa at estation.dk  Fri Jan 28 15:39:10 2011
From: oa at estation.dk (=?ISO-8859-1?Q?=D8yvind_Albrigtsen?=)
Date: Fri, 28 Jan 2011 13:39:10 +0100
Subject: [Freeswitch-users] Choppy VM when using loopback (on Debian lenny)
Message-ID: <4D42B8EE.7030208@estation.dk>

Hi,

I'm using latest git-version of Freeswitch, and when I go to voicemail 
when calling a number the sound playback is choppy and it skips some of 
the digits in the number I called.

If I set sleep to 2000 it sounds better, but it's still a little choppy 
and still skips some of the digits.

This is with the default settings coming with the git version. When I 
call 4000 to listen to voicemail it sounds fine.

The kernel is the default kernel which is compiled with CONFIG_HZ_250=y

Regards
Oyvind


From jmesquita at freeswitch.org  Fri Jan 28 16:01:38 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 28 Jan 2011 10:01:38 -0300
Subject: [Freeswitch-users] Portaudio Improvements
In-Reply-To: <AANLkTi=DQTDrFLv8HmMSOaL+PWKjg0B0htq+a=16QAUa@mail.gmail.com>
References: <AANLkTim9AmtEGZoK0NyG1yKVsmJ_MSwXAppi-6AcCqnz@mail.gmail.com>
	<AANLkTimm69f6adgkgDz03GUOq8xJw_oaVvrkAGxd1yjG@mail.gmail.com>
	<AANLkTikk==DKhXft67HWcj0ytya030bS=4vtt2o7+tD8@mail.gmail.com>
	<AANLkTi=T6a9nU54toi5+fMA7kq_xkmZrRnRT3Fa8Pao5@mail.gmail.com>
	<AANLkTimbULX-_ne4ZyirMFxcOcRBaEZwxTfude6=L8F_@mail.gmail.com>
	<AANLkTin_ue-i-Mw5CTQiZo3CTWi3Kcvt_4pdGYTS1jk7@mail.gmail.com>
	<AANLkTi=GRGJbQwE8yF=y_yUWJobUMnhdYfYpnGge5qf-@mail.gmail.com>
	<SNT124-W43801D7DD8A4B86A2A0B8EBBE10@phx.gbl>
	<AANLkTi=DQTDrFLv8HmMSOaL+PWKjg0B0htq+a=16QAUa@mail.gmail.com>
Message-ID: <AANLkTimyjXLgwf5_Vu092Wo2comyWWYEZLDx6rthH-+8@mail.gmail.com>

Paulo,

What is the current status of PA with non-blocking I/O? A little while ago
it was not moving forward and pablio was the only way out of it. What are
your plans? Maybe some of us can help?

Regards,
Jo?o Mesquita


On Fri, Jan 28, 2011 at 9:54 AM, Tamas Jalsovszky <jalsot at gmail.com> wrote:

> Wow, I didn't even know that somebody is working on PA2.
> Paulo, could you share some more information about that module? Why would
> somebody use it instead of the current one? Advantages/disadvantages?
> Is the source code available somewhere? Do you have a roadmap for the
> module? :)
>
> Regards,
>   Tamas
>
>
> On Fri, Jan 28, 2011 at 12:51 PM, Paulo Vicentini <vizentini at hotmail.com>wrote:
>
>>  Hi
>> You can try PortMixer for volume control that is intended to work along
>> with PortAudio
>>
>> I am developing/testing "mod_portaudio2" that does not rely on pablio,
>> maybe things will be better with this approach
>>
>> Paulo
>>
>> ------------------------------
>> From: mitch.capper at gmail.com
>> Date: Sat, 22 Jan 2011 21:05:22 -0800
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Portaudio Improvements
>>
>>
>> Ah well viewing volume levels had not been something I had looked into.
>> It wouldn't be too hard but the question is how do you constantly report
>> that (FS event every second? every half second? port audio commands to
>> return that value frequently?).  If your client is windows however again the
>> native windows volume handler for apps does also keep track of their current
>> output.
>>
>> ~Mitch
>>
>> On Sat, Jan 22, 2011 at 5:26 PM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> Thanks Mitch, I actually had tried the equivalent uuid_audio and looks
>> like it's the only way to do it for now. We had wanted to get energy
>> levels dynamically to show a live indication but we given up that
>> feature.
>>
>>
>> On Sun, Jan 23, 2011 at 2:52 AM, Mitch Capper <mitch.capper at gmail.com>
>> wrote:
>> > It is true PA lacks volume control, it had been something I was
>> debating.
>> > Adding gain is pretty simple as you are for the most part just
>> multiplying
>> > up the samples I believe so would not be a major item to add to
>> portaudio, I
>> > however decided not to go this route for two reasons.  1) In Windows
>> each
>> > application has its own volume setting (Vista and higher),  and apps are
>> > starting to tie into this than internal volume controls.  The downside
>> is it
>> > doesn't distort audio/have a gain option.   This is when I thought about
>> > allowing for a fixed gain amount added to portaudio.  2) Freeswitch
>> actually
>> > has its own built in volume control settings with set_audio_level.
>> > set_audio_level allows for 9 settings (4 lower 4 higher) and seems to do
>> a
>> > good job at digital volume control.
>> > Avoiding modifying the audio stream itself could also result in less
>> > distortion when using actual volume controls (headsets, etc).
>> >
>> > So if you want volume control I would say take a look at set_audio_level
>> as
>> > it may be the simplest method.   Adding it to portaudio would work but
>> one
>> > of the nice advantages of set_audio_level is the fact you can also do it
>> per
>> > channel rather than globally:)
>> >
>> > ~Mitch
>> >
>> > On Fri, Jan 21, 2011 at 5:44 PM, Seven Du <dujinfang at gmail.com> wrote:
>> >>
>> >> In addition to AEC. I also found PA lacks the ability to change the
>> >> sound device volume. I tried to control sound devices in QT but
>> >> haven't find a way. Ideas on this?
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> About: http://about.me/dujinfang
>> Blog: http://www.dujinfang.com
>> Proj:  http://www.freeswitch.org.cn
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________ FreeSWITCH-users mailing
>> list FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vermeulen.deon at gmail.com  Fri Jan 28 16:23:02 2011
From: vermeulen.deon at gmail.com (Deon Vermeulen)
Date: Fri, 28 Jan 2011 15:23:02 +0200
Subject: [Freeswitch-users] DI-TE122P Digium Single Span T1/E1 - PRI Card
Message-ID: <8C543835-EC2F-4536-B6BE-7F1A458C7281@gmail.com>

Hi List

I'm looking at purchasing this card and using it in my FS Server.

Is this in used in production by anyone?

Is there a guide on how to setup this card in FS?


Thank you very very much


Regards
Deon Vermeulen


From marcdecorny at gmail.com  Fri Jan 28 16:25:41 2011
From: marcdecorny at gmail.com (Marc de Corny)
Date: Fri, 28 Jan 2011 13:25:41 +0000
Subject: [Freeswitch-users] Send email on successful fax sending.
In-Reply-To: <AANLkTin7M1H_Rxeet8QC-JcqkupLmVYhONAijXkuQ0-u@mail.gmail.com>
References: <AANLkTinJ3N+zwg+ZK2RmdAZECEkLgcvOSKT0hf_mbpix@mail.gmail.com>
	<AANLkTi=5g-rOZtS9Dy4wq_J6OQU2roLzBDwhzhsuhQSG@mail.gmail.com>
	<4D3976FD.4090805@chaschperli.ch>
	<AANLkTimo0n3UP2c09F3AmgQdkRdU+DVAMxjYrw7cP9gm@mail.gmail.com>
	<AANLkTi=zBgX-trHwVUUgfGNZh3SmcJfru1tD=5-aE=MY@mail.gmail.com>
	<AANLkTimKgZDpuDUgNaqT1=61mqwOoJ66u4m-u4LBjxXM@mail.gmail.com>
	<AANLkTin7M1H_Rxeet8QC-JcqkupLmVYhONAijXkuQ0-u@mail.gmail.com>
Message-ID: <AANLkTi=2sMwKKRk3gu35KfMZ52Kow4CxViU=ZTtKmR1X@mail.gmail.com>

fantastic.
let me give it a go and see.

thanks for your help guys
Marc

On Fri, Jan 28, 2011 at 8:32 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Hi Marc,
>
> Brian's committed this patch to the trunk 27/1/2011 in 314a2a1e.
>
> You'd now be able to connect via ESL, do originate and then wait for the
> spandsp::txfaxresult event, then send the success/failure email.
>
> -Steve
>
>
> On 21 January 2011 13:10, Steven Ayre <steveayre at gmail.com> wrote:
>
>> I've just put a patch together for mod_spandsp that should add the
>> required event.
>>
>> It's on Jira, FS-3004.
>> http://jira.freeswitch.org/browse/FS-3004
>>
>> I've checked it compiles, but I have no T38 capable gateways so someone
>> else will need to test it out. Does anyone fancy trying it out?
>>
>> -Steve
>>
>>
>>
>>
>> On 21 January 2011 12:40, Marc de Corny <marcdecorny at gmail.com> wrote:
>>
>>> Thanks, all I have not investigated the ESL options yet, I will have a
>>> look and see,
>>>
>>>
>>> many thanks for the responses,
>>>
>>> Marc
>>>
>>>   On Fri, Jan 21, 2011 at 12:33 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>>
>>>> But since the channel will hangup, you'd need to wait for and check the
>>>> CDR to see that variable.
>>>>
>>>> mod_spandsp_fax.c:412 is "TODO Fire events"
>>>>
>>>> It looks like that it's planned for mod_spandsp to fire an event that'll
>>>> indicate that the fax has finished sending and would contain the status.
>>>>
>>>> If that was implemented, you could connect via ESL, do the originate and
>>>> wait for the event to tell you whether it worked or not.
>>>>
>>>> -Steve
>>>>
>>>>
>>>>
>>>>   On 21 January 2011 12:07, Thomas Mueller <thomas at chaschperli.ch>wrote:
>>>>
>>>>>   On 21.01.2011 09:14, Marc de Corny wrote:
>>>>> > Hi There,
>>>>> > Has anybody had any ideas on this ? I imagine you must all have the
>>>>> > same requirement in the Email to Fax scenario ?
>>>>>
>>>>>
>>>>> reading the sourcecode the channel var "fax_success" gets set to 1 for
>>>>> OK, to 0 for fail.
>>>>>
>>>>> - Thomas
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From avi at avimarcus.net  Fri Jan 28 16:33:08 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 28 Jan 2011 15:33:08 +0200
Subject: [Freeswitch-users] Virtualisation and freeswitch
In-Reply-To: <AANLkTimZTwZfUYMjC1ddAQMWHHWSLH=BLajv0cgSux4B@mail.gmail.com>
References: <AANLkTimZTwZfUYMjC1ddAQMWHHWSLH=BLajv0cgSux4B@mail.gmail.com>
Message-ID: <AANLkTinUFPDM8P4oU4SZ-DE4rd4Qg3fHQG-W+iQt5JXv@mail.gmail.com>

I'm using Linode's VPS which is run on xen, and all tests (80 moh channels,
playback into a conference) and light usage have been perfect.
Some have complained of Amazon's high latency, others say it's been an
awesome platform for them.

The devs, and many using FreeSWITCH in high usage, will say to stay away
from *all* virtualization for the best performance.
-Avi

On Fri, Jan 28, 2011 at 1:52 PM, Grygoriy Dobrovolskyy
<megahohol at gmail.com>wrote:

> Hello list, le me introduce myself first, I am Grigoriy Dobrovolskyy
> currently working as voip administrator and call centeres integrator. I
> worked a lot with asterisk, i think Fs is a great software but not easy to
> learn.
>
> I have googled my question and did not found a clear answer.
>
> With what technology i can virtualise freeswitch without timer problems ?
>
> I know that i can use OpenVZ but it's more 'jail' then virtualisation. I
> heard that Amazon EC with High CPU instance works (confirm?)
>
> Can someone give me info of a succesfull installations with Conferences,
> Moh  / mixing ? And a technology used ?
>
> Thank you.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From patrick.plattes at niemann-frey.info  Fri Jan 28 16:41:31 2011
From: patrick.plattes at niemann-frey.info (Patrick Plattes)
Date: Fri, 28 Jan 2011 14:41:31 +0100
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTin3Bu+8EEr_v_GaRGNPzOjfcnfzsV2Av2cgnWgO@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
	<AANLkTi=in3CLd-n0mT3SRKYA0kxLMGVWk9KeBx3_jhY6@mail.gmail.com>
	<AANLkTin3Bu+8EEr_v_GaRGNPzOjfcnfzsV2Av2cgnWgO@mail.gmail.com>
Message-ID: <AANLkTi=OC_jx0+nzK405VZURLDJPD25R4ZjyJ3RXJ1nk@mail.gmail.com>

Hi :-)

2011/1/27 Anthony Minessale <anthony.minessale at gmail.com>:
> try setting the variable presence_id to the blf entity you are
> subscribing to (include the domain)

I don't understand fully the variable presence_id, but i've tried a
few things without any success. If I use the following lines, the lamp
fill be enabled at the login for a few milliseconds.

<extension name="Queue Management">
  <condition field="destination_number"
expression="^queue-(\w+)-(\d+)$" break="on-false">
    <action application="set"
data="queue_name=${regex(${destination_number}|^queue-(\w+)-(\d+)$|$1)}"
inline="true"/>
    <action application="set"
data="queue_user=${regex(${destination_number}|^queue-(\w+)-(\d+)$|$2)}"
inline="true"/>
    <action application="log" data="INFO destination_number
${destination_number}"/>
    <action application="log" data="INFO queue_user ${queue_user}"/>
    <action application="log" data="INFO queue_name ${queue_name}"/>
  </condition>
  <condition field="${fifo(list ${queue_name}@${domain_name})}"
expression="${queue_user}">
    <action application="answer"/>
    <action application="set"
data="presence_id=queue-${queue_name}-${queue_user}@${domain_name}"/>
    <action application="set" data="result=${fifo_member(del
${queue_name}@${domain_name}
{fifo_member_wait=nowait}user/${queue_user} )}"/>
    <action application="log" data="INFO Delete FIFO agent result: ${result}"/>
    <action application="log" data="INFO User Logout: ${user_name}"/>
    <action application="sleep" data="600"/>
    <action application="playback" data="ivr/ivr-you_are_now_logged_out.wav"/>
    <action application="set" data="result=${presence(out
queue-${queue_name}-${queue_user}@${domain_name} inuse inuse)}"/>
    <action application="log" data="INFO Presence OUT ${result}"/>
    <action application="hangup"/>

    <anti-action application="answer"/>
    <anti-action application="set"
data="presence_id=queue-${queue_name}-${queue_user}@${domain_name}"
inline="true"/>
    <anti-action application="set" data="result=${fifo_member(add
${queue_name}@${domain_name}
{fifo_member_wait=nowait}user/${queue_user} )}" inline="true"/>
    <anti-action application="log" data="INFO Add FIFO agent result:
${result}"/>
    <anti-action application="log" data="INFO User Login: ${user_name}"/>
    <anti-action application="sleep" data="600"/>
    <anti-action application="playback"
data="ivr/ivr-you_are_now_logged_in.wav"/>
    <anti-action application="set" data="result=${presence(in
queue-${queue_name}-${queue_user}@${domain_name} inuse inuse)}"/>
    <anti-action application="log" data="INFO Presence IN ${result}"/>
    <anti-action application="hangup"/>
  </condition>
</extension>


From patrick.plattes at niemann-frey.info  Fri Jan 28 16:47:31 2011
From: patrick.plattes at niemann-frey.info (Patrick Plattes)
Date: Fri, 28 Jan 2011 14:47:31 +0100
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTi=OC_jx0+nzK405VZURLDJPD25R4ZjyJ3RXJ1nk@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
	<AANLkTi=in3CLd-n0mT3SRKYA0kxLMGVWk9KeBx3_jhY6@mail.gmail.com>
	<AANLkTin3Bu+8EEr_v_GaRGNPzOjfcnfzsV2Av2cgnWgO@mail.gmail.com>
	<AANLkTi=OC_jx0+nzK405VZURLDJPD25R4ZjyJ3RXJ1nk@mail.gmail.com>
Message-ID: <AANLkTi=Sm7CjoayiE8nHmL0Eehqknj1jdpbx0YBQJQGH@mail.gmail.com>

Formatted and colourized code: http://pastebin.freeswitch.org/15172
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From vipkilla at gmail.com  Fri Jan 28 16:48:51 2011
From: vipkilla at gmail.com (vip killa)
Date: Fri, 28 Jan 2011 08:48:51 -0500
Subject: [Freeswitch-users] Portaudio Improvements
In-Reply-To: <AANLkTimyjXLgwf5_Vu092Wo2comyWWYEZLDx6rthH-+8@mail.gmail.com>
References: <AANLkTim9AmtEGZoK0NyG1yKVsmJ_MSwXAppi-6AcCqnz@mail.gmail.com>
	<AANLkTimm69f6adgkgDz03GUOq8xJw_oaVvrkAGxd1yjG@mail.gmail.com>
	<AANLkTikk==DKhXft67HWcj0ytya030bS=4vtt2o7+tD8@mail.gmail.com>
	<AANLkTi=T6a9nU54toi5+fMA7kq_xkmZrRnRT3Fa8Pao5@mail.gmail.com>
	<AANLkTimbULX-_ne4ZyirMFxcOcRBaEZwxTfude6=L8F_@mail.gmail.com>
	<AANLkTin_ue-i-Mw5CTQiZo3CTWi3Kcvt_4pdGYTS1jk7@mail.gmail.com>
	<AANLkTi=GRGJbQwE8yF=y_yUWJobUMnhdYfYpnGge5qf-@mail.gmail.com>
	<SNT124-W43801D7DD8A4B86A2A0B8EBBE10@phx.gbl>
	<AANLkTi=DQTDrFLv8HmMSOaL+PWKjg0B0htq+a=16QAUa@mail.gmail.com>
	<AANLkTimyjXLgwf5_Vu092Wo2comyWWYEZLDx6rthH-+8@mail.gmail.com>
Message-ID: <AANLkTimOqdOUm4M2V6nArZgncVjwT=Fn+7Z2KB=rmLnx@mail.gmail.com>

It would be very cool to be able to send audio to portaudio from another
program using something like PulseAudio. For instance I am running a
freeswitch box with no soundcard but i want to be able to send audio from a
program such as MPD to freeswitch. Could something like this be possible in
PA2?

2011/1/28 Jo?o Mesquita <jmesquita at freeswitch.org>

> Paulo,
>
> What is the current status of PA with non-blocking I/O? A little while ago
> it was not moving forward and pablio was the only way out of it. What are
> your plans? Maybe some of us can help?
>
> Regards,
> Jo?o Mesquita
>
>
>
> On Fri, Jan 28, 2011 at 9:54 AM, Tamas Jalsovszky <jalsot at gmail.com>wrote:
>
>> Wow, I didn't even know that somebody is working on PA2.
>> Paulo, could you share some more information about that module? Why would
>> somebody use it instead of the current one? Advantages/disadvantages?
>> Is the source code available somewhere? Do you have a roadmap for the
>> module? :)
>>
>> Regards,
>>   Tamas
>>
>>
>> On Fri, Jan 28, 2011 at 12:51 PM, Paulo Vicentini <vizentini at hotmail.com>wrote:
>>
>>>  Hi
>>> You can try PortMixer for volume control that is intended to work along
>>> with PortAudio
>>>
>>> I am developing/testing "mod_portaudio2" that does not rely on pablio,
>>> maybe things will be better with this approach
>>>
>>> Paulo
>>>
>>> ------------------------------
>>> From: mitch.capper at gmail.com
>>> Date: Sat, 22 Jan 2011 21:05:22 -0800
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] Portaudio Improvements
>>>
>>>
>>> Ah well viewing volume levels had not been something I had looked into.
>>> It wouldn't be too hard but the question is how do you constantly report
>>> that (FS event every second? every half second? port audio commands to
>>> return that value frequently?).  If your client is windows however again the
>>> native windows volume handler for apps does also keep track of their current
>>> output.
>>>
>>> ~Mitch
>>>
>>> On Sat, Jan 22, 2011 at 5:26 PM, Seven Du <dujinfang at gmail.com> wrote:
>>>
>>> Thanks Mitch, I actually had tried the equivalent uuid_audio and looks
>>> like it's the only way to do it for now. We had wanted to get energy
>>> levels dynamically to show a live indication but we given up that
>>> feature.
>>>
>>>
>>> On Sun, Jan 23, 2011 at 2:52 AM, Mitch Capper <mitch.capper at gmail.com>
>>> wrote:
>>> > It is true PA lacks volume control, it had been something I was
>>> debating.
>>> > Adding gain is pretty simple as you are for the most part just
>>> multiplying
>>> > up the samples I believe so would not be a major item to add to
>>> portaudio, I
>>> > however decided not to go this route for two reasons.  1) In Windows
>>> each
>>> > application has its own volume setting (Vista and higher),  and apps
>>> are
>>> > starting to tie into this than internal volume controls.  The downside
>>> is it
>>> > doesn't distort audio/have a gain option.   This is when I thought
>>> about
>>> > allowing for a fixed gain amount added to portaudio.  2) Freeswitch
>>> actually
>>> > has its own built in volume control settings with set_audio_level.
>>> > set_audio_level allows for 9 settings (4 lower 4 higher) and seems to
>>> do a
>>> > good job at digital volume control.
>>> > Avoiding modifying the audio stream itself could also result in less
>>> > distortion when using actual volume controls (headsets, etc).
>>> >
>>> > So if you want volume control I would say take a look at
>>> set_audio_level as
>>> > it may be the simplest method.   Adding it to portaudio would work but
>>> one
>>> > of the nice advantages of set_audio_level is the fact you can also do
>>> it per
>>> > channel rather than globally:)
>>> >
>>> > ~Mitch
>>> >
>>> > On Fri, Jan 21, 2011 at 5:44 PM, Seven Du <dujinfang at gmail.com> wrote:
>>> >>
>>> >> In addition to AEC. I also found PA lacks the ability to change the
>>> >> sound device volume. I tried to control sound devices in QT but
>>> >> haven't find a way. Ideas on this?
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> About: http://about.me/dujinfang
>>> Blog: http://www.dujinfang.com
>>> Proj:  http://www.freeswitch.org.cn
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________ FreeSWITCH-users mailing
>>> list FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kris at kriskinc.com  Fri Jan 28 17:29:23 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Fri, 28 Jan 2011 09:29:23 -0500
Subject: [Freeswitch-users] 200 OK without SDP
In-Reply-To: <201101280730.17483.sos@sokhapkin.dyndns.org>
References: <201101280730.17483.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTimkQMBVD7xL9uRkCYWNu-1ng+dDA73i6XBjexqu@mail.gmail.com>

I've seen some issues with using bypass_media and session timers.
Disable one or the other and see if it goes away.  I'm currently
trying to reproduce it in order to work on it with the FS devs.

On Fri, Jan 28, 2011 at 7:30 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org> wrote:
> What are possible reasons for FS to send 200 OK without SDP in response to SST
> reinvite? bypass_media=true, initial INVITE and initial 200 OK response have
> more than one codec (in different order) if this matters.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner


From vizentini at hotmail.com  Fri Jan 28 17:30:47 2011
From: vizentini at hotmail.com (Paulo Vicentini)
Date: Fri, 28 Jan 2011 14:30:47 +0000
Subject: [Freeswitch-users] Portaudio Improvements
In-Reply-To: <AANLkTimOqdOUm4M2V6nArZgncVjwT=Fn+7Z2KB=rmLnx@mail.gmail.com>
References: <AANLkTim9AmtEGZoK0NyG1yKVsmJ_MSwXAppi-6AcCqnz@mail.gmail.com>,
	<AANLkTimm69f6adgkgDz03GUOq8xJw_oaVvrkAGxd1yjG@mail.gmail.com>,
	<AANLkTikk==DKhXft67HWcj0ytya030bS=4vtt2o7+tD8@mail.gmail.com>,
	<AANLkTi=T6a9nU54toi5+fMA7kq_xkmZrRnRT3Fa8Pao5@mail.gmail.com>,
	<AANLkTimbULX-_ne4ZyirMFxcOcRBaEZwxTfude6=L8F_@mail.gmail.com>,
	<AANLkTin_ue-i-Mw5CTQiZo3CTWi3Kcvt_4pdGYTS1jk7@mail.gmail.com>,
	<AANLkTi=GRGJbQwE8yF=y_yUWJobUMnhdYfYpnGge5qf-@mail.gmail.com>,
	<SNT124-W43801D7DD8A4B86A2A0B8EBBE10@phx.gbl>,
	<AANLkTi=DQTDrFLv8HmMSOaL+PWKjg0B0htq+a=16QAUa@mail.gmail.com>,
	<AANLkTimyjXLgwf5_Vu092Wo2comyWWYEZLDx6rthH-+8@mail.gmail.com>,
	<AANLkTimOqdOUm4M2V6nArZgncVjwT=Fn+7Z2KB=rmLnx@mail.gmail.com>
Message-ID: <SNT124-W522E1BE2F03639F055CF60BBE10@phx.gbl>


HiCode is quite experimental but I intend to release it asap (probably within a few weeks)Let's see how it goes
RegardsPaulo
Date: Fri, 28 Jan 2011 08:48:51 -0500
From: vipkilla at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Portaudio Improvements

It would be very cool to be able to send audio to portaudio from another program using something like PulseAudio. For instance I am running a freeswitch box with no soundcard but i want to be able to send audio from a program such as MPD to freeswitch. Could something like this be possible in PA2?


2011/1/28 Jo?o Mesquita <jmesquita at freeswitch.org>

Paulo,
What is the current status of PA with non-blocking I/O? A little while ago it was not moving forward and pablio was the only way out of it. What are your plans? Maybe some of us can help?


Regards,Jo?o Mesquita



On Fri, Jan 28, 2011 at 9:54 AM, Tamas Jalsovszky <jalsot at gmail.com> wrote:


Wow, I didn't even know that somebody is working on PA2.
Paulo, could you share some more information about that module? Why would somebody use it instead of the current one? Advantages/disadvantages?
Is the source code available somewhere? Do you have a roadmap for the module? :)




Regards,
  Tamas

On Fri, Jan 28, 2011 at 12:51 PM, Paulo Vicentini <vizentini at hotmail.com> wrote:








HiYou can try PortMixer for volume control that is intended to work along with PortAudio 
I am developing/testing "mod_portaudio2" that does not rely on pablio, maybe things will be better with this approach


  Paulo
From: mitch.capper at gmail.com
Date: Sat, 22 Jan 2011 21:05:22 -0800
To: freeswitch-users at lists.freeswitch.org



Subject: Re: [Freeswitch-users] Portaudio Improvements

Ah well viewing volume levels had not been something I had looked into.  It wouldn't be too hard but the question is how do you constantly report that (FS event every second? every half second? port audio commands to return that value frequently?).  If your client is windows however again the native windows volume handler for apps does also keep track of their current output.   






~Mitch

On Sat, Jan 22, 2011 at 5:26 PM, Seven Du <dujinfang at gmail.com> wrote:





Thanks Mitch, I actually had tried the equivalent uuid_audio and looks

like it's the only way to do it for now. We had wanted to get energy

levels dynamically to show a live indication but we given up that

feature.





On Sun, Jan 23, 2011 at 2:52 AM, Mitch Capper <mitch.capper at gmail.com> wrote:

> It is true PA lacks volume control, it had been something I was debating.

> Adding gain is pretty simple as you are for the most part just multiplying

> up the samples I believe so would not be a major item to add to portaudio, I

> however decided not to go this route for two reasons.  1) In Windows each

> application has its own volume setting (Vista and higher),  and apps are

> starting to tie into this than internal volume controls.  The downside is it

> doesn't distort audio/have a gain option.   This is when I thought about

> allowing for a fixed gain amount added to portaudio.  2) Freeswitch actually

> has its own built in volume control settings with set_audio_level.

> set_audio_level allows for 9 settings (4 lower 4 higher) and seems to do a

> good job at digital volume control.

> Avoiding modifying the audio stream itself could also result in less

> distortion when using actual volume controls (headsets, etc).

>

> So if you want volume control I would say take a look at set_audio_level as

> it may be the simplest method.   Adding it to portaudio would work but one

> of the nice advantages of set_audio_level is the fact you can also do it per

> channel rather than globally:)

>

> ~Mitch

>

> On Fri, Jan 21, 2011 at 5:44 PM, Seven Du <dujinfang at gmail.com> wrote:

>>

>> In addition to AEC. I also found PA lacks the ability to change the

>> sound device volume. I tried to control sound devices in QT but

>> haven't find a way. Ideas on this?

>

>

> _______________________________________________

> FreeSWITCH-users mailing list

> FreeSWITCH-users at lists.freeswitch.org

> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

> http://www.freeswitch.org

>

>







--

About: http://about.me/dujinfang

Blog: http://www.dujinfang.com

Proj:  http://www.freeswitch.org.cn



_______________________________________________

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From sos at sokhapkin.dyndns.org  Fri Jan 28 17:37:50 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 28 Jan 2011 09:37:50 -0500
Subject: [Freeswitch-users] 200 OK without SDP
In-Reply-To: <AANLkTimkQMBVD7xL9uRkCYWNu-1ng+dDA73i6XBjexqu@mail.gmail.com>
References: <201101280730.17483.sos@sokhapkin.dyndns.org>
	<AANLkTimkQMBVD7xL9uRkCYWNu-1ng+dDA73i6XBjexqu@mail.gmail.com>
Message-ID: <201101280937.50415.sos@sokhapkin.dyndns.org>

Unfortunately stay in audio path or disable SST is not an option to me.To my 
experience, the problem happens when leg B answers the call with more than 1 
codec in 200 OK response.


On Friday 28 January 2011, Kristian Kielhofner wrote:
> I've seen some issues with using bypass_media and session timers.
> Disable one or the other and see if it goes away.  I'm currently
> trying to reproduce it in order to work on it with the FS devs.
> 
> On Fri, Jan 28, 2011 at 7:30 AM, Sergey Okhapkin
> 
> <sos at sokhapkin.dyndns.org> wrote:
> > What are possible reasons for FS to send 200 OK without SDP in response
> > to SST reinvite? bypass_media=true, initial INVITE and initial 200 OK
> > response have more than one codec (in different order) if this matters.
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org



From helmut.kuper at ewetel.de  Fri Jan 28 18:09:25 2011
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Fri, 28 Jan 2011 16:09:25 +0100
Subject: [Freeswitch-users] intercom in lua dialplan
In-Reply-To: <4D3FD1D9.2@ewetel.de>
References: <4D3EF036.80701@ewetel.de>	<AANLkTik8jrN_hzom=VEQZfx39YqMgZ15OE8kzZWqJ8iO@mail.gmail.com>
	<4D3FD1D9.2@ewetel.de>
Message-ID: <4D42DC25.4090407@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Michael,

any news to this?


Am 26.01.2011 08:48, schrieb Helmut Kuper:
> Hi Michael,
> 
> you can find the complete log of this process in pastebin:
> http://pastebin.freeswitch.org/15150
> 
> 44180000 is A-Party calling 8367
> 8000 is C-Party which tries to intercept via **8367
> 
regards Helmut
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.10 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/

iEYEARECAAYFAk1C3CUACgkQ4tZeNddg3dxcPwCeJn0A2ypFGvaTU3iwbNAXbWSU
udcAn07r3ugbs5oBaVptI0DCJYAmPmG0
=u6Gp
-----END PGP SIGNATURE-----


From jeff at jefflenk.com  Fri Jan 28 18:25:29 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 28 Jan 2011 07:25:29 -0800 (PST)
Subject: [Freeswitch-users] mod_dingaling and the weekly windows build
In-Reply-To: <8C68232BC9314C40BBCDDAA480F7B01AEA55E1F8B6@harrow.exch.ad.byu.edu>
References: <8C68232BC9314C40BBCDDAA480F7B01AEA3B493B34@harrow.exch.ad.byu.edu>
	<1296150184550-5967186.post@n2.nabble.com>
	<8C68232BC9314C40BBCDDAA480F7B01AEA55E1F8B6@harrow.exch.ad.byu.edu>
Message-ID: <1296228329860-5970197.post@n2.nabble.com>


Its possible that mod_dingaling could be included but not with
GNUTLS(required for GV) builtin because thats not license compatible.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-dingaling-and-the-weekly-windows-build-tp5965119p5970197.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From imthiyaz at peopletech.co.in  Fri Jan 28 19:17:52 2011
From: imthiyaz at peopletech.co.in (Imthiyaz Ahmed)
Date: Fri, 28 Jan 2011 21:47:52 +0530
Subject: [Freeswitch-users] DI-TE122P Digium Single Span T1/E1 - PRI Card
In-Reply-To: <8C543835-EC2F-4536-B6BE-7F1A458C7281@gmail.com>
References: <8C543835-EC2F-4536-B6BE-7F1A458C7281@gmail.com>
Message-ID: <AANLkTi=DBvLNc8KLcOv6pWrEf+edUERNHtYMxDiQ3hQD@mail.gmail.com>

Hi Please refer

Synway offers some installation guidlines for T1/E1 card installation on
freeswitch
http://www.synway.net/Products/index_xx.aspx?id=59
On Fri, Jan 28, 2011 at 6:53 PM, Deon Vermeulen <vermeulen.deon at gmail.com>wrote:

> Hi List
>
> I'm looking at purchasing this card and using it in my FS Server.
>
> Is this in used in production by anyone?
>
> Is there a guide on how to setup this card in FS?
>
>
> Thank you very very much
>
>
> Regards
> Deon Vermeulen
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Best Regards
G.Imthiyaz Ahmed
PeopleTech systems (P) ltd
http://peopletech.co.in
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From anthony.minessale at gmail.com  Fri Jan 28 19:34:40 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 Jan 2011 10:34:40 -0600
Subject: [Freeswitch-users] Portaudio Improvements
In-Reply-To: <SNT124-W522E1BE2F03639F055CF60BBE10@phx.gbl>
References: <AANLkTim9AmtEGZoK0NyG1yKVsmJ_MSwXAppi-6AcCqnz@mail.gmail.com>
	<AANLkTimm69f6adgkgDz03GUOq8xJw_oaVvrkAGxd1yjG@mail.gmail.com>
	<AANLkTikk==DKhXft67HWcj0ytya030bS=4vtt2o7+tD8@mail.gmail.com>
	<AANLkTi=T6a9nU54toi5+fMA7kq_xkmZrRnRT3Fa8Pao5@mail.gmail.com>
	<AANLkTimbULX-_ne4ZyirMFxcOcRBaEZwxTfude6=L8F_@mail.gmail.com>
	<AANLkTin_ue-i-Mw5CTQiZo3CTWi3Kcvt_4pdGYTS1jk7@mail.gmail.com>
	<AANLkTi=GRGJbQwE8yF=y_yUWJobUMnhdYfYpnGge5qf-@mail.gmail.com>
	<SNT124-W43801D7DD8A4B86A2A0B8EBBE10@phx.gbl>
	<AANLkTi=DQTDrFLv8HmMSOaL+PWKjg0B0htq+a=16QAUa@mail.gmail.com>
	<AANLkTimyjXLgwf5_Vu092Wo2comyWWYEZLDx6rthH-+8@mail.gmail.com>
	<AANLkTimOqdOUm4M2V6nArZgncVjwT=Fn+7Z2KB=rmLnx@mail.gmail.com>
	<SNT124-W522E1BE2F03639F055CF60BBE10@phx.gbl>
Message-ID: <AANLkTime4o8CTqQ5uuOCC=0dcRkXgDQZsD+NGUSpRC_r@mail.gmail.com>

It would have been best to share it all along and let it be in contrib
to get testers and contributors.  There are many developers here who
can give guidance etc.  That's why we do this in open source.


On Fri, Jan 28, 2011 at 8:30 AM, Paulo Vicentini <vizentini at hotmail.com> wrote:
> Hi
> Code is quite experimental but I intend to release it asap (probably within
> a few weeks)
> Let's see how it goes
> Regards
> Paulo
> ________________________________
> Date: Fri, 28 Jan 2011 08:48:51 -0500
> From: vipkilla at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Portaudio Improvements
>
> It would be very cool to be able to send audio to portaudio from another
> program using something like PulseAudio. For instance I am running a
> freeswitch box with no soundcard but i want to be able to send audio from a
> program such as MPD to freeswitch. Could something like this be possible in
> PA2?
>
> 2011/1/28 Jo?o Mesquita <jmesquita at freeswitch.org>
>
> Paulo,
> What is the current status of PA with non-blocking I/O? A little while ago
> it was not moving forward and pablio was the only way out of it. What are
> your plans? Maybe some of us can help?
> Regards,
> Jo?o Mesquita
>
>
> On Fri, Jan 28, 2011 at 9:54 AM, Tamas Jalsovszky <jalsot at gmail.com> wrote:
>
> Wow, I didn't even know that somebody is working on PA2.
> Paulo, could you share some more information about that module? Why would
> somebody use it instead of the current one? Advantages/disadvantages?
> Is the source code available somewhere? Do you have a roadmap for the
> module? :)
>
> Regards,
> ? Tamas
>
> On Fri, Jan 28, 2011 at 12:51 PM, Paulo Vicentini <vizentini at hotmail.com>
> wrote:
>
> Hi
> You can try PortMixer for volume control that is intended to work along with
> PortAudio
> I am developing/testing "mod_portaudio2" that does not rely on pablio, maybe
> things will be better with this approach
>
> Paulo
> ________________________________
> From: mitch.capper at gmail.com
> Date: Sat, 22 Jan 2011 21:05:22 -0800
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Portaudio Improvements
>
> Ah well viewing volume levels had not been something I had looked into.? It
> wouldn't be too hard but the question is how do you constantly report that
> (FS event every second? every half second? port audio commands to return
> that value frequently?).? If your client is windows however again the native
> windows volume handler for apps does also keep track of their current
> output.
>
> ~Mitch
>
> On Sat, Jan 22, 2011 at 5:26 PM, Seven Du <dujinfang at gmail.com> wrote:
>
> Thanks Mitch, I actually had tried the equivalent uuid_audio and looks
> like it's the only way to do it for now. We had wanted to get energy
> levels dynamically to show a live indication but we given up that
> feature.
>
>
> On Sun, Jan 23, 2011 at 2:52 AM, Mitch Capper <mitch.capper at gmail.com>
> wrote:
>> It is true PA lacks volume control, it had been something I was debating.
>> Adding gain is pretty simple as you are for the most part just multiplying
>> up the samples I believe so would not be a major item to add to portaudio,
>> I
>> however decided not to go this route for two reasons.? 1) In Windows each
>> application has its own volume setting (Vista and higher),? and apps are
>> starting to tie into this than internal volume controls.? The downside is
>> it
>> doesn't distort audio/have a gain option.?? This is when I thought about
>> allowing for a fixed gain amount added to portaudio.? 2) Freeswitch
>> actually
>> has its own built in volume control settings with set_audio_level.
>> set_audio_level allows for 9 settings (4 lower 4 higher) and seems to do a
>> good job at digital volume control.
>> Avoiding modifying the audio stream itself could also result in less
>> distortion when using actual volume controls (headsets, etc).
>>
>> So if you want volume control I would say take a look at set_audio_level
>> as
>> it may be the simplest method.?? Adding it to portaudio would work but one
>> of the nice advantages of set_audio_level is the fact you can also do it
>> per
>> channel rather than globally:)
>>
>> ~Mitch
>>
>> On Fri, Jan 21, 2011 at 5:44 PM, Seven Du <dujinfang at gmail.com> wrote:
>>>
>>> In addition to AEC. I also found PA lacks the ability to change the
>>> sound device volume. I tried to control sound devices in QT but
>>> haven't find a way. Ideas on this?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:? http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Fri Jan 28 19:37:27 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 Jan 2011 10:37:27 -0600
Subject: [Freeswitch-users] Choppy VM when using loopback (on Debian
	lenny)
In-Reply-To: <4D42B8EE.7030208@estation.dk>
References: <4D42B8EE.7030208@estation.dk>
Message-ID: <AANLkTikLg73Tg+8LqA6J2C_LN7J181unM1bNROC8UL94@mail.gmail.com>

Do you realize the latest GIT is an hourly thing?  What revision do
you mean by latest?
We do not support anything but 1000hz kernels our platform of choice
is CentOS 5.2/5.3
Anything else you are experimenting......


On Fri, Jan 28, 2011 at 6:39 AM, ?yvind Albrigtsen <oa at estation.dk> wrote:
> Hi,
>
> I'm using latest git-version of Freeswitch, and when I go to voicemail
> when calling a number the sound playback is choppy and it skips some of
> the digits in the number I called.
>
> If I set sleep to 2000 it sounds better, but it's still a little choppy
> and still skips some of the digits.
>
> This is with the default settings coming with the git version. When I
> call 4000 to listen to voicemail it sounds fine.
>
> The kernel is the default kernel which is compiled with CONFIG_HZ_250=y
>
> Regards
> Oyvind
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
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FreeSWITCH Developer Conference
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pstn:+19193869900


From infos at madovsky.org  Fri Jan 28 19:42:52 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 28 Jan 2011 11:42:52 -0500
Subject: [Freeswitch-users] Send email on successful fax sending.
References: <AANLkTinJ3N+zwg+ZK2RmdAZECEkLgcvOSKT0hf_mbpix@mail.gmail.com><AANLkTi=5g-rOZtS9Dy4wq_J6OQU2roLzBDwhzhsuhQSG@mail.gmail.com><4D3976FD.4090805@chaschperli.ch><AANLkTimo0n3UP2c09F3AmgQdkRdU+DVAMxjYrw7cP9gm@mail.gmail.com><AANLkTi=zBgX-trHwVUUgfGNZh3SmcJfru1tD=5-aE=MY@mail.gmail.com><AANLkTimKgZDpuDUgNaqT1=61mqwOoJ66u4m-u4LBjxXM@mail.gmail.com><AANLkTin7M1H_Rxeet8QC-JcqkupLmVYhONAijXkuQ0-u@mail.gmail.com>
	<AANLkTi=2sMwKKRk3gu35KfMZ52Kow4CxViU=ZTtKmR1X@mail.gmail.com>
Message-ID: <AE5DB81B491B4313948566DF8BA89124@e1705>

That's a good 
  ----- Original Message ----- 
  From: Marc de Corny 
  To: FreeSWITCH Users Help 
  Sent: Friday, January 28, 2011 8:25 AM
  Subject: Re: [Freeswitch-users] Send email on successful fax sending.


  fantastic.
  let me give it a go and see.

  thanks for your help guys
  Marc


  On Fri, Jan 28, 2011 at 8:32 AM, Steven Ayre <steveayre at gmail.com> wrote:

    Hi Marc,

    Brian's committed this patch to the trunk 27/1/2011 in 314a2a1e.

    You'd now be able to connect via ESL, do originate and then wait for the spandsp::txfaxresult event, then send the success/failure email.

    -Steve



    On 21 January 2011 13:10, Steven Ayre <steveayre at gmail.com> wrote:

      I've just put a patch together for mod_spandsp that should add the required event.

      It's on Jira, FS-3004.
      http://jira.freeswitch.org/browse/FS-3004

      I've checked it compiles, but I have no T38 capable gateways so someone else will need to test it out. Does anyone fancy trying it out?

      -Steve 





      On 21 January 2011 12:40, Marc de Corny <marcdecorny at gmail.com> wrote:

        Thanks, all I have not investigated the ESL options yet, I will have a look and see,


        many thanks for the responses,

        Marc


        On Fri, Jan 21, 2011 at 12:33 PM, Steven Ayre <steveayre at gmail.com> wrote:

          But since the channel will hangup, you'd need to wait for and check the CDR to see that variable.

          mod_spandsp_fax.c:412 is "TODO Fire events"

          It looks like that it's planned for mod_spandsp to fire an event that'll indicate that the fax has finished sending and would contain the status.

          If that was implemented, you could connect via ESL, do the originate and wait for the event to tell you whether it worked or not.

          -Steve




          On 21 January 2011 12:07, Thomas Mueller <thomas at chaschperli.ch> wrote:

            On 21.01.2011 09:14, Marc de Corny wrote:
            > Hi There,
            > Has anybody had any ideas on this ? I imagine you must all have the
            > same requirement in the Email to Fax scenario ?



            reading the sourcecode the channel var "fax_success" gets set to 1 for
            OK, to 0 for fail.

            - Thomas




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From anthony.minessale at gmail.com  Fri Jan 28 19:49:34 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 Jan 2011 10:49:34 -0600
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTi=Sm7CjoayiE8nHmL0Eehqknj1jdpbx0YBQJQGH@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
	<AANLkTi=in3CLd-n0mT3SRKYA0kxLMGVWk9KeBx3_jhY6@mail.gmail.com>
	<AANLkTin3Bu+8EEr_v_GaRGNPzOjfcnfzsV2Av2cgnWgO@mail.gmail.com>
	<AANLkTi=OC_jx0+nzK405VZURLDJPD25R4ZjyJ3RXJ1nk@mail.gmail.com>
	<AANLkTi=Sm7CjoayiE8nHmL0Eehqknj1jdpbx0YBQJQGH@mail.gmail.com>
Message-ID: <AANLkTinr0RdgBs2UiS3=b3i+k6uCwkMihDy_FbAOFPc_@mail.gmail.com>

try with no presence_in and out commands at all

just set presence_id variable as soon as you can and that call will
report its state as that id for its entire duration.
Also make sure you are on latest GIT when you test this.


On Fri, Jan 28, 2011 at 7:47 AM, Patrick Plattes
<patrick.plattes at niemann-frey.info> wrote:
> Formatted and colourized code: http://pastebin.freeswitch.org/15172
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
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FreeSWITCH Developer Conference
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pstn:+19193869900


From anthony.minessale at gmail.com  Fri Jan 28 19:53:03 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 Jan 2011 10:53:03 -0600
Subject: [Freeswitch-users] blocking 183 w/o sdp
In-Reply-To: <AANLkTinH1s+Uy-EPvX9oV-x+aXYkyzUEBSYm+hQ+-+K-@mail.gmail.com>
References: <AANLkTinH1s+Uy-EPvX9oV-x+aXYkyzUEBSYm+hQ+-+K-@mail.gmail.com>
Message-ID: <AANLkTikJpEPvcQu2reAKXk1wbW7fV4myvXghBCuu2QHt@mail.gmail.com>

you need sip_ignore_183nosdp=true set on the b leg not the a leg.
Put it in the dial string in {}

{sip_ignore_183nosdp=true}sofia/foo/foo at bar.com


On Fri, Jan 28, 2011 at 12:41 AM, Sam <u2nsam at gmail.com> wrote:
> Hi,
>
> how can i ignore 183 without sdp,
> what happens is the provider sends 183 without sdp and by applying "<action
> application="set" data="sip_ignore_183nosdp=true"/>"? the FS sends 180 to
> the leg a.
> Here i want to block the 183 with SDP just like router as b2bua and send
> nothing to leg a, and when actual 183 with sdp comes it should send .
>
> Its because, providers are sending false signaling by sending 183 without
> sdp,and it hampers while @ production,
> Although by cisco sbc i have done this but i want to do it by FS,
> Take a scenario, when call is send 183 without sdp for 10 secs and then
> followed by 183 with sdp ( actual signal),
> but when some one dials invalid number it rings for 10 secs and then gives
> SIP cause 404, which is bad from the providers.
> So this is the reason i want to block it.
>
> Most of the providers do this, the way out is blocking.
>
> I have got an advice from Tihomir? to do "execute_on_ring and parse your 180
> / 183 messages in search of SDP,
> once you get 183 without SDP do not send it back to leg a and send signal
> only when you got 183 with sdp or 180 "
> And this could be valid call flow.
>
> This happens in many cases where the provider is having nextone as a sbc and
> that too tier 1 !
>
> Regards
> Sam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
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FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From infos at madovsky.org  Fri Jan 28 20:12:58 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 28 Jan 2011 12:12:58 -0500
Subject: [Freeswitch-users] execute_on_ring executing on answer
References: <1296215265.8986.147.camel@luna.tc.commsmundi.com>
Message-ID: <21B4FB4AAC2E428E938F99986CA09165@e1705>

what means the ^^ in your codec string ?

----- Original Message ----- 
From: "Fran?ois Delawarde" <fdelawarde at wirelessmundi.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, January 28, 2011 6:47 AM
Subject: [Freeswitch-users] execute_on_ring executing on answer


> Hi,
>
> Doing some testing with this morning's git (Fri Jan 28) I just found out
> that the execute_on_ring application runs when the destination answers
> instead of when it rings.
>
> So far, I can't seem to find out the reason. Could it be some
> configuration issue?
>
>
> Here a call log showing the phenomenon with a simple bridge:
>
> <action application="bridge" 
> data="[execute_on_ring=info,absolute_codec_string=^^:PCMA:G722]sofia/192.168.10.1/sip:2103 at 192.168.10.22:5060"/>
>
> http://pastebin.freeswitch.org/15168
>
>
> Thanks,
> Fran?ois.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From infos at madovsky.org  Fri Jan 28 20:13:35 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 28 Jan 2011 12:13:35 -0500
Subject: [Freeswitch-users] cancel a non answered bridge
Message-ID: <FED3BE97C223432DA8B2A4026E11F2CB@e1705>

How to cancel ok kil a non answered bridge ?

Thanks
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From infos at madovsky.org  Fri Jan 28 20:14:21 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 28 Jan 2011 12:14:21 -0500
Subject: [Freeswitch-users] cancel a non answered bridge
Message-ID: <1BCB0A82BAED443AA3579A62257A6C68@e1705>

Sorry I forgot

without to hangup ?
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, January 28, 2011 12:13 PM
  Subject: cancel a non answered bridge


  How to cancel ok kil a non answered bridge ?

  Thanks
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From fdelawarde at wirelessmundi.com  Fri Jan 28 20:18:15 2011
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Fri, 28 Jan 2011 18:18:15 +0100
Subject: [Freeswitch-users] execute_on_ring executing on answer
In-Reply-To: <21B4FB4AAC2E428E938F99986CA09165@e1705>
References: <1296215265.8986.147.camel@luna.tc.commsmundi.com>
	<21B4FB4AAC2E428E938F99986CA09165@e1705>
Message-ID: <1296235095.8986.167.camel@luna.tc.commsmundi.com>

It's some cool feature made by Anthony that allows me to specify the
separator.

in ^^:PCMA:G722
^^: means the separator is now : instead of ,

Useful in the [] or {} case because the coma is already used to separate
variables.

Fran?ois.

On Fri, 2011-01-28 at 12:12 -0500, Madovsky wrote:
> what means the ^^ in your codec string ?
> 
> ----- Original Message ----- 
> From: "Fran?ois Delawarde" <fdelawarde at wirelessmundi.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Friday, January 28, 2011 6:47 AM
> Subject: [Freeswitch-users] execute_on_ring executing on answer
> 
> 
> > Hi,
> >
> > Doing some testing with this morning's git (Fri Jan 28) I just found out
> > that the execute_on_ring application runs when the destination answers
> > instead of when it rings.
> >
> > So far, I can't seem to find out the reason. Could it be some
> > configuration issue?
> >
> >
> > Here a call log showing the phenomenon with a simple bridge:
> >
> > <action application="bridge" 
> > data="[execute_on_ring=info,absolute_codec_string=^^:PCMA:G722]sofia/192.168.10.1/sip:2103 at 192.168.10.22:5060"/>
> >
> > http://pastebin.freeswitch.org/15168
> >
> >
> > Thanks,
> > Fran?ois.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From infos at madovsky.org  Fri Jan 28 20:24:04 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 28 Jan 2011 12:24:04 -0500
Subject: [Freeswitch-users] execute_on_ring executing on answer
References: <1296215265.8986.147.camel@luna.tc.commsmundi.com><21B4FB4AAC2E428E938F99986CA09165@e1705>
	<1296235095.8986.167.camel@luna.tc.commsmundi.com>
Message-ID: <0E96944DB29F4CEB9ACA5994D19AB0D6@e1705>

ah ok, maybe a wiki update would be useful.



----- Original Message ----- 
From: "Fran?ois Delawarde" <fdelawarde at wirelessmundi.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, January 28, 2011 12:18 PM
Subject: Re: [Freeswitch-users] execute_on_ring executing on answer


> It's some cool feature made by Anthony that allows me to specify the
> separator.
>
> in ^^:PCMA:G722
> ^^: means the separator is now : instead of ,
>
> Useful in the [] or {} case because the coma is already used to separate
> variables.
>
> Fran?ois.
>
> On Fri, 2011-01-28 at 12:12 -0500, Madovsky wrote:
>> what means the ^^ in your codec string ?
>>
>> ----- Original Message ----- 
>> From: "Fran?ois Delawarde" <fdelawarde at wirelessmundi.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Friday, January 28, 2011 6:47 AM
>> Subject: [Freeswitch-users] execute_on_ring executing on answer
>>
>>
>> > Hi,
>> >
>> > Doing some testing with this morning's git (Fri Jan 28) I just found 
>> > out
>> > that the execute_on_ring application runs when the destination answers
>> > instead of when it rings.
>> >
>> > So far, I can't seem to find out the reason. Could it be some
>> > configuration issue?
>> >
>> >
>> > Here a call log showing the phenomenon with a simple bridge:
>> >
>> > <action application="bridge"
>> > data="[execute_on_ring=info,absolute_codec_string=^^:PCMA:G722]sofia/192.168.10.1/sip:2103 at 192.168.10.22:5060"/>
>> >
>> > http://pastebin.freeswitch.org/15168
>> >
>> >
>> > Thanks,
>> > Fran?ois.
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From david.ponzone at ipeva.fr  Fri Jan 28 20:27:03 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 28 Jan 2011 18:27:03 +0100
Subject: [Freeswitch-users] cancel a non answered bridge
In-Reply-To: <1BCB0A82BAED443AA3579A62257A6C68@e1705>
References: <1BCB0A82BAED443AA3579A62257A6C68@e1705>
Message-ID: <4D8FDEDE-BDE8-4DEE-AD18-B11C70D4A061@ipeva.fr>

Sorry, I fail to get the question...
Perhaps we need more details like: where do you want to do that ? from XML dialplan ? from script ?
From API, you can uuid_kill.
From XML, I am not sure I see what that would mean...perhaps you need to define a timeout, so that the bridge attempt ends after X seconds ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 28/01/2011 ? 18:14, Madovsky a ?crit :

> Sorry I forgot
>  
> without to hangup ?
> ----- Original Message -----
> From: Madovsky
> To: freeswitch-users at lists.freeswitch.org
> Sent: Friday, January 28, 2011 12:13 PM
> Subject: cancel a non answered bridge
> 
> How to cancel ok kil a non answered bridge ?
>  
> Thanks
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From infos at madovsky.org  Fri Jan 28 21:00:08 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 28 Jan 2011 13:00:08 -0500
Subject: [Freeswitch-users] cancel a non answered bridge
References: <1BCB0A82BAED443AA3579A62257A6C68@e1705>
	<4D8FDEDE-BDE8-4DEE-AD18-B11C70D4A061@ipeva.fr>
Message-ID: <11C441DB202F4E688C65A2E5C8153DFC@e1705>

the situation :

Aleg call external gateway
in dialplan I set on the fly the nibble_rate (nibble_account is already set)
and just after make a bridge to the gateway.
so nibblebill tries to ask DB if cahs is ok (so it takes a little time) but in the mean
time the bridge is created and Bleg is ringing.
nibblebill realize that cash is below the amount allowed and redirect to nofunds extension.

I'd like to avoid the Bleg ringing by kill the bridge (but not hanup Aleg) once the nofunds extension is reached.

Thanks
  ----- Original Message ----- 
  From: David Ponzone 
  To: FreeSWITCH Users Help 
  Sent: Friday, January 28, 2011 12:27 PM
  Subject: Re: [Freeswitch-users] cancel a non answered bridge


  Sorry, I fail to get the question...
  Perhaps we need more details like: where do you want to do that ? from XML dialplan ? from script ?
  From API, you can uuid_kill.
  From XML, I am not sure I see what that would mean...perhaps you need to define a timeout, so that the bridge attempt ends after X seconds ?


  David Ponzone  Direction Technique
  email: david.ponzone at ipeva.fr
  tel:      01 74 03 18 97
  gsm:   06 66 98 76 34


  Service Client IPeva
  tel:      0811 46 26 26
  www.ipeva.fr  -   www.ipeva-studio.com


  Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.







  Le 28/01/2011 ? 18:14, Madovsky a ?crit :


    Sorry I forgot

    without to hangup ?
      ----- Original Message -----
      From: Madovsky
      To: freeswitch-users at lists.freeswitch.org
      Sent: Friday, January 28, 2011 12:13 PM
      Subject: cancel a non answered bridge


      How to cancel ok kil a non answered bridge ?

      Thanks
    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





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From anthony.minessale at gmail.com  Fri Jan 28 21:34:15 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 Jan 2011 12:34:15 -0600
Subject: [Freeswitch-users] execute_on_ring executing on answer
In-Reply-To: <0E96944DB29F4CEB9ACA5994D19AB0D6@e1705>
References: <1296215265.8986.147.camel@luna.tc.commsmundi.com>
	<21B4FB4AAC2E428E938F99986CA09165@e1705>
	<1296235095.8986.167.camel@luna.tc.commsmundi.com>
	<0E96944DB29F4CEB9ACA5994D19AB0D6@e1705>
Message-ID: <AANLkTikHB-QnDyT0WPsEML8yv4AACnYnHrE37gpRsLS2@mail.gmail.com>

if it never rings, answer will still trigger it.


On Fri, Jan 28, 2011 at 11:24 AM, Madovsky <infos at madovsky.org> wrote:
> ah ok, maybe a wiki update would be useful.
>
>
>
> ----- Original Message -----
> From: "Fran?ois Delawarde" <fdelawarde at wirelessmundi.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Friday, January 28, 2011 12:18 PM
> Subject: Re: [Freeswitch-users] execute_on_ring executing on answer
>
>
>> It's some cool feature made by Anthony that allows me to specify the
>> separator.
>>
>> in ^^:PCMA:G722
>> ^^: means the separator is now : instead of ,
>>
>> Useful in the [] or {} case because the coma is already used to separate
>> variables.
>>
>> Fran?ois.
>>
>> On Fri, 2011-01-28 at 12:12 -0500, Madovsky wrote:
>>> what means the ^^ in your codec string ?
>>>
>>> ----- Original Message -----
>>> From: "Fran?ois Delawarde" <fdelawarde at wirelessmundi.com>
>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>> Sent: Friday, January 28, 2011 6:47 AM
>>> Subject: [Freeswitch-users] execute_on_ring executing on answer
>>>
>>>
>>> > Hi,
>>> >
>>> > Doing some testing with this morning's git (Fri Jan 28) I just found
>>> > out
>>> > that the execute_on_ring application runs when the destination answers
>>> > instead of when it rings.
>>> >
>>> > So far, I can't seem to find out the reason. Could it be some
>>> > configuration issue?
>>> >
>>> >
>>> > Here a call log showing the phenomenon with a simple bridge:
>>> >
>>> > <action application="bridge"
>>> > data="[execute_on_ring=info,absolute_codec_string=^^:PCMA:G722]sofia/192.168.10.1/sip:2103 at 192.168.10.22:5060"/>
>>> >
>>> > http://pastebin.freeswitch.org/15168
>>> >
>>> >
>>> > Thanks,
>>> > Fran?ois.
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From frank at carmickle.com  Fri Jan 28 21:49:51 2011
From: frank at carmickle.com (Frank Carmickle)
Date: Fri, 28 Jan 2011 13:49:51 -0500
Subject: [Freeswitch-users] Choppy VM when using loopback (on Debian
	lenny)
In-Reply-To: <4D42B8EE.7030208@estation.dk>
References: <4D42B8EE.7030208@estation.dk>
Message-ID: <7C6F3FC7-445F-4157-BD59-D76A705E4CE7@carmickle.com>

Hi

On Jan 28, 2011, at 7:39 AM, ?yvind Albrigtsen wrote:

> Hi,
> 
> I'm using latest git-version of Freeswitch, and when I go to voicemail 
> when calling a number the sound playback is choppy and it skips some of 
> the digits in the number I called.
> 
What kind of results do you get from timer_test at the fs_cli?  Are you running on hardware or are you virtualized?  What is your clock source set to and what are your available clock source options?  See /sys/devices/system/clocksource/clocksource0/available_clocksource and /sys/devices/system/clocksource/clocksource0/current_clocksource.  I am running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to hang at 19998/19999 which works very well for me.  When I was having problem it was reporting numbers all over the map from 17400 to 22600 with lots of randomness in between.  I have my clocksource set to jiffies and xen independent wallclock set to 1.  Of course at that point you need to have ntp running against a bunch of servers to drive your clock nice and steady.  I know my set up is probably a lot different than yours but I thought I'd toss it out there to show that some of the harshest conditions can be dealt with and don't give up trying.  If you are running on hardware with a cpu that doesn't have constant_tsc then you might have some problems.  Just play with the different timer options until you find the one that works.

HTH
--FC



From frank at carmickle.com  Fri Jan 28 22:03:25 2011
From: frank at carmickle.com (Frank Carmickle)
Date: Fri, 28 Jan 2011 14:03:25 -0500
Subject: [Freeswitch-users] Virtualisation and freeswitch
In-Reply-To: <AANLkTinUFPDM8P4oU4SZ-DE4rd4Qg3fHQG-W+iQt5JXv@mail.gmail.com>
References: <AANLkTimZTwZfUYMjC1ddAQMWHHWSLH=BLajv0cgSux4B@mail.gmail.com>
	<AANLkTinUFPDM8P4oU4SZ-DE4rd4Qg3fHQG-W+iQt5JXv@mail.gmail.com>
Message-ID: <F2168A27-ADEA-457E-8769-0285C4A52477@carmickle.com>

Hi

On Jan 28, 2011, at 8:33 AM, Avi Marcus wrote:

> I'm using Linode's VPS which is run on xen, and all tests (80 moh channels, playback into a conference) and light usage have been perfect.
> Some have complained of Amazon's high latency, others say it's been an awesome platform for them.
> 
Can second this.  Please see my other post just now on timing.  I was running fine for a year before some changes to the kernel that I was running made it become a problem.  I was able to hunt it down and fix it.

--FC



From frank at carmickle.com  Fri Jan 28 22:05:13 2011
From: frank at carmickle.com (Frank Carmickle)
Date: Fri, 28 Jan 2011 14:05:13 -0500
Subject: [Freeswitch-users] time_test on Centos 5.5
In-Reply-To: <0E4A540B-3E61-4940-8246-6FBF67CF91D8@ipeva.fr>
References: <AANLkTiksCg70iEdZiN1+CnfyBW87mg7T4jDHjeq7JW5Y@mail.gmail.com>
	<0E4A540B-3E61-4940-8246-6FBF67CF91D8@ipeva.fr>
Message-ID: <41F744A2-90FF-405F-AF62-5E7B8FB5128F@carmickle.com>


On Jan 28, 2011, at 6:35 AM, David Ponzone wrote:
Snip...

> Stay away from Debian, Centos is the right choice.
> You could eventually try to fallback to centos 5.3 or 5.4.
> 
Debian can work if that's what people want to use.  I have it working well on a few lenny machines.

--FC




From anthony.minessale at gmail.com  Fri Jan 28 22:33:01 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 Jan 2011 13:33:01 -0600
Subject: [Freeswitch-users] Virtualisation and freeswitch
In-Reply-To: <F2168A27-ADEA-457E-8769-0285C4A52477@carmickle.com>
References: <AANLkTimZTwZfUYMjC1ddAQMWHHWSLH=BLajv0cgSux4B@mail.gmail.com>
	<AANLkTinUFPDM8P4oU4SZ-DE4rd4Qg3fHQG-W+iQt5JXv@mail.gmail.com>
	<F2168A27-ADEA-457E-8769-0285C4A52477@carmickle.com>
Message-ID: <AANLkTikyGZ_yzX3YyyuWk3aywABGMtwarxSgCayssdge@mail.gmail.com>

openvz is by far the most reliable thanks to the jail vs virtual that
you already pointed out.
Other ones are mostly based on what people find out from experimenting.
Getting the most accurate timer settings as Frank pointed out is essential.
WE don't get much free time to play around with those so we rely on
the community to do this type of pioneering.



On Fri, Jan 28, 2011 at 1:03 PM, Frank Carmickle <frank at carmickle.com> wrote:
> Hi
>
> On Jan 28, 2011, at 8:33 AM, Avi Marcus wrote:
>
>> I'm using Linode's VPS which is run on xen, and all tests (80 moh channels, playback into a conference) and light usage have been perfect.
>> Some have complained of Amazon's high latency, others say it's been an awesome platform for them.
>>
> Can second this. ?Please see my other post just now on timing. ?I was running fine for a year before some changes to the kernel that I was running made it become a problem. ?I was able to hunt it down and fix it.
>
> --FC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From david.ponzone at ipeva.fr  Fri Jan 28 22:33:34 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 28 Jan 2011 20:33:34 +0100
Subject: [Freeswitch-users] Choppy VM when using loopback (on Debian
	lenny)
In-Reply-To: <7C6F3FC7-445F-4157-BD59-D76A705E4CE7@carmickle.com>
References: <4D42B8EE.7030208@estation.dk>
	<7C6F3FC7-445F-4157-BD59-D76A705E4CE7@carmickle.com>
Message-ID: <9894FEC6-E451-4993-A85A-2CD910E916B1@ipeva.fr>

Frank,

I fail to see the relationship between the hw timer and NTP.
Can you please elaborate ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 28/01/2011 ? 19:49, Frank Carmickle a ?crit :

> Hi
> 
> On Jan 28, 2011, at 7:39 AM, ?yvind Albrigtsen wrote:
> 
>> Hi,
>> 
>> I'm using latest git-version of Freeswitch, and when I go to voicemail 
>> when calling a number the sound playback is choppy and it skips some of 
>> the digits in the number I called.
>> 
> What kind of results do you get from timer_test at the fs_cli?  Are you running on hardware or are you virtualized?  What is your clock source set to and what are your available clock source options?  See /sys/devices/system/clocksource/clocksource0/available_clocksource and /sys/devices/system/clocksource/clocksource0/current_clocksource.  I am running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to hang at 19998/19999 which works very well for me.  When I was having problem it was reporting numbers all over the map from 17400 to 22600 with lots of randomness in between.  I have my clocksource set to jiffies and xen independent wallclock set to 1.  Of course at that point you need to have ntp running against a bunch of servers to drive your clock nice and steady.  I know my set up is probably a lot different than yours but I thought I'd toss it out there to show that some of the harshest conditions can be dealt with and don't give up trying.  If you are running on hardware with a cpu that doesn't have constant_tsc then you might have some problems.  Just play with the different timer options until you find the one that works.
> 
> HTH
> --FC
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From infos at madovsky.org  Fri Jan 28 23:18:20 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 28 Jan 2011 15:18:20 -0500
Subject: [Freeswitch-users] sofia_contact and sofia status profile
Message-ID: <7120B396EEC24BDAA37265B89A5C80E2@e1705>

in a script I run come CLI command
especially 

sofia_contact ext at default

so it returns the dialstring or user not registered.

but if I run 

sofia status profile [internal]  

I can see sometimes the user is not registrered but registered with sofia_contact.

is it a ODBC delay problem ?

Thanks

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From infos at madovsky.org  Fri Jan 28 23:22:46 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 28 Jan 2011 15:22:46 -0500
Subject: [Freeswitch-users] execute on answer on b leg
Message-ID: <318C6321F9844A5D98A0122C0E55EADB@e1705>

sometimes there are a delay when b leg answer and waiting a leg audio.
for this case I'd like to use a tts to say "please wait" until
the b leg receive a leg audio.

how can I do it ?

Thanks

Franck
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From chris.chen2004 at gmail.com  Fri Jan 28 23:23:10 2011
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Fri, 28 Jan 2011 15:23:10 -0500
Subject: [Freeswitch-users] Virtualisation and freeswitch
In-Reply-To: <AANLkTimZTwZfUYMjC1ddAQMWHHWSLH=BLajv0cgSux4B@mail.gmail.com>
References: <AANLkTimZTwZfUYMjC1ddAQMWHHWSLH=BLajv0cgSux4B@mail.gmail.com>
Message-ID: <AANLkTi=n_-dke-NLJAWRfE_RckbKU5nRc-nB2_aRA1+3@mail.gmail.com>

I have one small installation using  Amazon EC with Medium instance ( 32 Bit
but with 2 CPU) working without any customer complaints for more than 6
months, but when using default small instance I got a lot of complaints.
Hope this helps.

Thanks,
Chris

On Fri, Jan 28, 2011 at 6:52 AM, Grygoriy Dobrovolskyy
<megahohol at gmail.com>wrote:

> Hello list, le me introduce myself first, I am Grigoriy Dobrovolskyy
> currently working as voip administrator and call centeres integrator. I
> worked a lot with asterisk, i think Fs is a great software but not easy to
> learn.
>
> I have googled my question and did not found a clear answer.
>
> With what technology i can virtualise freeswitch without timer problems ?
>
> I know that i can use OpenVZ but it's more 'jail' then virtualisation. I
> heard that Amazon EC with High CPU instance works (confirm?)
>
> Can someone give me info of a succesfull installations with Conferences,
> Moh  / mixing ? And a technology used ?
>
> Thank you.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Fri Jan 28 23:25:40 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 28 Jan 2011 21:25:40 +0100
Subject: [Freeswitch-users] Virtualisation and freeswitch
In-Reply-To: <AANLkTi=n_-dke-NLJAWRfE_RckbKU5nRc-nB2_aRA1+3@mail.gmail.com>
References: <AANLkTimZTwZfUYMjC1ddAQMWHHWSLH=BLajv0cgSux4B@mail.gmail.com>
	<AANLkTi=n_-dke-NLJAWRfE_RckbKU5nRc-nB2_aRA1+3@mail.gmail.com>
Message-ID: <E3E629A5-6057-4531-8A2D-966AB7ADB23E@ipeva.fr>

Chris,

can you tell us how many concurrent calls and which applications you do with FreeSWITCH on this medium instance ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 28/01/2011 ? 21:23, Chris Chen a ?crit :

> I have one small installation using  Amazon EC with Medium instance ( 32 Bit but with 2 CPU) working without any customer complaints for more than 6 months, but when using default small instance I got a lot of complaints.
> Hope this helps.
> 
> Thanks,
> Chris
> 
> On Fri, Jan 28, 2011 at 6:52 AM, Grygoriy Dobrovolskyy <megahohol at gmail.com> wrote:
> Hello list, le me introduce myself first, I am Grigoriy Dobrovolskyy currently working as voip administrator and call centeres integrator. I worked a lot with asterisk, i think Fs is a great software but not easy to learn.
> 
> I have googled my question and did not found a clear answer.
> 
> With what technology i can virtualise freeswitch without timer problems ?
> 
> I know that i can use OpenVZ but it's more 'jail' then virtualisation. I heard that Amazon EC with High CPU instance works (confirm?)
> 
> Can someone give me info of a succesfull installations with Conferences, Moh  / mixing ? And a technology used ?
> 
> Thank you. 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Fri Jan 28 23:37:44 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 Jan 2011 14:37:44 -0600
Subject: [Freeswitch-users] execute on answer on b leg
In-Reply-To: <318C6321F9844A5D98A0122C0E55EADB@e1705>
References: <318C6321F9844A5D98A0122C0E55EADB@e1705>
Message-ID: <AANLkTi=Pq5dY0LSBOLJP29s1NxUY=O5_0bSoUNF0VxOH@mail.gmail.com>

add execute_on_answer to the originate vars
{execute_on_answer='<app> <args>'}sofa/foo/bar at baz.com


On Fri, Jan 28, 2011 at 2:22 PM, Madovsky <infos at madovsky.org> wrote:
> sometimes there are a delay when b leg answer and waiting a leg audio.
> for this case I'd like to use a tts to say "please wait" until
> the b leg receive a leg audio.
>
> how can I do it ?
>
> Thanks
> Franck
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From chris.chen2004 at gmail.com  Fri Jan 28 23:38:52 2011
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Fri, 28 Jan 2011 15:38:52 -0500
Subject: [Freeswitch-users] Virtualisation and freeswitch
In-Reply-To: <E3E629A5-6057-4531-8A2D-966AB7ADB23E@ipeva.fr>
References: <AANLkTimZTwZfUYMjC1ddAQMWHHWSLH=BLajv0cgSux4B@mail.gmail.com>
	<AANLkTi=n_-dke-NLJAWRfE_RckbKU5nRc-nB2_aRA1+3@mail.gmail.com>
	<E3E629A5-6057-4531-8A2D-966AB7ADB23E@ipeva.fr>
Message-ID: <AANLkTinLkp97_z=P7ZDvdztgUK9snEfn_sH8v4YRz0uR@mail.gmail.com>

No fancy stuff, just standard applications such as IVR, voice mail, incoming
calls and outgoing calls via SIP trunks, conference calls, although light
usage but supporting multi-tenant as well. I would say most time cpu idle at
99-100%, while with small instance cpu idle between 50-99% which was not
constant at all.
And Amazon has very good Internet connectivity, from the instance (at
east-US zone) to my home network always less than 30ms, which makes me feel
like any decent MAN/WAN connection. I have no complaint at all.
Thanks,
Chris

On Fri, Jan 28, 2011 at 3:25 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Chris,
>
> can you tell us how many concurrent calls and which applications you do
> with FreeSWITCH on this medium instance ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 28/01/2011 ? 21:23, Chris Chen a ?crit :
>
> I have one small installation using  Amazon EC with Medium instance ( 32
> Bit but with 2 CPU) working without any customer complaints for more than 6
> months, but when using default small instance I got a lot of complaints.
> Hope this helps.
>
> Thanks,
> Chris
>
> On Fri, Jan 28, 2011 at 6:52 AM, Grygoriy Dobrovolskyy <
> megahohol at gmail.com> wrote:
>
>> Hello list, le me introduce myself first, I am Grigoriy Dobrovolskyy
>> currently working as voip administrator and call centeres integrator. I
>> worked a lot with asterisk, i think Fs is a great software but not easy to
>> learn.
>>
>> I have googled my question and did not found a clear answer.
>>
>> With what technology i can virtualise freeswitch without timer problems ?
>>
>> I know that i can use OpenVZ but it's more 'jail' then virtualisation. I
>> heard that Amazon EC with High CPU instance works (confirm?)
>>
>> Can someone give me info of a succesfull installations with Conferences,
>> Moh  / mixing ? And a technology used ?
>>
>> Thank you.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ayhkor at gmail.com  Fri Jan 28 23:41:26 2011
From: ayhkor at gmail.com (deniro)
Date: Fri, 28 Jan 2011 15:41:26 -0500
Subject: [Freeswitch-users] freeswitch mysql handling
Message-ID: <AANLkTin7qtxg3heGtQFDqF7QwuTfa7FWF-gzoLzRrruV@mail.gmail.com>

Hi

I think lua is the recomended/preferred for database connections and
programming with freeswitch

1-- how can I connect to an existing mysql database (externally created and
managed by php program -- contructed tables -rows- fields etc...)

2-- how can I create  new mysql database  with freeswitch?

3-- how to check if required  lua components are  really  installed/working
for mysql db in freeswitch?

4-- if lua isn't a choice, which is the next best to handle  mysql
databases (javascript, perl etc..)

5-- any comprehensive  freeswitch  mysql db handling doc, links etc...?
thx
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From chris at cloudtel.com  Sat Jan 29 00:01:08 2011
From: chris at cloudtel.com (Chris Burns)
Date: Fri, 28 Jan 2011 16:01:08 -0500
Subject: [Freeswitch-users] Virtualisation and freeswitch
In-Reply-To: <AANLkTinLkp97_z=P7ZDvdztgUK9snEfn_sH8v4YRz0uR@mail.gmail.com>
References: <AANLkTimZTwZfUYMjC1ddAQMWHHWSLH=BLajv0cgSux4B@mail.gmail.com>
	<AANLkTi=n_-dke-NLJAWRfE_RckbKU5nRc-nB2_aRA1+3@mail.gmail.com>
	<E3E629A5-6057-4531-8A2D-966AB7ADB23E@ipeva.fr>
	<AANLkTinLkp97_z=P7ZDvdztgUK9snEfn_sH8v4YRz0uR@mail.gmail.com>
Message-ID: <AANLkTimUGiekiZy=0G3nFpMvLU7kycmE2ACu5-kAagZ_@mail.gmail.com>

I found the problem with small instances in EC2 not to be CPU usage, but how
acceptably bad your timing is allowed to be. If you dont handle media it
will be fine. Amazon also seems to have issues matching small instances with
higher resolution kernels. If anyone has a tip for that, Id like to hear it.
The bottleneck in my system is the bandwidth of g711u channels well before
the CPU, but the audio quality is poor even at small loads I assume from low
resolution timing.

On Fri, Jan 28, 2011 at 3:38 PM, Chris Chen <chris.chen2004 at gmail.com>wrote:

> No fancy stuff, just standard applications such as IVR, voice mail,
> incoming calls and outgoing calls via SIP trunks, conference calls, although
> light usage but supporting multi-tenant as well. I would say most time cpu
> idle at 99-100%, while with small instance cpu idle between 50-99% which was
> not constant at all.
> And Amazon has very good Internet connectivity, from the instance (at
> east-US zone) to my home network always less than 30ms, which makes me feel
> like any decent MAN/WAN connection. I have no complaint at all.
> Thanks,
> Chris
>
>
> On Fri, Jan 28, 2011 at 3:25 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>
>> Chris,
>>
>> can you tell us how many concurrent calls and which applications you do
>> with FreeSWITCH on this medium instance ?
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 28/01/2011 ? 21:23, Chris Chen a ?crit :
>>
>> I have one small installation using  Amazon EC with Medium instance ( 32
>> Bit but with 2 CPU) working without any customer complaints for more than 6
>> months, but when using default small instance I got a lot of complaints.
>> Hope this helps.
>>
>> Thanks,
>> Chris
>>
>> On Fri, Jan 28, 2011 at 6:52 AM, Grygoriy Dobrovolskyy <
>> megahohol at gmail.com> wrote:
>>
>>> Hello list, le me introduce myself first, I am Grigoriy Dobrovolskyy
>>> currently working as voip administrator and call centeres integrator. I
>>> worked a lot with asterisk, i think Fs is a great software but not easy to
>>> learn.
>>>
>>> I have googled my question and did not found a clear answer.
>>>
>>> With what technology i can virtualise freeswitch without timer problems ?
>>>
>>> I know that i can use OpenVZ but it's more 'jail' then virtualisation. I
>>> heard that Amazon EC with High CPU instance works (confirm?)
>>>
>>> Can someone give me info of a succesfull installations with Conferences,
>>> Moh  / mixing ? And a technology used ?
>>>
>>> Thank you.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kris at kriskinc.com  Sat Jan 29 00:07:59 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Fri, 28 Jan 2011 16:07:59 -0500
Subject: [Freeswitch-users] 200 OK without SDP
In-Reply-To: <201101280937.50415.sos@sokhapkin.dyndns.org>
References: <201101280730.17483.sos@sokhapkin.dyndns.org>
	<AANLkTimkQMBVD7xL9uRkCYWNu-1ng+dDA73i6XBjexqu@mail.gmail.com>
	<201101280937.50415.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTimF8M2tNk5+6su6gXR+ezEYOUk6p1_3KOoNk4S-@mail.gmail.com>

Sergey,

  I understand it may not be a permanent option but I was curious to
see if you could try it for a call or two to see if your problem goes
away...

On Fri, Jan 28, 2011 at 9:37 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org> wrote:
> Unfortunately stay in audio path or disable SST is not an option to me.To my
> experience, the problem happens when leg B answers the call with more than 1
> codec in 200 OK response.
>
>

-- 
Kristian Kielhofner


From phone.bytes at gmail.com  Sat Jan 29 00:09:05 2011
From: phone.bytes at gmail.com (Phone)
Date: Fri, 28 Jan 2011 14:09:05 -0700
Subject: [Freeswitch-users] freeswitch mysql handling
In-Reply-To: <AANLkTin7qtxg3heGtQFDqF7QwuTfa7FWF-gzoLzRrruV@mail.gmail.com>
References: <AANLkTin7qtxg3heGtQFDqF7QwuTfa7FWF-gzoLzRrruV@mail.gmail.com>
Message-ID: <4D433071.8020505@gmail.com>

Take a look here

http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh

and here

http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.Dbh

lots of good info to get you started.


On 01/28/2011 1:41 PM, deniro wrote:
> Hi
> I think lua is the recomended/preferred for database connections and 
> programming with freeswitch
> 1-- how can I connect to an existing mysql database (externally 
> created and managed by php program -- contructed tables -rows- fields 
> etc...)
> 2-- how can I create  new mysql database  with freeswitch?
> 3-- how to check if required  lua components are  really  
> installed/working  for mysql db in freeswitch?
> 4-- if lua isn't a choice, which is the next best to handle  mysql  
> databases (javascript, perl etc..)
> 5-- any comprehensive  freeswitch  mysql db handling doc, links etc...?
> thx
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From sos at sokhapkin.dyndns.org  Sat Jan 29 00:13:22 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 28 Jan 2011 16:13:22 -0500
Subject: [Freeswitch-users] 200 OK without SDP
In-Reply-To: <AANLkTimF8M2tNk5+6su6gXR+ezEYOUk6p1_3KOoNk4S-@mail.gmail.com>
References: <201101280730.17483.sos@sokhapkin.dyndns.org>
	<201101280937.50415.sos@sokhapkin.dyndns.org>
	<AANLkTimF8M2tNk5+6su6gXR+ezEYOUk6p1_3KOoNk4S-@mail.gmail.com>
Message-ID: <201101281613.22368.sos@sokhapkin.dyndns.org>

There is no problem if FS is in media path.

On Friday 28 January 2011, Kristian Kielhofner wrote:
> Sergey,
> 
>   I understand it may not be a permanent option but I was curious to
> see if you could try it for a call or two to see if your problem goes
> away...
> 
> On Fri, Jan 28, 2011 at 9:37 AM, Sergey Okhapkin
> 
> <sos at sokhapkin.dyndns.org> wrote:
> > Unfortunately stay in audio path or disable SST is not an option to me.To
> > my experience, the problem happens when leg B answers the call with more
> > than 1 codec in 200 OK response.



From kris at kriskinc.com  Sat Jan 29 00:27:23 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Fri, 28 Jan 2011 16:27:23 -0500
Subject: [Freeswitch-users] 200 OK without SDP
In-Reply-To: <201101281613.22368.sos@sokhapkin.dyndns.org>
References: <201101280730.17483.sos@sokhapkin.dyndns.org>
	<201101280937.50415.sos@sokhapkin.dyndns.org>
	<AANLkTimF8M2tNk5+6su6gXR+ezEYOUk6p1_3KOoNk4S-@mail.gmail.com>
	<201101281613.22368.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTi=KqOmroAYFRFyTfMPd_9_3Xjw0v3sdMtaSmdrc@mail.gmail.com>

That's what I'm seeing here too.

Thanks!

On Fri, Jan 28, 2011 at 4:13 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org> wrote:
> There is no problem if FS is in media path.
>

-- 
Kristian Kielhofner


From steveayre at gmail.com  Sat Jan 29 00:29:48 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 28 Jan 2011 21:29:48 +0000
Subject: [Freeswitch-users] freeswitch mysql handling
In-Reply-To: <4D433071.8020505@gmail.com>
References: <AANLkTin7qtxg3heGtQFDqF7QwuTfa7FWF-gzoLzRrruV@mail.gmail.com>
	<4D433071.8020505@gmail.com>
Message-ID: <AANLkTi=TU8dQ6+OOWaVxc64mJzdg18b-bm6vp8wuzTj3@mail.gmail.com>

Configure MySQL so you can connect via ODBC. That means installing MyODBC,
and unixodbc if you're on Linux, then setting up a DSN.

http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core

-Steve

On 28 January 2011 21:09, Phone <phone.bytes at gmail.com> wrote:

>  Take a look here
>
> http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh
>
> and here
>
> http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.Dbh
>
> lots of good info to get you started.
>
>
>
> On 01/28/2011 1:41 PM, deniro wrote:
>
> Hi
>
> I think lua is the recomended/preferred for database connections and
> programming with freeswitch
>
> 1-- how can I connect to an existing mysql database (externally created and
> managed by php program -- contructed tables -rows- fields etc...)
>
> 2-- how can I create  new mysql database  with freeswitch?
>
> 3-- how to check if required  lua components are  really
> installed/working  for mysql db in freeswitch?
>
> 4-- if lua isn't a choice, which is the next best to handle  mysql
> databases (javascript, perl etc..)
>
> 5-- any comprehensive  freeswitch  mysql db handling doc, links etc...?
> thx
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brent at overthewire.com.au  Sat Jan 29 01:03:27 2011
From: brent at overthewire.com.au (Brent Paddon)
Date: Sat, 29 Jan 2011 08:03:27 +1000
Subject: [Freeswitch-users] Virtualisation and freeswitch
In-Reply-To: <AANLkTimZTwZfUYMjC1ddAQMWHHWSLH=BLajv0cgSux4B@mail.gmail.com>
References: <AANLkTimZTwZfUYMjC1ddAQMWHHWSLH=BLajv0cgSux4B@mail.gmail.com>
Message-ID: <AANLkTikQbXLdCb4nNFEbfte-EAP8ytJyfURMALC6kwtK@mail.gmail.com>

Why would it possibly matter if it's more of a 'jail' than 'virtualisation'
?   Isn't the outcome the important thing ?

In regards OpenVZ there is an ISO installer called Proxmox which helps you
administer the whole system and gives you a pretty decent web interface for
management.  We use it extensively for voice applications.

http://www.proxmox.com/products/proxmox-ve

Brent

On Fri, Jan 28, 2011 at 9:52 PM, Grygoriy Dobrovolskyy
<megahohol at gmail.com>wrote:

> Hello list, le me introduce myself first, I am Grigoriy Dobrovolskyy
> currently working as voip administrator and call centeres integrator. I
> worked a lot with asterisk, i think Fs is a great software but not easy to
> learn.
>
> I have googled my question and did not found a clear answer.
>
> With what technology i can virtualise freeswitch without timer problems ?
>
> I know that i can use OpenVZ but it's more 'jail' then virtualisation. I
> heard that Amazon EC with High CPU instance works (confirm?)
>
> Can someone give me info of a succesfull installations with Conferences,
> Moh  / mixing ? And a technology used ?
>
> Thank you.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From ayhkor at gmail.com  Sat Jan 29 01:15:40 2011
From: ayhkor at gmail.com (deniro)
Date: Fri, 28 Jan 2011 17:15:40 -0500
Subject: [Freeswitch-users] freeswitch mysql handling
In-Reply-To: <AANLkTi=TU8dQ6+OOWaVxc64mJzdg18b-bm6vp8wuzTj3@mail.gmail.com>
References: <AANLkTin7qtxg3heGtQFDqF7QwuTfa7FWF-gzoLzRrruV@mail.gmail.com>
	<4D433071.8020505@gmail.com>
	<AANLkTi=TU8dQ6+OOWaVxc64mJzdg18b-bm6vp8wuzTj3@mail.gmail.com>
Message-ID: <AANLkTikSUVKiKLTAgrmNO_4Tor5XnqiHa-LAv_kuoOjL@mail.gmail.com>

thanks I looked these links and  I did some googling...
 I am using ubuntu 10.04  with mysql
couple of points to clear (sorry I am relativelly new  to freeswitch  world
and  concepts)

what is dsn?
I am using  ubuntu linux  so  it must be "unixodbc" to be installed?
what is this "unixodbc-dev " then ?
where spidermonkey come in picture? do I need spidermonkey too if I
installed   unixodbc?
what is "core" and how different from ODBC?

thx
deniro--







On Fri, Jan 28, 2011 at 4:29 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Configure MySQL so you can connect via ODBC. That means installing MyODBC,
> and unixodbc if you're on Linux, then setting up a DSN.
>
> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
>
> -Steve
>
>
> On 28 January 2011 21:09, Phone <phone.bytes at gmail.com> wrote:
>
>> Take a look here
>>
>> http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh
>>
>> and here
>>
>> http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.Dbh
>>
>> lots of good info to get you started.
>>
>>
>>
>> On 01/28/2011 1:41 PM, deniro wrote:
>>
>>   Hi
>>
>> I think lua is the recomended/preferred for database connections and
>> programming with freeswitch
>>
>> 1-- how can I connect to an existing mysql database (externally created
>> and managed by php program -- contructed tables -rows- fields etc...)
>>
>> 2-- how can I create  new mysql database  with freeswitch?
>>
>> 3-- how to check if required  lua components are  really
>> installed/working  for mysql db in freeswitch?
>>
>> 4-- if lua isn't a choice, which is the next best to handle  mysql
>> databases (javascript, perl etc..)
>>
>> 5-- any comprehensive  freeswitch  mysql db handling doc, links etc...?
>> thx
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From joaocarlosleme at gmail.com  Sat Jan 29 01:31:03 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Fri, 28 Jan 2011 14:31:03 -0800
Subject: [Freeswitch-users] Remote LogIn to Freeswitch?
In-Reply-To: <AANLkTinQiG5T6SrGW-Vr_kqiqSC8Mj9ZeMwnw4X+pyK1@mail.gmail.com>
References: <AANLkTinREKsePLFxXHhJobO0VkNBe0zt89araJJdBEnz@mail.gmail.com>
	<AANLkTinQiG5T6SrGW-Vr_kqiqSC8Mj9ZeMwnw4X+pyK1@mail.gmail.com>
Message-ID: <AANLkTi=CMg36dwcoRufQAA_7c14ZMmm+5ozeg4FKGDut@mail.gmail.com>

FINALLY I got it to work, I can now remotely log in on the internal profile
(port 5060) after placing the computer running Freeswitch on DMZ (changed on
Firewall), although that's not the ideal solution, so I was wondering why it
wasn't working before. I had the firewall set up according to the wiki on "
http://wiki.freeswitch.org/wiki/Firewall" and other than that I changed the
"domain" on vars.xml to my external ip of the router, it created an ALIAS so
that I can use as the "Domain" on X-lite to log on. So I had the internal
profile mod_sofia at 192.168.X.XX:5060 and the Alias 76.XXX.XX.XX, that was
before and I was able to LogIn, make calls but NO SOUND. NOW once I
forwarded all the port through DMZ, this particular router 2701HG-B Gateway
(Att) required me to change my (freeSwitch running computer) static ipv4
(192.XXX.X.XX) to DHCP and then assigned me the external IP (76.XXX.XX.XX)
as my a Ipv4. Now the internal profile (sofia status) is
mod_sofia at 76.XXX.XX.XX:5060 and I'm not using the ALIAS.

SO...I was wondering...it wasn't working before because 1) A Firewall
configuration missing? 2) A NAT/Firewall problem or 3) Freeswitch internal
profile running on local machine ipv4 192..... and I logging on using the
router external IP 76.XX and the Alias didn't do the job?????

Sorry there is some nonsense question, I'm a beginner and any help is
appreciated.
Thanks,
John

On Tue, Jan 11, 2011 at 1:37 AM, Steven Ayre <steveayre at gmail.com> wrote:

> If you're able to dial in but you're getting no sound, it's probably
> NAT stopping the audio get through.
>
> There's quite a bit of information on NAT on the Wiki that might be of use.
>
> http://wiki.freeswitch.org/wiki/NAT
>
> -Steve
>
>
> On 11 January 2011 00:39, Joao Leme <joaocarlosleme at gmail.com> wrote:
> > Hi There,
> > What do I have to do to be able to LogIn to Freeswitch from Home (server
> is
> > located at office) starting from the basic/original configuration?
> > I'm using X-Lite. I've been able to LogIn replacing the internal IP by
> the
> > external IP from the Office but the sound is not working so I wanted to
> know
> > what are the configuration changes that have to be done to allow it. Do I
> > have to create a different profile? I want be able to do the same just as
> if
> > I was at the Office.
> > Thanks,
> > John
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Sat Jan 29 01:50:03 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Jan 2011 14:50:03 -0800
Subject: [Freeswitch-users] time_test on Centos 5.5
In-Reply-To: <4908.1296215210@ccs.covici.com>
References: <AANLkTiksCg70iEdZiN1+CnfyBW87mg7T4jDHjeq7JW5Y@mail.gmail.com>
	<4908.1296215210@ccs.covici.com>
Message-ID: <AANLkTinj1O86ym6PngVpenrstVYa22DH2aWZccR2u5ex@mail.gmail.com>

On Fri, Jan 28, 2011 at 3:46 AM, <covici at ccs.covici.com> wrote:

> I don't even have time_test -- is this supposed to be a freeswitch
> command?
>
> It is an API you call from the fs_cli. It's been in tree for a few months
now.
-MC
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From ayhkor at gmail.com  Sat Jan 29 01:56:32 2011
From: ayhkor at gmail.com (deniro)
Date: Fri, 28 Jan 2011 17:56:32 -0500
Subject: [Freeswitch-users] freeswitch mysql handling
In-Reply-To: <AANLkTikSUVKiKLTAgrmNO_4Tor5XnqiHa-LAv_kuoOjL@mail.gmail.com>
References: <AANLkTin7qtxg3heGtQFDqF7QwuTfa7FWF-gzoLzRrruV@mail.gmail.com>
	<4D433071.8020505@gmail.com>
	<AANLkTi=TU8dQ6+OOWaVxc64mJzdg18b-bm6vp8wuzTj3@mail.gmail.com>
	<AANLkTikSUVKiKLTAgrmNO_4Tor5XnqiHa-LAv_kuoOjL@mail.gmail.com>
Message-ID: <AANLkTim17vW1Bughq4szv7HRyayu26WeckA7McWvnohf@mail.gmail.com>

ok I got dsn...

I am using  ubuntu linux & mqsql, so  is it "unixodbc" to be installed?
what is this "unixodbc-dev " then ?
where spidermonkey come in picture? do I need spidermonkey too if I
installed   unixodbc?
what is "core" and how different from ODBC?

thx
deniro--


On Fri, Jan 28, 2011 at 5:15 PM, deniro <ayhkor at gmail.com> wrote:

>
> thanks I looked these links and  I did some googling...
>  I am using ubuntu 10.04  with mysql
> couple of points to clear (sorry I am relativelly new  to freeswitch  world
> and  concepts)
>
> what is dsn?
> I am using  ubuntu linux  so  it must be "unixodbc" to be installed?
> what is this "unixodbc-dev " then ?
> where spidermonkey come in picture? do I need spidermonkey too if I
> installed   unixodbc?
> what is "core" and how different from ODBC?
>
> thx
> deniro--
>
>
>
>
>
>
>
> On Fri, Jan 28, 2011 at 4:29 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Configure MySQL so you can connect via ODBC. That means installing MyODBC,
>> and unixodbc if you're on Linux, then setting up a DSN.
>>
>> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
>>
>> -Steve
>>
>>
>> On 28 January 2011 21:09, Phone <phone.bytes at gmail.com> wrote:
>>
>>> Take a look here
>>>
>>> http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh
>>>
>>> and here
>>>
>>> http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.Dbh
>>>
>>> lots of good info to get you started.
>>>
>>>
>>>
>>> On 01/28/2011 1:41 PM, deniro wrote:
>>>
>>>   Hi
>>>
>>> I think lua is the recomended/preferred for database connections and
>>> programming with freeswitch
>>>
>>> 1-- how can I connect to an existing mysql database (externally created
>>> and managed by php program -- contructed tables -rows- fields etc...)
>>>
>>> 2-- how can I create  new mysql database  with freeswitch?
>>>
>>> 3-- how to check if required  lua components are  really
>>> installed/working  for mysql db in freeswitch?
>>>
>>> 4-- if lua isn't a choice, which is the next best to handle  mysql
>>> databases (javascript, perl etc..)
>>>
>>> 5-- any comprehensive  freeswitch  mysql db handling doc, links etc...?
>>> thx
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From chris at cloudtel.com  Sat Jan 29 02:00:52 2011
From: chris at cloudtel.com (Chris Burns)
Date: Fri, 28 Jan 2011 18:00:52 -0500
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTinr0RdgBs2UiS3=b3i+k6uCwkMihDy_FbAOFPc_@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
	<AANLkTi=in3CLd-n0mT3SRKYA0kxLMGVWk9KeBx3_jhY6@mail.gmail.com>
	<AANLkTin3Bu+8EEr_v_GaRGNPzOjfcnfzsV2Av2cgnWgO@mail.gmail.com>
	<AANLkTi=OC_jx0+nzK405VZURLDJPD25R4ZjyJ3RXJ1nk@mail.gmail.com>
	<AANLkTi=Sm7CjoayiE8nHmL0Eehqknj1jdpbx0YBQJQGH@mail.gmail.com>
	<AANLkTinr0RdgBs2UiS3=b3i+k6uCwkMihDy_FbAOFPc_@mail.gmail.com>
Message-ID: <AANLkTi=gcY9GckXECKHEis6pYCh5NQpsriGZgUXhe9rc@mail.gmail.com>

I have something set up in the lab right now that could be exactly what you
are trying to do. It is for a polycom deployment where 650s with BLF addons
monitor a collection of agents. The agents have multiple skillsets/languages
and belong to multiple inbound queues. The managers are used to having
things blink a certain specific way (as well as using polycom web apps that
tie into their intranet, but thats a different story x_X). Managers can
quickly see which agents are available on which queues just by lights, and
it can correct the state of BLFs if the phone gets rebooted.

If you still have issues and you are interested I could pull out the
relevant code into an example and pastebin it for you. It would be in php
using the ESL module.

On Fri, Jan 28, 2011 at 11:49 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> try with no presence_in and out commands at all
>
> just set presence_id variable as soon as you can and that call will
> report its state as that id for its entire duration.
> Also make sure you are on latest GIT when you test this.
>
>
> On Fri, Jan 28, 2011 at 7:47 AM, Patrick Plattes
> <patrick.plattes at niemann-frey.info> wrote:
> > Formatted and colourized code: http://pastebin.freeswitch.org/15172
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Sat Jan 29 03:02:18 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 29 Jan 2011 00:02:18 +0000
Subject: [Freeswitch-users] freeswitch mysql handling
In-Reply-To: <AANLkTim17vW1Bughq4szv7HRyayu26WeckA7McWvnohf@mail.gmail.com>
References: <AANLkTin7qtxg3heGtQFDqF7QwuTfa7FWF-gzoLzRrruV@mail.gmail.com>
	<4D433071.8020505@gmail.com>
	<AANLkTi=TU8dQ6+OOWaVxc64mJzdg18b-bm6vp8wuzTj3@mail.gmail.com>
	<AANLkTikSUVKiKLTAgrmNO_4Tor5XnqiHa-LAv_kuoOjL@mail.gmail.com>
	<AANLkTim17vW1Bughq4szv7HRyayu26WeckA7McWvnohf@mail.gmail.com>
Message-ID: <AANLkTikMufqaxZGXq6ETmRE8_8Puk9RQZSbqry1hnp4D@mail.gmail.com>

unixodbc is the unixodbc libraries
unixodbc-dev are the development headers for compiling freeswitch to be able
to speak to the unixodbc library

You need both unixodbc-dev and unixodbc on the box you're compiling
freeswitch on, but if you then distributed the binary files to another
server you'd only need unixodbc to run it.

Spidermonkey is just the module that runs javascript scripts, it isn't
needed at all for odbc.

"Core" is the database that freeswitch stores some internal data in...
normally that's in a sqlite file, but you can move it into ODBC too.

-Steve



On 28 January 2011 22:56, deniro <ayhkor at gmail.com> wrote:

> ok I got dsn...
>
> I am using  ubuntu linux & mqsql, so  is it "unixodbc" to be installed?
> what is this "unixodbc-dev " then ?
> where spidermonkey come in picture? do I need spidermonkey too if I
> installed   unixodbc?
> what is "core" and how different from ODBC?
>
> thx
> deniro--
>
>
> On Fri, Jan 28, 2011 at 5:15 PM, deniro <ayhkor at gmail.com> wrote:
>
>>
>> thanks I looked these links and  I did some googling...
>>  I am using ubuntu 10.04  with mysql
>> couple of points to clear (sorry I am relativelly new  to freeswitch
>> world and  concepts)
>>
>> what is dsn?
>> I am using  ubuntu linux  so  it must be "unixodbc" to be installed?
>> what is this "unixodbc-dev " then ?
>> where spidermonkey come in picture? do I need spidermonkey too if I
>> installed   unixodbc?
>> what is "core" and how different from ODBC?
>>
>> thx
>> deniro--
>>
>>
>>
>>
>>
>>
>>
>> On Fri, Jan 28, 2011 at 4:29 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> Configure MySQL so you can connect via ODBC. That means installing
>>> MyODBC, and unixodbc if you're on Linux, then setting up a DSN.
>>>
>>> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
>>>
>>> -Steve
>>>
>>>
>>> On 28 January 2011 21:09, Phone <phone.bytes at gmail.com> wrote:
>>>
>>>> Take a look here
>>>>
>>>> http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh
>>>>
>>>> and here
>>>>
>>>> http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.Dbh
>>>>
>>>> lots of good info to get you started.
>>>>
>>>>
>>>>
>>>> On 01/28/2011 1:41 PM, deniro wrote:
>>>>
>>>>   Hi
>>>>
>>>> I think lua is the recomended/preferred for database connections and
>>>> programming with freeswitch
>>>>
>>>> 1-- how can I connect to an existing mysql database (externally created
>>>> and managed by php program -- contructed tables -rows- fields etc...)
>>>>
>>>> 2-- how can I create  new mysql database  with freeswitch?
>>>>
>>>> 3-- how to check if required  lua components are  really
>>>> installed/working  for mysql db in freeswitch?
>>>>
>>>> 4-- if lua isn't a choice, which is the next best to handle  mysql
>>>> databases (javascript, perl etc..)
>>>>
>>>> 5-- any comprehensive  freeswitch  mysql db handling doc, links etc...?
>>>> thx
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Sat Jan 29 03:09:04 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 29 Jan 2011 00:09:04 +0000
Subject: [Freeswitch-users] Remote LogIn to Freeswitch?
In-Reply-To: <AANLkTi=CMg36dwcoRufQAA_7c14ZMmm+5ozeg4FKGDut@mail.gmail.com>
References: <AANLkTinREKsePLFxXHhJobO0VkNBe0zt89araJJdBEnz@mail.gmail.com>
	<AANLkTinQiG5T6SrGW-Vr_kqiqSC8Mj9ZeMwnw4X+pyK1@mail.gmail.com>
	<AANLkTi=CMg36dwcoRufQAA_7c14ZMmm+5ozeg4FKGDut@mail.gmail.com>
Message-ID: <AANLkTinzhQ_2kFo9ye1DEvF_sZA9cE64v2Z9DYMJj2Ti@mail.gmail.com>

Sounds like an NAT problem. You have two connections set up for a call - one
for the signalling and one for the media. It's common that behind NAT you
can have SIP working but RTP broken.

Off the top of my head there's a few possible reasons:
- The router is too dumb to spot the SIP and automatically open the RTP
ports: forwarding those manually may help resolve that.
- The router has a SIP ALG that tries to correct the IPs sent through SIP
and actually breaks things even worse: try disabling that feature.
- Freeswitch doesn't know the external IP address and will send the internal
IP in the SDP, so the other host doesn't know where to send RTP to: try
using STUN, or setting ext-sip-ip and ext-rtp-ip on the sip profile.

DMZ might help work around any of the above without actually solving your
real problem.

-Steve


On 28 January 2011 22:31, Joao Leme <joaocarlosleme at gmail.com> wrote:

> FINALLY I got it to work, I can now remotely log in on the internal profile
> (port 5060) after placing the computer running Freeswitch on DMZ (changed on
> Firewall), although that's not the ideal solution, so I was wondering why it
> wasn't working before. I had the firewall set up according to the wiki on "
> http://wiki.freeswitch.org/wiki/Firewall" and other than that I changed
> the "domain" on vars.xml to my external ip of the router, it created an
> ALIAS so that I can use as the "Domain" on X-lite to log on. So I had the
> internal profile mod_sofia at 192.168.X.XX:5060 and the Alias 76.XXX.XX.XX,
> that was before and I was able to LogIn, make calls but NO SOUND. NOW once I
> forwarded all the port through DMZ, this particular router 2701HG-B Gateway
> (Att) required me to change my (freeSwitch running computer) static ipv4
> (192.XXX.X.XX) to DHCP and then assigned me the external IP (76.XXX.XX.XX)
> as my a Ipv4. Now the internal profile (sofia status) is
> mod_sofia at 76.XXX.XX.XX:5060 and I'm not using the ALIAS.
>
> SO...I was wondering...it wasn't working before because 1) A Firewall
> configuration missing? 2) A NAT/Firewall problem or 3) Freeswitch internal
> profile running on local machine ipv4 192..... and I logging on using the
> router external IP 76.XX and the Alias didn't do the job?????
>
> Sorry there is some nonsense question, I'm a beginner and any help is
> appreciated.
> Thanks,
> John
>
> On Tue, Jan 11, 2011 at 1:37 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> If you're able to dial in but you're getting no sound, it's probably
>> NAT stopping the audio get through.
>>
>> There's quite a bit of information on NAT on the Wiki that might be of
>> use.
>>
>> http://wiki.freeswitch.org/wiki/NAT
>>
>> -Steve
>>
>>
>> On 11 January 2011 00:39, Joao Leme <joaocarlosleme at gmail.com> wrote:
>> > Hi There,
>> > What do I have to do to be able to LogIn to Freeswitch from Home (server
>> is
>> > located at office) starting from the basic/original configuration?
>> > I'm using X-Lite. I've been able to LogIn replacing the internal IP by
>> the
>> > external IP from the Office but the sound is not working so I wanted to
>> know
>> > what are the configuration changes that have to be done to allow it. Do
>> I
>> > have to create a different profile? I want be able to do the same just
>> as if
>> > I was at the Office.
>> > Thanks,
>> > John
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Sat Jan 29 03:10:36 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 29 Jan 2011 00:10:36 +0000
Subject: [Freeswitch-users] time_test on Centos 5.5
In-Reply-To: <41F744A2-90FF-405F-AF62-5E7B8FB5128F@carmickle.com>
References: <AANLkTiksCg70iEdZiN1+CnfyBW87mg7T4jDHjeq7JW5Y@mail.gmail.com>
	<0E4A540B-3E61-4940-8246-6FBF67CF91D8@ipeva.fr>
	<41F744A2-90FF-405F-AF62-5E7B8FB5128F@carmickle.com>
Message-ID: <AANLkTi=NEPwvbF70y3j0p_jfSgLg=xBxW7Es1k7RgAVm@mail.gmail.com>

I've been using it on Lenny with no problems for ~2 years, timing works
fine. It will work. CentOS is the reference platform though.

-Steve



On 28 January 2011 19:05, Frank Carmickle <frank at carmickle.com> wrote:

>
> On Jan 28, 2011, at 6:35 AM, David Ponzone wrote:
> Snip...
>
> > Stay away from Debian, Centos is the right choice.
> > You could eventually try to fallback to centos 5.3 or 5.4.
> >
> Debian can work if that's what people want to use.  I have it working well
> on a few lenny machines.
>
> --FC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From phone.bytes at gmail.com  Sat Jan 29 03:18:32 2011
From: phone.bytes at gmail.com (Phone)
Date: Fri, 28 Jan 2011 17:18:32 -0700
Subject: [Freeswitch-users] Disable Message Taking for a Voice Mail Box
Message-ID: <4D435CD8.5020106@gmail.com>

Is there a setting to disable a Voice Mail Box from taking a message?

I would like to set some up to play an announcement and then hang up.

Thanks


From dujinfang at gmail.com  Sat Jan 29 03:23:11 2011
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 29 Jan 2011 08:23:11 +0800
Subject: [Freeswitch-users] Virtualisation and freeswitch
In-Reply-To: <AANLkTikQbXLdCb4nNFEbfte-EAP8ytJyfURMALC6kwtK@mail.gmail.com>
References: <AANLkTimZTwZfUYMjC1ddAQMWHHWSLH=BLajv0cgSux4B@mail.gmail.com>
	<AANLkTikQbXLdCb4nNFEbfte-EAP8ytJyfURMALC6kwtK@mail.gmail.com>
Message-ID: <AANLkTinB17uNUKtEQAtuTgCGPf_61jnErGmv7+vXV_n+@mail.gmail.com>

We had used OpenVZ for a while, it was good given we never tested on load.

Another option besides OpenVZ is lxc , which is similar to OpenVZ. I
runs some lxc VMs for other projects but never tried with FS.

On Sat, Jan 29, 2011 at 6:03 AM, Brent Paddon <brent at overthewire.com.au> wrote:
> Why would it possibly matter if it's more of a 'jail' than 'virtualisation'
> ? ? Isn't the outcome the important thing ?
> In regards OpenVZ there is an ISO installer called Proxmox which helps you
> administer the whole system and gives you a pretty decent web interface for
> management. ?We use it extensively for voice applications.
> http://www.proxmox.com/products/proxmox-ve
> Brent
>
> On Fri, Jan 28, 2011 at 9:52 PM, Grygoriy Dobrovolskyy <megahohol at gmail.com>
> wrote:
>>
>> Hello list, le me introduce myself first, I am Grigoriy Dobrovolskyy
>> currently working as voip administrator and call centeres integrator. I
>> worked a lot with asterisk, i think Fs is a great software but not easy to
>> learn.
>> I have googled my question and did not found a clear answer.
>> With what technology i can virtualise freeswitch without timer problems ?
>> I know that i can use OpenVZ but it's more 'jail' then virtualisation. I
>> heard that Amazon EC with High CPU instance works (confirm?)
>> Can someone give me info of a succesfull installations with Conferences,
>> Moh ?/ mixing ? And a technology used ?
>> Thank you.
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> --
> Brent Paddon
>
> Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
> www.overthewire.com.au
> Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From dujinfang at gmail.com  Sat Jan 29 03:30:18 2011
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 29 Jan 2011 08:30:18 +0800
Subject: [Freeswitch-users] question about outbound socket async
In-Reply-To: <CE77FAA7-A13A-40BF-AD6E-7E0E10807AE3@visionutveckling.se>
References: <CE77FAA7-A13A-40BF-AD6E-7E0E10807AE3@visionutveckling.se>
Message-ID: <AANLkTikmGz0uwx8dpmmUjeh4MmDvX87aM18=z8T3MTeu@mail.gmail.com>

Thank you Peter. Yes I sendmsg break with event-lock, but the playback
is not using event-lock since I want to stop it at anytime, I need
more test to figure out. At the same time I still want to know if
there's an option to auto-break the running app on a channel.

Thanks.

On Fri, Jan 28, 2011 at 10:08 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> Are you using sendmsg break when issuing a break? That forces it to be queued in the execute queue the same way as playback, and it should be impossible to break the second file when not supposed to (since that should be the next one in queue).
>
> /Peter
>
>
> ----- Reply message -----
> Fr?n: "Seven Du" <dujinfang at gmail.com>
> Datum: fre, jan 28, 2011 07:59
> Rubrik: [Freeswitch-users] question about outbound socket async
> Till: "freeswitch-users" <freeswitch-users at lists.freeswitch.org>
>
> Hi all,
>
> I'm using mod_erlang_event and I think it's equivalent to event_socket
> async full.
>
> When I run sendmsg playback wav1 and then playback wav2, wav2 will
> stop wav1 and when wav2 is done, wav1 continues by default. Is there a
> way to automatically break wav1 when play wav2 ?
>
> I could send break to wav1 but because everything is async, sometimes
> break also breaks wav2 even with event-lock = true which made async
> mode a sort of pain.
>
> It seems like there is a var can control this but I cannot recall. Any
> help is appreciated.
>
> Thanks.
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj: ?http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4d42149232767334012844!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



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Blog: http://www.dujinfang.com
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From bwibowo at gmail.com  Sat Jan 29 03:44:47 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Sat, 29 Jan 2011 07:44:47 +0700
Subject: [Freeswitch-users] Number mapping (DID like)
In-Reply-To: <AANLkTikf2gS+=3NMUb=HW_QasRxiDZ3Ny+0a5=0d5VGK@mail.gmail.com>
References: <1806109314-1296182728-cardhu_decombobulator_blackberry.rim.net-210966170-@b25.c2.bise3.blackberry>
	<AANLkTinujT2YjspkUMn9fEnkk0w=vCH5wCWoZnQ07LLZ@mail.gmail.com>
	<AANLkTikf2gS+=3NMUb=HW_QasRxiDZ3Ny+0a5=0d5VGK@mail.gmail.com>
Message-ID: <AANLkTi=hbQNyrM-iFExykvpotGPqmcRvvz1J_Y=YvMmR@mail.gmail.com>

something like call forwarding,
found this http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect
<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect>but still not
work.
other question, is there any tool for mass provisioning/ bulk provisioning?
say i want to create user 1000 - 1100 without adding it one by one


regards

budi



On Fri, Jan 28, 2011 at 3:41 PM, Avi Marcus <avi at avimarcus.net> wrote:

> Propaby, but that could mean many things. Do you mean rewrite the number in
> the CDRs? Allow the owner of the extension to set a call forwarding / follow
> me? Or basically to use 1001 as a speeddial for the one user?
> I use mod_xml_curl extensively which allow me to query a database before I
> do just about anything with te normal dialplan.
> First, any 1-3 digit numbers it checks a extension-specific and then global
> speeddial table.
> Soon, I intend to have it query FusionPBX's follow-me settings so that I
> can implement that with poper billing.
> Some of tis might help you?
> -Avi Marcus
>
> > On Jan 28, 2011 4:46 AM, "Budi wibowo" <bwibowo at gmail.com> wrote:
> > > Hi
> > > I want to do B# (called) rewriting, for example
> > >>From user 1000 call to 1001, and freeswitch rewrite 1001 to other
> number (pstn).
> > > Is it doable with freeswitch?
> > >
> > > Regards
> > > Budi
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Sat Jan 29 05:09:52 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 28 Jan 2011 21:09:52 -0500
Subject: [Freeswitch-users] play_and_get_digits and TTS
Message-ID: <26BD45B1EB0445AE845E211126806FF2@e1705>

can I replace file and file_invalid args
with a TTS ?

Thanks
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From peter.olsson at visionutveckling.se  Sat Jan 29 05:10:57 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sat, 29 Jan 2011 03:10:57 +0100
Subject: [Freeswitch-users] question about outbound socket async
Message-ID: <F15F4ACE-2AEB-4701-A67D-F1C93172B25C@visionutveckling.se>

I don't think it's possible to "auto break" playback. However, break should work ok here, I use it like this all the time in an ESL application. Are you using latest git? About a year ago I submitted a patch to fix some timing for this - but I guess you're not running a version old like that :)

/Peter

----- Reply message -----
Fr?n: "Seven Du" <dujinfang at gmail.com>
Datum: l?r, jan 29, 2011 07:36
Rubrik: [Freeswitch-users] question about outbound socket async
Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

Thank you Peter. Yes I sendmsg break with event-lock, but the playback
is not using event-lock since I want to stop it at anytime, I need
more test to figure out. At the same time I still want to know if
there's an option to auto-break the running app on a channel.

Thanks.

On Fri, Jan 28, 2011 at 10:08 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> Are you using sendmsg break when issuing a break? That forces it to be queued in the execute queue the same way as playback, and it should be impossible to break the second file when not supposed to (since that should be the next one in queue).
>
> /Peter
>
>
> ----- Reply message -----
> Fr?n: "Seven Du" <dujinfang at gmail.com>
> Datum: fre, jan 28, 2011 07:59
> Rubrik: [Freeswitch-users] question about outbound socket async
> Till: "freeswitch-users" <freeswitch-users at lists.freeswitch.org>
>
> Hi all,
>
> I'm using mod_erlang_event and I think it's equivalent to event_socket
> async full.
>
> When I run sendmsg playback wav1 and then playback wav2, wav2 will
> stop wav1 and when wav2 is done, wav1 continues by default. Is there a
> way to automatically break wav1 when play wav2 ?
>
> I could send break to wav1 but because everything is async, sometimes
> break also breaks wav2 even with event-lock = true which made async
> mode a sort of pain.
>
> It seems like there is a var can control this but I cannot recall. Any
> help is appreciated.
>
> Thanks.
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



--
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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!DSPAM:4d4360a432761546914186!


From infos at madovsky.org  Sat Jan 29 05:37:17 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 28 Jan 2011 21:37:17 -0500
Subject: [Freeswitch-users] application socket
Message-ID: <DED70301F45845C5A1B1991F24F2314C@e1705>

in case of the socket fails after a while,
is is return to dialplan or maybe an api that check the status of the socket ?

Thanks
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From dujinfang at gmail.com  Sat Jan 29 05:54:17 2011
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 29 Jan 2011 10:54:17 +0800
Subject: [Freeswitch-users] play_and_get_digits and TTS
In-Reply-To: <26BD45B1EB0445AE845E211126806FF2@e1705>
References: <26BD45B1EB0445AE845E211126806FF2@e1705>
Message-ID: <AANLkTinY9=DQpKMbshaNboh3vAqqaL9b_BZ5bC7VLzHt@mail.gmail.com>

I think you can via phrase macro.

On Sat, Jan 29, 2011 at 10:09 AM, Madovsky <infos at madovsky.org> wrote:
> can I replace file and file_invalid args
> with a TTS ?
>
> Thanks
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



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Blog: http://www.dujinfang.com
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From u2nsam at gmail.com  Sat Jan 29 07:29:07 2011
From: u2nsam at gmail.com (Sam)
Date: Sat, 29 Jan 2011 09:59:07 +0530
Subject: [Freeswitch-users] blocking 183 w/o sdp
In-Reply-To: <AANLkTikJpEPvcQu2reAKXk1wbW7fV4myvXghBCuu2QHt@mail.gmail.com>
References: <AANLkTinH1s+Uy-EPvX9oV-x+aXYkyzUEBSYm+hQ+-+K-@mail.gmail.com>
	<AANLkTikJpEPvcQu2reAKXk1wbW7fV4myvXghBCuu2QHt@mail.gmail.com>
Message-ID: <AANLkTikuQFC2N3CYzreTvCdk8-28MYGUbrP=T83SfPtm@mail.gmail.com>

Hi,

So you say i need to put
<action application="bridge"
data="{sip_contact_user=${contact}}{sip_ignore_183nosdp=true}sofia/sbc/$1@
${dialed_domain}"/>

Regds
Sam




On Fri, Jan 28, 2011 at 10:23 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> you need sip_ignore_183nosdp=true set on the b leg not the a leg.
> Put it in the dial string in {}
>
> {sip_ignore_183nosdp=true}sofia/foo/foo at bar.com
>
>
> On Fri, Jan 28, 2011 at 12:41 AM, Sam <u2nsam at gmail.com> wrote:
> > Hi,
> >
> > how can i ignore 183 without sdp,
> > what happens is the provider sends 183 without sdp and by applying
> "<action
> > application="set" data="sip_ignore_183nosdp=true"/>"  the FS sends 180 to
> > the leg a.
> > Here i want to block the 183 with SDP just like router as b2bua and send
> > nothing to leg a, and when actual 183 with sdp comes it should send .
> >
> > Its because, providers are sending false signaling by sending 183 without
> > sdp,and it hampers while @ production,
> > Although by cisco sbc i have done this but i want to do it by FS,
> > Take a scenario, when call is send 183 without sdp for 10 secs and then
> > followed by 183 with sdp ( actual signal),
> > but when some one dials invalid number it rings for 10 secs and then
> gives
> > SIP cause 404, which is bad from the providers.
> > So this is the reason i want to block it.
> >
> > Most of the providers do this, the way out is blocking.
> >
> > I have got an advice from Tihomir  to do "execute_on_ring and parse your
> 180
> > / 183 messages in search of SDP,
> > once you get 183 without SDP do not send it back to leg a and send signal
> > only when you got 183 with sdp or 180 "
> > And this could be valid call flow.
> >
> > This happens in many cases where the provider is having nextone as a sbc
> and
> > that too tier 1 !
> >
> > Regards
> > Sam
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dome at tel.co.th  Sat Jan 29 12:30:57 2011
From: dome at tel.co.th (dome at tel.co.th)
Date: Sat, 29 Jan 2011 16:30:57 +0700
Subject: [Freeswitch-users] Virtualisation and freeswitch
In-Reply-To: <AANLkTinB17uNUKtEQAtuTgCGPf_61jnErGmv7+vXV_n+@mail.gmail.com>
References: <AANLkTimZTwZfUYMjC1ddAQMWHHWSLH=BLajv0cgSux4B@mail.gmail.com>
	<AANLkTikQbXLdCb4nNFEbfte-EAP8ytJyfURMALC6kwtK@mail.gmail.com>
	<AANLkTinB17uNUKtEQAtuTgCGPf_61jnErGmv7+vXV_n+@mail.gmail.com>
Message-ID: <AANLkTimtnMqFyrXFEYirmhLgqPJxaWsYWowYyiLi_k_=@mail.gmail.com>

(from wiki http://wiki.freeswitch.org/wiki/Virtualization)
if your FS running with high traffic please use
/etc/vz/conf/ve-unlimited.conf-sample for default template

Dome C.

On Sat, Jan 29, 2011 at 7:23 AM, Seven Du <dujinfang at gmail.com> wrote:
> We had used OpenVZ for a while, it was good given we never tested on load.
>
> Another option besides OpenVZ is lxc , which is similar to OpenVZ. I
> runs some lxc VMs for other projects but never tried with FS.
>
> On Sat, Jan 29, 2011 at 6:03 AM, Brent Paddon <brent at overthewire.com.au> wrote:
>> Why would it possibly matter if it's more of a 'jail' than 'virtualisation'
>> ? ? Isn't the outcome the important thing ?
>> In regards OpenVZ there is an ISO installer called Proxmox which helps you
>> administer the whole system and gives you a pretty decent web interface for
>> management. ?We use it extensively for voice applications.
>> http://www.proxmox.com/products/proxmox-ve
>> Brent
>>
>> On Fri, Jan 28, 2011 at 9:52 PM, Grygoriy Dobrovolskyy <megahohol at gmail.com>
>> wrote:
>>>
>>> Hello list, le me introduce myself first, I am Grigoriy Dobrovolskyy
>>> currently working as voip administrator and call centeres integrator. I
>>> worked a lot with asterisk, i think Fs is a great software but not easy to
>>> learn.
>>> I have googled my question and did not found a clear answer.
>>> With what technology i can virtualise freeswitch without timer problems ?
>>> I know that i can use OpenVZ but it's more 'jail' then virtualisation. I
>>> heard that Amazon EC with High CPU instance works (confirm?)
>>> Can someone give me info of a succesfull installations with Conferences,
>>> Moh ?/ mixing ? And a technology used ?
>>> Thank you.
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> --
>> Brent Paddon
>>
>> Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
>> www.overthewire.com.au
>> Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:? http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From steveayre at gmail.com  Sat Jan 29 12:46:54 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 29 Jan 2011 09:46:54 +0000
Subject: [Freeswitch-users] blocking 183 w/o sdp
In-Reply-To: <AANLkTikuQFC2N3CYzreTvCdk8-28MYGUbrP=T83SfPtm@mail.gmail.com>
References: <AANLkTinH1s+Uy-EPvX9oV-x+aXYkyzUEBSYm+hQ+-+K-@mail.gmail.com>
	<AANLkTikJpEPvcQu2reAKXk1wbW7fV4myvXghBCuu2QHt@mail.gmail.com>
	<AANLkTikuQFC2N3CYzreTvCdk8-28MYGUbrP=T83SfPtm@mail.gmail.com>
Message-ID: <AANLkTinyV-kAWCOiuOVL5jmnL7f9ck00=i_giHY_XGd_@mail.gmail.com>

Close. You can only have one set of {} brackets. You can separate multiple
variables with a comma.

<action application="bridge"
data="{sip_contact_user=${contact},sip_ignore_183nosdp=true}sofia/sbc/$1@
${dialed_domain}"/>

-Steve


On 29 January 2011 04:29, Sam <u2nsam at gmail.com> wrote:

> Hi,
>
> So you say i need to put
> <action application="bridge"
> data="{sip_contact_user=${contact}}{sip_ignore_183nosdp=true}sofia/sbc/$1@
> ${dialed_domain}"/>
>
> Regds
> Sam
>
>
>
>
>
> On Fri, Jan 28, 2011 at 10:23 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> you need sip_ignore_183nosdp=true set on the b leg not the a leg.
>> Put it in the dial string in {}
>>
>> {sip_ignore_183nosdp=true}sofia/foo/foo at bar.com
>>
>>
>> On Fri, Jan 28, 2011 at 12:41 AM, Sam <u2nsam at gmail.com> wrote:
>> > Hi,
>> >
>> > how can i ignore 183 without sdp,
>> > what happens is the provider sends 183 without sdp and by applying
>> "<action
>> > application="set" data="sip_ignore_183nosdp=true"/>"  the FS sends 180
>> to
>> > the leg a.
>> > Here i want to block the 183 with SDP just like router as b2bua and send
>> > nothing to leg a, and when actual 183 with sdp comes it should send .
>> >
>> > Its because, providers are sending false signaling by sending 183
>> without
>> > sdp,and it hampers while @ production,
>> > Although by cisco sbc i have done this but i want to do it by FS,
>> > Take a scenario, when call is send 183 without sdp for 10 secs and then
>> > followed by 183 with sdp ( actual signal),
>> > but when some one dials invalid number it rings for 10 secs and then
>> gives
>> > SIP cause 404, which is bad from the providers.
>> > So this is the reason i want to block it.
>> >
>> > Most of the providers do this, the way out is blocking.
>> >
>> > I have got an advice from Tihomir  to do "execute_on_ring and parse your
>> 180
>> > / 183 messages in search of SDP,
>> > once you get 183 without SDP do not send it back to leg a and send
>> signal
>> > only when you got 183 with sdp or 180 "
>> > And this could be valid call flow.
>> >
>> > This happens in many cases where the provider is having nextone as a sbc
>> and
>> > that too tier 1 !
>> >
>> > Regards
>> > Sam
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From list.subscription at alexrambau.com  Sat Jan 29 13:43:18 2011
From: list.subscription at alexrambau.com (Alex Rambau)
Date: Sat, 29 Jan 2011 03:43:18 -0700
Subject: [Freeswitch-users] Event Socket Timeout - Outbound
Message-ID: <000501cbbfa1$57d0e960$0772bc20$@subscription@alexrambau.com>

Is there any way to set the timeout on the application "socket" from the
dial plan so that if the remote socket isn't listening or is taking too long
the connect operation will timeout and the dial plan won't just hang? 

 

My dial plan entry is as follows:

 

    <extension name="testapp">

      <condition field="destination_number" expression="^2222$">

                <action application="set" data="continue_on_fail=true" />

                <action application="socket" data="192.168.1.108:4444 sync
full"/>

                <action application="transfer" data="8000 XML default"/>

      </condition>

    </extension>

 

Thanks in advance,

 

Alex

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From dujinfang at gmail.com  Sat Jan 29 14:54:33 2011
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 29 Jan 2011 19:54:33 +0800
Subject: [Freeswitch-users] question about outbound socket async
In-Reply-To: <F15F4ACE-2AEB-4701-A67D-F1C93172B25C@visionutveckling.se>
References: <F15F4ACE-2AEB-4701-A67D-F1C93172B25C@visionutveckling.se>
Message-ID: <AANLkTi=k2=Zuiq=RP0kHH31fm_1gv=aBY2fpwjt5Q-5b@mail.gmail.com>

For sure I'm using a recent git.

I test with nc and didn't found a problem, but in erlang it does has
problem I didn't test much so I just sleep some milliseconds after I
sendmsg break that works for now.

Let me tell you what I was trying to do:

an erlang FSM (like outbound event socket) controls a channel and can

playback to channel
listen to DTMF and send to a UDP server
listen to a UDP and break the current playback and play new voice on
new instruction

The problem is that it can get UDP instructions any time and can be
very quick( say, get 2 playback instruction at the same time).

So I want to listen to CHANNEL_EXECUTE to make sure a playback is
actually executing and CHANNEL_EXECUTE_COMPLETE to make sure if it
stopped. And also for the break APP.

The state machine would be complicated if I need to worry about all
the races. It might be better to implement in C but it's actually a
simple program which I think it's overkill for now.

Thanks for your help, a simple sleep works for now. Will worry about this later.

On Sat, Jan 29, 2011 at 10:10 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> I don't think it's possible to "auto break" playback. However, break should work ok here, I use it like this all the time in an ESL application. Are you using latest git? About a year ago I submitted a patch to fix some timing for this - but I guess you're not running a version old like that :)
>
> /Peter
>
> ----- Reply message -----
> Fr?n: "Seven Du" <dujinfang at gmail.com>
> Datum: l?r, jan 29, 2011 07:36
> Rubrik: [Freeswitch-users] question about outbound socket async
> Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>
> Thank you Peter. Yes I sendmsg break with event-lock, but the playback
> is not using event-lock since I want to stop it at anytime, I need
> more test to figure out. At the same time I still want to know if
> there's an option to auto-break the running app on a channel.
>
> Thanks.
>
> On Fri, Jan 28, 2011 at 10:08 AM, Peter Olsson
> <peter.olsson at visionutveckling.se> wrote:
>> Are you using sendmsg break when issuing a break? That forces it to be queued in the execute queue the same way as playback, and it should be impossible to break the second file when not supposed to (since that should be the next one in queue).
>>
>> /Peter
>>
>>
>> ----- Reply message -----
>> Fr?n: "Seven Du" <dujinfang at gmail.com>
>> Datum: fre, jan 28, 2011 07:59
>> Rubrik: [Freeswitch-users] question about outbound socket async
>> Till: "freeswitch-users" <freeswitch-users at lists.freeswitch.org>
>>
>> Hi all,
>>
>> I'm using mod_erlang_event and I think it's equivalent to event_socket
>> async full.
>>
>> When I run sendmsg playback wav1 and then playback wav2, wav2 will
>> stop wav1 and when wav2 is done, wav1 continues by default. Is there a
>> way to automatically break wav1 when play wav2 ?
>>
>> I could send break to wav1 but because everything is async, sometimes
>> break also breaks wav2 even with event-lock = true which made async
>> mode a sort of pain.
>>
>> It seems like there is a var can control this but I cannot recall. Any
>> help is appreciated.
>>
>> Thanks.
>>
>> --
>> About: http://about.me/dujinfang
>> Blog: http://www.dujinfang.com
>> Proj: ?http://www.freeswitch.org.cn
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>


From fdelawarde at wirelessmundi.com  Sat Jan 29 16:10:37 2011
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Sat, 29 Jan 2011 14:10:37 +0100
Subject: [Freeswitch-users] execute_on_ring executing on answer
In-Reply-To: <AANLkTikHB-QnDyT0WPsEML8yv4AACnYnHrE37gpRsLS2@mail.gmail.com>
References: <1296215265.8986.147.camel@luna.tc.commsmundi.com>
	<21B4FB4AAC2E428E938F99986CA09165@e1705>
	<1296235095.8986.167.camel@luna.tc.commsmundi.com>
	<0E96944DB29F4CEB9ACA5994D19AB0D6@e1705>
	<AANLkTikHB-QnDyT0WPsEML8yv4AACnYnHrE37gpRsLS2@mail.gmail.com>
Message-ID: <1296306637.8986.185.camel@luna.tc.commsmundi.com>

Nice to know, but in that case the destination actually rings (180).

See commented log:
http://pastebin.freeswitch.org/15168

Fran?ois.

On Fri, 2011-01-28 at 12:34 -0600, Anthony Minessale wrote:
> if it never rings, answer will still trigger it.
> 
> 
> On Fri, Jan 28, 2011 at 11:24 AM, Madovsky <infos at madovsky.org> wrote:
> > ah ok, maybe a wiki update would be useful.
> >
> >
> >
> > ----- Original Message -----
> > From: "Fran?ois Delawarde" <fdelawarde at wirelessmundi.com>
> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > Sent: Friday, January 28, 2011 12:18 PM
> > Subject: Re: [Freeswitch-users] execute_on_ring executing on answer
> >
> >
> >> It's some cool feature made by Anthony that allows me to specify the
> >> separator.
> >>
> >> in ^^:PCMA:G722
> >> ^^: means the separator is now : instead of ,
> >>
> >> Useful in the [] or {} case because the coma is already used to separate
> >> variables.
> >>
> >> Fran?ois.
> >>
> >> On Fri, 2011-01-28 at 12:12 -0500, Madovsky wrote:
> >>> what means the ^^ in your codec string ?
> >>>
> >>> ----- Original Message -----
> >>> From: "Fran?ois Delawarde" <fdelawarde at wirelessmundi.com>
> >>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> >>> Sent: Friday, January 28, 2011 6:47 AM
> >>> Subject: [Freeswitch-users] execute_on_ring executing on answer
> >>>
> >>>
> >>> > Hi,
> >>> >
> >>> > Doing some testing with this morning's git (Fri Jan 28) I just found
> >>> > out
> >>> > that the execute_on_ring application runs when the destination answers
> >>> > instead of when it rings.
> >>> >
> >>> > So far, I can't seem to find out the reason. Could it be some
> >>> > configuration issue?
> >>> >
> >>> >
> >>> > Here a call log showing the phenomenon with a simple bridge:
> >>> >
> >>> > <action application="bridge"
> >>> > data="[execute_on_ring=info,absolute_codec_string=^^:PCMA:G722]sofia/192.168.10.1/sip:2103 at 192.168.10.22:5060"/>
> >>> >
> >>> > http://pastebin.freeswitch.org/15168
> >>> >
> >>> >
> >>> > Thanks,
> >>> > Fran?ois.
> >>> >
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> 



From avi at avimarcus.net  Sat Jan 29 19:19:44 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sat, 29 Jan 2011 18:19:44 +0200
Subject: [Freeswitch-users] Number mapping (DID like)
In-Reply-To: <AANLkTi=hbQNyrM-iFExykvpotGPqmcRvvz1J_Y=YvMmR@mail.gmail.com>
References: <1806109314-1296182728-cardhu_decombobulator_blackberry.rim.net-210966170-@b25.c2.bise3.blackberry>
	<AANLkTinujT2YjspkUMn9fEnkk0w=vCH5wCWoZnQ07LLZ@mail.gmail.com>
	<AANLkTikf2gS+=3NMUb=HW_QasRxiDZ3Ny+0a5=0d5VGK@mail.gmail.com>
	<AANLkTi=hbQNyrM-iFExykvpotGPqmcRvvz1J_Y=YvMmR@mail.gmail.com>
Message-ID: <AANLkTikuDkxJv+uX_W2mvs6qb_d42LaNuSM1_pwCTtGq@mail.gmail.com>

FusionPBX (and maybe bluebox?) lets you create users in bulk, and has a
device provisioning ability. Only a few devices have templates ready to go,
but the system is there and it's pretty easy to add new ones.
-Avi Marcus

On Sat, Jan 29, 2011 at 2:44 AM, budi wibowo <bwibowo at gmail.com> wrote:

> something like call forwarding,
> found this http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect
> <http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect>but still
> not work.
> other question, is there any tool for mass provisioning/ bulk provisioning?
> say i want to create user 1000 - 1100 without adding it one by one
>
>
> regards
>
> budi
>
>
>
> On Fri, Jan 28, 2011 at 3:41 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> Propaby, but that could mean many things. Do you mean rewrite the number
>> in the CDRs? Allow the owner of the extension to set a call forwarding /
>> follow me? Or basically to use 1001 as a speeddial for the one user?
>> I use mod_xml_curl extensively which allow me to query a database before I
>> do just about anything with te normal dialplan.
>> First, any 1-3 digit numbers it checks a extension-specific and then
>> global speeddial table.
>> Soon, I intend to have it query FusionPBX's follow-me settings so that I
>> can implement that with poper billing.
>> Some of tis might help you?
>> -Avi Marcus
>>
>> > On Jan 28, 2011 4:46 AM, "Budi wibowo" <bwibowo at gmail.com> wrote:
>> > > Hi
>> > > I want to do B# (called) rewriting, for example
>> > >>From user 1000 call to 1001, and freeswitch rewrite 1001 to other
>> number (pstn).
>> > > Is it doable with freeswitch?
>> > >
>> > > Regards
>> > > Budi
>> > >
>> > > _______________________________________________
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From curriegrad2004 at gmail.com  Sat Jan 29 19:57:39 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 29 Jan 2011 08:57:39 -0800
Subject: [Freeswitch-users] Virtualisation and freeswitch
In-Reply-To: <AANLkTimtnMqFyrXFEYirmhLgqPJxaWsYWowYyiLi_k_=@mail.gmail.com>
References: <AANLkTimZTwZfUYMjC1ddAQMWHHWSLH=BLajv0cgSux4B@mail.gmail.com>
	<AANLkTikQbXLdCb4nNFEbfte-EAP8ytJyfURMALC6kwtK@mail.gmail.com>
	<AANLkTinB17uNUKtEQAtuTgCGPf_61jnErGmv7+vXV_n+@mail.gmail.com>
	<AANLkTimtnMqFyrXFEYirmhLgqPJxaWsYWowYyiLi_k_=@mail.gmail.com>
Message-ID: <AANLkTikhDhRL_hXcosBQzq9_vD75_=G0Ju8+3nJ9jyhn@mail.gmail.com>

I noticed the wiki telling people not to use Xen, why is that?

On Sat, Jan 29, 2011 at 1:30 AM, dome at tel.co.th <dome at tel.co.th> wrote:
> (from wiki http://wiki.freeswitch.org/wiki/Virtualization)
> if your FS running with high traffic please use
> /etc/vz/conf/ve-unlimited.conf-sample for default template
>
> Dome C.
>
> On Sat, Jan 29, 2011 at 7:23 AM, Seven Du <dujinfang at gmail.com> wrote:
>> We had used OpenVZ for a while, it was good given we never tested on load.
>>
>> Another option besides OpenVZ is lxc , which is similar to OpenVZ. I
>> runs some lxc VMs for other projects but never tried with FS.
>>
>> On Sat, Jan 29, 2011 at 6:03 AM, Brent Paddon <brent at overthewire.com.au> wrote:
>>> Why would it possibly matter if it's more of a 'jail' than 'virtualisation'
>>> ? ? Isn't the outcome the important thing ?
>>> In regards OpenVZ there is an ISO installer called Proxmox which helps you
>>> administer the whole system and gives you a pretty decent web interface for
>>> management. ?We use it extensively for voice applications.
>>> http://www.proxmox.com/products/proxmox-ve
>>> Brent
>>>
>>> On Fri, Jan 28, 2011 at 9:52 PM, Grygoriy Dobrovolskyy <megahohol at gmail.com>
>>> wrote:
>>>>
>>>> Hello list, le me introduce myself first, I am Grigoriy Dobrovolskyy
>>>> currently working as voip administrator and call centeres integrator. I
>>>> worked a lot with asterisk, i think Fs is a great software but not easy to
>>>> learn.
>>>> I have googled my question and did not found a clear answer.
>>>> With what technology i can virtualise freeswitch without timer problems ?
>>>> I know that i can use OpenVZ but it's more 'jail' then virtualisation. I
>>>> heard that Amazon EC with High CPU instance works (confirm?)
>>>> Can someone give me info of a succesfull installations with Conferences,
>>>> Moh ?/ mixing ? And a technology used ?
>>>> Thank you.
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> --
>>> Brent Paddon
>>>
>>> Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
>>> www.overthewire.com.au
>>> Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> About: http://about.me/dujinfang
>> Blog: http://www.dujinfang.com
>> Proj:? http://www.freeswitch.org.cn
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From gabe at gundy.org  Sat Jan 29 22:25:29 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 29 Jan 2011 12:25:29 -0700
Subject: [Freeswitch-users] Polycom and registering with domains
	(user@domain.tld@domain.tld)
Message-ID: <AANLkTikb2Mzw9oHagSHE=oe8ABSwOWOxwRE6CdNSO8WS@mail.gmail.com>

All,

I have a multi-tenant FreeSWITCH server up and running.  We have all
the proper SVR records setup for each domain and everything works
great.  Most SIP clients Just Work (tm), however, the new Polycom
phones I have don't seem to auth properly with domains.  I've tried
every combination of settings (via the web config, I haven't setup a
TFTP server yet).  The issue is that auth has the domain twice, like
so: user at domain.tld@domain.tld.

It seems like the Internet has threads of discussion where others
express frustration with this, but I can't seem to find a documented
solution.  It's maddening because every other User Agent Client works
swimmingly and no matter what I've tried, I can't get the right
combination of settings on the Polycom.  I love the phones, but this
is getting old :)

Anyway, any pointers for me?

Thanks,
Gabe

sofia.conf:
*************************************************************
[SNIP]

<domains>
  <domain name="all" alias="true" parse="false" />
</domains>
<settings>
  <param name="log-auth-failures" value="true" />
  <param name="forward-unsolicited-mwi-notify" value="false" />
  <param name="context" value="public" />
  <param name="sip-port" value="$${internal_sip_port}" />
  <param name="dialplan" value="XML" />
  <param name="inbound-codec-prefs" value="$${global_codec_prefs}" />
  <param name="outbound-codec-prefs" value="$${global_codec_prefs}" />
  <param name="rtp-ip" value="$${internal_ip}" />
  <param name="sip-ip" value="$${internal_ip}" />
  <param name="hold-music" value="$${hold_music}" />
  <param name="record-path" value="$${recordings_dir}" />
  <param name="record-template"
value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"
/>
  <param name="auth-calls" value="true" />
  <param name="ext-rtp-ip" value="auto-nat" />
  <param name="ext-sip-ip" value="auto-nat" />
</settings>

[/SNIP]


From anthony.minessale at gmail.com  Sat Jan 29 22:53:19 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 29 Jan 2011 13:53:19 -0600
Subject: [Freeswitch-users] 200 OK without SDP
In-Reply-To: <AANLkTi=KqOmroAYFRFyTfMPd_9_3Xjw0v3sdMtaSmdrc@mail.gmail.com>
References: <201101280730.17483.sos@sokhapkin.dyndns.org>
	<201101280937.50415.sos@sokhapkin.dyndns.org>
	<AANLkTimF8M2tNk5+6su6gXR+ezEYOUk6p1_3KOoNk4S-@mail.gmail.com>
	<201101281613.22368.sos@sokhapkin.dyndns.org>
	<AANLkTi=KqOmroAYFRFyTfMPd_9_3Xjw0v3sdMtaSmdrc@mail.gmail.com>
Message-ID: <AANLkTikzNqy3bOBXAsVdEumHyWfft_aPc-2jAfh+Jshi@mail.gmail.com>

Session timer code is part of the sofia sip libraries not FS.
If it's not compatible to use bypass and session timers together it
may not be something we can fix.

All I can think of to try is to set "enable-soa" to "false" in profile
params or sip_enable_soa=false channel var on both legs of the call.
This would tell sofia not to mess with the SDP at all and might serve
as a workaround.



On Fri, Jan 28, 2011 at 3:27 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> That's what I'm seeing here too.
>
> Thanks!
>
> On Fri, Jan 28, 2011 at 4:13 PM, Sergey Okhapkin
> <sos at sokhapkin.dyndns.org> wrote:
>> There is no problem if FS is in media path.
>>
>
> --
> Kristian Kielhofner
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Sat Jan 29 22:58:52 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 29 Jan 2011 13:58:52 -0600
Subject: [Freeswitch-users] Disable Message Taking for a Voice Mail Box
In-Reply-To: <4D435CD8.5020106@gmail.com>
References: <4D435CD8.5020106@gmail.com>
Message-ID: <AANLkTi=8CUecGMS1-RkTW_eTP7R6R2nc6J9VLK+AM_zs@mail.gmail.com>

couldn't you just do

extension 1000
"record", "file.wav"

extension 1001
"playback" "fille.wav"
"hangup"




On Fri, Jan 28, 2011 at 6:18 PM, Phone <phone.bytes at gmail.com> wrote:
> Is there a setting to disable a Voice Mail Box from taking a message?
>
> I would like to set some up to play an announcement and then hang up.
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From chris at cloudtel.com  Sat Jan 29 23:06:21 2011
From: chris at cloudtel.com (Chris Burns)
Date: Sat, 29 Jan 2011 15:06:21 -0500
Subject: [Freeswitch-users] Virtualisation and freeswitch
In-Reply-To: <AANLkTikhDhRL_hXcosBQzq9_vD75_=G0Ju8+3nJ9jyhn@mail.gmail.com>
References: <AANLkTimZTwZfUYMjC1ddAQMWHHWSLH=BLajv0cgSux4B@mail.gmail.com>
	<AANLkTikQbXLdCb4nNFEbfte-EAP8ytJyfURMALC6kwtK@mail.gmail.com>
	<AANLkTinB17uNUKtEQAtuTgCGPf_61jnErGmv7+vXV_n+@mail.gmail.com>
	<AANLkTimtnMqFyrXFEYirmhLgqPJxaWsYWowYyiLi_k_=@mail.gmail.com>
	<AANLkTikhDhRL_hXcosBQzq9_vD75_=G0Ju8+3nJ9jyhn@mail.gmail.com>
Message-ID: <AANLkTikiQUXWnrWZxoLHWOb3DL6MvXsYBmTdCUkcgjBK@mail.gmail.com>

Many Xen kernels are compiled with a lower resolution kernel timer. Most
applications will not notice this, but a VoIP application doing async RTP
probably will. It can make audio quality poor, especially if something other
than FreeSWITCH is trying to get CPU time while you are handling media. That
is the main reason that I found. I have had OK results with Xen with the
proper configuration.

On Sat, Jan 29, 2011 at 11:57 AM, curriegrad2004
<curriegrad2004 at gmail.com>wrote:

> I noticed the wiki telling people not to use Xen, why is that?
>
> On Sat, Jan 29, 2011 at 1:30 AM, dome at tel.co.th <dome at tel.co.th> wrote:
> > (from wiki http://wiki.freeswitch.org/wiki/Virtualization)
> > if your FS running with high traffic please use
> > /etc/vz/conf/ve-unlimited.conf-sample for default template
> >
> > Dome C.
> >
> > On Sat, Jan 29, 2011 at 7:23 AM, Seven Du <dujinfang at gmail.com> wrote:
> >> We had used OpenVZ for a while, it was good given we never tested on
> load.
> >>
> >> Another option besides OpenVZ is lxc , which is similar to OpenVZ. I
> >> runs some lxc VMs for other projects but never tried with FS.
> >>
> >> On Sat, Jan 29, 2011 at 6:03 AM, Brent Paddon <brent at overthewire.com.au>
> wrote:
> >>> Why would it possibly matter if it's more of a 'jail' than
> 'virtualisation'
> >>> ?   Isn't the outcome the important thing ?
> >>> In regards OpenVZ there is an ISO installer called Proxmox which helps
> you
> >>> administer the whole system and gives you a pretty decent web interface
> for
> >>> management.  We use it extensively for voice applications.
> >>> http://www.proxmox.com/products/proxmox-ve
> >>> Brent
> >>>
> >>> On Fri, Jan 28, 2011 at 9:52 PM, Grygoriy Dobrovolskyy <
> megahohol at gmail.com>
> >>> wrote:
> >>>>
> >>>> Hello list, le me introduce myself first, I am Grigoriy Dobrovolskyy
> >>>> currently working as voip administrator and call centeres integrator.
> I
> >>>> worked a lot with asterisk, i think Fs is a great software but not
> easy to
> >>>> learn.
> >>>> I have googled my question and did not found a clear answer.
> >>>> With what technology i can virtualise freeswitch without timer
> problems ?
> >>>> I know that i can use OpenVZ but it's more 'jail' then virtualisation.
> I
> >>>> heard that Amazon EC with High CPU instance works (confirm?)
> >>>> Can someone give me info of a succesfull installations with
> Conferences,
> >>>> Moh  / mixing ? And a technology used ?
> >>>> Thank you.
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>>
> >>>
> >>> --
> >>> --
> >>> Brent Paddon
> >>>
> >>> Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
> >>> www.overthewire.com.au
> >>> Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >>
> >>
> >> --
> >> About: http://about.me/dujinfang
> >> Blog: http://www.dujinfang.com
> >> Proj:  http://www.freeswitch.org.cn
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From frank at telonium.com  Sat Jan 29 23:12:04 2011
From: frank at telonium.com (Frank Park)
Date: Sat, 29 Jan 2011 15:12:04 -0500
Subject: [Freeswitch-users] Choppy VM when using loopback (on Debian
	lenny)
In-Reply-To: <9894FEC6-E451-4993-A85A-2CD910E916B1@ipeva.fr>
References: <4D42B8EE.7030208@estation.dk>
	<7C6F3FC7-445F-4157-BD59-D76A705E4CE7@carmickle.com>
	<9894FEC6-E451-4993-A85A-2CD910E916B1@ipeva.fr>
Message-ID: <AANLkTim+aBjRiAvxDUF1O1vut_CJpqgct2OqCNVxi0q4@mail.gmail.com>

Yeah. I, too, don't see the correlation between the NTP and hw timer..
I am not familiar with the timer_test command and what it's measuring, but
of the 50 tests it ran, min is 19089 and max is 20713.

Frank


On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Frank,
>
> I fail to see the relationship between the hw timer and NTP.
> Can you please elaborate ?
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 28/01/2011 ? 19:49, Frank Carmickle a ?crit :
>
> Hi
>
> On Jan 28, 2011, at 7:39 AM, ?yvind Albrigtsen wrote:
>
> Hi,
>
>
> I'm using latest git-version of Freeswitch, and when I go to voicemail
>
> when calling a number the sound playback is choppy and it skips some of
>
> the digits in the number I called.
>
>
> What kind of results do you get from timer_test at the fs_cli?  Are you
> running on hardware or are you virtualized?  What is your clock source set
> to and what are your available clock source options?  See
> /sys/devices/system/clocksource/clocksource0/available_clocksource and
> /sys/devices/system/clocksource/clocksource0/current_clocksource.  I am
> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to
> hang at 19998/19999 which works very well for me.  When I was having problem
> it was reporting numbers all over the map from 17400 to 22600 with lots of
> randomness in between.  I have my clocksource set to jiffies and xen
> independent wallclock set to 1.  Of course at that point you need to have
> ntp running against a bunch of servers to drive your clock nice and steady.
>  I know my set up is probably a lot different than yours but I thought I'd
> toss it out there to show that some of the harshest conditions can be dealt
> with and don't give up trying.  If you are running on hardware with a cpu
> that doesn't have constant_tsc then you might have some problems.  Just play
> with the different timer options until you find the one that works.
>
> HTH
> --FC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 

----=======================----
Frank Park
Telonium Communications, LLC
frank at telonium.com
http://www.telonium.com
Follow Us on Twitter: @GetTelonium
404-566-8888 x1001 Office
404-939-4242 Cell
----=======================----
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From phone.bytes at gmail.com  Sat Jan 29 23:13:05 2011
From: phone.bytes at gmail.com (phone.bytes)
Date: Sat, 29 Jan 2011 13:13:05 -0700
Subject: [Freeswitch-users] Disable Message Taking for a Voice Mail Box
In-Reply-To: <AANLkTi=8CUecGMS1-RkTW_eTP7R6R2nc6J9VLK+AM_zs@mail.gmail.com>
References: <4D435CD8.5020106@gmail.com>
	<AANLkTi=8CUecGMS1-RkTW_eTP7R6R2nc6J9VLK+AM_zs@mail.gmail.com>
Message-ID: <4D4474D1.1020501@gmail.com>

Sure that would work.  Just thought I had read about a flag that took 
care of it and would just use one regular mailbox.

Thanks for the reply . . and for all that you do!

On 1/29/2011 12:58 PM, Anthony Minessale wrote:
> couldn't you just do
>
> extension 1000
> "record", "file.wav"
>
> extension 1001
> "playback" "fille.wav"
> "hangup"
>
>
>
>
> On Fri, Jan 28, 2011 at 6:18 PM, Phone<phone.bytes at gmail.com>  wrote:
>> Is there a setting to disable a Voice Mail Box from taking a message?
>>
>> I would like to set some up to play an announcement and then hang up.
>>
>> Thanks
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>



From anthony.minessale at gmail.com  Sat Jan 29 23:48:08 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 29 Jan 2011 14:48:08 -0600
Subject: [Freeswitch-users] Choppy VM when using loopback (on Debian
	lenny)
In-Reply-To: <AANLkTim+aBjRiAvxDUF1O1vut_CJpqgct2OqCNVxi0q4@mail.gmail.com>
References: <4D42B8EE.7030208@estation.dk>
	<7C6F3FC7-445F-4157-BD59-D76A705E4CE7@carmickle.com>
	<9894FEC6-E451-4993-A85A-2CD910E916B1@ipeva.fr>
	<AANLkTim+aBjRiAvxDUF1O1vut_CJpqgct2OqCNVxi0q4@mail.gmail.com>
Message-ID: <AANLkTiktckovGv4Ksnn_56EzpvTTSnfG+iD1vPX1_4RM@mail.gmail.com>

Everyone should try latest GIT before pondering any further because I
added a patch like 2 days ago to adress this issue.


On Sat, Jan 29, 2011 at 2:12 PM, Frank Park <frank at telonium.com> wrote:
> Yeah. I, too, don't see the correlation between the NTP and hw timer..
> I am not familiar with the timer_test command and what it's measuring, but
> of the 50 tests it ran, min is 19089 and max is 20713.
> Frank
>
>
> On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone <david.ponzone at ipeva.fr>
> wrote:
>>
>> Frank,
>> I fail to see the relationship between the hw timer and NTP.
>> Can you please elaborate ?
>> David Ponzone ?Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel: ? ? ?01 74 03 18 97
>> gsm: ? 06 66 98 76 34
>> Service Client?IPeva
>> tel: ? ? ?0811 46 26 26
>> www.ipeva.fr? -? ?www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>> Le 28/01/2011 ? 19:49, Frank Carmickle a ?crit :
>>
>> Hi
>>
>> On Jan 28, 2011, at 7:39 AM, ?yvind Albrigtsen wrote:
>>
>> Hi,
>>
>> I'm using latest git-version of Freeswitch, and when I go to voicemail
>>
>> when calling a number the sound playback is choppy and it skips some of
>>
>> the digits in the number I called.
>>
>> What kind of results do you get from timer_test at the fs_cli? ?Are you
>> running on hardware or are you virtualized? ?What is your clock source set
>> to and what are your available clock source options? ?See
>> /sys/devices/system/clocksource/clocksource0/available_clocksource and
>> /sys/devices/system/clocksource/clocksource0/current_clocksource. ?I am
>> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to
>> hang at 19998/19999 which works very well for me. ?When I was having problem
>> it was reporting numbers all over the map from 17400 to 22600 with lots of
>> randomness in between. ?I have my clocksource set to jiffies and xen
>> independent wallclock set to 1. ?Of course at that point you need to have
>> ntp running against a bunch of servers to drive your clock nice and steady.
>> ?I know my set up is probably a lot different than yours but I thought I'd
>> toss it out there to show that some of the harshest conditions can be dealt
>> with and don't give up trying. ?If you are running on hardware with a cpu
>> that doesn't have constant_tsc then you might have some problems. ?Just play
>> with the different timer options until you find the one that works.
>>
>> HTH
>> --FC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> ----=======================----
> Frank Park
> Telonium Communications, LLC
> frank at telonium.com
> http://www.telonium.com
> Follow Us on Twitter: @GetTelonium
> 404-566-8888 x1001 Office
> 404-939-4242 Cell
> ----=======================----
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From lloyd.aloysius at gmail.com  Sun Jan 30 00:35:10 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sat, 29 Jan 2011 16:35:10 -0500
Subject: [Freeswitch-users] Polycom and registering with domains
	(user@domain.tld@domain.tld)
In-Reply-To: <AANLkTikb2Mzw9oHagSHE=oe8ABSwOWOxwRE6CdNSO8WS@mail.gmail.com>
References: <AANLkTikb2Mzw9oHagSHE=oe8ABSwOWOxwRE6CdNSO8WS@mail.gmail.com>
Message-ID: <AANLkTikieRwaDVgNhPRd1BSV1gHFkYyzdiUoASVLhbij@mail.gmail.com>

Gabriel,

I had the similar issues with Polycom phones. Stay away from the web
configuration.

All you need to setup the FTP or TFTP server and use the configuration
files. I am currently using polycom phones in a Multi-Tenant [SRV
Records] environment without any issue afterstart use the configuration
files.



Thanks
Lloyd


On Sat, Jan 29, 2011 at 2:25 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> All,
>
> I have a multi-tenant FreeSWITCH server up and running.  We have all
> the proper SVR records setup for each domain and everything works
> great.  Most SIP clients Just Work (tm), however, the new Polycom
> phones I have don't seem to auth properly with domains.  I've tried
> every combination of settings (via the web config, I haven't setup a
> TFTP server yet).  The issue is that auth has the domain twice, like
> so: user at domain.tld@domain.tld.
>
> It seems like the Internet has threads of discussion where others
> express frustration with this, but I can't seem to find a documented
> solution.  It's maddening because every other User Agent Client works
> swimmingly and no matter what I've tried, I can't get the right
> combination of settings on the Polycom.  I love the phones, but this
> is getting old :)
>
> Anyway, any pointers for me?
>
> Thanks,
> Gabe
>
> sofia.conf:
> *************************************************************
> [SNIP]
>
> <domains>
>  <domain name="all" alias="true" parse="false" />
> </domains>
> <settings>
>  <param name="log-auth-failures" value="true" />
>  <param name="forward-unsolicited-mwi-notify" value="false" />
>  <param name="context" value="public" />
>  <param name="sip-port" value="$${internal_sip_port}" />
>  <param name="dialplan" value="XML" />
>  <param name="inbound-codec-prefs" value="$${global_codec_prefs}" />
>  <param name="outbound-codec-prefs" value="$${global_codec_prefs}" />
>  <param name="rtp-ip" value="$${internal_ip}" />
>  <param name="sip-ip" value="$${internal_ip}" />
>  <param name="hold-music" value="$${hold_music}" />
>  <param name="record-path" value="$${recordings_dir}" />
>  <param name="record-template"
>
> value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"
> />
>  <param name="auth-calls" value="true" />
>  <param name="ext-rtp-ip" value="auto-nat" />
>  <param name="ext-sip-ip" value="auto-nat" />
> </settings>
>
> [/SNIP]
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Sun Jan 30 00:38:24 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 29 Jan 2011 16:38:24 -0500
Subject: [Freeswitch-users] esl outbound socket sendmsg
Message-ID: <7B10DDA67C574F49B13CC6E9F0E7138B@e1705>

Just for info sometimes it needs
to put usleep() between different sendmsg to let the process
wait the answer in case of async full
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From bektas at yahoo.com  Sat Jan 29 02:15:08 2011
From: bektas at yahoo.com (Kenan BEKTAS)
Date: Fri, 28 Jan 2011 15:15:08 -0800 (PST)
Subject: [Freeswitch-users] Remote LogIn to Freeswitch?
In-Reply-To: <AANLkTi=CMg36dwcoRufQAA_7c14ZMmm+5ozeg4FKGDut@mail.gmail.com>
Message-ID: <10183.49543.qm@web114709.mail.gq1.yahoo.com>

a) according to Wiki, you should set port 5080 for remote connections.
b) if no sound for some reason, then, it is most likely a RTP issue. Make sure RTP port ranges is open on your router.

---

Kenan
www.dbstreams.ca




--- On Fri, 1/28/11, Joao Leme <joaocarlosleme at gmail.com> wrote:

From: Joao Leme <joaocarlosleme at gmail.com>
Subject: Re: [Freeswitch-users] Remote LogIn to Freeswitch?
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Friday, January 28, 2011, 5:31 PM

FINALLY I got it to work, I can now remotely log in on the internal profile (port 5060) after placing the computer running Freeswitch on DMZ (changed on Firewall),?although that's not the ideal solution, so I was wondering why it wasn't working before. I had the firewall set up according to the wiki on "http://wiki.freeswitch.org/wiki/Firewall" and other than that I?changed the "domain" on vars.xml to my external ip of the router, it created an ALIAS so that I can use as the "Domain" on X-lite to log on. So I had the internal profile mod_sofia at 192.168.X.XX:5060 and the Alias 76.XXX.XX.XX, that was before and I was able to LogIn, make calls but NO SOUND. NOW once I forwarded all the port through DMZ, this particular router 2701HG-B Gateway (Att) required me to change my (freeSwitch running computer) static ipv4 (192.XXX.X.XX) to DHCP and then assigned me the external IP (76.XXX.XX.XX) as my a Ipv4. Now the internal profile (sofia status) is
 mod_sofia at 76.XXX.XX.XX:5060 and I'm not using the ALIAS.


SO...I was wondering...it wasn't working before because 1) A Firewall configuration missing? 2) A NAT/Firewall problem or 3) Freeswitch internal profile running on local machine ipv4 192..... and I logging on using the router external IP 76.XX and the Alias didn't do the job??????


Sorry there is some nonsense question, I'm a beginner and any help is appreciated.Thanks,John
On Tue, Jan 11, 2011 at 1:37 AM, Steven Ayre <steveayre at gmail.com> wrote:


If you're able to dial in but you're getting no sound, it's probably

NAT stopping the audio get through.



There's quite a bit of information on NAT on the Wiki that might be of use.



http://wiki.freeswitch.org/wiki/NAT



-Steve





On 11 January 2011 00:39, Joao Leme <joaocarlosleme at gmail.com> wrote:

> Hi There,

> What do I have to do to be able to LogIn to Freeswitch from Home (server is

> located at office) starting from the basic/original configuration?

> I'm using X-Lite. I've been able to LogIn replacing the internal IP by the

> external IP from the Office but the sound is not working so I wanted to know

> what are the configuration changes that have to be done to allow it. Do I

> have to create a different profile? I want be able to do the same just as if

> I was at the Office.

> Thanks,

> John

> _______________________________________________

> FreeSWITCH-users mailing list

> FreeSWITCH-users at lists.freeswitch.org

> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

> http://www.freeswitch.org

>

>



_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org




-----Inline Attachment Follows-----

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



      
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From megahohol at gmail.com  Sat Jan 29 03:36:11 2011
From: megahohol at gmail.com (Grygoriy Dobrovolskyy)
Date: Sat, 29 Jan 2011 01:36:11 +0100
Subject: [Freeswitch-users] Virtualisation and freeswitch
In-Reply-To: <AANLkTikQbXLdCb4nNFEbfte-EAP8ytJyfURMALC6kwtK@mail.gmail.com>
References: <AANLkTimZTwZfUYMjC1ddAQMWHHWSLH=BLajv0cgSux4B@mail.gmail.com>
	<AANLkTikQbXLdCb4nNFEbfte-EAP8ytJyfURMALC6kwtK@mail.gmail.com>
Message-ID: <AANLkTimwxcBnVigya0b_68sfG=CPDEb20ERJ26mLZkar@mail.gmail.com>

I've heard about proxmox, and even tested it a while ago. It's a good
solution.
The problem of Openvz is the maintenance mess. In clear you always need to
keep updated kernels, possible have compatibility issues and other stuff
when maintaining your virtual infrastructure up to date / migrating servers
back and forth. With a complete hypervisor this problems do not exist.



2011/1/28 Brent Paddon <brent at overthewire.com.au>

> Why would it possibly matter if it's more of a 'jail' than 'virtualisation'
> ?   Isn't the outcome the important thing ?
>
> In regards OpenVZ there is an ISO installer called Proxmox which helps you
> administer the whole system and gives you a pretty decent web interface for
> management.  We use it extensively for voice applications.
>
> http://www.proxmox.com/products/proxmox-ve
>
> Brent
>
> On Fri, Jan 28, 2011 at 9:52 PM, Grygoriy Dobrovolskyy <
> megahohol at gmail.com> wrote:
>
>> Hello list, le me introduce myself first, I am Grigoriy Dobrovolskyy
>> currently working as voip administrator and call centeres integrator. I
>> worked a lot with asterisk, i think Fs is a great software but not easy to
>> learn.
>>
>> I have googled my question and did not found a clear answer.
>>
>> With what technology i can virtualise freeswitch without timer problems ?
>>
>> I know that i can use OpenVZ but it's more 'jail' then virtualisation. I
>> heard that Amazon EC with High CPU instance works (confirm?)
>>
>> Can someone give me info of a succesfull installations with Conferences,
>> Moh  / mixing ? And a technology used ?
>>
>> Thank you.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> --
> Brent Paddon
>
> Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
> www.overthewire.com.au
> Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From megahohol at gmail.com  Sat Jan 29 03:39:29 2011
From: megahohol at gmail.com (Grygoriy Dobrovolskyy)
Date: Sat, 29 Jan 2011 01:39:29 +0100
Subject: [Freeswitch-users] Virtualisation and freeswitch
In-Reply-To: <AANLkTimwxcBnVigya0b_68sfG=CPDEb20ERJ26mLZkar@mail.gmail.com>
References: <AANLkTimZTwZfUYMjC1ddAQMWHHWSLH=BLajv0cgSux4B@mail.gmail.com>
	<AANLkTikQbXLdCb4nNFEbfte-EAP8ytJyfURMALC6kwtK@mail.gmail.com>
	<AANLkTimwxcBnVigya0b_68sfG=CPDEb20ERJ26mLZkar@mail.gmail.com>
Message-ID: <AANLkTi=fRLdjSpwpd7ny1Tk05jnctOrGwTsmpxUPQrjO@mail.gmail.com>

Another interesting point i have asked in near topic in opensips mailing
list about timer and Sa?l Ibarra Corretg? told me this:


About Asterisk, the timing issues were resolved long ago and works very well
with DAHDI if HPET is enabled. Latest releases support other timing sources:
TimerFD and pthreads based timing. AppKonference did the timing and on its
own, but it also brought another advantage: the possibility of not to use
Zaptel/DAHDI (MeetMe uses DAHDI mixing engine, so it's limited to 8KHz...).

So this virtualization timer problem is not so complicated to resolve if
others done it ?


2011/1/29 Grygoriy Dobrovolskyy <megahohol at gmail.com>

> I've heard about proxmox, and even tested it a while ago. It's a good
> solution.
> The problem of Openvz is the maintenance mess. In clear you always need to
> keep updated kernels, possible have compatibility issues and other stuff
> when maintaining your virtual infrastructure up to date / migrating servers
> back and forth. With a complete hypervisor this problems do not exist.
>
>
>
> 2011/1/28 Brent Paddon <brent at overthewire.com.au>
>
> Why would it possibly matter if it's more of a 'jail' than 'virtualisation'
>> ?   Isn't the outcome the important thing ?
>>
>> In regards OpenVZ there is an ISO installer called Proxmox which helps you
>> administer the whole system and gives you a pretty decent web interface for
>> management.  We use it extensively for voice applications.
>>
>> http://www.proxmox.com/products/proxmox-ve
>>
>> Brent
>>
>> On Fri, Jan 28, 2011 at 9:52 PM, Grygoriy Dobrovolskyy <
>> megahohol at gmail.com> wrote:
>>
>>> Hello list, le me introduce myself first, I am Grigoriy Dobrovolskyy
>>> currently working as voip administrator and call centeres integrator. I
>>> worked a lot with asterisk, i think Fs is a great software but not easy to
>>> learn.
>>>
>>> I have googled my question and did not found a clear answer.
>>>
>>> With what technology i can virtualise freeswitch without timer problems ?
>>>
>>> I know that i can use OpenVZ but it's more 'jail' then virtualisation. I
>>> heard that Amazon EC with High CPU instance works (confirm?)
>>>
>>> Can someone give me info of a succesfull installations with Conferences,
>>> Moh  / mixing ? And a technology used ?
>>>
>>> Thank you.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> --
>> Brent Paddon
>>
>> Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
>> www.overthewire.com.au
>> Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From marcin321 at hotmail.com  Sat Jan 29 03:54:19 2011
From: marcin321 at hotmail.com (Marcin Wojtowicz)
Date: Fri, 28 Jan 2011 19:54:19 -0500
Subject: [Freeswitch-users] Outbound only calls don't connect when
	bypass_media is true.
Message-ID: <COL118-W52E5226FC3337085AB6AFAE5E00@phx.gbl>



Hello,



I'm a new user of freeswitch, so please bear with me. I have the

following setup:

voip.ms <- SIP over UDP -> my desktop running freeswitch <- SIP over TCP 
-> my nokia cellphone on AT&T wireless. This setup is intended to 
conserve the battery usage.

I've managed to make everything work well when I'm calling in over any 
phone to my cell phone, and freeswitch is enabled to work in 
bypass_media = true, even though by cell is behind NAT on at&t's 
network. Things break when I pick up my cell and try to call my home 
phone (or any phone for that matter). This is the relevant snippet from 
my dialplan:

<extension name="outbound">

  <condition field="destination_number" 
expression="^1?([2-9]\d{2}[2-9]\d{6})$">

    <!--<action application="set" data="bypass_media=true"/>-->

    <action application="bridge" data="sofia/gateway/voip.ms/1$1"/>

  </condition>

</extension>



Like shown above, my call will go to my home phone. When I uncomment the 
bypass_media tag, my call will not connect. Here are the siptraces

I replaced my real home phone number in the with "MYPHONE".



recv 940 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.406250:

   ------------------------------------------------------------------------

   INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0

   Via: SIP/2.0/TCP 
10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport

   From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj

   To: <sip:MYPHONE at 192.168.1.100>

   Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>

   Supported: 100rel,timer

   CSeq: 5244503 INVITE

   Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn

   Allow: UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE

   User-Agent: S60 RM-624 v 20.2.042 (en)

   Expires: 120

   Privacy: None

   Session-Expires: 1800

   Max-Forwards: 70

   Content-Type: application/sdp

   Accept-Language: en

   Content-Length: 292



   v=0

   o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6

   s=-

   c=IN IP4 10.153.174.6

   t=0 0

   m=audio 49152 RTP/AVP 18 97 98

   a=sendrecv

   a=rtpmap:18 G729/8000

   a=ptime:20

   a=maxptime:40

   a=fmtp:18 annexb=no

   a=rtpmap:97 iLBC/8000

   a=rtpmap:98 telephone-event/8000

   a=fmtp:98 0-15

   ------------------------------------------------------------------------

send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:

   ------------------------------------------------------------------------

   SIP/2.0 100 Trying

   Via: SIP/2.0/TCP 
10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180

   From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj

   To: <sip:MYPHONE at 192.168.1.100>

   Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn

   CSeq: 5244503 INVITE

   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 
18-04-05 -0600

   Content-Length: 0



   ------------------------------------------------------------------------

send 875 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:

   ------------------------------------------------------------------------

   SIP/2.0 407 Proxy Authentication Required

   Via: SIP/2.0/TCP 
10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180

   From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj2011-01-28 
16:15:58.406250 [WARNING] sofia_reg.c:1241 SIP auth challenge (INVITE) 
on sofia profile 'internal' for [MYPHONE at 192.168.1.100] from ip 
32.136.78.180



   To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m

   Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn

   CSeq: 5244503 INVITE

   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 
18-04-05 -0600

   Accept: application/sdp

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

   Supported: timer, precondition, path, replaces

   Allow-Events: talk, hold, presence, dialog, line-seize, call-info, 
sla, include-session-description, presence.winfo, message-summary, refer

   Proxy-Authenticate: Digest realm="192.168.1.100", 
nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28", algorithm=MD5, qop="auth"

   Content-Length: 0



   ------------------------------------------------------------------------

recv 368 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.765625:

   ------------------------------------------------------------------------

   ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0

   Via: SIP/2.0/TCP 
10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport

   From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj

   To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m

   Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn

   CSeq: 5244503 ACK

   Supported: sec-agree

   Max-Forwards: 70

   Content-Length: 0



   ------------------------------------------------------------------------

recv 1222 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.406250:

   ------------------------------------------------------------------------

   INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0

   Via: SIP/2.0/TCP 
10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport

   From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj

   To: <sip:MYPHONE at 192.168.1.100>

   Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>

   Supported: 100rel,timer

   CSeq: 5244504 INVITE

   Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn

   Allow: UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE

   User-Agent: S60 RM-624 v 20.2.042 (en)

   Expires: 120

   Privacy: None

   Session-Expires: 1800

   Max-Forwards: 70

   Proxy-Authorization: Digest 
qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"

   Content-Type: application/sdp

   Accept-Language: en

   Content-Length: 292



   v=0

   o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6

   s=-

   c=IN IP4 10.153.174.6

   t=0 0

   m=audio 49152 RTP/AVP 18 97 98

   a=sendrecv

   a=rtpmap:18 G729/8000

   a=ptime:20

   a=maxptime:40

   a=fmtp:18 annexb=no

   a=rtpmap:97 iLBC/8000

   a=rtpmap:98 telephone-event/8000

   a=fmtp:98 0-15

   ------------------------------------------------------------------------

send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.406250:

   ------------------------------------------------------------------------

   SIP/2.0 100 Trying

   Via: SIP/2.0/TCP 
10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180

   From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj

   To: <sip:MYPHONE at 192.168.1.100>

   Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn

   CSeq: 5244504 INVITE

   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 
18-04-05 -0600

   Content-Length: 0



   ------------------------------------------------------------------------

2011-01-28 16:15:59.406250 [NOTICE] switch_channel.c:808 New Channel 
sofia/internal/1001 at 192.168.1.100 [e5841001-04bd-4e16-9519-64ff2c7a8c2f]

2011-01-28 16:15:59.421875 [INFO] mod_dialplan_xml.c:331 Processing 1001 
<1001>->MYPHONE in context default

2011-01-28 16:15:59.437500 [NOTICE] switch_channel.c:808 New Channel 
sofia/external/1MYPHONE [60940227-9ae0-4c0c-abdb-f3988852fae0]

send 1141 bytes to udp/[74.63.41.218]:5060 at 21:15:59.453125:

   ------------------------------------------------------------------------

   INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0

   Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS

   Max-Forwards: 69

   From: "Extension 1001" 
<sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S

   To: <sip:1MYPHONE at newyork.voip.ms>

   Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a

   CSeq: 7788615 INVITE

   Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>

   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 
18-04-05 -0600

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, REFER, NOTIFY

   Supported: timer, precondition, path, replaces

   Allow-Events: talk, hold, refer

   Content-Type: application/sdp

   Content-Disposition: session

   Content-Length: 280

   X-FS-Support: update_display

   Remote-Party-ID: "Extension 1001" 
<sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off



   v=0

   o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6

   s=-

   c=IN IP4 10.153.174.6

   t=0 0

   m=audio 49152 RTP/AVP 18 97 98

   a=rtpmap:18 G729/8000

   a=fmtp:18 annexb=no

   a=rtpmap:97 iLBC/8000

   a=rtpmap:98 telephone-event/8000

   a=fmtp:98 0-15

   a=ptime:20

   a=maxptime:40

   ------------------------------------------------------------------------

recv 570 bytes from udp/[74.63.41.218]:5060 at 21:15:59.468750:

   ------------------------------------------------------------------------

   SIP/2.0 407 Proxy Authentication Required

   Via: SIP/2.0/UDP 
69.125.20.15:5080;branch=z9hG4bKNp4ZFeKSD43tS;received=69.125.20.15;rport=5080

   From: "Extension 1001" 
<sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S

   To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843

   Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a

   CSeq: 7788615 INVITE

   User-Agent: VoIPMS/SERAST

   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

   Supported: replaces

   Proxy-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms", 
nonce="2d534dd6"

   Content-Length: 0



   ------------------------------------------------------------------------

send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:

   ------------------------------------------------------------------------

   ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0

   Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS

   Max-Forwards: 69

   From: "Extension 1001" 
<sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S

   To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843

   Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a

   CSeq: 7788615 ACK

   Content-Length: 0



   ------------------------------------------------------------------------

send 1344 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:

   ------------------------------------------------------------------------

   INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0

   Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN

   Max-Forwards: 69

   From: "Extension 1001" 
<sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S

   To: <sip:1MYPHONE at newyork.voip.ms>

   Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a

   CSeq: 7788616 INVITE

   Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>

   Expires: 300

   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 
18-04-05 -0600

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, REFER, NOTIFY

   Supported: timer, precondition, path, replaces

   Allow-Events: talk, hold, refer

   Proxy-Authorization: Digest username="121628", 
realm="newyork.voip.ms", nonce="2d534dd6", algorithm=MD5, 
uri="sip:1MYPHONE at newyork.voip.ms", 
response="16f3301efae13df926da7550f709d28a"

   Content-Type: application/sdp

   Content-Disposition: session

   Content-Length: 280

   X-FS-Support: update_display

   Remote-Party-ID: "Extension 1001" 
<sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off



   v=0

   o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6

   s=-

   c=IN IP4 10.153.174.6

   t=0 0

   m=audio 49152 RTP/AVP 18 97 98

   a=rtpmap:18 G729/8000

   a=fmtp:18 annexb=no

   a=rtpmap:97 iLBC/8000

   a=rtpmap:98 telephone-event/8000

   a=fmtp:98 0-15

   a=ptime:20

   a=maxptime:40

   ------------------------------------------------------------------------

recv 488 bytes from udp/[74.63.41.218]:5060 at 21:15:59.484375:

   ------------------------------------------------------------------------

   SIP/2.0 100 Trying

   Via: SIP/2.0/UDP 
69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080

   From: "Extension 1001" 
<sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S

   To: <sip:1MYPHONE at newyork.voip.ms>

   Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a

   CSeq: 7788616 INVITE

   User-Agent: VoIPMS/SERAST

   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

   Supported: replaces

   Contact: <sip:1MYPHONE at 74.63.41.218>

   Content-Length: 0



   ------------------------------------------------------------------------

recv 516 bytes from udp/[74.63.41.218]:5060 at 21:15:59.562500:

   ------------------------------------------------------------------------

   SIP/2.0 503 Service Unavailable

   Via: SIP/2.0/UDP 
69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080

   From: "Extension 1001" 
<sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S

   To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9

   Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a

   CSeq: 7788616 INVITE

   User-Agent: VoIPMS/SERAST

   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

   Supported: replaces

   Contact: <sip:1MYPHONE at 74.63.41.218>

   Content-Length: 0



   ------------------------------------------------------------------------

send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.562500:

   ------------------------------------------------------------------------

   ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0

   Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN

   Max-Forwards: 69

   From: "Extension 1001" 
<sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S

   To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9

   Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a

   CSeq: 7788616 ACK

   Content-Length: 0



   ------------------------------------------------------------------------

2011-01-28 16:15:59.562500 [INFO] mod_dptools.c:2612 Originate Failed. 
Cause: NO_ANSWER

2011-01-28 16:15:59.562500 [NOTICE] sofia.c:5286 Hangup 
sofia/external/1MYPHONE [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]

2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:189 
sofia/internal/1001 at 192.168.1.100 has executed the last dialplan 
instruction, hanging up.

2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:191 
Hangup sofia/internal/1001 at 192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING]

2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1306 Session 2 
(sofia/external/1MYPHONE) Ended

2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1308 Close 
Channel sofia/external/1MYPHONE [CS_DESTROY]

send 887 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.593750:

   ------------------------------------------------------------------------

   SIP/2.0 503 Service Unavailable

   Via: SIP/2.0/TCP 
10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180

   From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj

   To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg

   Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn

   CSeq: 5244504 INVITE

   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 
18-04-05 -0600

   Accept: application/sdp

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

   Supported: timer, precondition, path, replaces

   Allow-Events: talk, hold, presence, dialog, line-seize, call-info, 
sla, include-session-description, presence.winfo, message-summary, refer

   Reason: Q.850;cause=16;text="NORMAL_CLEARING"

2011-01-28 16:15:59.593750 [NOTICE] switch_core_session.c:1306 Session 1 
(sofia/internal/1001 at 192.168.1.100) Ended

   Content-Length: 02011-01-28 16:15:59.593750 [NOTICE] 
switch_core_session.c:1308 Close Channel 
sofia/internal/1001 at 192.168.1.100 [CS_DESTROY]



   Remote-Party-ID: "MYPHONE" 
<sip:MYPHONE at 192.168.1.100>;party=calling;privacy=off;screen=no



   ------------------------------------------------------------------------

recv 650 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.953125:

   ------------------------------------------------------------------------

   ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0

   Via: SIP/2.0/TCP 
10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport

   From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj

   To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg

   Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn

   CSeq: 5244504 ACK

   Supported: sec-agree

   Max-Forwards: 70

   Proxy-Authorization: Digest 
qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"

   Content-Length: 0



   ------------------------------------------------------------------------



Thank you in advance.

 		 	   		  
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From steveayre at gmail.com  Sun Jan 30 01:15:43 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 29 Jan 2011 22:15:43 +0000
Subject: [Freeswitch-users] Remote LogIn to Freeswitch?
In-Reply-To: <10183.49543.qm@web114709.mail.gq1.yahoo.com>
References: <10183.49543.qm@web114709.mail.gq1.yahoo.com>
Message-ID: <A6C135FB-AD49-4479-9A51-78A1A5341F43@gmail.com>

> a) according to Wiki, you should set port 5080 for remote connections
> 


That's for the default config and entirely depends on what you have configured. There is absolutely no reason it *must* be 5080.

Steve on iPhone

On 28 Jan 2011, at 23:15, Kenan BEKTAS <bektas at yahoo.com> wrote:

> a) according to Wiki, you should set port 5080 for remote connections.
> b) if no sound for some reason, then, it is most likely a RTP issue. Make sure RTP port ranges is open on your router.
> 
> ---
> Kenan
> www.dbstreams.ca
> 
> 
> 
> 
> --- On Fri, 1/28/11, Joao Leme <joaocarlosleme at gmail.com> wrote:
> 
> From: Joao Leme <joaocarlosleme at gmail.com>
> Subject: Re: [Freeswitch-users] Remote LogIn to Freeswitch?
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Friday, January 28, 2011, 5:31 PM
> 
> FINALLY I got it to work, I can now remotely log in on the internal profile (port 5060) after placing the computer running Freeswitch on DMZ (changed on Firewall), although that's not the ideal solution, so I was wondering why it wasn't working before. I had the firewall set up according to the wiki on "http://wiki.freeswitch.org/wiki/Firewall" and other than that I changed the "domain" on vars.xml to my external ip of the router, it created an ALIAS so that I can use as the "Domain" on X-lite to log on. So I had the internal profile mod_sofia at 192.168.X.XX:5060 and the Alias 76.XXX.XX.XX, that was before and I was able to LogIn, make calls but NO SOUND. NOW once I forwarded all the port through DMZ, this particular router 2701HG-B Gateway (Att) required me to change my (freeSwitch running computer) static ipv4 (192.XXX.X.XX) to DHCP and then assigned me the external IP (76.XXX.XX.XX) as my a Ipv4. Now the internal profile (sofia status) is mod_sofia at 76.XXX.XX.XX:5060 and I'm not using the ALIAS.
> 
> SO...I was wondering...it wasn't working before because 1) A Firewall configuration missing? 2) A NAT/Firewall problem or 3) Freeswitch internal profile running on local machine ipv4 192..... and I logging on using the router external IP 76.XX and the Alias didn't do the job????? 
> 
> Sorry there is some nonsense question, I'm a beginner and any help is appreciated.
> Thanks,
> John
> 
> On Tue, Jan 11, 2011 at 1:37 AM, Steven Ayre <steveayre at gmail.com> wrote:
> If you're able to dial in but you're getting no sound, it's probably
> NAT stopping the audio get through.
> 
> There's quite a bit of information on NAT on the Wiki that might be of use.
> 
> http://wiki.freeswitch.org/wiki/NAT
> 
> -Steve
> 
> 
> On 11 January 2011 00:39, Joao Leme <joaocarlosleme at gmail.com> wrote:
> > Hi There,
> > What do I have to do to be able to LogIn to Freeswitch from Home (server is
> > located at office) starting from the basic/original configuration?
> > I'm using X-Lite. I've been able to LogIn replacing the internal IP by the
> > external IP from the Office but the sound is not working so I wanted to know
> > what are the configuration changes that have to be done to allow it. Do I
> > have to create a different profile? I want be able to do the same just as if
> > I was at the Office.
> > Thanks,
> > John
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> -----Inline Attachment Follows-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From avi at avimarcus.net  Sun Jan 30 01:19:41 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 30 Jan 2011 00:19:41 +0200
Subject: [Freeswitch-users] Remote LogIn to Freeswitch?
In-Reply-To: <A6C135FB-AD49-4479-9A51-78A1A5341F43@gmail.com>
References: <10183.49543.qm@web114709.mail.gq1.yahoo.com>
	<A6C135FB-AD49-4479-9A51-78A1A5341F43@gmail.com>
Message-ID: <AANLkTinQrp9+Ntp+yvVrwSBERx2JHK9kR94QN=RFVLBi@mail.gmail.com>

5080 for "external" profile means for creating outbound connections to a
gateway, or for allowing incoming un-authed calls.
This question about "remote" login was referring to a login from outside the
local network, which should be going to the internal profile.

-Avi

On Sun, Jan 30, 2011 at 12:15 AM, Steven Ayre <steveayre at gmail.com> wrote:

> a) according to Wiki, you should set port 5080 for remote connections
>
>
> That's for the default config and entirely depends on what you have
> configured. There is absolutely no reason it *must* be 5080.
>
> Steve on iPhone
>
> On 28 Jan 2011, at 23:15, Kenan BEKTAS <bektas at yahoo.com> wrote:
>
> a) according to Wiki, you should set port 5080 for remote connections.
> b) if no sound for some reason, then, it is most likely a RTP issue. Make
> sure RTP port ranges is open on your router.
>
> ---
> Kenan
> www.dbstreams.ca
>
>
>
>
> --- On *Fri, 1/28/11, Joao Leme <joaocarlosleme at gmail.com>* wrote:
>
>
> From: Joao Leme <joaocarlosleme at gmail.com>
> Subject: Re: [Freeswitch-users] Remote LogIn to Freeswitch?
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Friday, January 28, 2011, 5:31 PM
>
> FINALLY I got it to work, I can now remotely log in on the internal profile
> (port 5060) after placing the computer running Freeswitch on DMZ (changed on
> Firewall), although that's not the ideal solution, so I was wondering why it
> wasn't working before. I had the firewall set up according to the wiki on "
> http://wiki.freeswitch.org/wiki/Firewall" and other than that I changed
> the "domain" on vars.xml to my external ip of the router, it created an
> ALIAS so that I can use as the "Domain" on X-lite to log on. So I had the
> internal profile mod_sofia at 192.168.X.XX:5060 and the Alias 76.XXX.XX.XX,
> that was before and I was able to LogIn, make calls but NO SOUND. NOW once I
> forwarded all the port through DMZ, this particular router 2701HG-B Gateway
> (Att) required me to change my (freeSwitch running computer) static ipv4
> (192.XXX.X.XX) to DHCP and then assigned me the external IP (76.XXX.XX.XX)
> as my a Ipv4. Now the internal profile (sofia status) is
> mod_sofia at 76.XXX.XX.XX:5060 and I'm not using the ALIAS.
>
> SO...I was wondering...it wasn't working before because 1) A Firewall
> configuration missing? 2) A NAT/Firewall problem or 3) Freeswitch internal
> profile running on local machine ipv4 192..... and I logging on using the
> router external IP 76.XX and the Alias didn't do the job?????
>
> Sorry there is some nonsense question, I'm a beginner and any help is
> appreciated.
> Thanks,
> John
>
> On Tue, Jan 11, 2011 at 1:37 AM, Steven Ayre <steveayre at gmail.com<http://mc/compose?to=steveayre at gmail.com>
> > wrote:
>
> If you're able to dial in but you're getting no sound, it's probably
> NAT stopping the audio get through.
>
> There's quite a bit of information on NAT on the Wiki that might be of use.
>
> http://wiki.freeswitch.org/wiki/NAT
>
> -Steve
>
>
> On 11 January 2011 00:39, Joao Leme <joaocarlosleme at gmail.com<http://mc/compose?to=joaocarlosleme at gmail.com>>
> wrote:
> > Hi There,
> > What do I have to do to be able to LogIn to Freeswitch from Home (server
> is
> > located at office) starting from the basic/original configuration?
> > I'm using X-Lite. I've been able to LogIn replacing the internal IP by
> the
> > external IP from the Office but the sound is not working so I wanted to
> know
> > what are the configuration changes that have to be done to allow it. Do I
> > have to create a different profile? I want be able to do the same just as
> if
> > I was at the Office.
> > Thanks,
> > John
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> -----Inline Attachment Follows-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gustavo.espeche at upper-soft.com  Sun Jan 30 02:23:42 2011
From: gustavo.espeche at upper-soft.com (Gustavo Espeche)
Date: Sat, 29 Jan 2011 20:23:42 -0300
Subject: [Freeswitch-users] open g729
Message-ID: <1296343422.2615.4.camel@gustavo-laptop>

Hello,
        some one can compile open g729 to work with freeswitch?
        http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
        
        i appreciate a lot if some one has some experience in it.
        Best Regards.
        
        Gustavo Espeche
        www.easyipcall.com
        
        
        



From rupa at rupa.com  Sun Jan 30 02:27:58 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 29 Jan 2011 17:27:58 -0600
Subject: [Freeswitch-users] open g729
In-Reply-To: <1296343422.2615.4.camel@gustavo-laptop>
References: <1296343422.2615.4.camel@gustavo-laptop>
Message-ID: <AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com>

How do you intend to pay the license fee?

On Sat, Jan 29, 2011 at 5:23 PM, Gustavo Espeche
<gustavo.espeche at upper-soft.com> wrote:
> Hello,
> ? ? ? ?some one can compile open g729 to work with freeswitch?
> ? ? ? ?http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
>
> ? ? ? ?i appreciate a lot if some one has some experience in it.
> ? ? ? ?Best Regards.
>
> ? ? ? ?Gustavo Espeche
> ? ? ? ?www.easyipcall.com
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa


From joaocarlosleme at gmail.com  Sun Jan 30 02:39:41 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Sat, 29 Jan 2011 15:39:41 -0800
Subject: [Freeswitch-users] Hacker Attack?
Message-ID: <AANLkTimcrJmG=CVTKMy-_4kZYA6hCU-_aORbbdRkO_P5@mail.gmail.com>

I just downloaded and compiled the latest Git and a little after starting
freeswitch I'm getting non stop the following:

[WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
[WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
[WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
[WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
[WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
[WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
[WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
[WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
[WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
[WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
[WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
[WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236

it's non-stop and doesn't let me do nothing else. After the first time I
went on to vars and changed the 1234 password....restarted and same thing
happened, I also try denying the ip on acl.conf (not sure if has something
to do with it but gave it a try):

<configuration name="acl.conf" description="Network Lists">
        <network-lists>
          <list name="test2" default="allow">
            <node type="deny" host="212.224.71.236" mask="255.255.255.0"/>
          </list>
        </network-lists>
      </configuration>


Restarted the computer but nothing, he (thomas I guess) was back on my console.

Any ideas??? p.s. My computer is on DMZ (I know DMZ is not ideal but is the
only way I got to be able to connect to the internal profile from out of the
office etc).
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From curriegrad2004 at gmail.com  Sun Jan 30 03:12:38 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 29 Jan 2011 16:12:38 -0800
Subject: [Freeswitch-users] Hacker Attack?
In-Reply-To: <AANLkTimcrJmG=CVTKMy-_4kZYA6hCU-_aORbbdRkO_P5@mail.gmail.com>
References: <AANLkTimcrJmG=CVTKMy-_4kZYA6hCU-_aORbbdRkO_P5@mail.gmail.com>
Message-ID: <AANLkTi=cvZ_bMi9ZWiHfWfbhsvms3ttbieB0NvXDoxvY@mail.gmail.com>

Try using iptables and block all incoming traffic from this specific host?

On Sat, Jan 29, 2011 at 3:39 PM, Joao Leme <joaocarlosleme at gmail.com> wrote:
> I just downloaded and compiled the latest Git and a little after starting
> freeswitch I'm getting non stop the following:
> [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
> ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
> ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
> ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
> ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
> ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
> ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
> ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
> ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
> ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
> ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
> ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
> ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> it's non-stop and doesn't let me do nothing else. After the first time I
> went on to vars and changed the 1234 password....restarted and same thing
> happened, I also try denying the ip on acl.conf (not sure if has something
> to do with it but gave it a try):
>
> <configuration name="acl.conf" description="Network Lists">
>         <network-lists>
>           <list name="test2" default="allow">
>             <node type="deny" host="212.224.71.236" mask="255.255.255.0"/>
>           </list>
>         </network-lists>
>       </configuration>
>
> Restarted the computer but nothing, he (thomas I guess) was back on my
> console.
>
> Any ideas??? p.s. My computer is on DMZ (I know DMZ is not ideal but is the
> only way I got to be able to connect to the internal profile from out of the
> office etc).
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From curriegrad2004 at gmail.com  Sun Jan 30 03:17:09 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 29 Jan 2011 16:17:09 -0800
Subject: [Freeswitch-users] Virtualisation and freeswitch
In-Reply-To: <AANLkTikiQUXWnrWZxoLHWOb3DL6MvXsYBmTdCUkcgjBK@mail.gmail.com>
References: <AANLkTimZTwZfUYMjC1ddAQMWHHWSLH=BLajv0cgSux4B@mail.gmail.com>
	<AANLkTikQbXLdCb4nNFEbfte-EAP8ytJyfURMALC6kwtK@mail.gmail.com>
	<AANLkTinB17uNUKtEQAtuTgCGPf_61jnErGmv7+vXV_n+@mail.gmail.com>
	<AANLkTimtnMqFyrXFEYirmhLgqPJxaWsYWowYyiLi_k_=@mail.gmail.com>
	<AANLkTikhDhRL_hXcosBQzq9_vD75_=G0Ju8+3nJ9jyhn@mail.gmail.com>
	<AANLkTikiQUXWnrWZxoLHWOb3DL6MvXsYBmTdCUkcgjBK@mail.gmail.com>
Message-ID: <AANLkTikUkK2x8HajWuGiTLZ+hvO7-S-G14U8bme708jJ@mail.gmail.com>

Thanks for that explanation, Chris. I didn't realize that Xen kernels
are compiled with lower timer resolution values. Well, that being
said, I'll probably attempt to run FreeSwitch (just for kicks, not in
a production envrionment!) under Hyper-V and see what happens.

On Sat, Jan 29, 2011 at 12:06 PM, Chris Burns <chris at cloudtel.com> wrote:
> Many Xen kernels are compiled with a lower resolution kernel timer. Most
> applications will not notice this, but a VoIP application doing async RTP
> probably will. It can make audio quality poor, especially if something other
> than FreeSWITCH is trying to get CPU time while you are handling media. That
> is the main reason that I found. I have had OK results with Xen with the
> proper configuration.
>
> On Sat, Jan 29, 2011 at 11:57 AM, curriegrad2004 <curriegrad2004 at gmail.com>
> wrote:
>>
>> I noticed the wiki telling people not to use Xen, why is that?
>>
>> On Sat, Jan 29, 2011 at 1:30 AM, dome at tel.co.th <dome at tel.co.th> wrote:
>> > (from wiki http://wiki.freeswitch.org/wiki/Virtualization)
>> > if your FS running with high traffic please use
>> > /etc/vz/conf/ve-unlimited.conf-sample for default template
>> >
>> > Dome C.
>> >
>> > On Sat, Jan 29, 2011 at 7:23 AM, Seven Du <dujinfang at gmail.com> wrote:
>> >> We had used OpenVZ for a while, it was good given we never tested on
>> >> load.
>> >>
>> >> Another option besides OpenVZ is lxc , which is similar to OpenVZ. I
>> >> runs some lxc VMs for other projects but never tried with FS.
>> >>
>> >> On Sat, Jan 29, 2011 at 6:03 AM, Brent Paddon
>> >> <brent at overthewire.com.au> wrote:
>> >>> Why would it possibly matter if it's more of a 'jail' than
>> >>> 'virtualisation'
>> >>> ? ? Isn't the outcome the important thing ?
>> >>> In regards OpenVZ there is an ISO installer called Proxmox which helps
>> >>> you
>> >>> administer the whole system and gives you a pretty decent web
>> >>> interface for
>> >>> management. ?We use it extensively for voice applications.
>> >>> http://www.proxmox.com/products/proxmox-ve
>> >>> Brent
>> >>>
>> >>> On Fri, Jan 28, 2011 at 9:52 PM, Grygoriy Dobrovolskyy
>> >>> <megahohol at gmail.com>
>> >>> wrote:
>> >>>>
>> >>>> Hello list, le me introduce myself first, I am Grigoriy Dobrovolskyy
>> >>>> currently working as voip administrator and call centeres integrator.
>> >>>> I
>> >>>> worked a lot with asterisk, i think Fs is a great software but not
>> >>>> easy to
>> >>>> learn.
>> >>>> I have googled my question and did not found a clear answer.
>> >>>> With what technology i can virtualise freeswitch without timer
>> >>>> problems ?
>> >>>> I know that i can use OpenVZ but it's more 'jail' then
>> >>>> virtualisation. I
>> >>>> heard that Amazon EC with High CPU instance works (confirm?)
>> >>>> Can someone give me info of a succesfull installations with
>> >>>> Conferences,
>> >>>> Moh ?/ mixing ? And a technology used ?
>> >>>> Thank you.
>> >>>> _______________________________________________
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>
>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>>
>> >>>
>> >>>
>> >>>
>> >>> --
>> >>> --
>> >>> Brent Paddon
>> >>>
>> >>> Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
>> >>> www.overthewire.com.au
>> >>> Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>
>> >>
>> >>
>> >> --
>> >> About: http://about.me/dujinfang
>> >> Blog: http://www.dujinfang.com
>> >> Proj:? http://www.freeswitch.org.cn
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From joaocarlosleme at gmail.com  Sun Jan 30 03:20:09 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Sat, 29 Jan 2011 16:20:09 -0800
Subject: [Freeswitch-users] Hacker Attack?
In-Reply-To: <AANLkTi=cvZ_bMi9ZWiHfWfbhsvms3ttbieB0NvXDoxvY@mail.gmail.com>
References: <AANLkTimcrJmG=CVTKMy-_4kZYA6hCU-_aORbbdRkO_P5@mail.gmail.com>
	<AANLkTi=cvZ_bMi9ZWiHfWfbhsvms3ttbieB0NvXDoxvY@mail.gmail.com>
Message-ID: <AANLkTinm4MzynbQhiRqUZm_+KXYx0_4sUYCFN0c85pez@mail.gmail.com>

How do I do that?
Thanks!

On Sat, Jan 29, 2011 at 4:12 PM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:

> Try using iptables and block all incoming traffic from this specific host?
>
> On Sat, Jan 29, 2011 at 3:39 PM, Joao Leme <joaocarlosleme at gmail.com>
> wrote:
> > I just downloaded and compiled the latest Git and a little after starting
> > freeswitch I'm getting non stop the following:
> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> > it's non-stop and doesn't let me do nothing else. After the first time I
> > went on to vars and changed the 1234 password....restarted and same thing
> > happened, I also try denying the ip on acl.conf (not sure if has
> something
> > to do with it but gave it a try):
> >
> > <configuration name="acl.conf" description="Network Lists">
> >         <network-lists>
> >           <list name="test2" default="allow">
> >             <node type="deny" host="212.224.71.236"
> mask="255.255.255.0"/>
> >           </list>
> >         </network-lists>
> >       </configuration>
> >
> > Restarted the computer but nothing, he (thomas I guess) was back on my
> > console.
> >
> > Any ideas??? p.s. My computer is on DMZ (I know DMZ is not ideal but is
> the
> > only way I got to be able to connect to the internal profile from out of
> the
> > office etc).
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From curriegrad2004 at gmail.com  Sun Jan 30 03:26:00 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 29 Jan 2011 16:26:00 -0800
Subject: [Freeswitch-users] Hacker Attack?
In-Reply-To: <AANLkTinm4MzynbQhiRqUZm_+KXYx0_4sUYCFN0c85pez@mail.gmail.com>
References: <AANLkTimcrJmG=CVTKMy-_4kZYA6hCU-_aORbbdRkO_P5@mail.gmail.com>
	<AANLkTi=cvZ_bMi9ZWiHfWfbhsvms3ttbieB0NvXDoxvY@mail.gmail.com>
	<AANLkTinm4MzynbQhiRqUZm_+KXYx0_4sUYCFN0c85pez@mail.gmail.com>
Message-ID: <AANLkTi=kttKAP_T61EX0tS7w+TSJQ8Ud-7zkLrKuFrAG@mail.gmail.com>

iptables -I INPUT -s [hackerip] -j DROP

A better solution is searching the wiki for fail2ban with FreeSwitch.

On Sat, Jan 29, 2011 at 4:20 PM, Joao Leme <joaocarlosleme at gmail.com> wrote:
> How do I do that?
> Thanks!
> On Sat, Jan 29, 2011 at 4:12 PM, curriegrad2004 <curriegrad2004 at gmail.com>
> wrote:
>>
>> Try using iptables and block all incoming traffic from this specific host?
>>
>> On Sat, Jan 29, 2011 at 3:39 PM, Joao Leme <joaocarlosleme at gmail.com>
>> wrote:
>> > I just downloaded and compiled the latest Git and a little after
>> > starting
>> > freeswitch I'm getting non stop the following:
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> > profile
>> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> > profile
>> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> > profile
>> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> > profile
>> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> > profile
>> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> > profile
>> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> > profile
>> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> > profile
>> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> > profile
>> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> > profile
>> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> > profile
>> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> > profile
>> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > it's non-stop and doesn't let me do nothing else. After the first time I
>> > went on to vars and changed the 1234 password....restarted and same
>> > thing
>> > happened, I also try denying the ip on acl.conf (not sure if has
>> > something
>> > to do with it but gave it a try):
>> >
>> > <configuration name="acl.conf" description="Network Lists">
>> > ? ? ? ? <network-lists>
>> > ? ? ? ? ? <list name="test2" default="allow">
>> > ? ? ? ? ? ? <node type="deny" host="212.224.71.236"
>> > mask="255.255.255.0"/>
>> > ? ? ? ? ? </list>
>> > ? ? ? ? </network-lists>
>> > ? ? ? </configuration>
>> >
>> > Restarted the computer but nothing, he (thomas I guess) was back on my
>> > console.
>> >
>> > Any ideas??? p.s. My computer is on DMZ (I know DMZ is not ideal but is
>> > the
>> > only way I got to be able to connect to the internal profile from out of
>> > the
>> > office etc).
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From joaocarlosleme at gmail.com  Sun Jan 30 03:42:43 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Sat, 29 Jan 2011 16:42:43 -0800
Subject: [Freeswitch-users] Hacker Attack?
In-Reply-To: <AANLkTi=kttKAP_T61EX0tS7w+TSJQ8Ud-7zkLrKuFrAG@mail.gmail.com>
References: <AANLkTimcrJmG=CVTKMy-_4kZYA6hCU-_aORbbdRkO_P5@mail.gmail.com>
	<AANLkTi=cvZ_bMi9ZWiHfWfbhsvms3ttbieB0NvXDoxvY@mail.gmail.com>
	<AANLkTinm4MzynbQhiRqUZm_+KXYx0_4sUYCFN0c85pez@mail.gmail.com>
	<AANLkTi=kttKAP_T61EX0tS7w+TSJQ8Ud-7zkLrKuFrAG@mail.gmail.com>
Message-ID: <AANLkTikvkD_TiTvaava=PhjVgnrUYOXXPNFaf8yTuo5z@mail.gmail.com>

Thanks I will try.
What do you thing he/it was trying to do (any idea)?

On Sat, Jan 29, 2011 at 4:26 PM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:

> iptables -I INPUT -s [hackerip] -j DROP
>
> A better solution is searching the wiki for fail2ban with FreeSwitch.
>
> On Sat, Jan 29, 2011 at 4:20 PM, Joao Leme <joaocarlosleme at gmail.com>
> wrote:
> > How do I do that?
> > Thanks!
> > On Sat, Jan 29, 2011 at 4:12 PM, curriegrad2004 <
> curriegrad2004 at gmail.com>
> > wrote:
> >>
> >> Try using iptables and block all incoming traffic from this specific
> host?
> >>
> >> On Sat, Jan 29, 2011 at 3:39 PM, Joao Leme <joaocarlosleme at gmail.com>
> >> wrote:
> >> > I just downloaded and compiled the latest Git and a little after
> >> > starting
> >> > freeswitch I'm getting non stop the following:
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > it's non-stop and doesn't let me do nothing else. After the first time
> I
> >> > went on to vars and changed the 1234 password....restarted and same
> >> > thing
> >> > happened, I also try denying the ip on acl.conf (not sure if has
> >> > something
> >> > to do with it but gave it a try):
> >> >
> >> > <configuration name="acl.conf" description="Network Lists">
> >> >         <network-lists>
> >> >           <list name="test2" default="allow">
> >> >             <node type="deny" host="212.224.71.236"
> >> > mask="255.255.255.0"/>
> >> >           </list>
> >> >         </network-lists>
> >> >       </configuration>
> >> >
> >> > Restarted the computer but nothing, he (thomas I guess) was back on my
> >> > console.
> >> >
> >> > Any ideas??? p.s. My computer is on DMZ (I know DMZ is not ideal but
> is
> >> > the
> >> > only way I got to be able to connect to the internal profile from out
> of
> >> > the
> >> > office etc).
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Sun Jan 30 03:45:26 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 29 Jan 2011 19:45:26 -0500
Subject: [Freeswitch-users] Hacker Attack?
References: <AANLkTimcrJmG=CVTKMy-_4kZYA6hCU-_aORbbdRkO_P5@mail.gmail.com><AANLkTi=cvZ_bMi9ZWiHfWfbhsvms3ttbieB0NvXDoxvY@mail.gmail.com>
	<AANLkTinm4MzynbQhiRqUZm_+KXYx0_4sUYCFN0c85pez@mail.gmail.com>
Message-ID: <545E726A843940729F16C602DB6B3605@e1705>

fail2ban on wiki
  ----- Original Message ----- 
  From: Joao Leme 
  To: FreeSWITCH Users Help 
  Sent: Saturday, January 29, 2011 7:20 PM
  Subject: Re: [Freeswitch-users] Hacker Attack?


  How do I do that?
  Thanks!


  On Sat, Jan 29, 2011 at 4:12 PM, curriegrad2004 <curriegrad2004 at gmail.com> wrote:

    Try using iptables and block all incoming traffic from this specific host?


    On Sat, Jan 29, 2011 at 3:39 PM, Joao Leme <joaocarlosleme at gmail.com> wrote:
    > I just downloaded and compiled the latest Git and a little after starting
    > freeswitch I'm getting non stop the following:
    > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
    > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
    > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
    > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
    > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
    > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
    > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
    > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
    > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
    > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
    > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
    > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
    > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
    > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
    > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
    > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
    > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
    > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
    > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
    > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
    > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
    > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
    > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile
    > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
    > it's non-stop and doesn't let me do nothing else. After the first time I
    > went on to vars and changed the 1234 password....restarted and same thing
    > happened, I also try denying the ip on acl.conf (not sure if has something
    > to do with it but gave it a try):
    >
    > <configuration name="acl.conf" description="Network Lists">
    >         <network-lists>
    >           <list name="test2" default="allow">
    >             <node type="deny" host="212.224.71.236" mask="255.255.255.0"/>
    >           </list>
    >         </network-lists>
    >       </configuration>
    >
    > Restarted the computer but nothing, he (thomas I guess) was back on my
    > console.
    >
    > Any ideas??? p.s. My computer is on DMZ (I know DMZ is not ideal but is the
    > only way I got to be able to connect to the internal profile from out of the
    > office etc).

    > _______________________________________________
    > FreeSWITCH-users mailing list
    > FreeSWITCH-users at lists.freeswitch.org
    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > http://www.freeswitch.org
    >
    >

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From frank at telonium.com  Sun Jan 30 04:48:35 2011
From: frank at telonium.com (Frank Park)
Date: Sat, 29 Jan 2011 20:48:35 -0500
Subject: [Freeswitch-users] Hacker Attack?
In-Reply-To: <545E726A843940729F16C602DB6B3605@e1705>
References: <AANLkTimcrJmG=CVTKMy-_4kZYA6hCU-_aORbbdRkO_P5@mail.gmail.com>
	<AANLkTi=cvZ_bMi9ZWiHfWfbhsvms3ttbieB0NvXDoxvY@mail.gmail.com>
	<AANLkTinm4MzynbQhiRqUZm_+KXYx0_4sUYCFN0c85pez@mail.gmail.com>
	<545E726A843940729F16C602DB6B3605@e1705>
Message-ID: <AANLkTi=E0nxp=SK+mCNYKcJJ+XLjse4hQcHD_7niYNZ0@mail.gmail.com>

It's pretty normal to see a crawler every so often that tries to brute-force
the username/password combo on SIP servers. Most of them are kiddie scripts
online and shouldn't last long. If you want to make sure they don't even
talk to the FS, iptable is a good way, but you can only do so much by
banning 1 ip address. fail2ban is a much better solution for the future
Denial of Service (DoS) attacks.

Regardless of any preventative you go with, make sure you don't have any sip
accounts with easy to guess passwords. Depending on their script, it
wouldn't take too long to brute-force a dictionary-based passwords.

Frank



On Sat, Jan 29, 2011 at 7:45 PM, Madovsky <infos at madovsky.org> wrote:

>  fail2ban on wiki
>
> ----- Original Message -----
> *From:* Joao Leme <joaocarlosleme at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Saturday, January 29, 2011 7:20 PM
> *Subject:* Re: [Freeswitch-users] Hacker Attack?
>
> How do I do that?
> Thanks!
>
> On Sat, Jan 29, 2011 at 4:12 PM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:
>
>> Try using iptables and block all incoming traffic from this specific host?
>>
>> On Sat, Jan 29, 2011 at 3:39 PM, Joao Leme <joaocarlosleme at gmail.com>
>> wrote:
>> > I just downloaded and compiled the latest Git and a little after
>> starting
>> > freeswitch I'm getting non stop the following:
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> profile
>> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> profile
>> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> profile
>> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> profile
>> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> profile
>> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> profile
>> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> profile
>> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> profile
>> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> profile
>> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> profile
>> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> profile
>> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> profile
>> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > it's non-stop and doesn't let me do nothing else. After the first time I
>> > went on to vars and changed the 1234 password....restarted and same
>> thing
>> > happened, I also try denying the ip on acl.conf (not sure if has
>> something
>> > to do with it but gave it a try):
>> >
>> > <configuration name="acl.conf" description="Network Lists">
>> >         <network-lists>
>> >           <list name="test2" default="allow">
>> >             <node type="deny" host="212.224.71.236"
>> mask="255.255.255.0"/>
>> >           </list>
>> >         </network-lists>
>> >       </configuration>
>> >
>> > Restarted the computer but nothing, he (thomas I guess) was back on my
>> > console.
>> >
>> > Any ideas??? p.s. My computer is on DMZ (I know DMZ is not ideal but is
>> the
>> > only way I got to be able to connect to the internal profile from out of
>> the
>> > office etc).
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 

----=======================----
Frank Park
Telonium Communications, LLC
frank at telonium.com
http://www.telonium.com
Follow Us on Twitter: @GetTelonium
404-566-8888 x1001 Office
404-939-4242 Cell
----=======================----
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From peter.olsson at visionutveckling.se  Sun Jan 30 05:54:43 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 30 Jan 2011 03:54:43 +0100
Subject: [Freeswitch-users] Virtualisation and freeswitch
In-Reply-To: <AANLkTi=fRLdjSpwpd7ny1Tk05jnctOrGwTsmpxUPQrjO@mail.gmail.com>
References: <AANLkTimZTwZfUYMjC1ddAQMWHHWSLH=BLajv0cgSux4B@mail.gmail.com>
	<AANLkTikQbXLdCb4nNFEbfte-EAP8ytJyfURMALC6kwtK@mail.gmail.com>
	<AANLkTimwxcBnVigya0b_68sfG=CPDEb20ERJ26mLZkar@mail.gmail.com>,
	<AANLkTi=fRLdjSpwpd7ny1Tk05jnctOrGwTsmpxUPQrjO@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57EC81C583@cooper>

Some people claim they have solved it, but it's still all dependant on the hw config, the VM config etc. There is not one generic solution that fixes timer problems in virtualized environments. And there is no "fix" available that makes timing in a VM as accurate as real hw systems.

I would say that FS timing works at least as good (or probably better) than Asterisk. But in virtualized systems the timing might be bad enough to produce bad audio. I would never use a virtual host for production, but I know there are people who use it that way, and are happy with the results.

The basic tricks I know of in ESXi is to only give the virtual host one CPU (SMP systems makes the clock less accurate in this case), and to make sure that this core is not shared with other hosts.

/Peter

________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Grygoriy Dobrovolskyy [megahohol at gmail.com]
Skickat: den 29 januari 2011 01:39
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Virtualisation and freeswitch

Another interesting point i have asked in near topic in opensips mailing list about timer and Sa?l Ibarra Corretg? told me this:


About Asterisk, the timing issues were resolved long ago and works very well with DAHDI if HPET is enabled. Latest releases support other timing sources: TimerFD and pthreads based timing. AppKonference did the timing and on its own, but it also brought another advantage: the possibility of not to use Zaptel/DAHDI (MeetMe uses DAHDI mixing engine, so it's limited to 8KHz...).

So this virtualization timer problem is not so complicated to resolve if others done it ?


2011/1/29 Grygoriy Dobrovolskyy <megahohol at gmail.com<mailto:megahohol at gmail.com>>
I've heard about proxmox, and even tested it a while ago. It's a good solution.
The problem of Openvz is the maintenance mess. In clear you always need to keep updated kernels, possible have compatibility issues and other stuff when maintaining your virtual infrastructure up to date / migrating servers back and forth. With a complete hypervisor this problems do not exist.



2011/1/28 Brent Paddon <brent at overthewire.com.au<mailto:brent at overthewire.com.au>>

Why would it possibly matter if it's more of a 'jail' than 'virtualisation' ?   Isn't the outcome the important thing ?

In regards OpenVZ there is an ISO installer called Proxmox which helps you administer the whole system and gives you a pretty decent web interface for management.  We use it extensively for voice applications.

http://www.proxmox.com/products/proxmox-ve

Brent

On Fri, Jan 28, 2011 at 9:52 PM, Grygoriy Dobrovolskyy <megahohol at gmail.com<mailto:megahohol at gmail.com>> wrote:
Hello list, le me introduce myself first, I am Grigoriy Dobrovolskyy currently working as voip administrator and call centeres integrator. I worked a lot with asterisk, i think Fs is a great software but not easy to learn.

I have googled my question and did not found a clear answer.

With what technology i can virtualise freeswitch without timer problems ?

I know that i can use OpenVZ but it's more 'jail' then virtualisation. I heard that Amazon EC with High CPU instance works (confirm?)

Can someone give me info of a succesfull installations with Conferences, Moh  / mixing ? And a technology used ?

Thank you.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au<mailto:brent.paddon at overthewire.com.au> | www.overthewire.com.au<http://www.overthewire.com.au>
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



!DSPAM:4d448f0d32766176048984!


From gabe at gundy.org  Sun Jan 30 06:24:00 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 29 Jan 2011 20:24:00 -0700
Subject: [Freeswitch-users] Hacker Attack?
In-Reply-To: <AANLkTi=E0nxp=SK+mCNYKcJJ+XLjse4hQcHD_7niYNZ0@mail.gmail.com>
References: <AANLkTimcrJmG=CVTKMy-_4kZYA6hCU-_aORbbdRkO_P5@mail.gmail.com>
	<AANLkTi=cvZ_bMi9ZWiHfWfbhsvms3ttbieB0NvXDoxvY@mail.gmail.com>
	<AANLkTinm4MzynbQhiRqUZm_+KXYx0_4sUYCFN0c85pez@mail.gmail.com>
	<545E726A843940729F16C602DB6B3605@e1705>
	<AANLkTi=E0nxp=SK+mCNYKcJJ+XLjse4hQcHD_7niYNZ0@mail.gmail.com>
Message-ID: <AANLkTi=S75GKc8N0r4mDSJa48ezCq6L_Wdv70BnwtoMY@mail.gmail.com>

On Sat, Jan 29, 2011 at 6:48 PM, Frank Park <frank at telonium.com> wrote:
> It's pretty normal to see a crawler every so often that tries to brute-force
> the username/password combo on SIP servers. Most of them are kiddie scripts
> online and shouldn't last long.

Has anyone seen enough of these to know if they're sophisticated
enough to use SRV records to find their target, or to they just plot
along at 5060 for some range of IP addresses?

Gabe


From infos at madovsky.org  Sun Jan 30 07:28:36 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 29 Jan 2011 23:28:36 -0500
Subject: [Freeswitch-users] clean UDP sockets
Message-ID: <DF2C3B08C24544419D16B3C546EFD1A0@e1705>

is there a way to clean inactive UDP sockets after
a timeout ?
when I do on my linux console
[bash] # netstat -tuvnap
I can see a lot of UDP sockets not removed after calls.
tried also to modify the kernel with nf_conntrack_udp_timeout without success.

Thanks
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From curriegrad2004 at gmail.com  Sun Jan 30 08:12:47 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 29 Jan 2011 21:12:47 -0800
Subject: [Freeswitch-users] Hacker Attack?
In-Reply-To: <AANLkTi=S75GKc8N0r4mDSJa48ezCq6L_Wdv70BnwtoMY@mail.gmail.com>
References: <AANLkTimcrJmG=CVTKMy-_4kZYA6hCU-_aORbbdRkO_P5@mail.gmail.com>
	<AANLkTi=cvZ_bMi9ZWiHfWfbhsvms3ttbieB0NvXDoxvY@mail.gmail.com>
	<AANLkTinm4MzynbQhiRqUZm_+KXYx0_4sUYCFN0c85pez@mail.gmail.com>
	<545E726A843940729F16C602DB6B3605@e1705>
	<AANLkTi=E0nxp=SK+mCNYKcJJ+XLjse4hQcHD_7niYNZ0@mail.gmail.com>
	<AANLkTi=S75GKc8N0r4mDSJa48ezCq6L_Wdv70BnwtoMY@mail.gmail.com>
Message-ID: <AANLkTi=sdmMsp+Zt7nDh0TtcgR6UznuiWpSwTk5Ys-RF@mail.gmail.com>

I'd suspect they just do a port scan and see who has port 5060 up and
running. Wouldn't be too surprised if they went through the SRV route

On Sat, Jan 29, 2011 at 7:24 PM, Gabriel Gunderson <gabe at gundy.org> wrote:
> On Sat, Jan 29, 2011 at 6:48 PM, Frank Park <frank at telonium.com> wrote:
>> It's pretty normal to see a crawler every so often that tries to brute-force
>> the username/password combo on SIP servers. Most of them are kiddie scripts
>> online and shouldn't last long.
>
> Has anyone seen enough of these to know if they're sophisticated
> enough to use SRV records to find their target, or to they just plot
> along at 5060 for some range of IP addresses?
>
> Gabe
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From joaocarlosleme at gmail.com  Sun Jan 30 09:47:30 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Sat, 29 Jan 2011 22:47:30 -0800
Subject: [Freeswitch-users] Hacker Attack?
In-Reply-To: <AANLkTi=kttKAP_T61EX0tS7w+TSJQ8Ud-7zkLrKuFrAG@mail.gmail.com>
References: <AANLkTimcrJmG=CVTKMy-_4kZYA6hCU-_aORbbdRkO_P5@mail.gmail.com>
	<AANLkTi=cvZ_bMi9ZWiHfWfbhsvms3ttbieB0NvXDoxvY@mail.gmail.com>
	<AANLkTinm4MzynbQhiRqUZm_+KXYx0_4sUYCFN0c85pez@mail.gmail.com>
	<AANLkTi=kttKAP_T61EX0tS7w+TSJQ8Ud-7zkLrKuFrAG@mail.gmail.com>
Message-ID: <AANLkTi=OFe8oyy6tL351OzRq9MG8L-0Uj08PD5U74bzZ@mail.gmail.com>

I tried "iptables -I INPUT -s [212.224.71.236] -j DROP" and got " Unknown
command: iptables...". Do I must install fail2ban to issue iptables command?
I'm on windows 7.
Thanks

On Sat, Jan 29, 2011 at 4:26 PM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:

> iptables -I INPUT -s [hackerip] -j DROP
>
> A better solution is searching the wiki for fail2ban with FreeSwitch.
>
> On Sat, Jan 29, 2011 at 4:20 PM, Joao Leme <joaocarlosleme at gmail.com>
> wrote:
> > How do I do that?
> > Thanks!
> > On Sat, Jan 29, 2011 at 4:12 PM, curriegrad2004 <
> curriegrad2004 at gmail.com>
> > wrote:
> >>
> >> Try using iptables and block all incoming traffic from this specific
> host?
> >>
> >> On Sat, Jan 29, 2011 at 3:39 PM, Joao Leme <joaocarlosleme at gmail.com>
> >> wrote:
> >> > I just downloaded and compiled the latest Git and a little after
> >> > starting
> >> > freeswitch I'm getting non stop the following:
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > it's non-stop and doesn't let me do nothing else. After the first time
> I
> >> > went on to vars and changed the 1234 password....restarted and same
> >> > thing
> >> > happened, I also try denying the ip on acl.conf (not sure if has
> >> > something
> >> > to do with it but gave it a try):
> >> >
> >> > <configuration name="acl.conf" description="Network Lists">
> >> >         <network-lists>
> >> >           <list name="test2" default="allow">
> >> >             <node type="deny" host="212.224.71.236"
> >> > mask="255.255.255.0"/>
> >> >           </list>
> >> >         </network-lists>
> >> >       </configuration>
> >> >
> >> > Restarted the computer but nothing, he (thomas I guess) was back on my
> >> > console.
> >> >
> >> > Any ideas??? p.s. My computer is on DMZ (I know DMZ is not ideal but
> is
> >> > the
> >> > only way I got to be able to connect to the internal profile from out
> of
> >> > the
> >> > office etc).
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From curriegrad2004 at gmail.com  Sun Jan 30 09:53:23 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 29 Jan 2011 22:53:23 -0800
Subject: [Freeswitch-users] Hacker Attack?
In-Reply-To: <AANLkTi=OFe8oyy6tL351OzRq9MG8L-0Uj08PD5U74bzZ@mail.gmail.com>
References: <AANLkTimcrJmG=CVTKMy-_4kZYA6hCU-_aORbbdRkO_P5@mail.gmail.com>
	<AANLkTi=cvZ_bMi9ZWiHfWfbhsvms3ttbieB0NvXDoxvY@mail.gmail.com>
	<AANLkTinm4MzynbQhiRqUZm_+KXYx0_4sUYCFN0c85pez@mail.gmail.com>
	<AANLkTi=kttKAP_T61EX0tS7w+TSJQ8Ud-7zkLrKuFrAG@mail.gmail.com>
	<AANLkTi=OFe8oyy6tL351OzRq9MG8L-0Uj08PD5U74bzZ@mail.gmail.com>
Message-ID: <AANLkTikOj7CBvos2MYCw-qFbR207GhmOox4fxV=ikcJ3@mail.gmail.com>

uhm... remove the braces and do iptables -I INPUT -s 212.224.71.236 -j
DROP instead. Sorry for not being clear what the braces meant...

On Sat, Jan 29, 2011 at 10:47 PM, Joao Leme <joaocarlosleme at gmail.com> wrote:
> I tried "iptables -I INPUT -s [212.224.71.236] -j DROP" and got " Unknown
> command: iptables...". Do I must install fail2ban to issue iptables command?
> I'm on windows 7.
> Thanks
>
> On Sat, Jan 29, 2011 at 4:26 PM, curriegrad2004 <curriegrad2004 at gmail.com>
> wrote:
>>
>> iptables -I INPUT -s [hackerip] -j DROP
>>
>> A better solution is searching the wiki for fail2ban with FreeSwitch.
>>
>> On Sat, Jan 29, 2011 at 4:20 PM, Joao Leme <joaocarlosleme at gmail.com>
>> wrote:
>> > How do I do that?
>> > Thanks!
>> > On Sat, Jan 29, 2011 at 4:12 PM, curriegrad2004
>> > <curriegrad2004 at gmail.com>
>> > wrote:
>> >>
>> >> Try using iptables and block all incoming traffic from this specific
>> >> host?
>> >>
>> >> On Sat, Jan 29, 2011 at 3:39 PM, Joao Leme <joaocarlosleme at gmail.com>
>> >> wrote:
>> >> > I just downloaded and compiled the latest Git and a little after
>> >> > starting
>> >> > freeswitch I'm getting non stop the following:
>> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> >> > profile
>> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> >> > profile
>> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> >> > profile
>> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> >> > profile
>> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> >> > profile
>> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> >> > profile
>> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> >> > profile
>> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> >> > profile
>> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> >> > profile
>> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> >> > profile
>> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> >> > profile
>> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> >> > profile
>> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> >> > it's non-stop and doesn't let me do nothing else. After the first
>> >> > time I
>> >> > went on to vars and changed the 1234 password....restarted and same
>> >> > thing
>> >> > happened, I also try denying the ip on acl.conf (not sure if has
>> >> > something
>> >> > to do with it but gave it a try):
>> >> >
>> >> > <configuration name="acl.conf" description="Network Lists">
>> >> > ? ? ? ? <network-lists>
>> >> > ? ? ? ? ? <list name="test2" default="allow">
>> >> > ? ? ? ? ? ? <node type="deny" host="212.224.71.236"
>> >> > mask="255.255.255.0"/>
>> >> > ? ? ? ? ? </list>
>> >> > ? ? ? ? </network-lists>
>> >> > ? ? ? </configuration>
>> >> >
>> >> > Restarted the computer but nothing, he (thomas I guess) was back on
>> >> > my
>> >> > console.
>> >> >
>> >> > Any ideas??? p.s. My computer is on DMZ (I know DMZ is not ideal but
>> >> > is
>> >> > the
>> >> > only way I got to be able to connect to the internal profile from out
>> >> > of
>> >> > the
>> >> > office etc).
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From infos at madovsky.org  Sun Jan 30 10:00:54 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 30 Jan 2011 02:00:54 -0500
Subject: [Freeswitch-users] sip_contact_uri
Message-ID: <5EF8A13E1BFB42638A55A7D9822DC912@e1705>

if a registered user make an outgouing sip call
the sip_contact_uri shows the IP address and not the domain.
how to change it to registered domain as this the callee can
see the domain and not the IP

Thanks
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From peter.olsson at visionutveckling.se  Sun Jan 30 13:10:30 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 30 Jan 2011 11:10:30 +0100
Subject: [Freeswitch-users] Hacker Attack?
Message-ID: <09EC9978-ED26-445F-9B0C-34D6D55DDA41@visionutveckling.se>

iptables is a Linux command.

/Peter


----- Reply message -----
Fr?n: "Joao Leme" <joaocarlosleme at gmail.com>
Datum: s?n, jan 30, 2011 13:56
Rubrik: [SPAM] - Re: [Freeswitch-users] Hacker Attack?
Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

I tried "iptables -I INPUT -s [212.224.71.236] -j DROP" and got " Unknown command: iptables...". Do I must install fail2ban to issue iptables command? I'm on windows 7.
Thanks

On Sat, Jan 29, 2011 at 4:26 PM, curriegrad2004 <curriegrad2004 at gmail.com<mailto:curriegrad2004 at gmail.com>> wrote:
iptables -I INPUT -s [hackerip] -j DROP

A better solution is searching the wiki for fail2ban with FreeSwitch.

On Sat, Jan 29, 2011 at 4:20 PM, Joao Leme <joaocarlosleme at gmail.com<mailto:joaocarlosleme at gmail.com>> wrote:
> How do I do that?
> Thanks!
> On Sat, Jan 29, 2011 at 4:12 PM, curriegrad2004 <curriegrad2004 at gmail.com<mailto:curriegrad2004 at gmail.com>>
> wrote:
>>
>> Try using iptables and block all incoming traffic from this specific host?
>>
>> On Sat, Jan 29, 2011 at 3:39 PM, Joao Leme <joaocarlosleme at gmail.com<mailto:joaocarlosleme at gmail.com>>
>> wrote:
>> > I just downloaded and compiled the latest Git and a little after
>> > starting
>> > freeswitch I'm getting non stop the following:
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> > profile
>> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> > profile
>> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> > profile
>> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> > profile
>> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> > profile
>> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> > profile
>> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> > profile
>> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> > profile
>> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> > profile
>> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> > profile
>> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> > profile
>> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
>> > profile
>> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
>> > it's non-stop and doesn't let me do nothing else. After the first time I
>> > went on to vars and changed the 1234 password....restarted and same
>> > thing
>> > happened, I also try denying the ip on acl.conf (not sure if has
>> > something
>> > to do with it but gave it a try):
>> >
>> > <configuration name="acl.conf" description="Network Lists">
>> >         <network-lists>
>> >           <list name="test2" default="allow">
>> >             <node type="deny" host="212.224.71.236"
>> > mask="255.255.255.0"/>
>> >           </list>
>> >         </network-lists>
>> >       </configuration>
>> >
>> > Restarted the computer but nothing, he (thomas I guess) was back on my
>> > console.
>> >
>> > Any ideas??? p.s. My computer is on DMZ (I know DMZ is not ideal but is
>> > the
>> > only way I got to be able to connect to the internal profile from out of
>> > the
>> > office etc).
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4d450b3232767678720833!


From steveayre at gmail.com  Sun Jan 30 14:06:56 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 30 Jan 2011 11:06:56 +0000
Subject: [Freeswitch-users] clean UDP sockets
In-Reply-To: <DF2C3B08C24544419D16B3C546EFD1A0@e1705>
References: <DF2C3B08C24544419D16B3C546EFD1A0@e1705>
Message-ID: <29EC78A7-59C7-4F08-898C-52F0F6CD5C03@gmail.com>

The kernel will clean them up in it's own good time. Don't worry about it.

Steve on iPhone

On 30 Jan 2011, at 04:28, "Madovsky" <infos at madovsky.org> wrote:

> is there a way to clean inactive UDP sockets after
> a timeout ?
> when I do on my linux console
> [bash] # netstat -tuvnap
> I can see a lot of UDP sockets not removed after  calls.
> tried also to modify the kernel with nf_conntrack_udp_timeout without success.
>  
> Thanks
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From sos at sokhapkin.dyndns.org  Sun Jan 30 17:40:31 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sun, 30 Jan 2011 09:40:31 -0500
Subject: [Freeswitch-users] 200 OK without SDP
In-Reply-To: <AANLkTikzNqy3bOBXAsVdEumHyWfft_aPc-2jAfh+Jshi@mail.gmail.com>
References: <201101280730.17483.sos@sokhapkin.dyndns.org>
	<AANLkTi=KqOmroAYFRFyTfMPd_9_3Xjw0v3sdMtaSmdrc@mail.gmail.com>
	<AANLkTikzNqy3bOBXAsVdEumHyWfft_aPc-2jAfh+Jshi@mail.gmail.com>
Message-ID: <201101300940.31174.sos@sokhapkin.dyndns.org>

I set "enable-soa" to "false" in profile settings and got the same problem in 
proxy media mode.

On Saturday 29 January 2011, Anthony Minessale wrote:
> Session timer code is part of the sofia sip libraries not FS.
> If it's not compatible to use bypass and session timers together it
> may not be something we can fix.
> 
> All I can think of to try is to set "enable-soa" to "false" in profile
> params or sip_enable_soa=false channel var on both legs of the call.
> This would tell sofia not to mess with the SDP at all and might serve
> as a workaround.
> 
> On Fri, Jan 28, 2011 at 3:27 PM, Kristian Kielhofner <kris at kriskinc.com> 
wrote:
> > That's what I'm seeing here too.
> > 
> > Thanks!
> > 
> > On Fri, Jan 28, 2011 at 4:13 PM, Sergey Okhapkin
> > 
> > <sos at sokhapkin.dyndns.org> wrote:
> >> There is no problem if FS is in media path.
> > 
> > --
> > Kristian Kielhofner
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org



From joaocarlosleme at gmail.com  Sun Jan 30 18:54:30 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Sun, 30 Jan 2011 07:54:30 -0800
Subject: [Freeswitch-users] Hacker Attack?
In-Reply-To: <09EC9978-ED26-445F-9B0C-34D6D55DDA41@visionutveckling.se>
References: <09EC9978-ED26-445F-9B0C-34D6D55DDA41@visionutveckling.se>
Message-ID: <AANLkTimNOHVYE=uaM8djk10mtqweZfaQLg9GUz4=osBA@mail.gmail.com>

I figured. Same for Fail2Ban I guess. Any suggestions for Windows?

Also I was wondering why it never happened on my 1.0.4 (14460) version
(precompiled version)? I had it running for a month 24hrs and had never seen
this before. And after starting the Git Head (below) from Yesterday it
happened in seconds all 3 times I restarted (restarted the computer to be
sure). Maybe something wrong with the current version? To be safe I went
back to my stable 1.0.4 version and haven't had any problems.

49a5effcdf2cea9e0ddcf146cf3fe85d1872e654
mod_callcenter: Add error response for queue load and queue reload (FS-2988)

Marc Olivier Chouinard
 2011-01-29 00:09:06

On Sun, Jan 30, 2011 at 2:10 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> iptables is a Linux command.
>
> /Peter
>
>
> ----- Reply message -----
> Fr?n: "Joao Leme" <joaocarlosleme at gmail.com>
> Datum: s?n, jan 30, 2011 13:56
> Rubrik: [SPAM] - Re: [Freeswitch-users] Hacker Attack?
> Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>
> I tried "iptables -I INPUT -s [212.224.71.236] -j DROP" and got " Unknown
> command: iptables...". Do I must install fail2ban to issue iptables command?
> I'm on windows 7.
> Thanks
>
> On Sat, Jan 29, 2011 at 4:26 PM, curriegrad2004 <curriegrad2004 at gmail.com
> <mailto:curriegrad2004 at gmail.com>> wrote:
> iptables -I INPUT -s [hackerip] -j DROP
>
> A better solution is searching the wiki for fail2ban with FreeSwitch.
>
> On Sat, Jan 29, 2011 at 4:20 PM, Joao Leme <joaocarlosleme at gmail.com
> <mailto:joaocarlosleme at gmail.com>> wrote:
> > How do I do that?
> > Thanks!
> > On Sat, Jan 29, 2011 at 4:12 PM, curriegrad2004 <
> curriegrad2004 at gmail.com<mailto:curriegrad2004 at gmail.com>>
> > wrote:
> >>
> >> Try using iptables and block all incoming traffic from this specific
> host?
> >>
> >> On Sat, Jan 29, 2011 at 3:39 PM, Joao Leme <joaocarlosleme at gmail.com
> <mailto:joaocarlosleme at gmail.com>>
> >> wrote:
> >> > I just downloaded and compiled the latest Git and a little after
> >> > starting
> >> > freeswitch I'm getting non stop the following:
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [140 at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia
> >> > profile
> >> > ?internal? for [thomas at 76.XXX.XX.XXX] from ip 212.224.71.236
> >> > it's non-stop and doesn't let me do nothing else. After the first time
> I
> >> > went on to vars and changed the 1234 password....restarted and same
> >> > thing
> >> > happened, I also try denying the ip on acl.conf (not sure if has
> >> > something
> >> > to do with it but gave it a try):
> >> >
> >> > <configuration name="acl.conf" description="Network Lists">
> >> >         <network-lists>
> >> >           <list name="test2" default="allow">
> >> >             <node type="deny" host="212.224.71.236"
> >> > mask="255.255.255.0"/>
> >> >           </list>
> >> >         </network-lists>
> >> >       </configuration>
> >> >
> >> > Restarted the computer but nothing, he (thomas I guess) was back on my
> >> > console.
> >> >
> >> > Any ideas??? p.s. My computer is on DMZ (I know DMZ is not ideal but
> is
> >> > the
> >> > only way I got to be able to connect to the internal profile from out
> of
> >> > the
> >> > office etc).
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4d450b3232767678720833!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Nabble at slickdeals.endjunk.com  Sun Jan 30 19:24:02 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 30 Jan 2011 08:24:02 -0800 (PST)
Subject: [Freeswitch-users] Hacker Attack?
In-Reply-To: <AANLkTimcrJmG=CVTKMy-_4kZYA6hCU-_aORbbdRkO_P5@mail.gmail.com>
References: <AANLkTimcrJmG=CVTKMy-_4kZYA6hCU-_aORbbdRkO_P5@mail.gmail.com>
Message-ID: <1296404642666-5974868.post@n2.nabble.com>



Joao Leme wrote:
> Any ideas??? p.s. My computer is on DMZ (I know DMZ is not ideal but is
> the
> only way I got to be able to connect to the internal profile from out of
> the
> office etc).
This kind of flexibility exposes your equipment to crackers (not hackers)
out in the Internet world. Why not remove the equipment from DMZ and use VPN
to connect from office? This way, you will still be able to connect to the
internal profile in a more secure manner.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Hacker-Attack-tp5973660p5974868.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jeff at jefflenk.com  Sun Jan 30 19:37:32 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sun, 30 Jan 2011 08:37:32 -0800 (PST)
Subject: [Freeswitch-users] Hacker Attack?
In-Reply-To: <AANLkTimNOHVYE=uaM8djk10mtqweZfaQLg9GUz4=osBA@mail.gmail.com>
References: <AANLkTimcrJmG=CVTKMy-_4kZYA6hCU-_aORbbdRkO_P5@mail.gmail.com>
	<09EC9978-ED26-445F-9B0C-34D6D55DDA41@visionutveckling.se>
	<AANLkTimNOHVYE=uaM8djk10mtqweZfaQLg9GUz4=osBA@mail.gmail.com>
Message-ID: <1296405452492-5974888.post@n2.nabble.com>


The old version simply does not log that the activity is occurring that
functionality was added more recently. If you dont need your fs box exposed
to the public internet then remove it from there(problem solved).


-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Hacker-Attack-tp5973660p5974888.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From infos at madovsky.org  Sun Jan 30 20:37:31 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 30 Jan 2011 12:37:31 -0500
Subject: [Freeswitch-users] clean UDP sockets
References: <DF2C3B08C24544419D16B3C546EFD1A0@e1705>
	<29EC78A7-59C7-4F08-898C-52F0F6CD5C03@gmail.com>
Message-ID: <7DA549328E86425B81E69406031EC24B@e1705>

right, after hours I checked it again and yes, the kernel removed them.

Thanks Steve
  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Sunday, January 30, 2011 6:06 AM
  Subject: Re: [Freeswitch-users] clean UDP sockets


  The kernel will clean them up in it's own good time. Don't worry about it.

  Steve on iPhone

  On 30 Jan 2011, at 04:28, "Madovsky" <infos at madovsky.org> wrote:


    is there a way to clean inactive UDP sockets after
    a timeout ?
    when I do on my linux console
    [bash] # netstat -tuvnap
    I can see a lot of UDP sockets not removed after calls.
    tried also to modify the kernel with nf_conntrack_udp_timeout without success.

    Thanks
    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org



------------------------------------------------------------------------------


  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Sun Jan 30 20:53:55 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 30 Jan 2011 12:53:55 -0500
Subject: [Freeswitch-users] udp stress tester
Message-ID: <59820F42226441B2BA7A1E2D27837102@e1705>

I found cool python scripts that
test the quality of your udp connection
so you can check your server behavior
in heavy load scenario

http://www.elifulkerson.com/projects/python-udp-stress-tester.php

 
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From tayeb.meftah at gmail.com  Sun Jan 30 21:08:38 2011
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sun, 30 Jan 2011 19:08:38 +0100
Subject: [Freeswitch-users] open g729
In-Reply-To: <1296343422.2615.4.camel@gustavo-laptop>
References: <1296343422.2615.4.camel@gustavo-laptop>
Message-ID: <4D45A926.50403@gmail.com>

do it and end up in jail.
Le 30/01/2011 00:23, Gustavo Espeche a ?crit :
> Hello,
>          some one can compile open g729 to work with freeswitch?
>          http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
>
>          i appreciate a lot if some one has some experience in it.
>          Best Regards.
>
>          Gustavo Espeche
>          www.easyipcall.com
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
phone: +13477595883



From infos at madovsky.org  Sun Jan 30 21:25:54 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 30 Jan 2011 13:25:54 -0500
Subject: [Freeswitch-users] open g729
References: <1296343422.2615.4.camel@gustavo-laptop> <4D45A926.50403@gmail.com>
Message-ID: <FCBAF73D64FA451CB352C01FADACD040@e1705>

it's companies that create
private licence for public use
you need to put in jail....

----- Original Message ----- 
From: "Meftah Tayeb" <tayeb.meftah at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Cc: "Gustavo Espeche" <gustavo.espeche at upper-soft.com>
Sent: Sunday, January 30, 2011 1:08 PM
Subject: Re: [Freeswitch-users] open g729


do it and end up in jail.
Le 30/01/2011 00:23, Gustavo Espeche a ?crit :
> Hello,
>          some one can compile open g729 to work with freeswitch?
>          http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
>
>          i appreciate a lot if some one has some experience in it.
>          Best Regards.
>
>          Gustavo Espeche
>          www.easyipcall.com
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


-- 
Meftah Tayeb
inum: +883510001288000
phone: +13477595883


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 



From curriegrad2004 at gmail.com  Sun Jan 30 21:30:58 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 30 Jan 2011 10:30:58 -0800
Subject: [Freeswitch-users] open g729
In-Reply-To: <FCBAF73D64FA451CB352C01FADACD040@e1705>
References: <1296343422.2615.4.camel@gustavo-laptop> <4D45A926.50403@gmail.com>
	<FCBAF73D64FA451CB352C01FADACD040@e1705>
Message-ID: <AANLkTimhCufZkGZKbFaoufnJ5Tqpr9kYK7VpSoaV0_tT@mail.gmail.com>

Meh, I think he wants a free g729 codec, really.

Why don't they just kill g729 completely and move on with Speex already...

On Sun, Jan 30, 2011 at 10:25 AM, Madovsky <infos at madovsky.org> wrote:
> it's companies that create
> private licence for public use
> you need to put in jail....
>
> ----- Original Message -----
> From: "Meftah Tayeb" <tayeb.meftah at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Cc: "Gustavo Espeche" <gustavo.espeche at upper-soft.com>
> Sent: Sunday, January 30, 2011 1:08 PM
> Subject: Re: [Freeswitch-users] open g729
>
>
> do it and end up in jail.
> Le 30/01/2011 00:23, Gustavo Espeche a ?crit :
>> Hello,
>> ? ? ? ? ?some one can compile open g729 to work with freeswitch?
>> ? ? ? ? ?http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
>>
>> ? ? ? ? ?i appreciate a lot if some one has some experience in it.
>> ? ? ? ? ?Best Regards.
>>
>> ? ? ? ? ?Gustavo Espeche
>> ? ? ? ? ?www.easyipcall.com
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> --
> Meftah Tayeb
> inum: +883510001288000
> phone: +13477595883
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From sos at sokhapkin.dyndns.org  Sun Jan 30 21:39:03 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sun, 30 Jan 2011 13:39:03 -0500
Subject: [Freeswitch-users] 200 OK without SDP
In-Reply-To: <AANLkTikzNqy3bOBXAsVdEumHyWfft_aPc-2jAfh+Jshi@mail.gmail.com>
References: <201101280730.17483.sos@sokhapkin.dyndns.org>
	<AANLkTi=KqOmroAYFRFyTfMPd_9_3Xjw0v3sdMtaSmdrc@mail.gmail.com>
	<AANLkTikzNqy3bOBXAsVdEumHyWfft_aPc-2jAfh+Jshi@mail.gmail.com>
Message-ID: <201101301339.03971.sos@sokhapkin.dyndns.org>

Does sofia lib call mod_sofia code when SST reinvite is received?

On Saturday 29 January 2011, Anthony Minessale wrote:
> Session timer code is part of the sofia sip libraries not FS.
> If it's not compatible to use bypass and session timers together it
> may not be something we can fix.
> 
> All I can think of to try is to set "enable-soa" to "false" in profile
> params or sip_enable_soa=false channel var on both legs of the call.
> This would tell sofia not to mess with the SDP at all and might serve
> as a workaround.
> 
> On Fri, Jan 28, 2011 at 3:27 PM, Kristian Kielhofner <kris at kriskinc.com> 
wrote:
> > That's what I'm seeing here too.
> > 
> > Thanks!
> > 
> > On Fri, Jan 28, 2011 at 4:13 PM, Sergey Okhapkin
> > 
> > <sos at sokhapkin.dyndns.org> wrote:
> >> There is no problem if FS is in media path.
> > 
> > --
> > Kristian Kielhofner
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org



From infos at madovsky.org  Sun Jan 30 21:46:01 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 30 Jan 2011 13:46:01 -0500
Subject: [Freeswitch-users] open g729
Message-ID: <0610594E3D2241C2B0BDDEC28EDBF605@e1705>

Imagine MP3 with the same licence of G729...
let's put all the world in the jail !!! :D

----- Original Message ----- 
From: "Madovsky" <infos at madovsky.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Sunday, January 30, 2011 1:25 PM
Subject: Re: [Freeswitch-users] open g729


> it's companies that create
> private licence for public use
> you need to put in jail....
>
> ----- Original Message ----- 
> From: "Meftah Tayeb" <tayeb.meftah at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Cc: "Gustavo Espeche" <gustavo.espeche at upper-soft.com>
> Sent: Sunday, January 30, 2011 1:08 PM
> Subject: Re: [Freeswitch-users] open g729
>
>
> do it and end up in jail.
> Le 30/01/2011 00:23, Gustavo Espeche a ?crit :
>> Hello,
>>          some one can compile open g729 to work with freeswitch?
>>          http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
>>
>>          i appreciate a lot if some one has some experience in it.
>>          Best Regards.
>>
>>          Gustavo Espeche
>>          www.easyipcall.com
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> -- 
> Meftah Tayeb
> inum: +883510001288000
> phone: +13477595883
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From curriegrad2004 at gmail.com  Sun Jan 30 22:45:30 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 30 Jan 2011 11:45:30 -0800
Subject: [Freeswitch-users] Compilation with -march options
Message-ID: <AANLkTi=civOgvpz55hxEifdkcmKWmZABsitj290cCkxN@mail.gmail.com>

Is compiling FreeSwitch with the -march option in gcc a good idea?
Because I just compiled the latest git using the -march option
targeted for a Pentium 3 Processor and it did seem to increase the
performance slightly in switching calls and transcoding calls.

So far I haven't seen any adverse effects of doing this, but I was
wondering if anybody else out there is also compiling FS with a march
option and saw some performance increase or adverse effects...


From steveayre at gmail.com  Sun Jan 30 23:32:29 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 30 Jan 2011 20:32:29 +0000
Subject: [Freeswitch-users] open g729
In-Reply-To: <0610594E3D2241C2B0BDDEC28EDBF605@e1705>
References: <0610594E3D2241C2B0BDDEC28EDBF605@e1705>
Message-ID: <AANLkTikng7_w0XcLb+B8AQ=3Xo1xsHDWkbmTz=8krWOt@mail.gmail.com>

Funny you mention it, but MP3 does have several patents covering aspects of
it that expire between 2007 (fine) and 2017 (not so fine). There have been
various companies at various times that have claimed licenses are required
to encode/decode it. Fraunhofer Institute being a good example - they earned
100million euros in 2005 from mp3 licenses. In1998 they sent a lot of
letters out to developers of software using mp3 stating that they needed a
licence.

I guess the difference from G729 is that a) enough people in the general
public use it and want to continue using it and b) the patent holders have
declined to enforce license fees on free/foss software, only focussing on
commerical software, so there's plenty of stuff around that can use it
without worrying too much about the patents.

http://en.wikipedia.org/wiki/MP3#Licensing_and_patent_issues

-Steve

On 30 January 2011 18:46, Madovsky <infos at madovsky.org> wrote:

> Imagine MP3 with the same licence of G729...
> let's put all the world in the jail !!! :D
>
> ----- Original Message -----
> From: "Madovsky" <infos at madovsky.org>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Sunday, January 30, 2011 1:25 PM
> Subject: Re: [Freeswitch-users] open g729
>
>
> > it's companies that create
> > private licence for public use
> > you need to put in jail....
> >
> > ----- Original Message -----
> > From: "Meftah Tayeb" <tayeb.meftah at gmail.com>
> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > Cc: "Gustavo Espeche" <gustavo.espeche at upper-soft.com>
> > Sent: Sunday, January 30, 2011 1:08 PM
> > Subject: Re: [Freeswitch-users] open g729
> >
> >
> > do it and end up in jail.
> > Le 30/01/2011 00:23, Gustavo Espeche a ?crit :
> >> Hello,
> >>          some one can compile open g729 to work with freeswitch?
> >>          http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
> >>
> >>          i appreciate a lot if some one has some experience in it.
> >>          Best Regards.
> >>
> >>          Gustavo Espeche
> >>          www.easyipcall.com
> >>
> >>
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > --
> > Meftah Tayeb
> > inum: +883510001288000
> > phone: +13477595883
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Sun Jan 30 23:34:19 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 30 Jan 2011 20:34:19 +0000
Subject: [Freeswitch-users] 200 OK without SDP
In-Reply-To: <201101301339.03971.sos@sokhapkin.dyndns.org>
References: <201101280730.17483.sos@sokhapkin.dyndns.org>
	<AANLkTi=KqOmroAYFRFyTfMPd_9_3Xjw0v3sdMtaSmdrc@mail.gmail.com>
	<AANLkTikzNqy3bOBXAsVdEumHyWfft_aPc-2jAfh+Jshi@mail.gmail.com>
	<201101301339.03971.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTikWeY6xbOovF+gcBE_p9UFUSuyiUndbP9R-1UZf@mail.gmail.com>

I think so, but a lot of the reinvite processing is done internally by Sofia
and not by mod_sofia.

-Steve



On 30 January 2011 18:39, Sergey Okhapkin <sos at sokhapkin.dyndns.org> wrote:

> Does sofia lib call mod_sofia code when SST reinvite is received?
>
> On Saturday 29 January 2011, Anthony Minessale wrote:
> > Session timer code is part of the sofia sip libraries not FS.
> > If it's not compatible to use bypass and session timers together it
> > may not be something we can fix.
> >
> > All I can think of to try is to set "enable-soa" to "false" in profile
> > params or sip_enable_soa=false channel var on both legs of the call.
> > This would tell sofia not to mess with the SDP at all and might serve
> > as a workaround.
> >
> > On Fri, Jan 28, 2011 at 3:27 PM, Kristian Kielhofner <kris at kriskinc.com>
> wrote:
> > > That's what I'm seeing here too.
> > >
> > > Thanks!
> > >
> > > On Fri, Jan 28, 2011 at 4:13 PM, Sergey Okhapkin
> > >
> > > <sos at sokhapkin.dyndns.org> wrote:
> > >> There is no problem if FS is in media path.
> > >
> > > --
> > > Kristian Kielhofner
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From frank at telonium.com  Sun Jan 30 23:11:52 2011
From: frank at telonium.com (Frank Park)
Date: Sun, 30 Jan 2011 15:11:52 -0500
Subject: [Freeswitch-users] MoH and bypass media
Message-ID: <AANLkTinJgVuR81AtMfr1ROhKgN21nxOSyG4ZkdZtygUX@mail.gmail.com>

Ok.. I know this topic's been covered several times, but I wanted to get the
latest information on this.

Brian said 2 years ago that...
"you can't get out of the media path when you unhold... FS will be in the
media path hold button moving forward unless you use the uuid_media command
to turn it back off. "

Now I have few questions:
1) Is this still the case?
2) If we have a console command to bypass the call on-the-fly, why can't
this be part of the built-in feature?
3) What other clever way have others come up with because using separate ESL
script to invoke such command every single time someone unholds a call?

Thanks!
Frank

-- 

----=======================----
Frank Park
Telonium Communications, LLC
frank at telonium.com
http://www.telonium.com
Follow Us on Twitter: @GetTelonium
404-566-8888 x1001 Office
404-939-4242 Cell
----=======================----
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From steveayre at gmail.com  Sun Jan 30 23:37:02 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 30 Jan 2011 20:37:02 +0000
Subject: [Freeswitch-users] clean UDP sockets
In-Reply-To: <7DA549328E86425B81E69406031EC24B@e1705>
References: <DF2C3B08C24544419D16B3C546EFD1A0@e1705>
	<29EC78A7-59C7-4F08-898C-52F0F6CD5C03@gmail.com>
	<7DA549328E86425B81E69406031EC24B@e1705>
Message-ID: <AANLkTinUcBpySK_DbcoNLJ-jL5TJsA4SJJH+hS0Co6wu@mail.gmail.com>

It's normal for the kernel to hold onto a port for a while after it's been
closed.

I believe it's so that you don't get another program grabbing the port
number and receiving data that was meant for the first program. Partly
that's good for security, and partly it stops the program getting data not
meant for it which may confuse it.

-Steve


On 30 January 2011 17:37, Madovsky <infos at madovsky.org> wrote:

>  right, after hours I checked it again and yes, the kernel removed them.
>
> Thanks Steve
>
> ----- Original Message -----
> *From:* Steven Ayre <steveayre at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Sunday, January 30, 2011 6:06 AM
> *Subject:* Re: [Freeswitch-users] clean UDP sockets
>
> The kernel will clean them up in it's own good time. Don't worry about it.
>
> Steve on iPhone
>
> On 30 Jan 2011, at 04:28, "Madovsky" <infos at madovsky.org> wrote:
>
>   is there a way to clean inactive UDP sockets after
> a timeout ?
> when I do on my linux console
> [bash] # netstat -tuvnap
> I can see a lot of UDP sockets not removed after calls.
> tried also to modify the kernel with nf_conntrack_udp_timeout without
> success.
>
> Thanks
>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From megahohol at gmail.com  Sun Jan 30 23:00:14 2011
From: megahohol at gmail.com (Grygoriy Dobrovolskyy)
Date: Sun, 30 Jan 2011 21:00:14 +0100
Subject: [Freeswitch-users] open g729
In-Reply-To: <0610594E3D2241C2B0BDDEC28EDBF605@e1705>
References: <0610594E3D2241C2B0BDDEC28EDBF605@e1705>
Message-ID: <AANLkTinA13kcMp5GU+gmW+exs7MBoaBdgtzeNdr2LstG@mail.gmail.com>

Hello, there is only one valid point for g.729 *free* It can be used
in countries where G.729 licence do not apply, without reinstrictions.
I can not name them, but i suppose you all know the list.

2011/1/30 Madovsky <infos at madovsky.org>:
> Imagine MP3 with the same licence of G729...
> let's put all the world in the jail !!! :D
>
> ----- Original Message -----
> From: "Madovsky" <infos at madovsky.org>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Sunday, January 30, 2011 1:25 PM
> Subject: Re: [Freeswitch-users] open g729
>
>
>> it's companies that create
>> private licence for public use
>> you need to put in jail....
>>
>> ----- Original Message -----
>> From: "Meftah Tayeb" <tayeb.meftah at gmail.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Cc: "Gustavo Espeche" <gustavo.espeche at upper-soft.com>
>> Sent: Sunday, January 30, 2011 1:08 PM
>> Subject: Re: [Freeswitch-users] open g729
>>
>>
>> do it and end up in jail.
>> Le 30/01/2011 00:23, Gustavo Espeche a ?crit :
>>> Hello,
>>> ? ? ? ? ?some one can compile open g729 to work with freeswitch?
>>> ? ? ? ? ?http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
>>>
>>> ? ? ? ? ?i appreciate a lot if some one has some experience in it.
>>> ? ? ? ? ?Best Regards.
>>>
>>> ? ? ? ? ?Gustavo Espeche
>>> ? ? ? ? ?www.easyipcall.com
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> --
>> Meftah Tayeb
>> inum: +883510001288000
>> phone: +13477595883
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From infos at madovsky.org  Mon Jan 31 00:06:49 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 30 Jan 2011 16:06:49 -0500
Subject: [Freeswitch-users] open g729
References: <0610594E3D2241C2B0BDDEC28EDBF605@e1705>
	<AANLkTinA13kcMp5GU+gmW+exs7MBoaBdgtzeNdr2LstG@mail.gmail.com>
Message-ID: <0B3F4B004C554A4684F47A0555730CE1@e1705>

this one doesn't work in production mode.
some calls are cut with this version of G729

----- Original Message ----- 
From: "Grygoriy Dobrovolskyy" <megahohol at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Sunday, January 30, 2011 3:00 PM
Subject: Re: [Freeswitch-users] open g729


Hello, there is only one valid point for g.729 *free* It can be used
in countries where G.729 licence do not apply, without reinstrictions.
I can not name them, but i suppose you all know the list.

2011/1/30 Madovsky <infos at madovsky.org>:
> Imagine MP3 with the same licence of G729...
> let's put all the world in the jail !!! :D
>
> ----- Original Message -----
> From: "Madovsky" <infos at madovsky.org>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Sunday, January 30, 2011 1:25 PM
> Subject: Re: [Freeswitch-users] open g729
>
>
>> it's companies that create
>> private licence for public use
>> you need to put in jail....
>>
>> ----- Original Message -----
>> From: "Meftah Tayeb" <tayeb.meftah at gmail.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Cc: "Gustavo Espeche" <gustavo.espeche at upper-soft.com>
>> Sent: Sunday, January 30, 2011 1:08 PM
>> Subject: Re: [Freeswitch-users] open g729
>>
>>
>> do it and end up in jail.
>> Le 30/01/2011 00:23, Gustavo Espeche a ?crit :
>>> Hello,
>>> some one can compile open g729 to work with freeswitch?
>>> http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
>>>
>>> i appreciate a lot if some one has some experience in it.
>>> Best Regards.
>>>
>>> Gustavo Espeche
>>> www.easyipcall.com
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> --
>> Meftah Tayeb
>> inum: +883510001288000
>> phone: +13477595883
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 



From infos at madovsky.org  Mon Jan 31 00:21:00 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 30 Jan 2011 16:21:00 -0500
Subject: [Freeswitch-users] open g729
References: <0610594E3D2241C2B0BDDEC28EDBF605@e1705>
	<AANLkTikng7_w0XcLb+B8AQ=3Xo1xsHDWkbmTz=8krWOt@mail.gmail.com>
Message-ID: <6BE716076665410F8FFB49899130DB7A@e1705>

yes I know the story,
but by the fact that even they made millions the first years,
they finally revoked to pursuit all developers and users.
that's the power of mass people...
it's like the transistor (invented by 2 germans).
imagine if they decided to get money of every transistor sold in the world
from a private patent. they would be maybe 100 times more rich that Bill Gates.
waht a wisdom when they decided (with a smile) to use a patent in public domain...
these guys rocked....
the opposite, apple gets royalties of every mouse made in the world....
  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Sunday, January 30, 2011 3:32 PM
  Subject: Re: [Freeswitch-users] open g729


  Funny you mention it, but MP3 does have several patents covering aspects of it that expire between 2007 (fine) and 2017 (not so fine). There have been various companies at various times that have claimed licenses are required to encode/decode it. Fraunhofer Institute being a good example - they earned 100million euros in 2005 from mp3 licenses. In1998 they sent a lot of letters out to developers of software using mp3 stating that they needed a licence.

  I guess the difference from G729 is that a) enough people in the general public use it and want to continue using it and b) the patent holders have declined to enforce license fees on free/foss software, only focussing on commerical software, so there's plenty of stuff around that can use it without worrying too much about the patents.

  http://en.wikipedia.org/wiki/MP3#Licensing_and_patent_issues

  -Steve


  On 30 January 2011 18:46, Madovsky <infos at madovsky.org> wrote:

    Imagine MP3 with the same licence of G729...
    let's put all the world in the jail !!! :D


    ----- Original Message -----
    From: "Madovsky" <infos at madovsky.org>
    To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

    Sent: Sunday, January 30, 2011 1:25 PM
    Subject: Re: [Freeswitch-users] open g729


    > it's companies that create
    > private licence for public use
    > you need to put in jail....
    >
    > ----- Original Message -----
    > From: "Meftah Tayeb" <tayeb.meftah at gmail.com>
    > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
    > Cc: "Gustavo Espeche" <gustavo.espeche at upper-soft.com>
    > Sent: Sunday, January 30, 2011 1:08 PM
    > Subject: Re: [Freeswitch-users] open g729
    >
    >
    > do it and end up in jail.
    > Le 30/01/2011 00:23, Gustavo Espeche a ?crit :
    >> Hello,
    >>          some one can compile open g729 to work with freeswitch?
    >>          http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
    >>
    >>          i appreciate a lot if some one has some experience in it.
    >>          Best Regards.
    >>
    >>          Gustavo Espeche
    >>          www.easyipcall.com
    >>
    >>
    >>
    >>
    >>
    >> _______________________________________________
    >> FreeSWITCH-users mailing list
    >> FreeSWITCH-users at lists.freeswitch.org
    >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    >> http://www.freeswitch.org
    >>
    >
    >
    > --
    > Meftah Tayeb
    > inum: +883510001288000
    > phone: +13477595883
    >
    >
    > _______________________________________________
    > FreeSWITCH-users mailing list
    > FreeSWITCH-users at lists.freeswitch.org
    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > http://www.freeswitch.org


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





------------------------------------------------------------------------------


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From anthony.minessale at gmail.com  Mon Jan 31 00:24:55 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 30 Jan 2011 15:24:55 -0600
Subject: [Freeswitch-users] 200 OK without SDP
In-Reply-To: <AANLkTikWeY6xbOovF+gcBE_p9UFUSuyiUndbP9R-1UZf@mail.gmail.com>
References: <201101280730.17483.sos@sokhapkin.dyndns.org>
	<AANLkTi=KqOmroAYFRFyTfMPd_9_3Xjw0v3sdMtaSmdrc@mail.gmail.com>
	<AANLkTikzNqy3bOBXAsVdEumHyWfft_aPc-2jAfh+Jshi@mail.gmail.com>
	<201101301339.03971.sos@sokhapkin.dyndns.org>
	<AANLkTikWeY6xbOovF+gcBE_p9UFUSuyiUndbP9R-1UZf@mail.gmail.com>
Message-ID: <AANLkTinD6ZER2NTq1uXk_hTc4a1Ab7o68Mbob+P=Lvt6@mail.gmail.com>

Can you repeat that pastebin with console loglevel debug?


On Sun, Jan 30, 2011 at 2:34 PM, Steven Ayre <steveayre at gmail.com> wrote:
> I think so, but a lot of the reinvite processing is done internally by Sofia
> and not by mod_sofia.
>
> -Steve
>
>
>
> On 30 January 2011 18:39, Sergey Okhapkin <sos at sokhapkin.dyndns.org> wrote:
>>
>> Does sofia lib call mod_sofia code when SST reinvite is received?
>>
>> On Saturday 29 January 2011, Anthony Minessale wrote:
>> > Session timer code is part of the sofia sip libraries not FS.
>> > If it's not compatible to use bypass and session timers together it
>> > may not be something we can fix.
>> >
>> > All I can think of to try is to set "enable-soa" to "false" in profile
>> > params or sip_enable_soa=false channel var on both legs of the call.
>> > This would tell sofia not to mess with the SDP at all and might serve
>> > as a workaround.
>> >
>> > On Fri, Jan 28, 2011 at 3:27 PM, Kristian Kielhofner <kris at kriskinc.com>
>> wrote:
>> > > That's what I'm seeing here too.
>> > >
>> > > Thanks!
>> > >
>> > > On Fri, Jan 28, 2011 at 4:13 PM, Sergey Okhapkin
>> > >
>> > > <sos at sokhapkin.dyndns.org> wrote:
>> > >> There is no problem if FS is in media path.
>> > >
>> > > --
>> > > Kristian Kielhofner
>> > >
>> > > _______________________________________________
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > >
>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Mon Jan 31 00:50:10 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 30 Jan 2011 15:50:10 -0600
Subject: [Freeswitch-users] Outbound only calls don't connect when
 bypass_media is true.
In-Reply-To: <COL118-W52E5226FC3337085AB6AFAE5E00@phx.gbl>
References: <COL118-W52E5226FC3337085AB6AFAE5E00@phx.gbl>
Message-ID: <AANLkTinEpyR0k+0jpx5dhRECKc4X+ok9WbSfuM8qAKU=@mail.gmail.com>

bypass media does not work through nat in many situations because
there is nothing FreeSWITCH can do to fix it since it's bypassing its
chance by design.


On Fri, Jan 28, 2011 at 6:54 PM, Marcin Wojtowicz <marcin321 at hotmail.com> wrote:
> Hello,
>
> I'm a new user of freeswitch, so please bear with me. I have the
> following setup:
> voip.ms <- SIP over UDP -> my desktop running freeswitch <- SIP over TCP ->
> my nokia cellphone on AT&T wireless. This setup is intended to conserve the
> battery usage.
> I've managed to make everything work well when I'm calling in over any phone
> to my cell phone, and freeswitch is enabled to work in bypass_media = true,
> even though by cell is behind NAT on at&t's network. Things break when I
> pick up my cell and try to call my home phone (or any phone for that
> matter). This is the relevant snippet from my dialplan:
> <extension name="outbound">
> ? <condition field="destination_number"
> expression="^1?([2-9]\d{2}[2-9]\d{6})$">
> ??? <!--<action application="set" data="bypass_media=true"/>-->
> ??? <action application="bridge" data="sofia/gateway/voip.ms/1$1"/>
> ? </condition>
> </extension>
>
> Like shown above, my call will go to my home phone. When I uncomment the
> bypass_media tag, my call will not connect. Here are the siptraces
> I replaced my real home phone number in the with "MYPHONE".
>
> recv 940 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.406250:
> ?? ------------------------------------------------------------------------
> ?? INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> ?? Via: SIP/2.0/TCP
> 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport
> ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> ?? To: <sip:MYPHONE at 192.168.1.100>
> ?? Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>
> ?? Supported: 100rel,timer
> ?? CSeq: 5244503 INVITE
> ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> ?? Allow: UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE
> ?? User-Agent: S60 RM-624 v 20.2.042 (en)
> ?? Expires: 120
> ?? Privacy: None
> ?? Session-Expires: 1800
> ?? Max-Forwards: 70
> ?? Content-Type: application/sdp
> ?? Accept-Language: en
> ?? Content-Length: 292
>
> ?? v=0
> ?? o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> ?? s=-
> ?? c=IN IP4 10.153.174.6
> ?? t=0 0
> ?? m=audio 49152 RTP/AVP 18 97 98
> ?? a=sendrecv
> ?? a=rtpmap:18 G729/8000
> ?? a=ptime:20
> ?? a=maxptime:40
> ?? a=fmtp:18 annexb=no
> ?? a=rtpmap:97 iLBC/8000
> ?? a=rtpmap:98 telephone-event/8000
> ?? a=fmtp:98 0-15
> ?? ------------------------------------------------------------------------
> send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:
> ?? ------------------------------------------------------------------------
> ?? SIP/2.0 100 Trying
> ?? Via: SIP/2.0/TCP
> 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180
> ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> ?? To: <sip:MYPHONE at 192.168.1.100>
> ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> ?? CSeq: 5244503 INVITE
> ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> -0600
> ?? Content-Length: 0
>
> ?? ------------------------------------------------------------------------
> send 875 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:
> ?? ------------------------------------------------------------------------
> ?? SIP/2.0 407 Proxy Authentication Required
> ?? Via: SIP/2.0/TCP
> 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180
> ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj2011-01-28
> 16:15:58.406250 [WARNING] sofia_reg.c:1241 SIP auth challenge (INVITE) on
> sofia profile 'internal' for [MYPHONE at 192.168.1.100] from ip 32.136.78.180
>
> ?? To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m
> ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> ?? CSeq: 5244503 INVITE
> ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> -0600
> ?? Accept: application/sdp
> ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> ?? Supported: timer, precondition, path, replaces
> ?? Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
> ?? Proxy-Authenticate: Digest realm="192.168.1.100",
> nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28", algorithm=MD5, qop="auth"
> ?? Content-Length: 0
>
> ?? ------------------------------------------------------------------------
> recv 368 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.765625:
> ?? ------------------------------------------------------------------------
> ?? ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> ?? Via: SIP/2.0/TCP
> 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport
> ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> ?? To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m
> ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> ?? CSeq: 5244503 ACK
> ?? Supported: sec-agree
> ?? Max-Forwards: 70
> ?? Content-Length: 0
>
> ?? ------------------------------------------------------------------------
> recv 1222 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.406250:
> ?? ------------------------------------------------------------------------
> ?? INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> ?? Via: SIP/2.0/TCP
> 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport
> ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> ?? To: <sip:MYPHONE at 192.168.1.100>
> ?? Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>
> ?? Supported: 100rel,timer
> ?? CSeq: 5244504 INVITE
> ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> ?? Allow: UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE
> ?? User-Agent: S60 RM-624 v 20.2.042 (en)
> ?? Expires: 120
> ?? Privacy: None
> ?? Session-Expires: 1800
> ?? Max-Forwards: 70
> ?? Proxy-Authorization: Digest
> qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"
> ?? Content-Type: application/sdp
> ?? Accept-Language: en
> ?? Content-Length: 292
>
> ?? v=0
> ?? o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> ?? s=-
> ?? c=IN IP4 10.153.174.6
> ?? t=0 0
> ?? m=audio 49152 RTP/AVP 18 97 98
> ?? a=sendrecv
> ?? a=rtpmap:18 G729/8000
> ?? a=ptime:20
> ?? a=maxptime:40
> ?? a=fmtp:18 annexb=no
> ?? a=rtpmap:97 iLBC/8000
> ?? a=rtpmap:98 telephone-event/8000
> ?? a=fmtp:98 0-15
> ?? ------------------------------------------------------------------------
> send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.406250:
> ?? ------------------------------------------------------------------------
> ?? SIP/2.0 100 Trying
> ?? Via: SIP/2.0/TCP
> 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180
> ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> ?? To: <sip:MYPHONE at 192.168.1.100>
> ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> ?? CSeq: 5244504 INVITE
> ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> -0600
> ?? Content-Length: 0
>
> ?? ------------------------------------------------------------------------
> 2011-01-28 16:15:59.406250 [NOTICE] switch_channel.c:808 New Channel
> sofia/internal/1001 at 192.168.1.100 [e5841001-04bd-4e16-9519-64ff2c7a8c2f]
> 2011-01-28 16:15:59.421875 [INFO] mod_dialplan_xml.c:331 Processing 1001
> <1001>->MYPHONE in context default
> 2011-01-28 16:15:59.437500 [NOTICE] switch_channel.c:808 New Channel
> sofia/external/1MYPHONE [60940227-9ae0-4c0c-abdb-f3988852fae0]
> send 1141 bytes to udp/[74.63.41.218]:5060 at 21:15:59.453125:
> ?? ------------------------------------------------------------------------
> ?? INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0
> ?? Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS
> ?? Max-Forwards: 69
> ?? From: "Extension 1001"
> <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> ?? To: <sip:1MYPHONE at newyork.voip.ms>
> ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> ?? CSeq: 7788615 INVITE
> ?? Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
> ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> -0600
> ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> ?? Supported: timer, precondition, path, replaces
> ?? Allow-Events: talk, hold, refer
> ?? Content-Type: application/sdp
> ?? Content-Disposition: session
> ?? Content-Length: 280
> ?? X-FS-Support: update_display
> ?? Remote-Party-ID: "Extension 1001"
> <sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off
>
> ?? v=0
> ?? o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> ?? s=-
> ?? c=IN IP4 10.153.174.6
> ?? t=0 0
> ?? m=audio 49152 RTP/AVP 18 97 98
> ?? a=rtpmap:18 G729/8000
> ?? a=fmtp:18 annexb=no
> ?? a=rtpmap:97 iLBC/8000
> ?? a=rtpmap:98 telephone-event/8000
> ?? a=fmtp:98 0-15
> ?? a=ptime:20
> ?? a=maxptime:40
> ?? ------------------------------------------------------------------------
> recv 570 bytes from udp/[74.63.41.218]:5060 at 21:15:59.468750:
> ?? ------------------------------------------------------------------------
> ?? SIP/2.0 407 Proxy Authentication Required
> ?? Via: SIP/2.0/UDP
> 69.125.20.15:5080;branch=z9hG4bKNp4ZFeKSD43tS;received=69.125.20.15;rport=5080
> ?? From: "Extension 1001"
> <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> ?? To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843
> ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> ?? CSeq: 7788615 INVITE
> ?? User-Agent: VoIPMS/SERAST
> ?? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> ?? Supported: replaces
> ?? Proxy-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms",
> nonce="2d534dd6"
> ?? Content-Length: 0
>
> ?? ------------------------------------------------------------------------
> send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:
> ?? ------------------------------------------------------------------------
> ?? ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0
> ?? Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS
> ?? Max-Forwards: 69
> ?? From: "Extension 1001"
> <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> ?? To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843
> ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> ?? CSeq: 7788615 ACK
> ?? Content-Length: 0
>
> ?? ------------------------------------------------------------------------
> send 1344 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:
> ?? ------------------------------------------------------------------------
> ?? INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0
> ?? Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN
> ?? Max-Forwards: 69
> ?? From: "Extension 1001"
> <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> ?? To: <sip:1MYPHONE at newyork.voip.ms>
> ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> ?? CSeq: 7788616 INVITE
> ?? Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
> ?? Expires: 300
> ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> -0600
> ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> ?? Supported: timer, precondition, path, replaces
> ?? Allow-Events: talk, hold, refer
> ?? Proxy-Authorization: Digest username="121628", realm="newyork.voip.ms",
> nonce="2d534dd6", algorithm=MD5, uri="sip:1MYPHONE at newyork.voip.ms",
> response="16f3301efae13df926da7550f709d28a"
> ?? Content-Type: application/sdp
> ?? Content-Disposition: session
> ?? Content-Length: 280
> ?? X-FS-Support: update_display
> ?? Remote-Party-ID: "Extension 1001"
> <sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off
>
> ?? v=0
> ?? o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> ?? s=-
> ?? c=IN IP4 10.153.174.6
> ?? t=0 0
> ?? m=audio 49152 RTP/AVP 18 97 98
> ?? a=rtpmap:18 G729/8000
> ?? a=fmtp:18 annexb=no
> ?? a=rtpmap:97 iLBC/8000
> ?? a=rtpmap:98 telephone-event/8000
> ?? a=fmtp:98 0-15
> ?? a=ptime:20
> ?? a=maxptime:40
> ?? ------------------------------------------------------------------------
> recv 488 bytes from udp/[74.63.41.218]:5060 at 21:15:59.484375:
> ?? ------------------------------------------------------------------------
> ?? SIP/2.0 100 Trying
> ?? Via: SIP/2.0/UDP
> 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080
> ?? From: "Extension 1001"
> <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> ?? To: <sip:1MYPHONE at newyork.voip.ms>
> ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> ?? CSeq: 7788616 INVITE
> ?? User-Agent: VoIPMS/SERAST
> ?? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> ?? Supported: replaces
> ?? Contact: <sip:1MYPHONE at 74.63.41.218>
> ?? Content-Length: 0
>
> ?? ------------------------------------------------------------------------
> recv 516 bytes from udp/[74.63.41.218]:5060 at 21:15:59.562500:
> ?? ------------------------------------------------------------------------
> ?? SIP/2.0 503 Service Unavailable
> ?? Via: SIP/2.0/UDP
> 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080
> ?? From: "Extension 1001"
> <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> ?? To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9
> ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> ?? CSeq: 7788616 INVITE
> ?? User-Agent: VoIPMS/SERAST
> ?? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> ?? Supported: replaces
> ?? Contact: <sip:1MYPHONE at 74.63.41.218>
> ?? Content-Length: 0
>
> ?? ------------------------------------------------------------------------
> send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.562500:
> ?? ------------------------------------------------------------------------
> ?? ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0
> ?? Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN
> ?? Max-Forwards: 69
> ?? From: "Extension 1001"
> <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> ?? To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9
> ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> ?? CSeq: 7788616 ACK
> ?? Content-Length: 0
>
> ?? ------------------------------------------------------------------------
> 2011-01-28 16:15:59.562500 [INFO] mod_dptools.c:2612 Originate Failed.
> Cause: NO_ANSWER
> 2011-01-28 16:15:59.562500 [NOTICE] sofia.c:5286 Hangup
> sofia/external/1MYPHONE [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
> 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:189
> sofia/internal/1001 at 192.168.1.100 has executed the last dialplan
> instruction, hanging up.
> 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:191 Hangup
> sofia/internal/1001 at 192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1306 Session 2
> (sofia/external/1MYPHONE) Ended
> 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1308 Close Channel
> sofia/external/1MYPHONE [CS_DESTROY]
> send 887 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.593750:
> ?? ------------------------------------------------------------------------
> ?? SIP/2.0 503 Service Unavailable
> ?? Via: SIP/2.0/TCP
> 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180
> ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> ?? To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg
> ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> ?? CSeq: 5244504 INVITE
> ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> -0600
> ?? Accept: application/sdp
> ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> ?? Supported: timer, precondition, path, replaces
> ?? Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
> ?? Reason: Q.850;cause=16;text="NORMAL_CLEARING"
> 2011-01-28 16:15:59.593750 [NOTICE] switch_core_session.c:1306 Session 1
> (sofia/internal/1001 at 192.168.1.100) Ended
> ?? Content-Length: 02011-01-28 16:15:59.593750 [NOTICE]
> switch_core_session.c:1308 Close Channel sofia/internal/1001 at 192.168.1.100
> [CS_DESTROY]
>
> ?? Remote-Party-ID: "MYPHONE"
> <sip:MYPHONE at 192.168.1.100>;party=calling;privacy=off;screen=no
>
> ?? ------------------------------------------------------------------------
> recv 650 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.953125:
> ?? ------------------------------------------------------------------------
> ?? ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> ?? Via: SIP/2.0/TCP
> 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport
> ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> ?? To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg
> ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> ?? CSeq: 5244504 ACK
> ?? Supported: sec-agree
> ?? Max-Forwards: 70
> ?? Proxy-Authorization: Digest
> qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"
> ?? Content-Length: 0
>
> ?? ------------------------------------------------------------------------
>
> Thank you in advance.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
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Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
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FreeSWITCH Developer Conference
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pstn:+19193869900


From marcin321 at hotmail.com  Mon Jan 31 00:44:45 2011
From: marcin321 at hotmail.com (Marcin Wojtowicz)
Date: Sun, 30 Jan 2011 16:44:45 -0500
Subject: [Freeswitch-users] Outbound only calls don't connect when
 bypass_media is true.
In-Reply-To: <COL118-W52E5226FC3337085AB6AFAE5E00@phx.gbl>
References: <COL118-W52E5226FC3337085AB6AFAE5E00@phx.gbl>
Message-ID: <COL118-W38F26048D441DAA2777942E5E30@phx.gbl>


I just want to add that I enabled STUN on my cell so now the SDP message in the INVITE to voip.ms contains the public IP of my phone, but it still doesn't work.

From: marcin321 at hotmail.com
To: freeswitch-users at lists.freeswitch.org
Date: Fri, 28 Jan 2011 19:54:19 -0500
Subject: [Freeswitch-users] Outbound only calls don't connect when	bypass_media is true.









Hello,



I'm a new user of freeswitch, so please bear with me. I have the

following setup:

voip.ms <- SIP over UDP -> my desktop running freeswitch <- SIP over TCP 
-> my nokia cellphone on AT&T wireless. This setup is intended to 
conserve the battery usage.

I've managed to make everything work well when I'm calling in over any 
phone to my cell phone, and freeswitch is enabled to work in 
bypass_media = true, even though by cell is behind NAT on at&t's 
network. Things break when I pick up my cell and try to call my home 
phone (or any phone for that matter). This is the relevant snippet from 
my dialplan:

<extension name="outbound">

  <condition field="destination_number" 
expression="^1?([2-9]\d{2}[2-9]\d{6})$">

    <!--<action application="set" data="bypass_media=true"/>-->

    <action application="bridge" data="sofia/gateway/voip.ms/1$1"/>

  </condition>

</extension>



Like shown above, my call will go to my home phone. When I uncomment the 
bypass_media tag, my call will not connect. Here are the siptraces

I replaced my real home phone number in the with "MYPHONE".



recv 940 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.406250:

   ------------------------------------------------------------------------

   INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0

   Via: SIP/2.0/TCP 
10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport

   From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj

   To: <sip:MYPHONE at 192.168.1.100>

   Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>

   Supported: 100rel,timer

   CSeq: 5244503 INVITE

   Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn

   Allow: UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE

   User-Agent: S60 RM-624 v 20.2.042 (en)

   Expires: 120

   Privacy: None

   Session-Expires: 1800

   Max-Forwards: 70

   Content-Type: application/sdp

   Accept-Language: en

   Content-Length: 292



   v=0

   o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6

   s=-

   c=IN IP4 10.153.174.6

   t=0 0

   m=audio 49152 RTP/AVP 18 97 98

   a=sendrecv

   a=rtpmap:18 G729/8000

   a=ptime:20

   a=maxptime:40

   a=fmtp:18 annexb=no

   a=rtpmap:97 iLBC/8000

   a=rtpmap:98 telephone-event/8000

   a=fmtp:98 0-15

   ------------------------------------------------------------------------

send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:

   ------------------------------------------------------------------------

   SIP/2.0 100 Trying

   Via: SIP/2.0/TCP 
10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180

   From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj

   To: <sip:MYPHONE at 192.168.1.100>

   Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn

   CSeq: 5244503 INVITE

   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 
18-04-05 -0600

   Content-Length: 0



   ------------------------------------------------------------------------

send 875 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:

   ------------------------------------------------------------------------

   SIP/2.0 407 Proxy Authentication Required

   Via: SIP/2.0/TCP 
10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180

   From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj2011-01-28 
16:15:58.406250 [WARNING] sofia_reg.c:1241 SIP auth challenge (INVITE) 
on sofia profile 'internal' for [MYPHONE at 192.168.1.100] from ip 
32.136.78.180



   To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m

   Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn

   CSeq: 5244503 INVITE

   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 
18-04-05 -0600

   Accept: application/sdp

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

   Supported: timer, precondition, path, replaces

   Allow-Events: talk, hold, presence, dialog, line-seize, call-info, 
sla, include-session-description, presence.winfo, message-summary, refer

   Proxy-Authenticate: Digest realm="192.168.1.100", 
nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28", algorithm=MD5, qop="auth"

   Content-Length: 0



   ------------------------------------------------------------------------

recv 368 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.765625:

   ------------------------------------------------------------------------

   ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0

   Via: SIP/2.0/TCP 
10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport

   From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj

   To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m

   Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn

   CSeq: 5244503 ACK

   Supported: sec-agree

   Max-Forwards: 70

   Content-Length: 0



   ------------------------------------------------------------------------

recv 1222 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.406250:

   ------------------------------------------------------------------------

   INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0

   Via: SIP/2.0/TCP 
10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport

   From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj

   To: <sip:MYPHONE at 192.168.1.100>

   Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>

   Supported: 100rel,timer

   CSeq: 5244504 INVITE

   Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn

   Allow: UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE

   User-Agent: S60 RM-624 v 20.2.042 (en)

   Expires: 120

   Privacy: None

   Session-Expires: 1800

   Max-Forwards: 70

   Proxy-Authorization: Digest 
qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"

   Content-Type: application/sdp

   Accept-Language: en

   Content-Length: 292



   v=0

   o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6

   s=-

   c=IN IP4 10.153.174.6

   t=0 0

   m=audio 49152 RTP/AVP 18 97 98

   a=sendrecv

   a=rtpmap:18 G729/8000

   a=ptime:20

   a=maxptime:40

   a=fmtp:18 annexb=no

   a=rtpmap:97 iLBC/8000

   a=rtpmap:98 telephone-event/8000

   a=fmtp:98 0-15

   ------------------------------------------------------------------------

send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.406250:

   ------------------------------------------------------------------------

   SIP/2.0 100 Trying

   Via: SIP/2.0/TCP 
10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180

   From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj

   To: <sip:MYPHONE at 192.168.1.100>

   Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn

   CSeq: 5244504 INVITE

   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 
18-04-05 -0600

   Content-Length: 0



   ------------------------------------------------------------------------

2011-01-28 16:15:59.406250 [NOTICE] switch_channel.c:808 New Channel 
sofia/internal/1001 at 192.168.1.100 [e5841001-04bd-4e16-9519-64ff2c7a8c2f]

2011-01-28 16:15:59.421875 [INFO] mod_dialplan_xml.c:331 Processing 1001 
<1001>->MYPHONE in context default

2011-01-28 16:15:59.437500 [NOTICE] switch_channel.c:808 New Channel 
sofia/external/1MYPHONE [60940227-9ae0-4c0c-abdb-f3988852fae0]

send 1141 bytes to udp/[74.63.41.218]:5060 at 21:15:59.453125:

   ------------------------------------------------------------------------

   INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0

   Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS

   Max-Forwards: 69

   From: "Extension 1001" 
<sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S

   To: <sip:1MYPHONE at newyork.voip.ms>

   Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a

   CSeq: 7788615 INVITE

   Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>

   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 
18-04-05 -0600

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, REFER, NOTIFY

   Supported: timer, precondition, path, replaces

   Allow-Events: talk, hold, refer

   Content-Type: application/sdp

   Content-Disposition: session

   Content-Length: 280

   X-FS-Support: update_display

   Remote-Party-ID: "Extension 1001" 
<sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off



   v=0

   o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6

   s=-

   c=IN IP4 10.153.174.6

   t=0 0

   m=audio 49152 RTP/AVP 18 97 98

   a=rtpmap:18 G729/8000

   a=fmtp:18 annexb=no

   a=rtpmap:97 iLBC/8000

   a=rtpmap:98 telephone-event/8000

   a=fmtp:98 0-15

   a=ptime:20

   a=maxptime:40

   ------------------------------------------------------------------------

recv 570 bytes from udp/[74.63.41.218]:5060 at 21:15:59.468750:

   ------------------------------------------------------------------------

   SIP/2.0 407 Proxy Authentication Required

   Via: SIP/2.0/UDP 
69.125.20.15:5080;branch=z9hG4bKNp4ZFeKSD43tS;received=69.125.20.15;rport=5080

   From: "Extension 1001" 
<sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S

   To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843

   Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a

   CSeq: 7788615 INVITE

   User-Agent: VoIPMS/SERAST

   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

   Supported: replaces

   Proxy-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms", 
nonce="2d534dd6"

   Content-Length: 0



   ------------------------------------------------------------------------

send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:

   ------------------------------------------------------------------------

   ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0

   Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS

   Max-Forwards: 69

   From: "Extension 1001" 
<sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S

   To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843

   Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a

   CSeq: 7788615 ACK

   Content-Length: 0



   ------------------------------------------------------------------------

send 1344 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:

   ------------------------------------------------------------------------

   INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0

   Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN

   Max-Forwards: 69

   From: "Extension 1001" 
<sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S

   To: <sip:1MYPHONE at newyork.voip.ms>

   Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a

   CSeq: 7788616 INVITE

   Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>

   Expires: 300

   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 
18-04-05 -0600

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, REFER, NOTIFY

   Supported: timer, precondition, path, replaces

   Allow-Events: talk, hold, refer

   Proxy-Authorization: Digest username="121628", 
realm="newyork.voip.ms", nonce="2d534dd6", algorithm=MD5, 
uri="sip:1MYPHONE at newyork.voip.ms", 
response="16f3301efae13df926da7550f709d28a"

   Content-Type: application/sdp

   Content-Disposition: session

   Content-Length: 280

   X-FS-Support: update_display

   Remote-Party-ID: "Extension 1001" 
<sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off



   v=0

   o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6

   s=-

   c=IN IP4 10.153.174.6

   t=0 0

   m=audio 49152 RTP/AVP 18 97 98

   a=rtpmap:18 G729/8000

   a=fmtp:18 annexb=no

   a=rtpmap:97 iLBC/8000

   a=rtpmap:98 telephone-event/8000

   a=fmtp:98 0-15

   a=ptime:20

   a=maxptime:40

   ------------------------------------------------------------------------

recv 488 bytes from udp/[74.63.41.218]:5060 at 21:15:59.484375:

   ------------------------------------------------------------------------

   SIP/2.0 100 Trying

   Via: SIP/2.0/UDP 
69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080

   From: "Extension 1001" 
<sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S

   To: <sip:1MYPHONE at newyork.voip.ms>

   Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a

   CSeq: 7788616 INVITE

   User-Agent: VoIPMS/SERAST

   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

   Supported: replaces

   Contact: <sip:1MYPHONE at 74.63.41.218>

   Content-Length: 0



   ------------------------------------------------------------------------

recv 516 bytes from udp/[74.63.41.218]:5060 at 21:15:59.562500:

   ------------------------------------------------------------------------

   SIP/2.0 503 Service Unavailable

   Via: SIP/2.0/UDP 
69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080

   From: "Extension 1001" 
<sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S

   To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9

   Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a

   CSeq: 7788616 INVITE

   User-Agent: VoIPMS/SERAST

   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

   Supported: replaces

   Contact: <sip:1MYPHONE at 74.63.41.218>

   Content-Length: 0



   ------------------------------------------------------------------------

send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.562500:

   ------------------------------------------------------------------------

   ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0

   Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN

   Max-Forwards: 69

   From: "Extension 1001" 
<sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S

   To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9

   Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a

   CSeq: 7788616 ACK

   Content-Length: 0



   ------------------------------------------------------------------------

2011-01-28 16:15:59.562500 [INFO] mod_dptools.c:2612 Originate Failed. 
Cause: NO_ANSWER

2011-01-28 16:15:59.562500 [NOTICE] sofia.c:5286 Hangup 
sofia/external/1MYPHONE [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]

2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:189 
sofia/internal/1001 at 192.168.1.100 has executed the last dialplan 
instruction, hanging up.

2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:191 
Hangup sofia/internal/1001 at 192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING]

2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1306 Session 2 
(sofia/external/1MYPHONE) Ended

2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1308 Close 
Channel sofia/external/1MYPHONE [CS_DESTROY]

send 887 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.593750:

   ------------------------------------------------------------------------

   SIP/2.0 503 Service Unavailable

   Via: SIP/2.0/TCP 
10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180

   From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj

   To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg

   Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn

   CSeq: 5244504 INVITE

   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 
18-04-05 -0600

   Accept: application/sdp

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

   Supported: timer, precondition, path, replaces

   Allow-Events: talk, hold, presence, dialog, line-seize, call-info, 
sla, include-session-description, presence.winfo, message-summary, refer

   Reason: Q.850;cause=16;text="NORMAL_CLEARING"

2011-01-28 16:15:59.593750 [NOTICE] switch_core_session.c:1306 Session 1 
(sofia/internal/1001 at 192.168.1.100) Ended

   Content-Length: 02011-01-28 16:15:59.593750 [NOTICE] 
switch_core_session.c:1308 Close Channel 
sofia/internal/1001 at 192.168.1.100 [CS_DESTROY]



   Remote-Party-ID: "MYPHONE" 
<sip:MYPHONE at 192.168.1.100>;party=calling;privacy=off;screen=no



   ------------------------------------------------------------------------

recv 650 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.953125:

   ------------------------------------------------------------------------

   ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0

   Via: SIP/2.0/TCP 
10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport

   From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj

   To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg

   Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn

   CSeq: 5244504 ACK

   Supported: sec-agree

   Max-Forwards: 70

   Proxy-Authorization: Digest 
qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"

   Content-Length: 0



   ------------------------------------------------------------------------



Thank you in advance.

 		 	   		  

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From anthony.minessale at gmail.com  Mon Jan 31 01:34:09 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 30 Jan 2011 16:34:09 -0600
Subject: [Freeswitch-users] Outbound only calls don't connect when
 bypass_media is true.
In-Reply-To: <COL118-W38F26048D441DAA2777942E5E30@phx.gbl>
References: <COL118-W52E5226FC3337085AB6AFAE5E00@phx.gbl>
	<COL118-W38F26048D441DAA2777942E5E30@phx.gbl>
Message-ID: <AANLkTinbiU4zNc-ZM+u+iST7E2aZ3zisgh5Lu2FTuBLt@mail.gmail.com>

Just do not use bypass media.
That is all you can do in that situation.


On Sun, Jan 30, 2011 at 3:44 PM, Marcin Wojtowicz <marcin321 at hotmail.com> wrote:
> I just want to add that I enabled STUN on my cell so now the SDP message in
> the INVITE to voip.ms contains the public IP of my phone, but it still
> doesn't work.
>
> ________________________________
> From: marcin321 at hotmail.com
> To: freeswitch-users at lists.freeswitch.org
> Date: Fri, 28 Jan 2011 19:54:19 -0500
> Subject: [Freeswitch-users] Outbound only calls don't connect when
> bypass_media is true.
>
> Hello,
>
> I'm a new user of freeswitch, so please bear with me. I have the
> following setup:
> voip.ms <- SIP over UDP -> my desktop running freeswitch <- SIP over TCP ->
> my nokia cellphone on AT&T wireless. This setup is intended to conserve the
> battery usage.
> I've managed to make everything work well when I'm calling in over any phone
> to my cell phone, and freeswitch is enabled to work in bypass_media = true,
> even though by cell is behind NAT on at&t's network. Things break when I
> pick up my cell and try to call my home phone (or any phone for that
> matter). This is the relevant snippet from my dialplan:
> <extension name="outbound">
> ? <condition field="destination_number"
> expression="^1?([2-9]\d{2}[2-9]\d{6})$">
> ??? <!--<action application="set" data="bypass_media=true"/>-->
> ??? <action application="bridge" data="sofia/gateway/voip.ms/1$1"/>
> ? </condition>
> </extension>
>
> Like shown above, my call will go to my home phone. When I uncomment the
> bypass_media tag, my call will not connect. Here are the siptraces
> I replaced my real home phone number in the with "MYPHONE".
>
> recv 940 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.406250:
> ?? ------------------------------------------------------------------------
> ?? INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> ?? Via: SIP/2.0/TCP
> 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport
> ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> ?? To: <sip:MYPHONE at 192.168.1.100>
> ?? Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>
> ?? Supported: 100rel,timer
> ?? CSeq: 5244503 INVITE
> ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> ?? Allow: UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE
> ?? User-Agent: S60 RM-624 v 20.2.042 (en)
> ?? Expires: 120
> ?? Privacy: None
> ?? Session-Expires: 1800
> ?? Max-Forwards: 70
> ?? Content-Type: application/sdp
> ?? Accept-Language: en
> ?? Content-Length: 292
>
> ?? v=0
> ?? o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> ?? s=-
> ?? c=IN IP4 10.153.174.6
> ?? t=0 0
> ?? m=audio 49152 RTP/AVP 18 97 98
> ?? a=sendrecv
> ?? a=rtpmap:18 G729/8000
> ?? a=ptime:20
> ?? a=maxptime:40
> ?? a=fmtp:18 annexb=no
> ?? a=rtpmap:97 iLBC/8000
> ?? a=rtpmap:98 telephone-event/8000
> ?? a=fmtp:98 0-15
> ?? ------------------------------------------------------------------------
> send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:
> ?? ------------------------------------------------------------------------
> ?? SIP/2.0 100 Trying
> ?? Via: SIP/2.0/TCP
> 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180
> ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> ?? To: <sip:MYPHONE at 192.168.1.100>
> ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> ?? CSeq: 5244503 INVITE
> ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> -0600
> ?? Content-Length: 0
>
> ?? ------------------------------------------------------------------------
> send 875 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:
> ?? ------------------------------------------------------------------------
> ?? SIP/2.0 407 Proxy Authentication Required
> ?? Via: SIP/2.0/TCP
> 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180
> ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj2011-01-28
> 16:15:58.406250 [WARNING] sofia_reg.c:1241 SIP auth challenge (INVITE) on
> sofia profile 'internal' for [MYPHONE at 192.168.1.100] from ip 32.136.78.180
>
> ?? To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m
> ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> ?? CSeq: 5244503 INVITE
> ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> -0600
> ?? Accept: application/sdp
> ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> ?? Supported: timer, precondition, path, replaces
> ?? Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
> ?? Proxy-Authenticate: Digest realm="192.168.1.100",
> nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28", algorithm=MD5, qop="auth"
> ?? Content-Length: 0
>
> ?? ------------------------------------------------------------------------
> recv 368 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.765625:
> ?? ------------------------------------------------------------------------
> ?? ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> ?? Via: SIP/2.0/TCP
> 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport
> ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> ?? To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m
> ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> ?? CSeq: 5244503 ACK
> ?? Supported: sec-agree
> ?? Max-Forwards: 70
> ?? Content-Length: 0
>
> ?? ------------------------------------------------------------------------
> recv 1222 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.406250:
> ?? ------------------------------------------------------------------------
> ?? INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> ?? Via: SIP/2.0/TCP
> 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport
> ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> ?? To: <sip:MYPHONE at 192.168.1.100>
> ?? Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>
> ?? Supported: 100rel,timer
> ?? CSeq: 5244504 INVITE
> ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> ?? Allow: UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE
> ?? User-Agent: S60 RM-624 v 20.2.042 (en)
> ?? Expires: 120
> ?? Privacy: None
> ?? Session-Expires: 1800
> ?? Max-Forwards: 70
> ?? Proxy-Authorization: Digest
> qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"
> ?? Content-Type: application/sdp
> ?? Accept-Language: en
> ?? Content-Length: 292
>
> ?? v=0
> ?? o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> ?? s=-
> ?? c=IN IP4 10.153.174.6
> ?? t=0 0
> ?? m=audio 49152 RTP/AVP 18 97 98
> ?? a=sendrecv
> ?? a=rtpmap:18 G729/8000
> ?? a=ptime:20
> ?? a=maxptime:40
> ?? a=fmtp:18 annexb=no
> ?? a=rtpmap:97 iLBC/8000
> ?? a=rtpmap:98 telephone-event/8000
> ?? a=fmtp:98 0-15
> ?? ------------------------------------------------------------------------
> send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.406250:
> ?? ------------------------------------------------------------------------
> ?? SIP/2.0 100 Trying
> ?? Via: SIP/2.0/TCP
> 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180
> ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> ?? To: <sip:MYPHONE at 192.168.1.100>
> ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> ?? CSeq: 5244504 INVITE
> ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> -0600
> ?? Content-Length: 0
>
> ?? ------------------------------------------------------------------------
> 2011-01-28 16:15:59.406250 [NOTICE] switch_channel.c:808 New Channel
> sofia/internal/1001 at 192.168.1.100 [e5841001-04bd-4e16-9519-64ff2c7a8c2f]
> 2011-01-28 16:15:59.421875 [INFO] mod_dialplan_xml.c:331 Processing 1001
> <1001>->MYPHONE in context default
> 2011-01-28 16:15:59.437500 [NOTICE] switch_channel.c:808 New Channel
> sofia/external/1MYPHONE [60940227-9ae0-4c0c-abdb-f3988852fae0]
> send 1141 bytes to udp/[74.63.41.218]:5060 at 21:15:59.453125:
> ?? ------------------------------------------------------------------------
> ?? INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0
> ?? Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS
> ?? Max-Forwards: 69
> ?? From: "Extension 1001"
> <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> ?? To: <sip:1MYPHONE at newyork.voip.ms>
> ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> ?? CSeq: 7788615 INVITE
> ?? Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
> ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> -0600
> ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> ?? Supported: timer, precondition, path, replaces
> ?? Allow-Events: talk, hold, refer
> ?? Content-Type: application/sdp
> ?? Content-Disposition: session
> ?? Content-Length: 280
> ?? X-FS-Support: update_display
> ?? Remote-Party-ID: "Extension 1001"
> <sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off
>
> ?? v=0
> ?? o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> ?? s=-
> ?? c=IN IP4 10.153.174.6
> ?? t=0 0
> ?? m=audio 49152 RTP/AVP 18 97 98
> ?? a=rtpmap:18 G729/8000
> ?? a=fmtp:18 annexb=no
> ?? a=rtpmap:97 iLBC/8000
> ?? a=rtpmap:98 telephone-event/8000
> ?? a=fmtp:98 0-15
> ?? a=ptime:20
> ?? a=maxptime:40
> ?? ------------------------------------------------------------------------
> recv 570 bytes from udp/[74.63.41.218]:5060 at 21:15:59.468750:
> ?? ------------------------------------------------------------------------
> ?? SIP/2.0 407 Proxy Authentication Required
> ?? Via: SIP/2.0/UDP
> 69.125.20.15:5080;branch=z9hG4bKNp4ZFeKSD43tS;received=69.125.20.15;rport=5080
> ?? From: "Extension 1001"
> <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> ?? To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843
> ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> ?? CSeq: 7788615 INVITE
> ?? User-Agent: VoIPMS/SERAST
> ?? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> ?? Supported: replaces
> ?? Proxy-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms",
> nonce="2d534dd6"
> ?? Content-Length: 0
>
> ?? ------------------------------------------------------------------------
> send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:
> ?? ------------------------------------------------------------------------
> ?? ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0
> ?? Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS
> ?? Max-Forwards: 69
> ?? From: "Extension 1001"
> <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> ?? To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843
> ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> ?? CSeq: 7788615 ACK
> ?? Content-Length: 0
>
> ?? ------------------------------------------------------------------------
> send 1344 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:
> ?? ------------------------------------------------------------------------
> ?? INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0
> ?? Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN
> ?? Max-Forwards: 69
> ?? From: "Extension 1001"
> <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> ?? To: <sip:1MYPHONE at newyork.voip.ms>
> ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> ?? CSeq: 7788616 INVITE
> ?? Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
> ?? Expires: 300
> ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> -0600
> ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> ?? Supported: timer, precondition, path, replaces
> ?? Allow-Events: talk, hold, refer
> ?? Proxy-Authorization: Digest username="121628", realm="newyork.voip.ms",
> nonce="2d534dd6", algorithm=MD5, uri="sip:1MYPHONE at newyork.voip.ms",
> response="16f3301efae13df926da7550f709d28a"
> ?? Content-Type: application/sdp
> ?? Content-Disposition: session
> ?? Content-Length: 280
> ?? X-FS-Support: update_display
> ?? Remote-Party-ID: "Extension 1001"
> <sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off
>
> ?? v=0
> ?? o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> ?? s=-
> ?? c=IN IP4 10.153.174.6
> ?? t=0 0
> ?? m=audio 49152 RTP/AVP 18 97 98
> ?? a=rtpmap:18 G729/8000
> ?? a=fmtp:18 annexb=no
> ?? a=rtpmap:97 iLBC/8000
> ?? a=rtpmap:98 telephone-event/8000
> ?? a=fmtp:98 0-15
> ?? a=ptime:20
> ?? a=maxptime:40
> ?? ------------------------------------------------------------------------
> recv 488 bytes from udp/[74.63.41.218]:5060 at 21:15:59.484375:
> ?? ------------------------------------------------------------------------
> ?? SIP/2.0 100 Trying
> ?? Via: SIP/2.0/UDP
> 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080
> ?? From: "Extension 1001"
> <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> ?? To: <sip:1MYPHONE at newyork.voip.ms>
> ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> ?? CSeq: 7788616 INVITE
> ?? User-Agent: VoIPMS/SERAST
> ?? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> ?? Supported: replaces
> ?? Contact: <sip:1MYPHONE at 74.63.41.218>
> ?? Content-Length: 0
>
> ?? ------------------------------------------------------------------------
> recv 516 bytes from udp/[74.63.41.218]:5060 at 21:15:59.562500:
> ?? ------------------------------------------------------------------------
> ?? SIP/2.0 503 Service Unavailable
> ?? Via: SIP/2.0/UDP
> 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080
> ?? From: "Extension 1001"
> <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> ?? To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9
> ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> ?? CSeq: 7788616 INVITE
> ?? User-Agent: VoIPMS/SERAST
> ?? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> ?? Supported: replaces
> ?? Contact: <sip:1MYPHONE at 74.63.41.218>
> ?? Content-Length: 0
>
> ?? ------------------------------------------------------------------------
> send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.562500:
> ?? ------------------------------------------------------------------------
> ?? ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0
> ?? Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN
> ?? Max-Forwards: 69
> ?? From: "Extension 1001"
> <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> ?? To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9
> ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> ?? CSeq: 7788616 ACK
> ?? Content-Length: 0
>
> ?? ------------------------------------------------------------------------
> 2011-01-28 16:15:59.562500 [INFO] mod_dptools.c:2612 Originate Failed.
> Cause: NO_ANSWER
> 2011-01-28 16:15:59.562500 [NOTICE] sofia.c:5286 Hangup
> sofia/external/1MYPHONE [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
> 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:189
> sofia/internal/1001 at 192.168.1.100 has executed the last dialplan
> instruction, hanging up.
> 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:191 Hangup
> sofia/internal/1001 at 192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1306 Session 2
> (sofia/external/1MYPHONE) Ended
> 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1308 Close Channel
> sofia/external/1MYPHONE [CS_DESTROY]
> send 887 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.593750:
> ?? ------------------------------------------------------------------------
> ?? SIP/2.0 503 Service Unavailable
> ?? Via: SIP/2.0/TCP
> 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180
> ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> ?? To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg
> ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> ?? CSeq: 5244504 INVITE
> ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> -0600
> ?? Accept: application/sdp
> ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> ?? Supported: timer, precondition, path, replaces
> ?? Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
> ?? Reason: Q.850;cause=16;text="NORMAL_CLEARING"
> 2011-01-28 16:15:59.593750 [NOTICE] switch_core_session.c:1306 Session 1
> (sofia/internal/1001 at 192.168.1.100) Ended
> ?? Content-Length: 02011-01-28 16:15:59.593750 [NOTICE]
> switch_core_session.c:1308 Close Channel sofia/internal/1001 at 192.168.1.100
> [CS_DESTROY]
>
> ?? Remote-Party-ID: "MYPHONE"
> <sip:MYPHONE at 192.168.1.100>;party=calling;privacy=off;screen=no
>
> ?? ------------------------------------------------------------------------
> recv 650 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.953125:
> ?? ------------------------------------------------------------------------
> ?? ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> ?? Via: SIP/2.0/TCP
> 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport
> ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> ?? To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg
> ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> ?? CSeq: 5244504 ACK
> ?? Supported: sec-agree
> ?? Max-Forwards: 70
> ?? Proxy-Authorization: Digest
> qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"
> ?? Content-Length: 0
>
> ?? ------------------------------------------------------------------------
>
> Thank you in advance.
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From brokendash at gmail.com  Mon Jan 31 01:39:11 2011
From: brokendash at gmail.com (broken dash)
Date: Sun, 30 Jan 2011 16:39:11 -0600
Subject: [Freeswitch-users] Agent String/Version
	1.0.7-hacked-20110129T044740Z ??
Message-ID: <AANLkTimevt5AG=0r0bHnQG+Eg7+X7ynXnjn=Xo7PrwcE@mail.gmail.com>

I recently re-installed FS from src a few days ago using the 1.0.7
version from http://latest.freeswitch.org/....  Can anyone tell me if
these tarballs are snapshots or something? I assumed that latest was
the most current stable release and overlooked the file timestamps...
is some continuous automated build process dropping these out there?
It's just that "hacked" caught my attention making the ole hamster
wheel start turning... :-)  I've done svn and git pulls before and I
know that the compile time version strings reflected that it was a git
pull,snapshot,blah etc..  Surely I'm just being a paranoid n00b right?


Cheers,
B



Registrations:
=================================================================================================
Call-ID:    	acf2aec1-f3c2-4be2-ba05-469c817e4f54
User:       	100 at 75.XX.XX.XX
Contact:    	"user"
Agent:      	FreeSWITCH-mod_sofia/1.0.7-hacked-20110129T044740Z
Status:     	Registered(UDP-NAT)(unknown) EXP(2011-01-30 16:33:14)
Host:       	wormhole.home
IP:         	192.168.X.X
Port:       	5080
Auth-User:  	100
Auth-Realm: 	75.XX.XX.XX

=================================================================================================


From paul at cupis.co.uk  Mon Jan 31 02:13:10 2011
From: paul at cupis.co.uk (Paul Cupis)
Date: Sun, 30 Jan 2011 23:13:10 +0000
Subject: [Freeswitch-users] Agent
 String/Version	1.0.7-hacked-20110129T044740Z ??
In-Reply-To: <AANLkTimevt5AG=0r0bHnQG+Eg7+X7ynXnjn=Xo7PrwcE@mail.gmail.com>
References: <AANLkTimevt5AG=0r0bHnQG+Eg7+X7ynXnjn=Xo7PrwcE@mail.gmail.com>
Message-ID: <4D45F086.7090504@cupis.co.uk>

On 30/01/11 22:39, broken dash wrote:
> I recently re-installed FS from src a few days ago using the 1.0.7
> version from http://latest.freeswitch.org/....  Can anyone tell me if
> these tarballs are snapshots or something?

I believe 1.0.7 is a nightly snapshot - almost a release candidate. When
1.0.8 is ready it will be a final/standard release.



From brokendash at gmail.com  Mon Jan 31 03:15:48 2011
From: brokendash at gmail.com (broken dash)
Date: Sun, 30 Jan 2011 18:15:48 -0600
Subject: [Freeswitch-users] Agent String/Version
 1.0.7-hacked-20110129T044740Z ??
In-Reply-To: <4D45F086.7090504@cupis.co.uk>
References: <AANLkTimevt5AG=0r0bHnQG+Eg7+X7ynXnjn=Xo7PrwcE@mail.gmail.com>
	<4D45F086.7090504@cupis.co.uk>
Message-ID: <AANLkTikO=kiCypvgLqZGGMXsh=GDQdS3Fx+gZYx4YKKa@mail.gmail.com>

Thanks :-)   I'm just so damn paranoid that I've got some gaping hole
open to the outside world for some reason...

Cheers,
B

On Sun, Jan 30, 2011 at 5:13 PM, Paul Cupis <paul at cupis.co.uk> wrote:
> On 30/01/11 22:39, broken dash wrote:
>> I recently re-installed FS from src a few days ago using the 1.0.7
>> version from http://latest.freeswitch.org/.... ?Can anyone tell me if
>> these tarballs are snapshots or something?
>
> I believe 1.0.7 is a nightly snapshot - almost a release candidate. When
> 1.0.8 is ready it will be a final/standard release.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From bwibowo at gmail.com  Mon Jan 31 03:32:23 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Mon, 31 Jan 2011 07:32:23 +0700
Subject: [Freeswitch-users] dingalingissue
Message-ID: <AANLkTikbhmD4sB=tOtkYL4oy4Hw7HYf_DAYyYC7Wp7Hk@mail.gmail.com>

hi, anybody has "no voice" experience when call using mod dingaling? i try
from my gmail account and call still normal but from FS always no voice. it
was working perfectly before


thx

budi
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From Nabble at slickdeals.endjunk.com  Mon Jan 31 03:34:41 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 30 Jan 2011 16:34:41 -0800 (PST)
Subject: [Freeswitch-users] Agent String/Version
	1.0.7-hacked-20110129T044740Z ??
In-Reply-To: <AANLkTimevt5AG=0r0bHnQG+Eg7+X7ynXnjn=Xo7PrwcE@mail.gmail.com>
References: <AANLkTimevt5AG=0r0bHnQG+Eg7+X7ynXnjn=Xo7PrwcE@mail.gmail.com>
Message-ID: <1296434081870-5975831.post@n2.nabble.com>


http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-1-0-7-td5922408.html
Here  is the latest information on FS 1.0.7.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Agent-String-Version-1-0-7-hacked-20110129T044740Z-tp5975646p5975831.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveu at coppice.org  Mon Jan 31 03:45:25 2011
From: steveu at coppice.org (Steve Underwood)
Date: Mon, 31 Jan 2011 08:45:25 +0800
Subject: [Freeswitch-users] open g729
In-Reply-To: <AANLkTikng7_w0XcLb+B8AQ=3Xo1xsHDWkbmTz=8krWOt@mail.gmail.com>
References: <0610594E3D2241C2B0BDDEC28EDBF605@e1705>
	<AANLkTikng7_w0XcLb+B8AQ=3Xo1xsHDWkbmTz=8krWOt@mail.gmail.com>
Message-ID: <4D460625.4020502@coppice.org>

On 01/31/2011 04:32 AM, Steven Ayre wrote:
> Funny you mention it, but MP3 does have several patents covering 
> aspects of it that expire between 2007 (fine) and 2017 (not so fine). 
> There have been various companies at various times that have claimed 
> licenses are required to encode/decode it. Fraunhofer Institute being 
> a good example - they earned 100million euros in 2005 from mp3 
> licenses. In1998 they sent a lot of letters out to developers of 
> software using mp3 stating that they needed a licence.
>
> I guess the difference from G729 is that a) enough people in the 
> general public use it and want to continue using it and b) the patent 
> holders have declined to enforce license fees on free/foss software, 
> only focussing on commerical software, so there's plenty of stuff 
> around that can use it without worrying too much about the patents.
>
> http://en.wikipedia.org/wiki/MP3#Licensing_and_patent_issues
>
> -Steve
I think the main difference is the MP3 patent holders had to take a 
somewhat relaxed attitude to patent enforcement in the early day, to get 
MP3 usage to the point where people were fairly well locked in. Even 
then, Apple chose not to use MP3, so there was never the level of lock 
in they would have liked. This is lead to a somewhat relaxed attitude to 
patent enforcement for small offenders.

G.729 got a tight grip on IP phones from the very first IP phones. The 
market very quickly got to the point where the only codecs you could 
rely on two talk between two phones were G.711 and G.729. With that 
level of lock-in, the G.729 patent holders have had no reason to hold 
back, and lawyers letters go out to any minor offenders they can find.

Steve



From chris.chen2004 at gmail.com  Mon Jan 31 04:59:01 2011
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Sun, 30 Jan 2011 20:59:01 -0500
Subject: [Freeswitch-users] dingalingissue
In-Reply-To: <AANLkTikbhmD4sB=tOtkYL4oy4Hw7HYf_DAYyYC7Wp7Hk@mail.gmail.com>
References: <AANLkTikbhmD4sB=tOtkYL4oy4Hw7HYf_DAYyYC7Wp7Hk@mail.gmail.com>
Message-ID: <AANLkTinCDE+1655D+c4T9NwSzLBFot-FDREZ677OJzDt@mail.gmail.com>

I have the same issue as you since the official tarball release of 1.0.7,
the mod_dingaling.c establish the jingle session, however it loops between
line 2941 and 3275 until it times out without any RTP traffic.
2011-01-30 20:52:26.973945 [DEBUG] mod_dingaling.c:2941 using Existing
session for 3028119339
2011-01-30 20:52:26.973945 [DEBUG] mod_dingaling.c:3275 Already picked an IP
[99.xxx.xxx.xxx]
I tested with google talk to SIP, or gmail call out using mod_dingaling,
both have the same behavior.

It was working st least up to Jan 11 2011. I had this issue since official
release of tarball 1.0.7 and every latest GIT version I tested up to today.

Best regards,

Chris Chen


On Sun, Jan 30, 2011 at 7:32 PM, budi wibowo <bwibowo at gmail.com> wrote:

> hi, anybody has "no voice" experience when call using mod dingaling? i try
> from my gmail account and call still normal but from FS always no voice. it
> was working perfectly before
>
>
> thx
>
> budi
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fvillarroel at yahoo.com  Mon Jan 31 05:04:12 2011
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Sun, 30 Jan 2011 18:04:12 -0800 (PST)
Subject: [Freeswitch-users] Accountcode
Message-ID: <923586.77791.qm@web34304.mail.mud.yahoo.com>

Hi All.

If i received traffic from a customer and i forward this traffic to a provider, How i can know the traffic inbound and outbound (customer and provider)?

Like accountcode of Asterisk.

Regards.


      


From curriegrad2004 at gmail.com  Mon Jan 31 05:09:59 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 30 Jan 2011 18:09:59 -0800
Subject: [Freeswitch-users] Accountcode
In-Reply-To: <923586.77791.qm@web34304.mail.mud.yahoo.com>
References: <923586.77791.qm@web34304.mail.mud.yahoo.com>
Message-ID: <AANLkTi=gAdAthcTNN8xkHk=ubQqrMDQ2vBj+FbTpTXoR@mail.gmail.com>

Actually, I think FS does have account code. That should be defined in
the user directory page itself, I'm not too sure myself, but I won't
be surprised if there was account code in the directory page.

On Sun, Jan 30, 2011 at 6:04 PM, FERNANDO VILLARROEL
<fvillarroel at yahoo.com> wrote:
> Hi All.
>
> If i received traffic from a customer and i forward this traffic to a provider, How i can know the traffic inbound and outbound (customer and provider)?
>
> Like accountcode of Asterisk.
>
> Regards.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From djbinter at gmail.com  Mon Jan 31 05:54:34 2011
From: djbinter at gmail.com (DJB International)
Date: Sun, 30 Jan 2011 18:54:34 -0800
Subject: [Freeswitch-users] Accountcode
In-Reply-To: <AANLkTi=gAdAthcTNN8xkHk=ubQqrMDQ2vBj+FbTpTXoR@mail.gmail.com>
References: <923586.77791.qm@web34304.mail.mud.yahoo.com>
	<AANLkTi=gAdAthcTNN8xkHk=ubQqrMDQ2vBj+FbTpTXoR@mail.gmail.com>
Message-ID: <AANLkTikGgwORvUNxsbK8d4fYB9XNWWYU6MKV0C=RBVsG@mail.gmail.com>

There are many ways to implement this.  The easiest way would be to add
something like:
<action application="set" data="accountcode=XXX"/>

or, you can set your own variable like:
<action application="set" data="src_gw_name=XXX"/>

-djbinter


On Sun, Jan 30, 2011 at 6:09 PM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:

> Actually, I think FS does have account code. That should be defined in
> the user directory page itself, I'm not too sure myself, but I won't
> be surprised if there was account code in the directory page.
>
> On Sun, Jan 30, 2011 at 6:04 PM, FERNANDO VILLARROEL
> <fvillarroel at yahoo.com> wrote:
> > Hi All.
> >
> > If i received traffic from a customer and i forward this traffic to a
> provider, How i can know the traffic inbound and outbound (customer and
> provider)?
> >
> > Like accountcode of Asterisk.
> >
> > Regards.
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lloyd.aloysius at gmail.com  Mon Jan 31 06:00:39 2011
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 30 Jan 2011 22:00:39 -0500
Subject: [Freeswitch-users] Accountcode
In-Reply-To: <AANLkTi=gAdAthcTNN8xkHk=ubQqrMDQ2vBj+FbTpTXoR@mail.gmail.com>
References: <923586.77791.qm@web34304.mail.mud.yahoo.com>
	<AANLkTi=gAdAthcTNN8xkHk=ubQqrMDQ2vBj+FbTpTXoR@mail.gmail.com>
Message-ID: <AANLkTik0y4xnei1T=sdr2Dp1LgRV67MtgL5DqfFWLgKK@mail.gmail.com>

You can define the account code in user directory like below

<variable name="accountcode" value="930145"/>

Also in xml_cdr you have the full information. including inbound and
outbound.


<cdr>
  <channel_data>
    <state>CS_REPORTING</state>
    <direction>inbound</direction>


http://wiki.freeswitch.org/wiki/Mod_xml_cdr

Thanks
Lloyd

On Sun, Jan 30, 2011 at 9:09 PM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:

> Actually, I think FS does have account code. That should be defined in
> the user directory page itself, I'm not too sure myself, but I won't
> be surprised if there was account code in the directory page.
>
> On Sun, Jan 30, 2011 at 6:04 PM, FERNANDO VILLARROEL
> <fvillarroel at yahoo.com> wrote:
> > Hi All.
> >
> > If i received traffic from a customer and i forward this traffic to a
> provider, How i can know the traffic inbound and outbound (customer and
> provider)?
> >
> > Like accountcode of Asterisk.
> >
> > Regards.
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Nabble at slickdeals.endjunk.com  Mon Jan 31 06:26:10 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 30 Jan 2011 19:26:10 -0800 (PST)
Subject: [Freeswitch-users] dingalingissue
In-Reply-To: <AANLkTinCDE+1655D+c4T9NwSzLBFot-FDREZ677OJzDt@mail.gmail.com>
References: <AANLkTikbhmD4sB=tOtkYL4oy4Hw7HYf_DAYyYC7Wp7Hk@mail.gmail.com>
	<AANLkTinCDE+1655D+c4T9NwSzLBFot-FDREZ677OJzDt@mail.gmail.com>
Message-ID: <1296444370502-5976081.post@n2.nabble.com>



Chris Chen-4 wrote:
> 
> I have the same issue as you since the official tarball release of 1.0.7,
> the mod_dingaling.c establish the jingle session, however it loops between
> line 2941 and 3275 until it times out without any RTP traffic.
> 2011-01-30 20:52:26.973945 [DEBUG] mod_dingaling.c:2941 using Existing
> session for 3028119339
> 2011-01-30 20:52:26.973945 [DEBUG] mod_dingaling.c:3275 Already picked an
> IP
> [99.xxx.xxx.xxx]
> I tested with google talk to SIP, or gmail call out using mod_dingaling,
> both have the same behavior.
> 
> It was working st least up to Jan 11 2011. I had this issue since official
> release of tarball 1.0.7 and every latest GIT version I tested up to
> today.
> 
> Best regards,
> 
> Chris Chen
I can confirm what you said above. However, I have updated/recompiled my
local FS git repository yesterday and now it looks like mod_dingaling is
back working again. Mine is from a FreeSWITCH Version 1.0.head (git-49a5eff
2011-01-29 03-09-06 -0500) on a Seagate DockStar. The only problem my FS is
facing is no audio in both ways if the caller places a call to my GV DID#
from the Google Chat inside GMail.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/dingalingissue-tp5975836p5976081.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Mon Jan 31 06:33:35 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 30 Jan 2011 19:33:35 -0800 (PST)
Subject: [Freeswitch-users] Compilation with -march options
In-Reply-To: <AANLkTi=civOgvpz55hxEifdkcmKWmZABsitj290cCkxN@mail.gmail.com>
References: <AANLkTi=civOgvpz55hxEifdkcmKWmZABsitj290cCkxN@mail.gmail.com>
Message-ID: <1296444815434-5976090.post@n2.nabble.com>



curriegrad2004 wrote:
> ..., but I was
> wondering if anybody else out there is also compiling FS with a march
> option and saw some performance increase or adverse effects...
I use -march=armv5te switch to cross-compile FS for an ARM platform sans any
problem.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Compilation-with-march-options-tp5975263p5976090.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From u2nsam at gmail.com  Mon Jan 31 07:00:04 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 31 Jan 2011 09:30:04 +0530
Subject: [Freeswitch-users] blocking 183 w/o sdp
In-Reply-To: <AANLkTinyV-kAWCOiuOVL5jmnL7f9ck00=i_giHY_XGd_@mail.gmail.com>
References: <AANLkTinH1s+Uy-EPvX9oV-x+aXYkyzUEBSYm+hQ+-+K-@mail.gmail.com>
	<AANLkTikJpEPvcQu2reAKXk1wbW7fV4myvXghBCuu2QHt@mail.gmail.com>
	<AANLkTikuQFC2N3CYzreTvCdk8-28MYGUbrP=T83SfPtm@mail.gmail.com>
	<AANLkTinyV-kAWCOiuOVL5jmnL7f9ck00=i_giHY_XGd_@mail.gmail.com>
Message-ID: <AANLkTik6dUTrSLY+FYCBWQ2gsNQViCK4GW2-UnQPYmE=@mail.gmail.com>

Hi,

After using , <action application="bridge"
data="{sip_contact_user=${contact},sip_ignore_183nosdp=true}sofia/sbc/$1@
${dialed_domain}"/>

the 183 without udp is not blocked/ignored .

Below are the traces to visualize:
192.168.2.98 is provider
192.168.2.16 is FS


U 192.168.2.98:5060 -> 192.168.2.16:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 192.168.2.16;rport;branch=z9hG4bKjQBQg7vy0y2SF.
To: <sip:599261244747199 at 192.168.2.16 <sip%3A599261244747199 at 192.168.2.16>
>;tag=3505434022-138257.
From: "0280910101" <sip:0280910101 at 192.168.2.16<sip%3A0280910101 at 192.168.2.16>
>;tag=51SjQQQUX14QF.
Call-ID: 6ade3e61-a78e-122e-9698-00137256e1a2.
CSeq: 7886492 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
SUBSCRIBE, PRACK, UPDATE.
Contact: <sip:599261244747199 at 192.168.2.98:5060>.
Call-Info:
<sip:192.168.2.98>;method="NOTIFY;Event=telephone-event;Duration=1000".
Content-Length: 0.
.


U 192.168.2.16:5060 -> 192.168.2.6:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.2.6;branch=z9hG4bK3fb.51f5e6e1.0.
Via: SIP/2.0/UDP 192.168.2.158:5060
;received=192.168.2.158;rport=5060;branch=z9hG4bK-f33ff5a-631d780c-50647c36.
Record-Route:
<sip:192.168.2.6;lr=on;ftag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a>.
From: "0280910101" <sip:0280910101 at 192.168.2.158<sip%3A0280910101 at 192.168.2.158>
>;tag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a.
To: <sip:61244747199 at 192.168.2.6 <sip%3A61244747199 at 192.168.2.6>
>;tag=3F70K1Nm3Frjr.
Call-ID: 100f6c88-9e3599cb-13c4-50029-f33ff5a-631c036b-f33ff5a.
CSeq: 1 INVITE.
Contact: <sip:61244747199 at 192.168.2.16:5060;transport=udp>.
User-Agent:  SBC.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, refer.
Content-Length: 0.
Remote-Party-ID: "599261244747199"
<sip:599261244747199 at 192.168.2.16<sip%3A599261244747199 at 192.168.2.16>
>;party=calling;privacy=off;screen=no.
.


U 192.168.2.98:5060 -> 192.168.2.16:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.2.16;rport;branch=z9hG4bKjQBQg7vy0y2SF.
To: <sip:599261244747199 at 192.168.2.98 <sip%3A599261244747199 at 192.168.2.98>
>;tag=3505434022-138257.
From: "0280910101" <sip:0280910101 at 192.168.2.16<sip%3A0280910101 at 192.168.2.16>
>;tag=51SjQQQUX14QF.
Call-ID: 6ade3e61-a78e-122e-9698-00137256e1a2.
CSeq: 7886492 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
SUBSCRIBE, PRACK, UPDATE.
Contact: <sip:599261244747199 at 192.168.2.98:5060>.
Call-Info:
<sip:192.168.2.98>;method="NOTIFY;Event=telephone-event;Duration=1000".
Content-Type: application/sdp.
Content-Length: 209.
.
v=0.
o=vsnl2 770 13521 IN IP4 192.168.2.98.
s=sip call.
c=IN IP4 115.113.121.99.
t=0 0.
m=audio 49034 RTP/AVP 18 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
a=ptime:20.
a=rtpmap:18 G729/8000/1.


U 192.168.2.16:5060 -> 192.168.2.6:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 192.168.2.6;branch=z9hG4bK3fb.51f5e6e1.0.
Via: SIP/2.0/UDP 192.168.2.158:5060
;received=192.168.2.158;rport=5060;branch=z9hG4bK-f33ff5a-631d780c-50647c36.
Record-Route:
<sip:192.168.2.6;lr=on;ftag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a>.
From: "0280910101" <sip:0280910101 at 192.168.2.158<sip%3A0280910101 at 192.168.2.158>
>;tag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a.
To: <sip:61244747199 at 192.168.2.6 <sip%3A61244747199 at 192.168.2.6>
>;tag=3F70K1Nm3Frjr.
Call-ID: 100f6c88-9e3599cb-13c4-50029-f33ff5a-631c036b-f33ff5a.
CSeq: 1 INVITE.
Contact: <sip:61244747199 at 192.168.2.16:5060;transport=udp>.
User-Agent:  SBC.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 212.
Remote-Party-ID: "599261244747199"
<sip:599261244747199 at 192.168.2.16<sip%3A599261244747199 at 192.168.2.16>
>;party=calling;privacy=off;screen=no.
.
v=0.
o=SBC 1019267468 1019267469 IN IP4 192.168.2.16.
s=SBC.
c=IN IP4 192.168.2.16.
t=0 0.
m=audio 16922 RTP/AVP 18 101.
a=rtpmap:18 G729/8000/1.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
a=ptime:20.




Regds
Sam





On Sat, Jan 29, 2011 at 3:16 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Close. You can only have one set of {} brackets. You can separate multiple
> variables with a comma.
>
>
> <action application="bridge"
> data="{sip_contact_user=${contact},sip_ignore_183nosdp=true}sofia/sbc/$1@
> ${dialed_domain}"/>
>
> -Steve
>
>
>
> On 29 January 2011 04:29, Sam <u2nsam at gmail.com> wrote:
>
>> Hi,
>>
>> So you say i need to put
>> <action application="bridge"
>> data="{sip_contact_user=${contact}}{sip_ignore_183nosdp=true}sofia/sbc/$1@
>> ${dialed_domain}"/>
>>
>> Regds
>> Sam
>>
>>
>>
>>
>>
>> On Fri, Jan 28, 2011 at 10:23 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> you need sip_ignore_183nosdp=true set on the b leg not the a leg.
>>> Put it in the dial string in {}
>>>
>>> {sip_ignore_183nosdp=true}sofia/foo/foo at bar.com
>>>
>>>
>>> On Fri, Jan 28, 2011 at 12:41 AM, Sam <u2nsam at gmail.com> wrote:
>>> > Hi,
>>> >
>>> > how can i ignore 183 without sdp,
>>> > what happens is the provider sends 183 without sdp and by applying
>>> "<action
>>> > application="set" data="sip_ignore_183nosdp=true"/>"  the FS sends 180
>>> to
>>> > the leg a.
>>> > Here i want to block the 183 with SDP just like router as b2bua and
>>> send
>>> > nothing to leg a, and when actual 183 with sdp comes it should send .
>>> >
>>> > Its because, providers are sending false signaling by sending 183
>>> without
>>> > sdp,and it hampers while @ production,
>>> > Although by cisco sbc i have done this but i want to do it by FS,
>>> > Take a scenario, when call is send 183 without sdp for 10 secs and then
>>> > followed by 183 with sdp ( actual signal),
>>> > but when some one dials invalid number it rings for 10 secs and then
>>> gives
>>> > SIP cause 404, which is bad from the providers.
>>> > So this is the reason i want to block it.
>>> >
>>> > Most of the providers do this, the way out is blocking.
>>> >
>>> > I have got an advice from Tihomir  to do "execute_on_ring and parse
>>> your 180
>>> > / 183 messages in search of SDP,
>>> > once you get 183 without SDP do not send it back to leg a and send
>>> signal
>>> > only when you got 183 with sdp or 180 "
>>> > And this could be valid call flow.
>>> >
>>> > This happens in many cases where the provider is having nextone as a
>>> sbc and
>>> > that too tier 1 !
>>> >
>>> > Regards
>>> > Sam
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Mon Jan 31 07:06:31 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 31 Jan 2011 09:36:31 +0530
Subject: [Freeswitch-users] deflect
In-Reply-To: <AANLkTi=OnWrFDS_JbijJnpQ+_EmPm6r9=yM_aYVNjLVA@mail.gmail.com>
References: <AANLkTiksYLOeki2_XDJC5YwMW6hjUxvdpT7-Mvg7vmKZ@mail.gmail.com>
	<AANLkTi=EMn55D1xGBrWuGgVE_dW_5itoJ4WpzByyagTE@mail.gmail.com>
	<AANLkTi=F5Lhsk1tjcG5z5LTUjZb4e-cQdNXQjK3OsVD7@mail.gmail.com>
	<AANLkTinV=xZ_ahs=8DH7n8hADGL4hTOR05hNCWCasA85@mail.gmail.com>
	<AANLkTimfCVsJFCVDv1Az-8ERh8UZRqx1cKG1hY+GfKwd@mail.gmail.com>
	<AANLkTi=OnWrFDS_JbijJnpQ+_EmPm6r9=yM_aYVNjLVA@mail.gmail.com>
Message-ID: <AANLkTik-vBpwkikio_2HqkE1hiBxCuwKUarOOOT1C+Ak@mail.gmail.com>

Hi,

Any method to just blind transfer with media to the proxy ahead so that call
remains connected to 12127773456 ?
or does it requires only b2bua in the next transferred hop ?

Regards
Sam



On Thu, Jan 27, 2011 at 1:10 PM, Sam <u2nsam at gmail.com> wrote:

> Hi Michael,
>
> Here is it. http://pastebin.freeswitch.org/15156
>
> Regds
> Sam
>
>
> On Wed, Jan 26, 2011 at 12:23 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> I strongly recommend that you capture the debug output and drop it into a
>> pastebin at pastebin.freeswitch.org. You may also wish to capture the sip
>> traffic as well. If you are using fs_cli then you already see the debug
>> level console output. To get the sip traffic inline with the debug output
>> just do "sofia global siptrace on".
>>
>> -MC
>>
>>
>> On Tue, Jan 25, 2011 at 8:24 AM, Sam <u2nsam at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> Is it possible in this scenario,
>>>
>>> I have a call (leg a) to an IVR on FS1 , after the ivr the below
>>> statement is executed,
>>> <action application="bridge" data="{api_on_answer='uuid_transfer  ${uuid}
>>> both'}sofia/external/12127773456 at 192.168.2.130 <9999 at 192.168.2.130>"/>
>>>
>>> As the FS1 sends invite to 192.168.2.130 and the call is connected to the
>>> moviephone IVR,
>>> but here what happens is the call is getting disconnected from leg a and
>>> the movie phone ivr 12127773456.
>>>
>>>
>>>
>>> Regds
>>>
>>> Sam
>>>
>>>
>>>
>>> On Mon, Jan 24, 2011 at 3:12 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>>
>>>> You could try uuid_simplify with the api_on_answer hook
>>>>
>>>> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_simplify
>>>> http://wiki.freeswitch.org/wiki/Variable_api_on_answer
>>>>
>>>> -Steve
>>>>
>>>>
>>>>
>>>> On 24 January 2011 09:05, Sam <u2nsam at gmail.com> wrote:
>>>>
>>>>>  Hi,
>>>>>
>>>>> Is it possible by having b2bua in between , would the leg A be
>>>>> deflected to the another FS server from first server ?
>>>>>
>>>>> Regds
>>>>> Sam
>>>>>
>>>>>
>>>>> On Wed, Jan 12, 2011 at 11:42 AM, Sam <u2nsam at gmail.com> wrote:
>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> When call comes on 1 server and plays an application and after
>>>>>> execution of the
>>>>>> application the call is bridge to the other server ,but here after
>>>>>> bridging the call
>>>>>> should refer/deflect to other server, how this can be done ?
>>>>>>
>>>>>> Here just using the deflect variable is not recommended as there is
>>>>>> proxy in between,
>>>>>> so once the call is bridge the next step would be deflect the leg
>>>>>> totally to another server via proxy.
>>>>>>
>>>>>> Regards
>>>>>> Sam
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From chat2jesse at gmail.com  Mon Jan 31 09:00:23 2011
From: chat2jesse at gmail.com (jesse)
Date: Sun, 30 Jan 2011 22:00:23 -0800
Subject: [Freeswitch-users] keeps receiving 407
Message-ID: <AANLkTinWqR0PwrhjScdaBzXb4UoH5CmfzzVxwFrQWk5B@mail.gmail.com>

I set up a sipp uac client. here is the call flow messages:
http://freeswitch.pastebin.com/Sx5Z4egM
why does FS send 407 after 200 OK?

thanks!
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From kapil.rastogi at telemune.net  Mon Jan 31 09:57:43 2011
From: kapil.rastogi at telemune.net (kapil.rastogi)
Date: Sun, 30 Jan 2011 22:57:43 -0800 (PST)
Subject: [Freeswitch-users] How to play Background music during conference?
Message-ID: <1296457063838-5976296.post@n2.nabble.com>


Dear Friends,

I am using javascript as a scripting language in freeSWITCH.

I want to play the background music during conference without mute any
member in the conference room.

I tried "perpetual-sound" but it is not working. Can anyone help me about
this? If yes, then please also send me the example code to play background
music.

-----
Regards,
Kapil Rastogi
Telemune Software Solutions P Ltd.
kapil.rastogi at telemune.net
+919013204760
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-play-Background-music-during-conference-tp5976296p5976296.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kapil.rastogi at telemune.net  Mon Jan 31 10:02:28 2011
From: kapil.rastogi at telemune.net (kapil.rastogi)
Date: Sun, 30 Jan 2011 23:02:28 -0800 (PST)
Subject: [Freeswitch-users] How to play wav files from other path?
Message-ID: <1296457348066-5976304.post@n2.nabble.com>


Hi,

I want to play wav files from the different path. As now it is playing the
wav files from:
"/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/"
path.

Please tell me how to play wav files from other path using javascript
application.

Thanks in advance....

-----
Regards,
Kapil Rastogi
Telemune Software Solutions P Ltd.
kapil.rastogi at telemune.net
+919013204760
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-play-wav-files-from-other-path-tp5976304p5976304.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From marcin321 at hotmail.com  Mon Jan 31 07:40:12 2011
From: marcin321 at hotmail.com (Marcin Wojtowicz)
Date: Sun, 30 Jan 2011 23:40:12 -0500
Subject: [Freeswitch-users] Outbound only calls don't connect when
 bypass_media is true.
In-Reply-To: <AANLkTinbiU4zNc-ZM+u+iST7E2aZ3zisgh5Lu2FTuBLt@mail.gmail.com>
References: <COL118-W52E5226FC3337085AB6AFAE5E00@phx.gbl>,
	<COL118-W38F26048D441DAA2777942E5E30@phx.gbl>,
	<AANLkTinbiU4zNc-ZM+u+iST7E2aZ3zisgh5Lu2FTuBLt@mail.gmail.com>
Message-ID: <COL118-W196844C55E71088CAD1FC1E5E20@phx.gbl>


OK, so I gave up on bypass media, but now I have another problem. This time I set up freeswitch to communicate with voip.ms using PCMU codec (configured in my external profile), and use iLBC on my phone (codec configured in my internal profile, where the phone registers). When I call my mobile it rings, but when I pick up all I hear is a high pitched squeal. Am I missing something here?

> Date: Sun, 30 Jan 2011 16:34:09 -0600
> From: anthony.minessale at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Outbound only calls don't connect when bypass_media is true.
> 
> Just do not use bypass media.
> That is all you can do in that situation.
> 
> 
> On Sun, Jan 30, 2011 at 3:44 PM, Marcin Wojtowicz <marcin321 at hotmail.com> wrote:
> > I just want to add that I enabled STUN on my cell so now the SDP message in
> > the INVITE to voip.ms contains the public IP of my phone, but it still
> > doesn't work.
> >
> > ________________________________
> > From: marcin321 at hotmail.com
> > To: freeswitch-users at lists.freeswitch.org
> > Date: Fri, 28 Jan 2011 19:54:19 -0500
> > Subject: [Freeswitch-users] Outbound only calls don't connect when
> > bypass_media is true.
> >
> > Hello,
> >
> > I'm a new user of freeswitch, so please bear with me. I have the
> > following setup:
> > voip.ms <- SIP over UDP -> my desktop running freeswitch <- SIP over TCP ->
> > my nokia cellphone on AT&T wireless. This setup is intended to conserve the
> > battery usage.
> > I've managed to make everything work well when I'm calling in over any phone
> > to my cell phone, and freeswitch is enabled to work in bypass_media = true,
> > even though by cell is behind NAT on at&t's network. Things break when I
> > pick up my cell and try to call my home phone (or any phone for that
> > matter). This is the relevant snippet from my dialplan:
> > <extension name="outbound">
> >   <condition field="destination_number"
> > expression="^1?([2-9]\d{2}[2-9]\d{6})$">
> >     <!--<action application="set" data="bypass_media=true"/>-->
> >     <action application="bridge" data="sofia/gateway/voip.ms/1$1"/>
> >   </condition>
> > </extension>
> >
> > Like shown above, my call will go to my home phone. When I uncomment the
> > bypass_media tag, my call will not connect. Here are the siptraces
> > I replaced my real home phone number in the with "MYPHONE".
> >
> > recv 940 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.406250:
> >    ------------------------------------------------------------------------
> >    INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> >    Via: SIP/2.0/TCP
> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport
> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >    To: <sip:MYPHONE at 192.168.1.100>
> >    Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>
> >    Supported: 100rel,timer
> >    CSeq: 5244503 INVITE
> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >    Allow: UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE
> >    User-Agent: S60 RM-624 v 20.2.042 (en)
> >    Expires: 120
> >    Privacy: None
> >    Session-Expires: 1800
> >    Max-Forwards: 70
> >    Content-Type: application/sdp
> >    Accept-Language: en
> >    Content-Length: 292
> >
> >    v=0
> >    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> >    s=-
> >    c=IN IP4 10.153.174.6
> >    t=0 0
> >    m=audio 49152 RTP/AVP 18 97 98
> >    a=sendrecv
> >    a=rtpmap:18 G729/8000
> >    a=ptime:20
> >    a=maxptime:40
> >    a=fmtp:18 annexb=no
> >    a=rtpmap:97 iLBC/8000
> >    a=rtpmap:98 telephone-event/8000
> >    a=fmtp:98 0-15
> >    ------------------------------------------------------------------------
> > send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:
> >    ------------------------------------------------------------------------
> >    SIP/2.0 100 Trying
> >    Via: SIP/2.0/TCP
> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180
> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >    To: <sip:MYPHONE at 192.168.1.100>
> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >    CSeq: 5244503 INVITE
> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> > -0600
> >    Content-Length: 0
> >
> >    ------------------------------------------------------------------------
> > send 875 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:
> >    ------------------------------------------------------------------------
> >    SIP/2.0 407 Proxy Authentication Required
> >    Via: SIP/2.0/TCP
> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180
> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj2011-01-28
> > 16:15:58.406250 [WARNING] sofia_reg.c:1241 SIP auth challenge (INVITE) on
> > sofia profile 'internal' for [MYPHONE at 192.168.1.100] from ip 32.136.78.180
> >
> >    To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m
> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >    CSeq: 5244503 INVITE
> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> > -0600
> >    Accept: application/sdp
> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >    Supported: timer, precondition, path, replaces
> >    Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
> > include-session-description, presence.winfo, message-summary, refer
> >    Proxy-Authenticate: Digest realm="192.168.1.100",
> > nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28", algorithm=MD5, qop="auth"
> >    Content-Length: 0
> >
> >    ------------------------------------------------------------------------
> > recv 368 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.765625:
> >    ------------------------------------------------------------------------
> >    ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> >    Via: SIP/2.0/TCP
> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport
> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >    To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m
> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >    CSeq: 5244503 ACK
> >    Supported: sec-agree
> >    Max-Forwards: 70
> >    Content-Length: 0
> >
> >    ------------------------------------------------------------------------
> > recv 1222 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.406250:
> >    ------------------------------------------------------------------------
> >    INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> >    Via: SIP/2.0/TCP
> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport
> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >    To: <sip:MYPHONE at 192.168.1.100>
> >    Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>
> >    Supported: 100rel,timer
> >    CSeq: 5244504 INVITE
> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >    Allow: UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE
> >    User-Agent: S60 RM-624 v 20.2.042 (en)
> >    Expires: 120
> >    Privacy: None
> >    Session-Expires: 1800
> >    Max-Forwards: 70
> >    Proxy-Authorization: Digest
> > qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"
> >    Content-Type: application/sdp
> >    Accept-Language: en
> >    Content-Length: 292
> >
> >    v=0
> >    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> >    s=-
> >    c=IN IP4 10.153.174.6
> >    t=0 0
> >    m=audio 49152 RTP/AVP 18 97 98
> >    a=sendrecv
> >    a=rtpmap:18 G729/8000
> >    a=ptime:20
> >    a=maxptime:40
> >    a=fmtp:18 annexb=no
> >    a=rtpmap:97 iLBC/8000
> >    a=rtpmap:98 telephone-event/8000
> >    a=fmtp:98 0-15
> >    ------------------------------------------------------------------------
> > send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.406250:
> >    ------------------------------------------------------------------------
> >    SIP/2.0 100 Trying
> >    Via: SIP/2.0/TCP
> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180
> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >    To: <sip:MYPHONE at 192.168.1.100>
> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >    CSeq: 5244504 INVITE
> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> > -0600
> >    Content-Length: 0
> >
> >    ------------------------------------------------------------------------
> > 2011-01-28 16:15:59.406250 [NOTICE] switch_channel.c:808 New Channel
> > sofia/internal/1001 at 192.168.1.100 [e5841001-04bd-4e16-9519-64ff2c7a8c2f]
> > 2011-01-28 16:15:59.421875 [INFO] mod_dialplan_xml.c:331 Processing 1001
> > <1001>->MYPHONE in context default
> > 2011-01-28 16:15:59.437500 [NOTICE] switch_channel.c:808 New Channel
> > sofia/external/1MYPHONE [60940227-9ae0-4c0c-abdb-f3988852fae0]
> > send 1141 bytes to udp/[74.63.41.218]:5060 at 21:15:59.453125:
> >    ------------------------------------------------------------------------
> >    INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0
> >    Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS
> >    Max-Forwards: 69
> >    From: "Extension 1001"
> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >    To: <sip:1MYPHONE at newyork.voip.ms>
> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >    CSeq: 7788615 INVITE
> >    Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> > -0600
> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> > REGISTER, REFER, NOTIFY
> >    Supported: timer, precondition, path, replaces
> >    Allow-Events: talk, hold, refer
> >    Content-Type: application/sdp
> >    Content-Disposition: session
> >    Content-Length: 280
> >    X-FS-Support: update_display
> >    Remote-Party-ID: "Extension 1001"
> > <sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off
> >
> >    v=0
> >    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> >    s=-
> >    c=IN IP4 10.153.174.6
> >    t=0 0
> >    m=audio 49152 RTP/AVP 18 97 98
> >    a=rtpmap:18 G729/8000
> >    a=fmtp:18 annexb=no
> >    a=rtpmap:97 iLBC/8000
> >    a=rtpmap:98 telephone-event/8000
> >    a=fmtp:98 0-15
> >    a=ptime:20
> >    a=maxptime:40
> >    ------------------------------------------------------------------------
> > recv 570 bytes from udp/[74.63.41.218]:5060 at 21:15:59.468750:
> >    ------------------------------------------------------------------------
> >    SIP/2.0 407 Proxy Authentication Required
> >    Via: SIP/2.0/UDP
> > 69.125.20.15:5080;branch=z9hG4bKNp4ZFeKSD43tS;received=69.125.20.15;rport=5080
> >    From: "Extension 1001"
> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843
> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >    CSeq: 7788615 INVITE
> >    User-Agent: VoIPMS/SERAST
> >    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> >    Supported: replaces
> >    Proxy-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms",
> > nonce="2d534dd6"
> >    Content-Length: 0
> >
> >    ------------------------------------------------------------------------
> > send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:
> >    ------------------------------------------------------------------------
> >    ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0
> >    Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS
> >    Max-Forwards: 69
> >    From: "Extension 1001"
> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843
> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >    CSeq: 7788615 ACK
> >    Content-Length: 0
> >
> >    ------------------------------------------------------------------------
> > send 1344 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:
> >    ------------------------------------------------------------------------
> >    INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0
> >    Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN
> >    Max-Forwards: 69
> >    From: "Extension 1001"
> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >    To: <sip:1MYPHONE at newyork.voip.ms>
> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >    CSeq: 7788616 INVITE
> >    Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
> >    Expires: 300
> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> > -0600
> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> > REGISTER, REFER, NOTIFY
> >    Supported: timer, precondition, path, replaces
> >    Allow-Events: talk, hold, refer
> >    Proxy-Authorization: Digest username="121628", realm="newyork.voip.ms",
> > nonce="2d534dd6", algorithm=MD5, uri="sip:1MYPHONE at newyork.voip.ms",
> > response="16f3301efae13df926da7550f709d28a"
> >    Content-Type: application/sdp
> >    Content-Disposition: session
> >    Content-Length: 280
> >    X-FS-Support: update_display
> >    Remote-Party-ID: "Extension 1001"
> > <sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off
> >
> >    v=0
> >    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> >    s=-
> >    c=IN IP4 10.153.174.6
> >    t=0 0
> >    m=audio 49152 RTP/AVP 18 97 98
> >    a=rtpmap:18 G729/8000
> >    a=fmtp:18 annexb=no
> >    a=rtpmap:97 iLBC/8000
> >    a=rtpmap:98 telephone-event/8000
> >    a=fmtp:98 0-15
> >    a=ptime:20
> >    a=maxptime:40
> >    ------------------------------------------------------------------------
> > recv 488 bytes from udp/[74.63.41.218]:5060 at 21:15:59.484375:
> >    ------------------------------------------------------------------------
> >    SIP/2.0 100 Trying
> >    Via: SIP/2.0/UDP
> > 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080
> >    From: "Extension 1001"
> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >    To: <sip:1MYPHONE at newyork.voip.ms>
> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >    CSeq: 7788616 INVITE
> >    User-Agent: VoIPMS/SERAST
> >    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> >    Supported: replaces
> >    Contact: <sip:1MYPHONE at 74.63.41.218>
> >    Content-Length: 0
> >
> >    ------------------------------------------------------------------------
> > recv 516 bytes from udp/[74.63.41.218]:5060 at 21:15:59.562500:
> >    ------------------------------------------------------------------------
> >    SIP/2.0 503 Service Unavailable
> >    Via: SIP/2.0/UDP
> > 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080
> >    From: "Extension 1001"
> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9
> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >    CSeq: 7788616 INVITE
> >    User-Agent: VoIPMS/SERAST
> >    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> >    Supported: replaces
> >    Contact: <sip:1MYPHONE at 74.63.41.218>
> >    Content-Length: 0
> >
> >    ------------------------------------------------------------------------
> > send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.562500:
> >    ------------------------------------------------------------------------
> >    ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0
> >    Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN
> >    Max-Forwards: 69
> >    From: "Extension 1001"
> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9
> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >    CSeq: 7788616 ACK
> >    Content-Length: 0
> >
> >    ------------------------------------------------------------------------
> > 2011-01-28 16:15:59.562500 [INFO] mod_dptools.c:2612 Originate Failed.
> > Cause: NO_ANSWER
> > 2011-01-28 16:15:59.562500 [NOTICE] sofia.c:5286 Hangup
> > sofia/external/1MYPHONE [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
> > 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:189
> > sofia/internal/1001 at 192.168.1.100 has executed the last dialplan
> > instruction, hanging up.
> > 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:191 Hangup
> > sofia/internal/1001 at 192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING]
> > 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1306 Session 2
> > (sofia/external/1MYPHONE) Ended
> > 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1308 Close Channel
> > sofia/external/1MYPHONE [CS_DESTROY]
> > send 887 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.593750:
> >    ------------------------------------------------------------------------
> >    SIP/2.0 503 Service Unavailable
> >    Via: SIP/2.0/TCP
> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180
> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >    To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg
> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >    CSeq: 5244504 INVITE
> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> > -0600
> >    Accept: application/sdp
> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >    Supported: timer, precondition, path, replaces
> >    Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
> > include-session-description, presence.winfo, message-summary, refer
> >    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
> > 2011-01-28 16:15:59.593750 [NOTICE] switch_core_session.c:1306 Session 1
> > (sofia/internal/1001 at 192.168.1.100) Ended
> >    Content-Length: 02011-01-28 16:15:59.593750 [NOTICE]
> > switch_core_session.c:1308 Close Channel sofia/internal/1001 at 192.168.1.100
> > [CS_DESTROY]
> >
> >    Remote-Party-ID: "MYPHONE"
> > <sip:MYPHONE at 192.168.1.100>;party=calling;privacy=off;screen=no
> >
> >    ------------------------------------------------------------------------
> > recv 650 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.953125:
> >    ------------------------------------------------------------------------
> >    ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> >    Via: SIP/2.0/TCP
> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport
> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >    To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg
> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >    CSeq: 5244504 ACK
> >    Supported: sec-agree
> >    Max-Forwards: 70
> >    Proxy-Authorization: Digest
> > qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"
> >    Content-Length: 0
> >
> >    ------------------------------------------------------------------------
> >
> > Thank you in advance.
> >
> > _______________________________________________ FreeSWITCH-users mailing
> > list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
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From steveayre at gmail.com  Mon Jan 31 11:19:30 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 31 Jan 2011 08:19:30 +0000
Subject: [Freeswitch-users] Accountcode
In-Reply-To: <923586.77791.qm@web34304.mail.mud.yahoo.com>
References: <923586.77791.qm@web34304.mail.mud.yahoo.com>
Message-ID: <AANLkTinHkXouBvRMYz2Re3Xu6upo6emqnY1ZkF4ewRXU@mail.gmail.com>

For inbound (customer), I set a account_code variable on the aleg. You do
that either from dialplan or user directory.

For outbound (provider), I set gateway_name on the bleg in the dialstring,
e.g.
"[gateway_name=gw1]sofia/gateway/gw1/$1,[gateway_name=gw2]sofia/profile/outbound/$1 at a.b.c.d"
- mod_lcr does this with carrier_name for example.

You're free to use whatever variable names suit you best.

-Steve



On 31 January 2011 02:04, FERNANDO VILLARROEL <fvillarroel at yahoo.com> wrote:

> Hi All.
>
> If i received traffic from a customer and i forward this traffic to a
> provider, How i can know the traffic inbound and outbound (customer and
> provider)?
>
> Like accountcode of Asterisk.
>
> Regards.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.ponzone at ipeva.fr  Mon Jan 31 11:33:34 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 31 Jan 2011 09:33:34 +0100
Subject: [Freeswitch-users] Accountcode
In-Reply-To: <AANLkTinHkXouBvRMYz2Re3Xu6upo6emqnY1ZkF4ewRXU@mail.gmail.com>
References: <923586.77791.qm@web34304.mail.mud.yahoo.com>
	<AANLkTinHkXouBvRMYz2Re3Xu6upo6emqnY1ZkF4ewRXU@mail.gmail.com>
Message-ID: <210D5247-96DC-480D-9419-F49AC33DAEE4@ipeva.fr>

I do the same as Steven, except as I use only pre-defined gateways, I already have the ${sip_gateway_name} variable available for me to use in the XML or CSV CDR.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 31/01/2011 ? 09:19, Steven Ayre a ?crit :

> For inbound (customer), I set a account_code variable on the aleg. You do that either from dialplan or user directory.
> 
> For outbound (provider), I set gateway_name on the bleg in the dialstring, e.g. "[gateway_name=gw1]sofia/gateway/gw1/$1,[gateway_name=gw2]sofia/profile/outbound/$1 at a.b.c.d" - mod_lcr does this with carrier_name for example.
> 
> You're free to use whatever variable names suit you best.
> 
> -Steve
> 
> 
> 
> On 31 January 2011 02:04, FERNANDO VILLARROEL <fvillarroel at yahoo.com> wrote:
> Hi All.
> 
> If i received traffic from a customer and i forward this traffic to a provider, How i can know the traffic inbound and outbound (customer and provider)?
> 
> Like accountcode of Asterisk.
> 
> Regards.
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jonas.gauffin at gmail.com  Mon Jan 31 13:34:28 2011
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Mon, 31 Jan 2011 11:34:28 +0100
Subject: [Freeswitch-users] Outbound faxes fail
Message-ID: <AANLkTin7g6g9CwvvLhq_smi4BZvUEMy9OB-k34-g0qfJ@mail.gmail.com>

Hello,

A lot of my outbound faxes fail with an error message saying "Timed out
waiting for initial communication".
Can someone help me determine the cause?

I got a log here: http://pastebin.freeswitch.org/15187

//Jonas
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From vik_kom at mail.ru  Mon Jan 31 13:35:50 2011
From: vik_kom at mail.ru (=?koi8-r?Q?=D7=C9=CB=5F=CB=CF=CD_=CB=CF=CD=CD=C9=D3=C1=D2=CF=D7?=)
Date: Mon, 31 Jan 2011 13:35:50 +0300
Subject: [Freeswitch-users] =?koi8-r?b?U2FuZ29tYSBBMTA0RA==?=
In-Reply-To: <4D408506.207@sangoma.com>
References: <1296054869082-5962992.post@n2.nabble.com>
	<4D408506.207@sangoma.com>
Message-ID: <E1Pjr6s-0003DT-00.vik_kom-mail-ru@f282.mail.ru>

Thank You David! I resolved this problem by changing RRING and RTIP, TRING and TTIP between himself. And i remember, when a was an scool, us IT teacher say to us: Boys! Before you tell me your questions, get up your hand, then blood reached your head, and maybe need of your question get out. :)


From david.ponzone at ipeva.fr  Mon Jan 31 13:46:12 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 31 Jan 2011 11:46:12 +0100
Subject: [Freeswitch-users] Outbound faxes fail
In-Reply-To: <AANLkTin7g6g9CwvvLhq_smi4BZvUEMy9OB-k34-g0qfJ@mail.gmail.com>
References: <AANLkTin7g6g9CwvvLhq_smi4BZvUEMy9OB-k34-g0qfJ@mail.gmail.com>
Message-ID: <13F4AC41-3E40-43A3-B155-2A6FE638B432@ipeva.fr>

it seems you are using Fax over G711.
Can you switch to T.38 ?
What happens if you make a voice call to the same number ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 31/01/2011 ? 11:34, Jonas Gauffin a ?crit :

> Hello,
> 
> A lot of my outbound faxes fail with an error message saying "Timed out waiting for initial communication". 
> Can someone help me determine the cause?
> 
> I got a log here: http://pastebin.freeswitch.org/15187
> 
> //Jonas
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jonas.gauffin at gmail.com  Mon Jan 31 14:15:47 2011
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Mon, 31 Jan 2011 12:15:47 +0100
Subject: [Freeswitch-users] Outbound faxes fail
In-Reply-To: <13F4AC41-3E40-43A3-B155-2A6FE638B432@ipeva.fr>
References: <AANLkTin7g6g9CwvvLhq_smi4BZvUEMy9OB-k34-g0qfJ@mail.gmail.com>
	<13F4AC41-3E40-43A3-B155-2A6FE638B432@ipeva.fr>
Message-ID: <AANLkTimiYu1d8q5=+-o5zh71Fw9b=90DbjCpR_phPTVG@mail.gmail.com>

I got a fax tone if I called the number manually.

I've added <param name="enable-t38" value="true" /> to my fax.conf.xml. What
else do I need to do?

I'm doing this for outbound faxes:

1. Invoke an originate though eventsocket bgapi
2. The originate runs a mod_managed application
3. The managed application invokes session.execute with txfax(xxxxxx)


On Mon, Jan 31, 2011 at 11:46 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> it seems you are using Fax over G711.
> Can you switch to T.38 ?
> What happens if you make a voice call to the same number ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 31/01/2011 ? 11:34, Jonas Gauffin a ?crit :
>
> Hello,
>
> A lot of my outbound faxes fail with an error message saying "Timed out
> waiting for initial communication".
> Can someone help me determine the cause?
>
> I got a log here: http://pastebin.freeswitch.org/15187
>
> //Jonas
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Mon Jan 31 14:25:11 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 31 Jan 2011 12:25:11 +0100
Subject: [Freeswitch-users] Outbound faxes fail
In-Reply-To: <AANLkTimiYu1d8q5=+-o5zh71Fw9b=90DbjCpR_phPTVG@mail.gmail.com>
References: <AANLkTin7g6g9CwvvLhq_smi4BZvUEMy9OB-k34-g0qfJ@mail.gmail.com>
	<13F4AC41-3E40-43A3-B155-2A6FE638B432@ipeva.fr>
	<AANLkTimiYu1d8q5=+-o5zh71Fw9b=90DbjCpR_phPTVG@mail.gmail.com>
Message-ID: <AE397829-B4F5-411B-90FB-06B8E387F098@ipeva.fr>

Well you need to make sure your outbound gateway supports T38.
You could also try to reduce your fax transmit speed to 9600.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 31/01/2011 ? 12:15, Jonas Gauffin a ?crit :

> I got a fax tone if I called the number manually.
> 
> I've added <param name="enable-t38" value="true" /> to my fax.conf.xml. What else do I need to do?
> 
> I'm doing this for outbound faxes:
> 
> 1. Invoke an originate though eventsocket bgapi
> 2. The originate runs a mod_managed application 
> 3. The managed application invokes session.execute with txfax(xxxxxx)
> 
> 
> On Mon, Jan 31, 2011 at 11:46 AM, David Ponzone <david.ponzone at ipeva.fr> wrote:
> it seems you are using Fax over G711.
> Can you switch to T.38 ?
> What happens if you make a voice call to the same number ?
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 31/01/2011 ? 11:34, Jonas Gauffin a ?crit :
> 
>> Hello,
>> 
>> A lot of my outbound faxes fail with an error message saying "Timed out waiting for initial communication". 
>> Can someone help me determine the cause?
>> 
>> I got a log here: http://pastebin.freeswitch.org/15187
>> 
>> //Jonas
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From steveayre at gmail.com  Mon Jan 31 15:14:39 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 31 Jan 2011 12:14:39 +0000
Subject: [Freeswitch-users] Accountcode
In-Reply-To: <210D5247-96DC-480D-9419-F49AC33DAEE4@ipeva.fr>
References: <923586.77791.qm@web34304.mail.mud.yahoo.com>
	<AANLkTinHkXouBvRMYz2Re3Xu6upo6emqnY1ZkF4ewRXU@mail.gmail.com>
	<210D5247-96DC-480D-9419-F49AC33DAEE4@ipeva.fr>
Message-ID: <AANLkTi=4DS5wkY2654Sio_TN01XJ9utcV7XO0HBWctpk@mail.gmail.com>

Of course that only goes for Sofia gateways so ignores connecting via the IP
or using anything other than SIP. But for most people that'd be good enough.
:)

-Steve



On 31 January 2011 08:33, David Ponzone <david.ponzone at ipeva.fr> wrote:

> I do the same as Steven, except as I use only pre-defined gateways, I
> already have the ${sip_gateway_name} variable available for me to use in the
> XML or CSV CDR.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 31/01/2011 ? 09:19, Steven Ayre a ?crit :
>
> For inbound (customer), I set a account_code variable on the aleg. You do
> that either from dialplan or user directory.
>
> For outbound (provider), I set gateway_name on the bleg in the dialstring,
> e.g.
> "[gateway_name=gw1]sofia/gateway/gw1/$1,[gateway_name=gw2]sofia/profile/outbound/$1 at a.b.c.d"
> - mod_lcr does this with carrier_name for example.
>
> You're free to use whatever variable names suit you best.
>
> -Steve
>
>
>
> On 31 January 2011 02:04, FERNANDO VILLARROEL <fvillarroel at yahoo.com>wrote:
>
>> Hi All.
>>
>> If i received traffic from a customer and i forward this traffic to a
>> provider, How i can know the traffic inbound and outbound (customer and
>> provider)?
>>
>> Like accountcode of Asterisk.
>>
>> Regards.
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gustavo.espeche at upper-soft.com  Mon Jan 31 15:44:37 2011
From: gustavo.espeche at upper-soft.com (Gustavo Espeche)
Date: Mon, 31 Jan 2011 09:44:37 -0300
Subject: [Freeswitch-users] open g729
In-Reply-To: <AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com>
References: <1296343422.2615.4.camel@gustavo-laptop>
	<AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com>
Message-ID: <1296477877.3589.24.camel@gustavo-laptop>

I just need test the performance of trans-coding before to recommend to
may customer what server it need and buy the codec, because of then i
need testing in my lab how much cpu and memory require the trans-coding
in freeswitch for g729.
And searching i found the open-g729/g723 for asterisk.
I relay don't wish to start a discussion about the g729 license i read
and understated it. But i think that the license price per channel is to
hight taking care that this codec has more than 10 years but the way
this is only my opinion.

Best Regards.
Gustavo Espeche
www.easyipcall.com


On Sat, 2011-01-29 at 17:27 -0600, Rupa Schomaker wrote:
> How do you intend to pay the license fee?
> 
> On Sat, Jan 29, 2011 at 5:23 PM, Gustavo Espeche
> <gustavo.espeche at upper-soft.com> wrote:
> > Hello,
> >    some one can compile open g729 to work with freeswitch?
> >    http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
> >
> >    i appreciate a lot if some one has some experience in it.
> >    Best Regards.
> >
> >    Gustavo Espeche
> >    www.easyipcall.com
> >
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> 




From lakindia89 at gmail.com  Mon Jan 31 15:48:27 2011
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Mon, 31 Jan 2011 18:18:27 +0530
Subject: [Freeswitch-users] Blind Transfer Not working
Message-ID: <AANLkTim7-A=NRjwND3tyzJeDk4boiuCrar5xv-7SB428@mail.gmail.com>

Hi all,

here is my dialplan
        <context name="default">
                <extension name="outbound_soc">
                        <condition field="destination_number"
expression="^.*$">
                                <action application="log" data="ERR
${domain_name}"/>
                                <action application="set"
data="continue_on_fail=true"/>
                                <action application="set"
data="bypass_media=false"/>
                                <action application="ring_ready"/>
                                <action application="set"
data="ignore_early_media=true"/>
                                <action application="set"
data="exec_after_bridge_app=park"/>
                                <action application="export"
data="RECORD_STEREO=true"/>
                                <action application="bridge"
data="user/1000"/>
                        </condition>
                </extension>
        </context>

I made a call from 9952248266 to 39114601. As expected it called to the 1000
extension and I answered the call. I was using twinkle as a softphone.
Now from twinkle I initiated "Transfer". I've given "9976975781" as a number
and I choose "Blind Transfer".
But the call didn't get proceeded.

Here is the log with sofia trace enabled.
http://pastebin.freeswitch.org/15189

Once REFER is received by freeswitch, It replied with "Accept" for that
refer, but after that the call was not proceeding!!
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From erik.dekkers at wvds.nl  Mon Jan 31 15:51:38 2011
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Mon, 31 Jan 2011 13:51:38 +0100
Subject: [Freeswitch-users] open g729
In-Reply-To: <4D45A926.50403@gmail.com>
References: <1296343422.2615.4.camel@gustavo-laptop> <4D45A926.50403@gmail.com>
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE0787301834@FS1.wvds.local>

Or in Algeria, but that's almost the same :)

-----Oorspronkelijk bericht-----
Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Meftah Tayeb
Verzonden: zondag 30 januari 2011 19:09
Aan: FreeSWITCH Users Help
CC: Gustavo Espeche
Onderwerp: Re: [Freeswitch-users] open g729

do it and end up in jail.
Le 30/01/2011 00:23, Gustavo Espeche a ?crit :
> Hello,
>          some one can compile open g729 to work with freeswitch?
>          http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
>
>          i appreciate a lot if some one has some experience in it.
>          Best Regards.
>
>          Gustavo Espeche
>          www.easyipcall.com
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
phone: +13477595883


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From steveayre at gmail.com  Mon Jan 31 15:53:48 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 31 Jan 2011 12:53:48 +0000
Subject: [Freeswitch-users] open g729
In-Reply-To: <1296477877.3589.24.camel@gustavo-laptop>
References: <1296343422.2615.4.camel@gustavo-laptop>
	<AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com>
	<1296477877.3589.24.camel@gustavo-laptop>
Message-ID: <AANLkTinq9AmN=ZqjGLNUxucw2nJXBLpNHKLTA5jLWfgj@mail.gmail.com>

The price is set by the patent holders, there's not much that can be done
about that.

If  you want to evaluate the g729 transcoding performance you **must** test
with the one you plan to recommend - the mod_com_g729, mod_sangoma_codec and
ipp g729 codecs have very difference performance characteristics. It is
pointless to evaluate one and then recommend another.

- mod_com_g729 is the official $10/channel license. It has good performance.
To evaluate it you should try it for yourself.
- The unofficial illegal FS module based ipp g729 which I shall not name has
stability problems and uses far more CPU than mod_com_g729. Do not use it.
- mod_sangoma_codec uses almost no CPU, but you do have to purchase a
transcoding card.

-Steve



On 31 January 2011 12:44, Gustavo Espeche <gustavo.espeche at upper-soft.com>wrote:

> I just need test the performance of trans-coding before to recommend to
> may customer what server it need and buy the codec, because of then i
> need testing in my lab how much cpu and memory require the trans-coding
> in freeswitch for g729.
> And searching i found the open-g729/g723 for asterisk.
> I relay don't wish to start a discussion about the g729 license i read
> and understated it. But i think that the license price per channel is to
> hight taking care that this codec has more than 10 years but the way
> this is only my opinion.
>
> Best Regards.
> Gustavo Espeche
> www.easyipcall.com
>
>
> On Sat, 2011-01-29 at 17:27 -0600, Rupa Schomaker wrote:
> > How do you intend to pay the license fee?
> >
> > On Sat, Jan 29, 2011 at 5:23 PM, Gustavo Espeche
> > <gustavo.espeche at upper-soft.com> wrote:
> > > Hello,
> > >    some one can compile open g729 to work with freeswitch?
> > >    http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
> > >
> > >    i appreciate a lot if some one has some experience in it.
> > >    Best Regards.
> > >
> > >    Gustavo Espeche
> > >    www.easyipcall.com
> > >
> > >
> > >
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> >
> >
> >
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From patrick.plattes at niemann-frey.info  Mon Jan 31 16:10:47 2011
From: patrick.plattes at niemann-frey.info (Patrick Plattes)
Date: Mon, 31 Jan 2011 14:10:47 +0100
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTinr0RdgBs2UiS3=b3i+k6uCwkMihDy_FbAOFPc_@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
	<AANLkTi=in3CLd-n0mT3SRKYA0kxLMGVWk9KeBx3_jhY6@mail.gmail.com>
	<AANLkTin3Bu+8EEr_v_GaRGNPzOjfcnfzsV2Av2cgnWgO@mail.gmail.com>
	<AANLkTi=OC_jx0+nzK405VZURLDJPD25R4ZjyJ3RXJ1nk@mail.gmail.com>
	<AANLkTi=Sm7CjoayiE8nHmL0Eehqknj1jdpbx0YBQJQGH@mail.gmail.com>
	<AANLkTinr0RdgBs2UiS3=b3i+k6uCwkMihDy_FbAOFPc_@mail.gmail.com>
Message-ID: <AANLkTi=+aObW3kVBZq5ox4b4E1uFTSgtXUL72tHLOzPi@mail.gmail.com>

Hi,

thanks for your answer. I can't get it to work on 'current'. I'm also
not able to use presence in in the current, but it works fine with
1.0.6. Is there anything broken?

I've done exactly the same with exactly the same configuration.

FS 1.0.6: http://fs.kwixo.de/fs-1.0.6.pcap
presence in queue-sales_fr-263 at 192.168.56.189 inuse 'inuse'
presence out queue-sales_fr-263 at 192.168.56.189 inuse 'inuse'

FS current: http://fs.kwixo.de/fs-current-20110131.pcap
presence in queue-sales_fr-263 at 192.168.56.189 inuse 'inuse'
presence out queue-sales_fr-263 at 192.168.56.189 inuse 'inuse'


Thanks,
 Patrick

2011/1/28 Anthony Minessale <anthony.minessale at gmail.com>:
> try with no presence_in and out commands at all
>
> just set presence_id variable as soon as you can and that call will
> report its state as that id for its entire duration.
> Also make sure you are on latest GIT when you test this.
>
>
> On Fri, Jan 28, 2011 at 7:47 AM, Patrick Plattes
> <patrick.plattes at niemann-frey.info> wrote:
>> Formatted and colourized code: http://pastebin.freeswitch.org/15172
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Patrick Plattes
IT - Projektleiter

Niemann + Frey GmbH
Adolf-Dembach-Str. 24
47829 Krefeld

Tel.: +49/2151 - 5554-263
Fax : +49/2151 - 5554-123
patrick.plattes at niemann-frey.info

Gesch?ftsf?hrer: Gerd Frey
Sitz und Registergericht: Krefeld HRB 10851

www.niemann-frey.de


From patrick.plattes at niemann-frey.info  Mon Jan 31 16:15:05 2011
From: patrick.plattes at niemann-frey.info (Patrick Plattes)
Date: Mon, 31 Jan 2011 14:15:05 +0100
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTi=gcY9GckXECKHEis6pYCh5NQpsriGZgUXhe9rc@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
	<AANLkTi=in3CLd-n0mT3SRKYA0kxLMGVWk9KeBx3_jhY6@mail.gmail.com>
	<AANLkTin3Bu+8EEr_v_GaRGNPzOjfcnfzsV2Av2cgnWgO@mail.gmail.com>
	<AANLkTi=OC_jx0+nzK405VZURLDJPD25R4ZjyJ3RXJ1nk@mail.gmail.com>
	<AANLkTi=Sm7CjoayiE8nHmL0Eehqknj1jdpbx0YBQJQGH@mail.gmail.com>
	<AANLkTinr0RdgBs2UiS3=b3i+k6uCwkMihDy_FbAOFPc_@mail.gmail.com>
	<AANLkTi=gcY9GckXECKHEis6pYCh5NQpsriGZgUXhe9rc@mail.gmail.com>
Message-ID: <AANLkTin0ofNUgxeOM9jrqp5G=ZEtftY0QrwRE2THvx58@mail.gmail.com>

Hi Chris :-),

sounds very interesting :-)

Is the Phone after a reboot of the phone again a member of the queue?
Will the correct state of the BLF automaticly send to the phone?

Thanks,
 Patrick

2011/1/29 Chris Burns <chris at cloudtel.com>:
> I have something set up in the lab right now that could be exactly what you
> are trying to do. It is for a polycom deployment where 650s with BLF addons
> monitor a collection of agents. The agents have multiple skillsets/languages
> and belong to multiple inbound queues. The managers are used to having
> things blink a certain specific way (as well as using polycom web apps that
> tie into their intranet, but thats a different story x_X). Managers can
> quickly see which agents are available on which queues just by lights, and
> it can correct the state of BLFs if the phone gets rebooted.
>
> If you still have issues and you are interested I could pull out the
> relevant code into an example and pastebin it for you. It would be in php
> using the ESL module.
>
> On Fri, Jan 28, 2011 at 11:49 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> try with no presence_in and out commands at all
>>
>> just set presence_id variable as soon as you can and that call will
>> report its state as that id for its entire duration.
>> Also make sure you are on latest GIT when you test this.
>>
>>
>> On Fri, Jan 28, 2011 at 7:47 AM, Patrick Plattes
>> <patrick.plattes at niemann-frey.info> wrote:
>> > Formatted and colourized code: http://pastebin.freeswitch.org/15172
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Patrick Plattes
IT - Projektleiter

Niemann + Frey GmbH
Adolf-Dembach-Str. 24
47829 Krefeld

Tel.: +49/2151 - 5554-263
Fax : +49/2151 - 5554-123
patrick.plattes at niemann-frey.info

Gesch?ftsf?hrer: Gerd Frey
Sitz und Registergericht: Krefeld HRB 10851

www.niemann-frey.de


From Stefan.Weigel at allianz-warranty.com  Mon Jan 31 16:37:36 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Mon, 31 Jan 2011 14:37:36 +0100
Subject: [Freeswitch-users] mod_callcenter and effective_caller_id_name
Message-ID: <5003D7D3E06F514E8C682F18D223265C04717DC7F0@AZWSMS03.azwarranty.int>

 Hi all,

when modifying 'effective_caller_id_name' it's working for user extensions. When routing to application mod_callcenter the change doesn't take effect. Is it a bug or am I missing something ?


Thanks and best regards

Stefan
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From steveu at coppice.org  Mon Jan 31 16:44:55 2011
From: steveu at coppice.org (Steve Underwood)
Date: Mon, 31 Jan 2011 21:44:55 +0800
Subject: [Freeswitch-users] open g729
In-Reply-To: <1296477877.3589.24.camel@gustavo-laptop>
References: <1296343422.2615.4.camel@gustavo-laptop>	<AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com>
	<1296477877.3589.24.camel@gustavo-laptop>
Message-ID: <4D46BCD7.9080206@coppice.org>

On 01/31/2011 08:44 PM, Gustavo Espeche wrote:
> I just need test the performance of trans-coding before to recommend to
> may customer what server it need and buy the codec, because of then i
> need testing in my lab how much cpu and memory require the trans-coding
> in freeswitch for g729.
> And searching i found the open-g729/g723 for asterisk.
> I relay don't wish to start a discussion about the g729 license i read
> and understated it. But i think that the license price per channel is to
> hight taking care that this codec has more than 10 years but the way
> this is only my opinion.
That is not a plausible reason for wanting to use the unlicenced code. 
It is completely different from the licenced code. The performance of 
the two is very different.

The price won't really change until the codec is 20 years old, and the 
patents expire.

Steve



From Michal.Muszalski at telekomunikacja.pl  Mon Jan 31 17:05:03 2011
From: Michal.Muszalski at telekomunikacja.pl (=?iso-8859-2?Q?Muszalski_Micha=B3_-_Korpo_TP?=)
Date: Mon, 31 Jan 2011 15:05:03 +0100
Subject: [Freeswitch-users] ODP:  PD: SIP -> XMPP calls
In-Reply-To: <1296145512923-5966876.post@n2.nabble.com>
Message-ID: <964C8735B9EC444A99C4EC298B1BA0AB0209DE68@OPEXCN07.tp.gk.corp.tepenet>

Hello!

I've installed FreeSWITCH Version 1.0.head (git-cf253c3 2011-01-11 15-31-40
-0600) and still have the same problem :(
Any ideas?

Regards,
Michal

-----Wiadomo?? oryginalna-----
Od: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] W imieniu mazilo
Wys?ano: 27 stycznia 2011 17:25
Do: freeswitch-users at lists.freeswitch.org
Temat: Re: [Freeswitch-users] PD: SIP -> XMPP calls



Muszalski Micha? - Korpo TP wrote:
> 
> 
>> Hello,
>> 
>> We are trying to make some kind of a gateway between SIP and XMPP domain.
>> We have an environment with FreeSWITCH and OpenFire (FreeSWITCH is 
>> registered as a component in OF). Calls from XMPP to SIP are working 
>> fine.
>> The problem is with calls from SIP to XMPP. The caller (SIP) has a 
>> ringing tone, a the callee (XMPP) has a 'connecting...' message after 
>> answer the call.
I see this problem with a self-built FS git as of 01/26/2011, but not on a
FreeSWITCH Version 1.0.head (git-cf253c3 2011-01-11 15-31-40 -0600).

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
--
View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/PD-SIP-XMPP-calls-tp5966636p59
66876.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From max.bridgewater at gmail.com  Mon Jan 31 17:21:29 2011
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 31 Jan 2011 09:21:29 -0500
Subject: [Freeswitch-users] Freeswitch and MongoDB
Message-ID: <AANLkTikw1_S0uZGE_jG9nU27XvPhLzvDSuBCG4w9SPY7@mail.gmail.com>

Hi,

I was wondering whether it's possible to have Freeswitch work with a
non relational database such as MongoDB? Or when I write a Javascript
application for Freeswitch, can I make it access MongoDB?instead of
MySQL via ODBC?

Thanks,
Max.


From tim at compnetwork.net  Mon Jan 31 17:49:01 2011
From: tim at compnetwork.net (Tim King)
Date: Mon, 31 Jan 2011 09:49:01 -0500
Subject: [Freeswitch-users] CDR Fields
Message-ID: <AANLkTik8anSQqoXFOb6jyX0Rx6f5BzTrDvoii2J_nPA=@mail.gmail.com>

Guys I have been digging for a while and I am struggling to find a page the
lists all of the possible fields (preferably with brief descriptions) that
are available with the various CDR records aka(XML and CSv).

I am also curious if anyone as found a way to record quality information
aka(Jitter, Latency, etc..) in the CDR record for the call.

Thanks

Tim
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From steveayre at gmail.com  Mon Jan 31 18:06:54 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 31 Jan 2011 15:06:54 +0000
Subject: [Freeswitch-users] CDR Fields
In-Reply-To: <AANLkTik8anSQqoXFOb6jyX0Rx6f5BzTrDvoii2J_nPA=@mail.gmail.com>
References: <AANLkTik8anSQqoXFOb6jyX0Rx6f5BzTrDvoii2J_nPA=@mail.gmail.com>
Message-ID: <AANLkTim6xqDOGJ+s7B+kLPz2V+UBYG=Em6zKiS_UOHKn@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Channel_Variables
XML CDR also contains extra information such as callflow.

Can't really give you an answer on the quality information...

-Steve


On 31 January 2011 14:49, Tim King <tim at compnetwork.net> wrote:

> Guys I have been digging for a while and I am struggling to find a page the
> lists all of the possible fields (preferably with brief descriptions) that
> are available with the various CDR records aka(XML and CSv).
>
> I am also curious if anyone as found a way to record quality information
> aka(Jitter, Latency, etc..) in the CDR record for the call.
>
> Thanks
>
> Tim
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Nabble at slickdeals.endjunk.com  Mon Jan 31 18:19:51 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 31 Jan 2011 07:19:51 -0800 (PST)
Subject: [Freeswitch-users] open g729
In-Reply-To: <4D46BCD7.9080206@coppice.org>
References: <1296343422.2615.4.camel@gustavo-laptop>
	<AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com>
	<1296477877.3589.24.camel@gustavo-laptop>
	<4D46BCD7.9080206@coppice.org>
Message-ID: <1296487191683-5977433.post@n2.nabble.com>



Steve Underwood wrote:
> The price won't really change until the codec is 20 years old, and the 
> patents expire.
What happens when the patents get renewed?

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/open-g729-tp5973641p5977433.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From curriegrad2004 at gmail.com  Mon Jan 31 18:26:12 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 31 Jan 2011 07:26:12 -0800
Subject: [Freeswitch-users] open g729
In-Reply-To: <1296487191683-5977433.post@n2.nabble.com>
References: <1296343422.2615.4.camel@gustavo-laptop>
	<AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com>
	<1296477877.3589.24.camel@gustavo-laptop>
	<4D46BCD7.9080206@coppice.org>
	<1296487191683-5977433.post@n2.nabble.com>
Message-ID: <AANLkTikRg6LcyasebweNq=UD9YK4MNNvbGzXi5w6AXKt@mail.gmail.com>

Legally, they would have to make substantial changes to the G.729 code
before they can even re-patent it after the expiry. Now I'm not a
lawyer on this, but this seems to be the case with most of the patents
out there.

On Mon, Jan 31, 2011 at 7:19 AM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
>
> Steve Underwood wrote:
>> The price won't really change until the codec is 20 years old, and the
>> patents expire.
> What happens when the patents get renewed?
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/open-g729-tp5973641p5977433.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From infos at madovsky.org  Mon Jan 31 18:48:33 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 31 Jan 2011 10:48:33 -0500
Subject: [Freeswitch-users] open g729
References: <1296343422.2615.4.camel@gustavo-laptop><AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com><1296477877.3589.24.camel@gustavo-laptop>
	<AANLkTinq9AmN=ZqjGLNUxucw2nJXBLpNHKLTA5jLWfgj@mail.gmail.com>
Message-ID: <9D7F812E93DC47A99F6A30638FB8092F@e1705>

so the best is,
don't use G729 and go to buy food with your $10 ;)
  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 31, 2011 7:53 AM
  Subject: Re: [Freeswitch-users] open g729


  The price is set by the patent holders, there's not much that can be done about that.

  If  you want to evaluate the g729 transcoding performance you *must* test with the one you plan to recommend - the mod_com_g729, mod_sangoma_codec and ipp g729 codecs have very difference performance characteristics. It is pointless to evaluate one and then recommend another.

  - mod_com_g729 is the official $10/channel license. It has good performance. To evaluate it you should try it for yourself.
  - The unofficial illegal FS module based ipp g729 which I shall not name has stability problems and uses far more CPU than mod_com_g729. Do not use it.
  - mod_sangoma_codec uses almost no CPU, but you do have to purchase a transcoding card.

  -Steve




  On 31 January 2011 12:44, Gustavo Espeche <gustavo.espeche at upper-soft.com> wrote:

    I just need test the performance of trans-coding before to recommend to
    may customer what server it need and buy the codec, because of then i
    need testing in my lab how much cpu and memory require the trans-coding
    in freeswitch for g729.
    And searching i found the open-g729/g723 for asterisk.
    I relay don't wish to start a discussion about the g729 license i read
    and understated it. But i think that the license price per channel is to
    hight taking care that this codec has more than 10 years but the way
    this is only my opinion.


    Best Regards.
    Gustavo Espeche
    www.easyipcall.com



    On Sat, 2011-01-29 at 17:27 -0600, Rupa Schomaker wrote:
    > How do you intend to pay the license fee?
    >
    > On Sat, Jan 29, 2011 at 5:23 PM, Gustavo Espeche
    > <gustavo.espeche at upper-soft.com> wrote:
    > > Hello,
    > >    some one can compile open g729 to work with freeswitch?
    > >    http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
    > >
    > >    i appreciate a lot if some one has some experience in it.
    > >    Best Regards.
    > >
    > >    Gustavo Espeche
    > >    www.easyipcall.com
    > >
    > >
    > >
    > >
    > >
    > > _______________________________________________
    > > FreeSWITCH-users mailing list
    > > FreeSWITCH-users at lists.freeswitch.org
    > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > > http://www.freeswitch.org
    > >
    >
    >
    >



    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anton.vazir at gmail.com  Mon Jan 31 18:54:10 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Mon, 31 Jan 2011 20:54:10 +0500
Subject: [Freeswitch-users] open g729
In-Reply-To: <AANLkTinq9AmN=ZqjGLNUxucw2nJXBLpNHKLTA5jLWfgj@mail.gmail.com>
References: <1296343422.2615.4.camel@gustavo-laptop>
	<AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com>
	<1296477877.3589.24.camel@gustavo-laptop>
	<AANLkTinq9AmN=ZqjGLNUxucw2nJXBLpNHKLTA5jLWfgj@mail.gmail.com>
Message-ID: <AANLkTikNcipSGxfofujQwSNn4E16F-97u51xLB8a=cLB@mail.gmail.com>

IPP has a fairly good performance, at least on asterisk it performs
as good as proprietary one, works perfectly fine, has no stability
problems, and I highly doubt that FS situation differs, unless someone
can prove it by posting real performance test numbers.

2011/1/31 Steven Ayre <steveayre at gmail.com>:
> The price is set by the patent holders, there's not much that can be done
> about that.
>
> If? you want to evaluate the g729 transcoding performance you *must* test
> with the one you plan to recommend - the mod_com_g729, mod_sangoma_codec and
> ipp g729 codecs have very difference performance characteristics. It is
> pointless to evaluate one and then recommend another.
>
> - mod_com_g729 is the official $10/channel license. It has good performance.
> To evaluate it you should try it for yourself.
> - The unofficial illegal FS module based ipp g729 which I shall not name has
> stability problems and uses far more CPU than mod_com_g729. Do not use it.
> - mod_sangoma_codec uses almost no CPU, but you do have to purchase a
> transcoding card.
>
> -Steve
>
>
>
> On 31 January 2011 12:44, Gustavo Espeche <gustavo.espeche at upper-soft.com>
> wrote:
>>
>> I just need test the performance of trans-coding before to recommend to
>> may customer what server it need and buy the codec, because of then i
>> need testing in my lab how much cpu and memory require the trans-coding
>> in freeswitch for g729.
>> And searching i found the open-g729/g723 for asterisk.
>> I relay don't wish to start a discussion about the g729 license i read
>> and understated it. But i think that the license price per channel is to
>> hight taking care that this codec has more than 10 years but the way
>> this is only my opinion.
>>
>> Best Regards.
>> Gustavo Espeche
>> www.easyipcall.com
>>
>>
>> On Sat, 2011-01-29 at 17:27 -0600, Rupa Schomaker wrote:
>> > How do you intend to pay the license fee?
>> >
>> > On Sat, Jan 29, 2011 at 5:23 PM, Gustavo Espeche
>> > <gustavo.espeche at upper-soft.com> wrote:
>> > > Hello,
>> > > ? ?some one can compile open g729 to work with freeswitch?
>> > > ? ?http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
>> > >
>> > > ? ?i appreciate a lot if some one has some experience in it.
>> > > ? ?Best Regards.
>> > >
>> > > ? ?Gustavo Espeche
>> > > ? ?www.easyipcall.com
>> > >
>> > >
>> > >
>> > >
>> > >
>> > > _______________________________________________
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > >
>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> > >
>> >
>> >
>> >
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steveayre at gmail.com  Mon Jan 31 18:54:54 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 31 Jan 2011 15:54:54 +0000
Subject: [Freeswitch-users] open g729
In-Reply-To: <9D7F812E93DC47A99F6A30638FB8092F@e1705>
References: <1296343422.2615.4.camel@gustavo-laptop>
	<AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com>
	<1296477877.3589.24.camel@gustavo-laptop>
	<AANLkTinq9AmN=ZqjGLNUxucw2nJXBLpNHKLTA5jLWfgj@mail.gmail.com>
	<9D7F812E93DC47A99F6A30638FB8092F@e1705>
Message-ID: <AANLkTikkLnf4BHYgL1k8th=VWk44vpH5gRYSAFW5YfW1@mail.gmail.com>

Not always possible, for instance in wholesale it's pretty much the standard
codec used.

-Steve


On 31 January 2011 15:48, Madovsky <infos at madovsky.org> wrote:

>  so the best is,
> don't use G729 and go to buy food with your $10 ;)
>
> ----- Original Message -----
> *From:* Steven Ayre <steveayre at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, January 31, 2011 7:53 AM
> *Subject:* Re: [Freeswitch-users] open g729
>
> The price is set by the patent holders, there's not much that can be done
> about that.
>
> If  you want to evaluate the g729 transcoding performance you **must**test with the one you plan to recommend - the mod_com_g729,
> mod_sangoma_codec and ipp g729 codecs have very difference performance
> characteristics. It is pointless to evaluate one and then recommend another.
>
> - mod_com_g729 is the official $10/channel license. It has good
> performance. To evaluate it you should try it for yourself.
> - The unofficial illegal FS module based ipp g729 which I shall not name
> has stability problems and uses far more CPU than mod_com_g729. Do not use
> it.
> - mod_sangoma_codec uses almost no CPU, but you do have to purchase a
> transcoding card.
>
> -Steve
>
>
>
> On 31 January 2011 12:44, Gustavo Espeche <gustavo.espeche at upper-soft.com>wrote:
>
>> I just need test the performance of trans-coding before to recommend to
>> may customer what server it need and buy the codec, because of then i
>> need testing in my lab how much cpu and memory require the trans-coding
>> in freeswitch for g729.
>> And searching i found the open-g729/g723 for asterisk.
>> I relay don't wish to start a discussion about the g729 license i read
>> and understated it. But i think that the license price per channel is to
>> hight taking care that this codec has more than 10 years but the way
>> this is only my opinion.
>>
>> Best Regards.
>> Gustavo Espeche
>> www.easyipcall.com
>>
>>
>>  On Sat, 2011-01-29 at 17:27 -0600, Rupa Schomaker wrote:
>> > How do you intend to pay the license fee?
>> >
>> > On Sat, Jan 29, 2011 at 5:23 PM, Gustavo Espeche
>> > <gustavo.espeche at upper-soft.com> wrote:
>> > > Hello,
>> > >    some one can compile open g729 to work with freeswitch?
>> > >    http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
>> > >
>> > >    i appreciate a lot if some one has some experience in it.
>> > >    Best Regards.
>> > >
>> > >    Gustavo Espeche
>> > >    www.easyipcall.com
>> > >
>> > >
>> > >
>> > >
>> > >
>> > > _______________________________________________
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> > >
>> >
>> >
>> >
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dyatsin at sangoma.com  Mon Jan 31 19:00:06 2011
From: dyatsin at sangoma.com (David Yat Sin)
Date: Mon, 31 Jan 2011 11:00:06 -0500
Subject: [Freeswitch-users] freetdm (sangoma A101): isdn-sip
 display	name interworking
In-Reply-To: <20110127091509.592AF119FB@mail.nstel.ru>
References: <20110127091509.592AF119FB@mail.nstel.ru>
Message-ID: <4D46DC86.30700@sangoma.com>

Hi Nicolay,
I don't think there is a print in the Freeswitch logs for the the 
Calling Name, so you will not see a mention of "sip isdn name 
interworking" in the logs.

The Calling Name is encoded in Display IE in IA5 characters based on Q.931.

Can you enable Q931 trace before making an outbound call and I can check 
whether there are any problems with the encoding?

To enable Q.931 trace:
ftdm sangoma_isdn trace q931 <span name>

And add this line in your dialplan before bridging to freetdm:
<action application="log" data="caller_id_name:[${caller_id_name}]"/>

Make an outbound call.

And send me the Freeswitch logs.

Regards,
David

*David Yat Sin, BEng*
*/Senior Software Engineer/*
Sangoma Technologies
100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada

	



t.   +1 800 388 2475 x119
t.   +1 905 474 1990 x119
f.   +1 905 474 9223

	

Description: SANGOMA 
<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>

Products 
<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 
| Solutions 
<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 
| Events 
<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 
| Contact 
<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 
| Wiki 
<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 
| Facebook 
<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 
| Twitter 
<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>


On 1/27/2011 4:15 AM, Nikolay Kondratyev wrote:
> sip isdn name interworking
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From infos at madovsky.org  Mon Jan 31 19:00:50 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 31 Jan 2011 11:00:50 -0500
Subject: [Freeswitch-users] open g729
References: <1296343422.2615.4.camel@gustavo-laptop><AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com><1296477877.3589.24.camel@gustavo-laptop><AANLkTinq9AmN=ZqjGLNUxucw2nJXBLpNHKLTA5jLWfgj@mail.gmail.com><9D7F812E93DC47A99F6A30638FB8092F@e1705>
	<AANLkTikkLnf4BHYgL1k8th=VWk44vpH5gRYSAFW5YfW1@mail.gmail.com>
Message-ID: <A64FADA67C6D4C1C8B041EEFBB8C1DBB@e1705>

wholesale offer also GSM and G726
  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 31, 2011 10:54 AM
  Subject: Re: [Freeswitch-users] open g729


  Not always possible, for instance in wholesale it's pretty much the standard codec used.

  -Steve



  On 31 January 2011 15:48, Madovsky <infos at madovsky.org> wrote:

    so the best is,
    don't use G729 and go to buy food with your $10 ;)
      ----- Original Message ----- 
      From: Steven Ayre 
      To: FreeSWITCH Users Help 
      Sent: Monday, January 31, 2011 7:53 AM
      Subject: Re: [Freeswitch-users] open g729


      The price is set by the patent holders, there's not much that can be done about that.

      If  you want to evaluate the g729 transcoding performance you *must* test with the one you plan to recommend - the mod_com_g729, mod_sangoma_codec and ipp g729 codecs have very difference performance characteristics. It is pointless to evaluate one and then recommend another.

      - mod_com_g729 is the official $10/channel license. It has good performance. To evaluate it you should try it for yourself.
      - The unofficial illegal FS module based ipp g729 which I shall not name has stability problems and uses far more CPU than mod_com_g729. Do not use it.
      - mod_sangoma_codec uses almost no CPU, but you do have to purchase a transcoding card.

      -Steve




      On 31 January 2011 12:44, Gustavo Espeche <gustavo.espeche at upper-soft.com> wrote:

        I just need test the performance of trans-coding before to recommend to
        may customer what server it need and buy the codec, because of then i
        need testing in my lab how much cpu and memory require the trans-coding
        in freeswitch for g729.
        And searching i found the open-g729/g723 for asterisk.
        I relay don't wish to start a discussion about the g729 license i read
        and understated it. But i think that the license price per channel is to
        hight taking care that this codec has more than 10 years but the way
        this is only my opinion.


        Best Regards.
        Gustavo Espeche
        www.easyipcall.com



        On Sat, 2011-01-29 at 17:27 -0600, Rupa Schomaker wrote:
        > How do you intend to pay the license fee?
        >
        > On Sat, Jan 29, 2011 at 5:23 PM, Gustavo Espeche
        > <gustavo.espeche at upper-soft.com> wrote:
        > > Hello,
        > >    some one can compile open g729 to work with freeswitch?
        > >    http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
        > >
        > >    i appreciate a lot if some one has some experience in it.
        > >    Best Regards.
        > >
        > >    Gustavo Espeche
        > >    www.easyipcall.com
        > >
        > >
        > >
        > >
        > >
        > > _______________________________________________
        > > FreeSWITCH-users mailing list
        > > FreeSWITCH-users at lists.freeswitch.org
        > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        > > http://www.freeswitch.org
        > >
        >
        >
        >



        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org





--------------------------------------------------------------------------


      _______________________________________________
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      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org


    _______________________________________________
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    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org






------------------------------------------------------------------------------


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From steveayre at gmail.com  Mon Jan 31 19:31:31 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 31 Jan 2011 16:31:31 +0000
Subject: [Freeswitch-users] open g729
In-Reply-To: <AANLkTikNcipSGxfofujQwSNn4E16F-97u51xLB8a=cLB@mail.gmail.com>
References: <1296343422.2615.4.camel@gustavo-laptop>
	<AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com>
	<1296477877.3589.24.camel@gustavo-laptop>
	<AANLkTinq9AmN=ZqjGLNUxucw2nJXBLpNHKLTA5jLWfgj@mail.gmail.com>
	<AANLkTikNcipSGxfofujQwSNn4E16F-97u51xLB8a=cLB@mail.gmail.com>
Message-ID: <AANLkTinMsOeVicnszEaOB-28bvychOPW5HBAF+WidTt_@mail.gmail.com>

I was talking more about the FS module that I won't name that's based on
IPP, not IPP itself.

-Steve


On 31 January 2011 15:54, Anton VG <anton.vazir at gmail.com> wrote:

> IPP has a fairly good performance, at least on asterisk it performs
> as good as proprietary one, works perfectly fine, has no stability
> problems, and I highly doubt that FS situation differs, unless someone
> can prove it by posting real performance test numbers.
>
> 2011/1/31 Steven Ayre <steveayre at gmail.com>:
> > The price is set by the patent holders, there's not much that can be done
> > about that.
> >
> > If  you want to evaluate the g729 transcoding performance you *must* test
> > with the one you plan to recommend - the mod_com_g729, mod_sangoma_codec
> and
> > ipp g729 codecs have very difference performance characteristics. It is
> > pointless to evaluate one and then recommend another.
> >
> > - mod_com_g729 is the official $10/channel license. It has good
> performance.
> > To evaluate it you should try it for yourself.
> > - The unofficial illegal FS module based ipp g729 which I shall not name
> has
> > stability problems and uses far more CPU than mod_com_g729. Do not use
> it.
> > - mod_sangoma_codec uses almost no CPU, but you do have to purchase a
> > transcoding card.
> >
> > -Steve
> >
> >
> >
> > On 31 January 2011 12:44, Gustavo Espeche <
> gustavo.espeche at upper-soft.com>
> > wrote:
> >>
> >> I just need test the performance of trans-coding before to recommend to
> >> may customer what server it need and buy the codec, because of then i
> >> need testing in my lab how much cpu and memory require the trans-coding
> >> in freeswitch for g729.
> >> And searching i found the open-g729/g723 for asterisk.
> >> I relay don't wish to start a discussion about the g729 license i read
> >> and understated it. But i think that the license price per channel is to
> >> hight taking care that this codec has more than 10 years but the way
> >> this is only my opinion.
> >>
> >> Best Regards.
> >> Gustavo Espeche
> >> www.easyipcall.com
> >>
> >>
> >> On Sat, 2011-01-29 at 17:27 -0600, Rupa Schomaker wrote:
> >> > How do you intend to pay the license fee?
> >> >
> >> > On Sat, Jan 29, 2011 at 5:23 PM, Gustavo Espeche
> >> > <gustavo.espeche at upper-soft.com> wrote:
> >> > > Hello,
> >> > >    some one can compile open g729 to work with freeswitch?
> >> > >    http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
> >> > >
> >> > >    i appreciate a lot if some one has some experience in it.
> >> > >    Best Regards.
> >> > >
> >> > >    Gustavo Espeche
> >> > >    www.easyipcall.com
> >> > >
> >> > >
> >> > >
> >> > >
> >> > >
> >> > > _______________________________________________
> >> > > FreeSWITCH-users mailing list
> >> > > FreeSWITCH-users at lists.freeswitch.org
> >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > >
> >> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > > http://www.freeswitch.org
> >> > >
> >> >
> >> >
> >> >
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Mon Jan 31 19:32:47 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 31 Jan 2011 16:32:47 +0000
Subject: [Freeswitch-users] open g729
In-Reply-To: <A64FADA67C6D4C1C8B041EEFBB8C1DBB@e1705>
References: <1296343422.2615.4.camel@gustavo-laptop>
	<AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com>
	<1296477877.3589.24.camel@gustavo-laptop>
	<AANLkTinq9AmN=ZqjGLNUxucw2nJXBLpNHKLTA5jLWfgj@mail.gmail.com>
	<9D7F812E93DC47A99F6A30638FB8092F@e1705>
	<AANLkTikkLnf4BHYgL1k8th=VWk44vpH5gRYSAFW5YfW1@mail.gmail.com>
	<A64FADA67C6D4C1C8B041EEFBB8C1DBB@e1705>
Message-ID: <AANLkTi=1Fr2e-sz-n5Ds=q6KVntRzfzgrAQOgc0rCYgF@mail.gmail.com>

Depends on who your customer/provider is.

90% of our customers only support G711 and G729. Which is just fine since
most of our gateways only support G711, G729 and G723.1 so we stick with
G729 and don't need to transcode. :)

-Steve


On 31 January 2011 16:00, Madovsky <infos at madovsky.org> wrote:

>  wholesale offer also GSM and G726
>
> ----- Original Message -----
> *From:* Steven Ayre <steveayre at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, January 31, 2011 10:54 AM
> *Subject:* Re: [Freeswitch-users] open g729
>
> Not always possible, for instance in wholesale it's pretty much the
> standard codec used.
>
> -Steve
>
>
> On 31 January 2011 15:48, Madovsky <infos at madovsky.org> wrote:
>
>>  so the best is,
>> don't use G729 and go to buy food with your $10 ;)
>>
>>  ----- Original Message -----
>> *From:* Steven Ayre <steveayre at gmail.com>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>   *Sent:* Monday, January 31, 2011 7:53 AM
>> *Subject:* Re: [Freeswitch-users] open g729
>>
>> The price is set by the patent holders, there's not much that can be done
>> about that.
>>
>> If  you want to evaluate the g729 transcoding performance you **must**test with the one you plan to recommend - the mod_com_g729,
>> mod_sangoma_codec and ipp g729 codecs have very difference performance
>> characteristics. It is pointless to evaluate one and then recommend another.
>>
>> - mod_com_g729 is the official $10/channel license. It has good
>> performance. To evaluate it you should try it for yourself.
>> - The unofficial illegal FS module based ipp g729 which I shall not name
>> has stability problems and uses far more CPU than mod_com_g729. Do not use
>> it.
>> - mod_sangoma_codec uses almost no CPU, but you do have to purchase a
>> transcoding card.
>>
>> -Steve
>>
>>
>>
>> On 31 January 2011 12:44, Gustavo Espeche <gustavo.espeche at upper-soft.com
>> > wrote:
>>
>>> I just need test the performance of trans-coding before to recommend to
>>> may customer what server it need and buy the codec, because of then i
>>> need testing in my lab how much cpu and memory require the trans-coding
>>> in freeswitch for g729.
>>> And searching i found the open-g729/g723 for asterisk.
>>> I relay don't wish to start a discussion about the g729 license i read
>>> and understated it. But i think that the license price per channel is to
>>> hight taking care that this codec has more than 10 years but the way
>>> this is only my opinion.
>>>
>>> Best Regards.
>>> Gustavo Espeche
>>> www.easyipcall.com
>>>
>>>
>>>  On Sat, 2011-01-29 at 17:27 -0600, Rupa Schomaker wrote:
>>> > How do you intend to pay the license fee?
>>> >
>>> > On Sat, Jan 29, 2011 at 5:23 PM, Gustavo Espeche
>>> > <gustavo.espeche at upper-soft.com> wrote:
>>> > > Hello,
>>> > >    some one can compile open g729 to work with freeswitch?
>>> > >    http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
>>> > >
>>> > >    i appreciate a lot if some one has some experience in it.
>>> > >    Best Regards.
>>> > >
>>> > >    Gustavo Espeche
>>> > >    www.easyipcall.com
>>> > >
>>> > >
>>> > >
>>> > >
>>> > >
>>> > > _______________________________________________
>>> > > FreeSWITCH-users mailing list
>>> > > FreeSWITCH-users at lists.freeswitch.org
>>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > > http://www.freeswitch.org
>>> > >
>>> >
>>> >
>>> >
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>  ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Mon Jan 31 19:33:41 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 31 Jan 2011 10:33:41 -0600
Subject: [Freeswitch-users] blocking 183 w/o sdp
In-Reply-To: <AANLkTik6dUTrSLY+FYCBWQ2gsNQViCK4GW2-UnQPYmE=@mail.gmail.com>
References: <AANLkTinH1s+Uy-EPvX9oV-x+aXYkyzUEBSYm+hQ+-+K-@mail.gmail.com>
	<AANLkTikJpEPvcQu2reAKXk1wbW7fV4myvXghBCuu2QHt@mail.gmail.com>
	<AANLkTikuQFC2N3CYzreTvCdk8-28MYGUbrP=T83SfPtm@mail.gmail.com>
	<AANLkTinyV-kAWCOiuOVL5jmnL7f9ck00=i_giHY_XGd_@mail.gmail.com>
	<AANLkTik6dUTrSLY+FYCBWQ2gsNQViCK4GW2-UnQPYmE=@mail.gmail.com>
Message-ID: <AANLkTikc1prUQweEsbHt94G6AUtPAkHrzf9KB2Dc3QEE@mail.gmail.com>

If it does not work for you, your version of FreeSWITCH may be too old
for this particular feature.
Did you try with the latest release snapshot?


On Sun, Jan 30, 2011 at 10:00 PM, Sam <u2nsam at gmail.com> wrote:
> Hi,
>
> After using , <action application="bridge"
> data="{sip_contact_user=${contact},sip_ignore_183nosdp=true}sofia/sbc/$1@${dialed_domain}"/>
>
> the 183 without udp is not blocked/ignored .
>
> Below are the traces to visualize:
> 192.168.2.98 is provider
> 192.168.2.16 is FS
>
>
> U 192.168.2.98:5060 -> 192.168.2.16:5060
> SIP/2.0 183 Session Progress.
> Via: SIP/2.0/UDP 192.168.2.16;rport;branch=z9hG4bKjQBQg7vy0y2SF.
> To: <sip:599261244747199 at 192.168.2.16>;tag=3505434022-138257.
> From: "0280910101" <sip:0280910101 at 192.168.2.16>;tag=51SjQQQUX14QF.
> Call-ID: 6ade3e61-a78e-122e-9698-00137256e1a2.
> CSeq: 7886492 INVITE.
> Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
> SUBSCRIBE, PRACK, UPDATE.
> Contact: <sip:599261244747199 at 192.168.2.98:5060>.
> Call-Info:
> <sip:192.168.2.98>;method="NOTIFY;Event=telephone-event;Duration=1000".
> Content-Length: 0.
> .
>
>
> U 192.168.2.16:5060 -> 192.168.2.6:5060
> SIP/2.0 180 Ringing.
> Via: SIP/2.0/UDP 192.168.2.6;branch=z9hG4bK3fb.51f5e6e1.0.
> Via: SIP/2.0/UDP
> 192.168.2.158:5060;received=192.168.2.158;rport=5060;branch=z9hG4bK-f33ff5a-631d780c-50647c36.
> Record-Route:
> <sip:192.168.2.6;lr=on;ftag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a>.
> From: "0280910101"
> <sip:0280910101 at 192.168.2.158>;tag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a.
> To: <sip:61244747199 at 192.168.2.6>;tag=3F70K1Nm3Frjr.
> Call-ID: 100f6c88-9e3599cb-13c4-50029-f33ff5a-631c036b-f33ff5a.
> CSeq: 1 INVITE.
> Contact: <sip:61244747199 at 192.168.2.16:5060;transport=udp>.
> User-Agent:? SBC.
> Accept: application/sdp.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, refer.
> Content-Length: 0.
> Remote-Party-ID: "599261244747199"
> <sip:599261244747199 at 192.168.2.16>;party=calling;privacy=off;screen=no.
> .
>
>
> U 192.168.2.98:5060 -> 192.168.2.16:5060
> SIP/2.0 180 Ringing.
> Via: SIP/2.0/UDP 192.168.2.16;rport;branch=z9hG4bKjQBQg7vy0y2SF.
> To: <sip:599261244747199 at 192.168.2.98>;tag=3505434022-138257.
> From: "0280910101" <sip:0280910101 at 192.168.2.16>;tag=51SjQQQUX14QF.
> Call-ID: 6ade3e61-a78e-122e-9698-00137256e1a2.
> CSeq: 7886492 INVITE.
> Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
> SUBSCRIBE, PRACK, UPDATE.
> Contact: <sip:599261244747199 at 192.168.2.98:5060>.
> Call-Info:
> <sip:192.168.2.98>;method="NOTIFY;Event=telephone-event;Duration=1000".
> Content-Type: application/sdp.
> Content-Length: 209.
> .
> v=0.
> o=vsnl2 770 13521 IN IP4 192.168.2.98.
> s=sip call.
> c=IN IP4 115.113.121.99.
> t=0 0.
> m=audio 49034 RTP/AVP 18 101.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-11.
> a=ptime:20.
> a=rtpmap:18 G729/8000/1.
>
>
> U 192.168.2.16:5060 -> 192.168.2.6:5060
> SIP/2.0 183 Session Progress.
> Via: SIP/2.0/UDP 192.168.2.6;branch=z9hG4bK3fb.51f5e6e1.0.
> Via: SIP/2.0/UDP
> 192.168.2.158:5060;received=192.168.2.158;rport=5060;branch=z9hG4bK-f33ff5a-631d780c-50647c36.
> Record-Route:
> <sip:192.168.2.6;lr=on;ftag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a>.
> From: "0280910101"
> <sip:0280910101 at 192.168.2.158>;tag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a.
> To: <sip:61244747199 at 192.168.2.6>;tag=3F70K1Nm3Frjr.
> Call-ID: 100f6c88-9e3599cb-13c4-50029-f33ff5a-631c036b-f33ff5a.
> CSeq: 1 INVITE.
> Contact: <sip:61244747199 at 192.168.2.16:5060;transport=udp>.
> User-Agent:? SBC.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 212.
> Remote-Party-ID: "599261244747199"
> <sip:599261244747199 at 192.168.2.16>;party=calling;privacy=off;screen=no.
> .
> v=0.
> o=SBC 1019267468 1019267469 IN IP4 192.168.2.16.
> s=SBC.
> c=IN IP4 192.168.2.16.
> t=0 0.
> m=audio 16922 RTP/AVP 18 101.
> a=rtpmap:18 G729/8000/1.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-11.
> a=ptime:20.
>
>
>
>
> Regds
> Sam
>
>
>
>
>
> On Sat, Jan 29, 2011 at 3:16 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> Close. You can only have one set of {} brackets. You can separate multiple
>> variables with a comma.
>>
>> <action application="bridge"
>> data="{sip_contact_user=${contact},sip_ignore_183nosdp=true}sofia/sbc/$1@${dialed_domain}"/>
>>
>> -Steve
>>
>>
>> On 29 January 2011 04:29, Sam <u2nsam at gmail.com> wrote:
>>>
>>> Hi,
>>>
>>> So you say i need to put
>>> <action application="bridge"
>>> data="{sip_contact_user=${contact}}{sip_ignore_183nosdp=true}sofia/sbc/$1@${dialed_domain}"/>
>>>
>>> Regds
>>> Sam
>>>
>>>
>>>
>>>
>>> On Fri, Jan 28, 2011 at 10:23 PM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>>>
>>>> you need sip_ignore_183nosdp=true set on the b leg not the a leg.
>>>> Put it in the dial string in {}
>>>>
>>>> {sip_ignore_183nosdp=true}sofia/foo/foo at bar.com
>>>>
>>>>
>>>> On Fri, Jan 28, 2011 at 12:41 AM, Sam <u2nsam at gmail.com> wrote:
>>>> > Hi,
>>>> >
>>>> > how can i ignore 183 without sdp,
>>>> > what happens is the provider sends 183 without sdp and by applying
>>>> > "<action
>>>> > application="set" data="sip_ignore_183nosdp=true"/>"? the FS sends 180
>>>> > to
>>>> > the leg a.
>>>> > Here i want to block the 183 with SDP just like router as b2bua and
>>>> > send
>>>> > nothing to leg a, and when actual 183 with sdp comes it should send .
>>>> >
>>>> > Its because, providers are sending false signaling by sending 183
>>>> > without
>>>> > sdp,and it hampers while @ production,
>>>> > Although by cisco sbc i have done this but i want to do it by FS,
>>>> > Take a scenario, when call is send 183 without sdp for 10 secs and
>>>> > then
>>>> > followed by 183 with sdp ( actual signal),
>>>> > but when some one dials invalid number it rings for 10 secs and then
>>>> > gives
>>>> > SIP cause 404, which is bad from the providers.
>>>> > So this is the reason i want to block it.
>>>> >
>>>> > Most of the providers do this, the way out is blocking.
>>>> >
>>>> > I have got an advice from Tihomir? to do "execute_on_ring and parse
>>>> > your 180
>>>> > / 183 messages in search of SDP,
>>>> > once you get 183 without SDP do not send it back to leg a and send
>>>> > signal
>>>> > only when you got 183 with sdp or 180 "
>>>> > And this could be valid call flow.
>>>> >
>>>> > This happens in many cases where the provider is having nextone as a
>>>> > sbc and
>>>> > that too tier 1 !
>>>> >
>>>> > Regards
>>>> > Sam
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >
>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Mon Jan 31 19:34:11 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 31 Jan 2011 10:34:11 -0600
Subject: [Freeswitch-users] blocking 183 w/o sdp
In-Reply-To: <AANLkTikc1prUQweEsbHt94G6AUtPAkHrzf9KB2Dc3QEE@mail.gmail.com>
References: <AANLkTinH1s+Uy-EPvX9oV-x+aXYkyzUEBSYm+hQ+-+K-@mail.gmail.com>
	<AANLkTikJpEPvcQu2reAKXk1wbW7fV4myvXghBCuu2QHt@mail.gmail.com>
	<AANLkTikuQFC2N3CYzreTvCdk8-28MYGUbrP=T83SfPtm@mail.gmail.com>
	<AANLkTinyV-kAWCOiuOVL5jmnL7f9ck00=i_giHY_XGd_@mail.gmail.com>
	<AANLkTik6dUTrSLY+FYCBWQ2gsNQViCK4GW2-UnQPYmE=@mail.gmail.com>
	<AANLkTikc1prUQweEsbHt94G6AUtPAkHrzf9KB2Dc3QEE@mail.gmail.com>
Message-ID: <AANLkTikX=zn5+iB1hY_6No-QwfaYtVBPoSufsXgPKdUr@mail.gmail.com>

Also your trace is incomplete,

you are best off with

sofia global siptrace on
console loglevel debug



On Mon, Jan 31, 2011 at 10:33 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> If it does not work for you, your version of FreeSWITCH may be too old
> for this particular feature.
> Did you try with the latest release snapshot?
>
>
> On Sun, Jan 30, 2011 at 10:00 PM, Sam <u2nsam at gmail.com> wrote:
>> Hi,
>>
>> After using , <action application="bridge"
>> data="{sip_contact_user=${contact},sip_ignore_183nosdp=true}sofia/sbc/$1@${dialed_domain}"/>
>>
>> the 183 without udp is not blocked/ignored .
>>
>> Below are the traces to visualize:
>> 192.168.2.98 is provider
>> 192.168.2.16 is FS
>>
>>
>> U 192.168.2.98:5060 -> 192.168.2.16:5060
>> SIP/2.0 183 Session Progress.
>> Via: SIP/2.0/UDP 192.168.2.16;rport;branch=z9hG4bKjQBQg7vy0y2SF.
>> To: <sip:599261244747199 at 192.168.2.16>;tag=3505434022-138257.
>> From: "0280910101" <sip:0280910101 at 192.168.2.16>;tag=51SjQQQUX14QF.
>> Call-ID: 6ade3e61-a78e-122e-9698-00137256e1a2.
>> CSeq: 7886492 INVITE.
>> Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
>> SUBSCRIBE, PRACK, UPDATE.
>> Contact: <sip:599261244747199 at 192.168.2.98:5060>.
>> Call-Info:
>> <sip:192.168.2.98>;method="NOTIFY;Event=telephone-event;Duration=1000".
>> Content-Length: 0.
>> .
>>
>>
>> U 192.168.2.16:5060 -> 192.168.2.6:5060
>> SIP/2.0 180 Ringing.
>> Via: SIP/2.0/UDP 192.168.2.6;branch=z9hG4bK3fb.51f5e6e1.0.
>> Via: SIP/2.0/UDP
>> 192.168.2.158:5060;received=192.168.2.158;rport=5060;branch=z9hG4bK-f33ff5a-631d780c-50647c36.
>> Record-Route:
>> <sip:192.168.2.6;lr=on;ftag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a>.
>> From: "0280910101"
>> <sip:0280910101 at 192.168.2.158>;tag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a.
>> To: <sip:61244747199 at 192.168.2.6>;tag=3F70K1Nm3Frjr.
>> Call-ID: 100f6c88-9e3599cb-13c4-50029-f33ff5a-631c036b-f33ff5a.
>> CSeq: 1 INVITE.
>> Contact: <sip:61244747199 at 192.168.2.16:5060;transport=udp>.
>> User-Agent:? SBC.
>> Accept: application/sdp.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
>> REFER, NOTIFY.
>> Supported: timer, precondition, path, replaces.
>> Allow-Events: talk, refer.
>> Content-Length: 0.
>> Remote-Party-ID: "599261244747199"
>> <sip:599261244747199 at 192.168.2.16>;party=calling;privacy=off;screen=no.
>> .
>>
>>
>> U 192.168.2.98:5060 -> 192.168.2.16:5060
>> SIP/2.0 180 Ringing.
>> Via: SIP/2.0/UDP 192.168.2.16;rport;branch=z9hG4bKjQBQg7vy0y2SF.
>> To: <sip:599261244747199 at 192.168.2.98>;tag=3505434022-138257.
>> From: "0280910101" <sip:0280910101 at 192.168.2.16>;tag=51SjQQQUX14QF.
>> Call-ID: 6ade3e61-a78e-122e-9698-00137256e1a2.
>> CSeq: 7886492 INVITE.
>> Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
>> SUBSCRIBE, PRACK, UPDATE.
>> Contact: <sip:599261244747199 at 192.168.2.98:5060>.
>> Call-Info:
>> <sip:192.168.2.98>;method="NOTIFY;Event=telephone-event;Duration=1000".
>> Content-Type: application/sdp.
>> Content-Length: 209.
>> .
>> v=0.
>> o=vsnl2 770 13521 IN IP4 192.168.2.98.
>> s=sip call.
>> c=IN IP4 115.113.121.99.
>> t=0 0.
>> m=audio 49034 RTP/AVP 18 101.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-11.
>> a=ptime:20.
>> a=rtpmap:18 G729/8000/1.
>>
>>
>> U 192.168.2.16:5060 -> 192.168.2.6:5060
>> SIP/2.0 183 Session Progress.
>> Via: SIP/2.0/UDP 192.168.2.6;branch=z9hG4bK3fb.51f5e6e1.0.
>> Via: SIP/2.0/UDP
>> 192.168.2.158:5060;received=192.168.2.158;rport=5060;branch=z9hG4bK-f33ff5a-631d780c-50647c36.
>> Record-Route:
>> <sip:192.168.2.6;lr=on;ftag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a>.
>> From: "0280910101"
>> <sip:0280910101 at 192.168.2.158>;tag=100ea820-9e3599cb-13c4-50029-f33ff5a-62495c41-f33ff5a.
>> To: <sip:61244747199 at 192.168.2.6>;tag=3F70K1Nm3Frjr.
>> Call-ID: 100f6c88-9e3599cb-13c4-50029-f33ff5a-631c036b-f33ff5a.
>> CSeq: 1 INVITE.
>> Contact: <sip:61244747199 at 192.168.2.16:5060;transport=udp>.
>> User-Agent:? SBC.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
>> REFER, NOTIFY.
>> Supported: timer, precondition, path, replaces.
>> Allow-Events: talk, refer.
>> Content-Type: application/sdp.
>> Content-Disposition: session.
>> Content-Length: 212.
>> Remote-Party-ID: "599261244747199"
>> <sip:599261244747199 at 192.168.2.16>;party=calling;privacy=off;screen=no.
>> .
>> v=0.
>> o=SBC 1019267468 1019267469 IN IP4 192.168.2.16.
>> s=SBC.
>> c=IN IP4 192.168.2.16.
>> t=0 0.
>> m=audio 16922 RTP/AVP 18 101.
>> a=rtpmap:18 G729/8000/1.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-11.
>> a=ptime:20.
>>
>>
>>
>>
>> Regds
>> Sam
>>
>>
>>
>>
>>
>> On Sat, Jan 29, 2011 at 3:16 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>>
>>> Close. You can only have one set of {} brackets. You can separate multiple
>>> variables with a comma.
>>>
>>> <action application="bridge"
>>> data="{sip_contact_user=${contact},sip_ignore_183nosdp=true}sofia/sbc/$1@${dialed_domain}"/>
>>>
>>> -Steve
>>>
>>>
>>> On 29 January 2011 04:29, Sam <u2nsam at gmail.com> wrote:
>>>>
>>>> Hi,
>>>>
>>>> So you say i need to put
>>>> <action application="bridge"
>>>> data="{sip_contact_user=${contact}}{sip_ignore_183nosdp=true}sofia/sbc/$1@${dialed_domain}"/>
>>>>
>>>> Regds
>>>> Sam
>>>>
>>>>
>>>>
>>>>
>>>> On Fri, Jan 28, 2011 at 10:23 PM, Anthony Minessale
>>>> <anthony.minessale at gmail.com> wrote:
>>>>>
>>>>> you need sip_ignore_183nosdp=true set on the b leg not the a leg.
>>>>> Put it in the dial string in {}
>>>>>
>>>>> {sip_ignore_183nosdp=true}sofia/foo/foo at bar.com
>>>>>
>>>>>
>>>>> On Fri, Jan 28, 2011 at 12:41 AM, Sam <u2nsam at gmail.com> wrote:
>>>>> > Hi,
>>>>> >
>>>>> > how can i ignore 183 without sdp,
>>>>> > what happens is the provider sends 183 without sdp and by applying
>>>>> > "<action
>>>>> > application="set" data="sip_ignore_183nosdp=true"/>"? the FS sends 180
>>>>> > to
>>>>> > the leg a.
>>>>> > Here i want to block the 183 with SDP just like router as b2bua and
>>>>> > send
>>>>> > nothing to leg a, and when actual 183 with sdp comes it should send .
>>>>> >
>>>>> > Its because, providers are sending false signaling by sending 183
>>>>> > without
>>>>> > sdp,and it hampers while @ production,
>>>>> > Although by cisco sbc i have done this but i want to do it by FS,
>>>>> > Take a scenario, when call is send 183 without sdp for 10 secs and
>>>>> > then
>>>>> > followed by 183 with sdp ( actual signal),
>>>>> > but when some one dials invalid number it rings for 10 secs and then
>>>>> > gives
>>>>> > SIP cause 404, which is bad from the providers.
>>>>> > So this is the reason i want to block it.
>>>>> >
>>>>> > Most of the providers do this, the way out is blocking.
>>>>> >
>>>>> > I have got an advice from Tihomir? to do "execute_on_ring and parse
>>>>> > your 180
>>>>> > / 183 messages in search of SDP,
>>>>> > once you get 183 without SDP do not send it back to leg a and send
>>>>> > signal
>>>>> > only when you got 183 with sdp or 180 "
>>>>> > And this could be valid call flow.
>>>>> >
>>>>> > This happens in many cases where the provider is having nextone as a
>>>>> > sbc and
>>>>> > that too tier 1 !
>>>>> >
>>>>> > Regards
>>>>> > Sam
>>>>> >
>>>>> > _______________________________________________
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >
>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>> >
>>>>> >
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org
>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From brian at freeswitch.org  Mon Jan 31 19:33:42 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 31 Jan 2011 10:33:42 -0600
Subject: [Freeswitch-users] open g729
In-Reply-To: <1296343422.2615.4.camel@gustavo-laptop>
References: <1296343422.2615.4.camel@gustavo-laptop>
Message-ID: <491C8800-742B-4D69-87B8-E4F3A62B35EF@freeswitch.org>

Please refrain from posting these types of illegal things.  You put our project and us in a bad position doing this.  It is not legal for you to use G729 in most countries around the world even if you believe otherwise without a license to do so.

/b

On Jan 29, 2011, at 5:23 PM, Gustavo Espeche wrote:

> Hello,
>        some one can compile open g729 to work with freeswitch?
>      
>        i appreciate a lot if some one has some experience in it.
>        Best Regards.
> 
>        Gustavo Espeche
>        www.easyipcall.com

-------------- next part --------------
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From Nabble at slickdeals.endjunk.com  Mon Jan 31 19:37:43 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 31 Jan 2011 08:37:43 -0800 (PST)
Subject: [Freeswitch-users] Problem with mod_shout on FS git
Message-ID: <1296491863148-5977722.post@n2.nabble.com>


Last time, when I ran into problem with mod_shout, I posted 
http://freeswitch-users.2379917.n2.nabble.com/Playing-Google-translation-tts-tp5916138p5920422.html
here . Since then, I did an update with FreeSWITCH Version 1.0.head
(git-49a5eff 2011-01-29 03-09-06 -0500). My same diialplan
(conf/dialplan/default/2115_Google_TTS.xml), copied from 
http://freeswitch-users.2379917.n2.nabble.com/Playing-Google-translation-tts-tp5916138p5916461.html
here , is shown below:
<extension name="Free_Google_Text_To_Speech">
  <condition field="destination_number" expression="^2115$">
    <action application="answer" data=""/>
    <action application="playback"
data="shout://translate.google.com/translate_tts?tl=en&q=FreeSWI
  </condition>
</extension>
The fs_cli dumped the following error messages when I called 2115#:

2011-01-31 11:13:16.647405 [WARNING] sofia_reg.c:1247 SIP auth challenge
(INVITE) on sofia profile 'internal' for [2115 at 192.168.1.123] from ip
192.168.1.15
2011-01-31 11:13:18.656994 [NOTICE] switch_channel.c:808 New Channel
sofia/internal/1003 at 192.168.1.15 [18fd8311-60b0-4c93-b022-51204ae6dd17]
2011-01-31 11:13:20.091366 [INFO] mod_dialplan_xml.c:331 Processing 1003
<1003>->2115 in context default
2011-01-31 11:13:22.339858 [NOTICE] mod_dptools.c:920 Channel
[sofia/internal/1003 at 192.168.1.15] has been answered
2011-01-31 11:13:22.354824 [ERR] mod_shout.c:804 Error: MPG123 Error at
/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.31_eabi/freeswitch_git/src/mod/formats/mod_shout/mod_shout.c:627.
2011-01-31 11:13:22.354824 [ERR] mod_shout.c:807 Error from mpg123: Invalid
mpg123 handle. (code 10)
2011-01-31 11:13:22.360467 [NOTICE] switch_core_state_machine.c:189
sofia/internal/1003 at 192.168.1.15 has executed the last dialplan instruction,
hanging up.
2011-01-31 11:13:22.367752 [NOTICE] switch_core_state_machine.c:191 Hangup
sofia/internal/1003 at 192.168.1.15 [CS_EXECUTE] [NORMAL_CLEARING]
2011-01-31 11:13:24.215896 [NOTICE] switch_core_session.c:1306 Session 7
(sofia/internal/1003 at 192.168.1.15) Ended
2011-01-31 11:13:24.215896 [NOTICE] switch_core_session.c:1308 Close Channel
sofia/internal/1003 at 192.168.1.15 [CS_DESTROY]
The above [ERR] line shows an MPG123 error on mod_shout.c file @line 627. I
don't suppose an MPG123 lib is needed mainly because the mpeg123 codes are
statically linked to mod_shout.so. Can anyone confirm this?

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-with-mod-shout-on-FS-git-tp5977722p5977722.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Mon Jan 31 19:37:44 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 31 Jan 2011 10:37:44 -0600
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTi=+aObW3kVBZq5ox4b4E1uFTSgtXUL72tHLOzPi@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
	<AANLkTi=in3CLd-n0mT3SRKYA0kxLMGVWk9KeBx3_jhY6@mail.gmail.com>
	<AANLkTin3Bu+8EEr_v_GaRGNPzOjfcnfzsV2Av2cgnWgO@mail.gmail.com>
	<AANLkTi=OC_jx0+nzK405VZURLDJPD25R4ZjyJ3RXJ1nk@mail.gmail.com>
	<AANLkTi=Sm7CjoayiE8nHmL0Eehqknj1jdpbx0YBQJQGH@mail.gmail.com>
	<AANLkTinr0RdgBs2UiS3=b3i+k6uCwkMihDy_FbAOFPc_@mail.gmail.com>
	<AANLkTi=+aObW3kVBZq5ox4b4E1uFTSgtXUL72tHLOzPi@mail.gmail.com>
Message-ID: <AANLkTikVDKS40dVsf8x=+5=qnW59LqJbSv1H=vtROFaM@mail.gmail.com>

You need to have it working on the latest GIT and you need to
eliminate presence_in and presence_out 100% and use presence_id
variable as explained.  This is how you do it and it will work the way
you want.
If you think presence does not work on latest, you need to report what
phone and model you are using.  I will guess since you say you are in
Germany and you have presence that is not working that it's a SNOM.
Is that correct?


On Mon, Jan 31, 2011 at 7:10 AM, Patrick Plattes
<patrick.plattes at niemann-frey.info> wrote:
> Hi,
>
> thanks for your answer. I can't get it to work on 'current'. I'm also
> not able to use presence in in the current, but it works fine with
> 1.0.6. Is there anything broken?
>
> I've done exactly the same with exactly the same configuration.
>
> FS 1.0.6: http://fs.kwixo.de/fs-1.0.6.pcap
> presence in queue-sales_fr-263 at 192.168.56.189 inuse 'inuse'
> presence out queue-sales_fr-263 at 192.168.56.189 inuse 'inuse'
>
> FS current: http://fs.kwixo.de/fs-current-20110131.pcap
> presence in queue-sales_fr-263 at 192.168.56.189 inuse 'inuse'
> presence out queue-sales_fr-263 at 192.168.56.189 inuse 'inuse'
>
>
> Thanks,
> ?Patrick
>
> 2011/1/28 Anthony Minessale <anthony.minessale at gmail.com>:
>> try with no presence_in and out commands at all
>>
>> just set presence_id variable as soon as you can and that call will
>> report its state as that id for its entire duration.
>> Also make sure you are on latest GIT when you test this.
>>
>>
>> On Fri, Jan 28, 2011 at 7:47 AM, Patrick Plattes
>> <patrick.plattes at niemann-frey.info> wrote:
>>> Formatted and colourized code: http://pastebin.freeswitch.org/15172
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Patrick Plattes
> IT - Projektleiter
>
> Niemann + Frey GmbH
> Adolf-Dembach-Str. 24
> 47829 Krefeld
>
> Tel.: +49/2151 - 5554-263
> Fax : +49/2151 - 5554-123
> patrick.plattes at niemann-frey.info
>
> Gesch?ftsf?hrer: Gerd Frey
> Sitz und Registergericht: Krefeld HRB 10851
>
> www.niemann-frey.de
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Mon Jan 31 19:40:10 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 31 Jan 2011 10:40:10 -0600
Subject: [Freeswitch-users] Outbound only calls don't connect when
 bypass_media is true.
In-Reply-To: <COL118-W196844C55E71088CAD1FC1E5E20@phx.gbl>
References: <COL118-W52E5226FC3337085AB6AFAE5E00@phx.gbl>
	<COL118-W38F26048D441DAA2777942E5E30@phx.gbl>
	<AANLkTinbiU4zNc-ZM+u+iST7E2aZ3zisgh5Lu2FTuBLt@mail.gmail.com>
	<COL118-W196844C55E71088CAD1FC1E5E20@phx.gbl>
Message-ID: <AANLkTi=t0ixs6L3SgUMQzEPtgvFowbFYXCVOtmvzLTEW@mail.gmail.com>

Many things have problems doing iLBC right.
I recommend you define it in your configs as iLBC at 30i or it will try
using the 20ms version which is not compatible with many other
platforms.  Also make sure you are on the latest version of FS since
we have tweaked iLBC behavior to compensate for problems like this.



On Sun, Jan 30, 2011 at 10:40 PM, Marcin Wojtowicz
<marcin321 at hotmail.com> wrote:
> OK, so I gave up on bypass media, but now I have another problem. This time
> I set up freeswitch to communicate with voip.ms using PCMU codec (configured
> in my external profile), and use iLBC on my phone (codec configured in my
> internal profile, where the phone registers). When I call my mobile it
> rings, but when I pick up all I hear is a high pitched squeal. Am I missing
> something here?
>
>> Date: Sun, 30 Jan 2011 16:34:09 -0600
>> From: anthony.minessale at gmail.com
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Outbound only calls don't connect when
>> bypass_media is true.
>>
>> Just do not use bypass media.
>> That is all you can do in that situation.
>>
>>
>> On Sun, Jan 30, 2011 at 3:44 PM, Marcin Wojtowicz <marcin321 at hotmail.com>
>> wrote:
>> > I just want to add that I enabled STUN on my cell so now the SDP message
>> > in
>> > the INVITE to voip.ms contains the public IP of my phone, but it still
>> > doesn't work.
>> >
>> > ________________________________
>> > From: marcin321 at hotmail.com
>> > To: freeswitch-users at lists.freeswitch.org
>> > Date: Fri, 28 Jan 2011 19:54:19 -0500
>> > Subject: [Freeswitch-users] Outbound only calls don't connect when
>> > bypass_media is true.
>> >
>> > Hello,
>> >
>> > I'm a new user of freeswitch, so please bear with me. I have the
>> > following setup:
>> > voip.ms <- SIP over UDP -> my desktop running freeswitch <- SIP over TCP
>> > ->
>> > my nokia cellphone on AT&T wireless. This setup is intended to conserve
>> > the
>> > battery usage.
>> > I've managed to make everything work well when I'm calling in over any
>> > phone
>> > to my cell phone, and freeswitch is enabled to work in bypass_media =
>> > true,
>> > even though by cell is behind NAT on at&t's network. Things break when I
>> > pick up my cell and try to call my home phone (or any phone for that
>> > matter). This is the relevant snippet from my dialplan:
>> > <extension name="outbound">
>> > ? <condition field="destination_number"
>> > expression="^1?([2-9]\d{2}[2-9]\d{6})$">
>> > ??? <!--<action application="set" data="bypass_media=true"/>-->
>> > ??? <action application="bridge" data="sofia/gateway/voip.ms/1$1"/>
>> > ? </condition>
>> > </extension>
>> >
>> > Like shown above, my call will go to my home phone. When I uncomment the
>> > bypass_media tag, my call will not connect. Here are the siptraces
>> > I replaced my real home phone number in the with "MYPHONE".
>> >
>> > recv 940 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.406250:
>> >
>> > ------------------------------------------------------------------------
>> > ?? INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
>> > ?? Via: SIP/2.0/TCP
>> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport
>> > ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>> > ?? To: <sip:MYPHONE at 192.168.1.100>
>> > ?? Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>
>> > ?? Supported: 100rel,timer
>> > ?? CSeq: 5244503 INVITE
>> > ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
>> > ?? Allow:
>> > UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE
>> > ?? User-Agent: S60 RM-624 v 20.2.042 (en)
>> > ?? Expires: 120
>> > ?? Privacy: None
>> > ?? Session-Expires: 1800
>> > ?? Max-Forwards: 70
>> > ?? Content-Type: application/sdp
>> > ?? Accept-Language: en
>> > ?? Content-Length: 292
>> >
>> > ?? v=0
>> > ?? o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
>> > ?? s=-
>> > ?? c=IN IP4 10.153.174.6
>> > ?? t=0 0
>> > ?? m=audio 49152 RTP/AVP 18 97 98
>> > ?? a=sendrecv
>> > ?? a=rtpmap:18 G729/8000
>> > ?? a=ptime:20
>> > ?? a=maxptime:40
>> > ?? a=fmtp:18 annexb=no
>> > ?? a=rtpmap:97 iLBC/8000
>> > ?? a=rtpmap:98 telephone-event/8000
>> > ?? a=fmtp:98 0-15
>> >
>> > ------------------------------------------------------------------------
>> > send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:
>> >
>> > ------------------------------------------------------------------------
>> > ?? SIP/2.0 100 Trying
>> > ?? Via: SIP/2.0/TCP
>> >
>> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180
>> > ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>> > ?? To: <sip:MYPHONE at 192.168.1.100>
>> > ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
>> > ?? CSeq: 5244503 INVITE
>> > ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
>> > 18-04-05
>> > -0600
>> > ?? Content-Length: 0
>> >
>> >
>> > ------------------------------------------------------------------------
>> > send 875 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:
>> >
>> > ------------------------------------------------------------------------
>> > ?? SIP/2.0 407 Proxy Authentication Required
>> > ?? Via: SIP/2.0/TCP
>> >
>> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180
>> > ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj2011-01-28
>> > 16:15:58.406250 [WARNING] sofia_reg.c:1241 SIP auth challenge (INVITE)
>> > on
>> > sofia profile 'internal' for [MYPHONE at 192.168.1.100] from ip
>> > 32.136.78.180
>> >
>> > ?? To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m
>> > ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
>> > ?? CSeq: 5244503 INVITE
>> > ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
>> > 18-04-05
>> > -0600
>> > ?? Accept: application/sdp
>> > ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> > ?? Supported: timer, precondition, path, replaces
>> > ?? Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>> > sla,
>> > include-session-description, presence.winfo, message-summary, refer
>> > ?? Proxy-Authenticate: Digest realm="192.168.1.100",
>> > nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28", algorithm=MD5, qop="auth"
>> > ?? Content-Length: 0
>> >
>> >
>> > ------------------------------------------------------------------------
>> > recv 368 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.765625:
>> >
>> > ------------------------------------------------------------------------
>> > ?? ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
>> > ?? Via: SIP/2.0/TCP
>> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport
>> > ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>> > ?? To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m
>> > ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
>> > ?? CSeq: 5244503 ACK
>> > ?? Supported: sec-agree
>> > ?? Max-Forwards: 70
>> > ?? Content-Length: 0
>> >
>> >
>> > ------------------------------------------------------------------------
>> > recv 1222 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.406250:
>> >
>> > ------------------------------------------------------------------------
>> > ?? INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
>> > ?? Via: SIP/2.0/TCP
>> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport
>> > ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>> > ?? To: <sip:MYPHONE at 192.168.1.100>
>> > ?? Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>
>> > ?? Supported: 100rel,timer
>> > ?? CSeq: 5244504 INVITE
>> > ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
>> > ?? Allow:
>> > UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE
>> > ?? User-Agent: S60 RM-624 v 20.2.042 (en)
>> > ?? Expires: 120
>> > ?? Privacy: None
>> > ?? Session-Expires: 1800
>> > ?? Max-Forwards: 70
>> > ?? Proxy-Authorization: Digest
>> >
>> > qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"
>> > ?? Content-Type: application/sdp
>> > ?? Accept-Language: en
>> > ?? Content-Length: 292
>> >
>> > ?? v=0
>> > ?? o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
>> > ?? s=-
>> > ?? c=IN IP4 10.153.174.6
>> > ?? t=0 0
>> > ?? m=audio 49152 RTP/AVP 18 97 98
>> > ?? a=sendrecv
>> > ?? a=rtpmap:18 G729/8000
>> > ?? a=ptime:20
>> > ?? a=maxptime:40
>> > ?? a=fmtp:18 annexb=no
>> > ?? a=rtpmap:97 iLBC/8000
>> > ?? a=rtpmap:98 telephone-event/8000
>> > ?? a=fmtp:98 0-15
>> >
>> > ------------------------------------------------------------------------
>> > send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.406250:
>> >
>> > ------------------------------------------------------------------------
>> > ?? SIP/2.0 100 Trying
>> > ?? Via: SIP/2.0/TCP
>> >
>> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180
>> > ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>> > ?? To: <sip:MYPHONE at 192.168.1.100>
>> > ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
>> > ?? CSeq: 5244504 INVITE
>> > ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
>> > 18-04-05
>> > -0600
>> > ?? Content-Length: 0
>> >
>> >
>> > ------------------------------------------------------------------------
>> > 2011-01-28 16:15:59.406250 [NOTICE] switch_channel.c:808 New Channel
>> > sofia/internal/1001 at 192.168.1.100 [e5841001-04bd-4e16-9519-64ff2c7a8c2f]
>> > 2011-01-28 16:15:59.421875 [INFO] mod_dialplan_xml.c:331 Processing 1001
>> > <1001>->MYPHONE in context default
>> > 2011-01-28 16:15:59.437500 [NOTICE] switch_channel.c:808 New Channel
>> > sofia/external/1MYPHONE [60940227-9ae0-4c0c-abdb-f3988852fae0]
>> > send 1141 bytes to udp/[74.63.41.218]:5060 at 21:15:59.453125:
>> >
>> > ------------------------------------------------------------------------
>> > ?? INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0
>> > ?? Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS
>> > ?? Max-Forwards: 69
>> > ?? From: "Extension 1001"
>> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>> > ?? To: <sip:1MYPHONE at newyork.voip.ms>
>> > ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
>> > ?? CSeq: 7788615 INVITE
>> > ?? Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
>> > ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
>> > 18-04-05
>> > -0600
>> > ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> > REGISTER, REFER, NOTIFY
>> > ?? Supported: timer, precondition, path, replaces
>> > ?? Allow-Events: talk, hold, refer
>> > ?? Content-Type: application/sdp
>> > ?? Content-Disposition: session
>> > ?? Content-Length: 280
>> > ?? X-FS-Support: update_display
>> > ?? Remote-Party-ID: "Extension 1001"
>> > <sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off
>> >
>> > ?? v=0
>> > ?? o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
>> > ?? s=-
>> > ?? c=IN IP4 10.153.174.6
>> > ?? t=0 0
>> > ?? m=audio 49152 RTP/AVP 18 97 98
>> > ?? a=rtpmap:18 G729/8000
>> > ?? a=fmtp:18 annexb=no
>> > ?? a=rtpmap:97 iLBC/8000
>> > ?? a=rtpmap:98 telephone-event/8000
>> > ?? a=fmtp:98 0-15
>> > ?? a=ptime:20
>> > ?? a=maxptime:40
>> >
>> > ------------------------------------------------------------------------
>> > recv 570 bytes from udp/[74.63.41.218]:5060 at 21:15:59.468750:
>> >
>> > ------------------------------------------------------------------------
>> > ?? SIP/2.0 407 Proxy Authentication Required
>> > ?? Via: SIP/2.0/UDP
>> >
>> > 69.125.20.15:5080;branch=z9hG4bKNp4ZFeKSD43tS;received=69.125.20.15;rport=5080
>> > ?? From: "Extension 1001"
>> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>> > ?? To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843
>> > ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
>> > ?? CSeq: 7788615 INVITE
>> > ?? User-Agent: VoIPMS/SERAST
>> > ?? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> > ?? Supported: replaces
>> > ?? Proxy-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms",
>> > nonce="2d534dd6"
>> > ?? Content-Length: 0
>> >
>> >
>> > ------------------------------------------------------------------------
>> > send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:
>> >
>> > ------------------------------------------------------------------------
>> > ?? ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0
>> > ?? Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS
>> > ?? Max-Forwards: 69
>> > ?? From: "Extension 1001"
>> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>> > ?? To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843
>> > ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
>> > ?? CSeq: 7788615 ACK
>> > ?? Content-Length: 0
>> >
>> >
>> > ------------------------------------------------------------------------
>> > send 1344 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:
>> >
>> > ------------------------------------------------------------------------
>> > ?? INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0
>> > ?? Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN
>> > ?? Max-Forwards: 69
>> > ?? From: "Extension 1001"
>> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>> > ?? To: <sip:1MYPHONE at newyork.voip.ms>
>> > ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
>> > ?? CSeq: 7788616 INVITE
>> > ?? Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
>> > ?? Expires: 300
>> > ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
>> > 18-04-05
>> > -0600
>> > ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> > REGISTER, REFER, NOTIFY
>> > ?? Supported: timer, precondition, path, replaces
>> > ?? Allow-Events: talk, hold, refer
>> > ?? Proxy-Authorization: Digest username="121628",
>> > realm="newyork.voip.ms",
>> > nonce="2d534dd6", algorithm=MD5, uri="sip:1MYPHONE at newyork.voip.ms",
>> > response="16f3301efae13df926da7550f709d28a"
>> > ?? Content-Type: application/sdp
>> > ?? Content-Disposition: session
>> > ?? Content-Length: 280
>> > ?? X-FS-Support: update_display
>> > ?? Remote-Party-ID: "Extension 1001"
>> > <sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off
>> >
>> > ?? v=0
>> > ?? o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
>> > ?? s=-
>> > ?? c=IN IP4 10.153.174.6
>> > ?? t=0 0
>> > ?? m=audio 49152 RTP/AVP 18 97 98
>> > ?? a=rtpmap:18 G729/8000
>> > ?? a=fmtp:18 annexb=no
>> > ?? a=rtpmap:97 iLBC/8000
>> > ?? a=rtpmap:98 telephone-event/8000
>> > ?? a=fmtp:98 0-15
>> > ?? a=ptime:20
>> > ?? a=maxptime:40
>> >
>> > ------------------------------------------------------------------------
>> > recv 488 bytes from udp/[74.63.41.218]:5060 at 21:15:59.484375:
>> >
>> > ------------------------------------------------------------------------
>> > ?? SIP/2.0 100 Trying
>> > ?? Via: SIP/2.0/UDP
>> >
>> > 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080
>> > ?? From: "Extension 1001"
>> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>> > ?? To: <sip:1MYPHONE at newyork.voip.ms>
>> > ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
>> > ?? CSeq: 7788616 INVITE
>> > ?? User-Agent: VoIPMS/SERAST
>> > ?? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> > ?? Supported: replaces
>> > ?? Contact: <sip:1MYPHONE at 74.63.41.218>
>> > ?? Content-Length: 0
>> >
>> >
>> > ------------------------------------------------------------------------
>> > recv 516 bytes from udp/[74.63.41.218]:5060 at 21:15:59.562500:
>> >
>> > ------------------------------------------------------------------------
>> > ?? SIP/2.0 503 Service Unavailable
>> > ?? Via: SIP/2.0/UDP
>> >
>> > 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080
>> > ?? From: "Extension 1001"
>> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>> > ?? To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9
>> > ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
>> > ?? CSeq: 7788616 INVITE
>> > ?? User-Agent: VoIPMS/SERAST
>> > ?? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> > ?? Supported: replaces
>> > ?? Contact: <sip:1MYPHONE at 74.63.41.218>
>> > ?? Content-Length: 0
>> >
>> >
>> > ------------------------------------------------------------------------
>> > send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.562500:
>> >
>> > ------------------------------------------------------------------------
>> > ?? ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0
>> > ?? Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN
>> > ?? Max-Forwards: 69
>> > ?? From: "Extension 1001"
>> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>> > ?? To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9
>> > ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
>> > ?? CSeq: 7788616 ACK
>> > ?? Content-Length: 0
>> >
>> >
>> > ------------------------------------------------------------------------
>> > 2011-01-28 16:15:59.562500 [INFO] mod_dptools.c:2612 Originate Failed.
>> > Cause: NO_ANSWER
>> > 2011-01-28 16:15:59.562500 [NOTICE] sofia.c:5286 Hangup
>> > sofia/external/1MYPHONE [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
>> > 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:189
>> > sofia/internal/1001 at 192.168.1.100 has executed the last dialplan
>> > instruction, hanging up.
>> > 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:191
>> > Hangup
>> > sofia/internal/1001 at 192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING]
>> > 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1306 Session 2
>> > (sofia/external/1MYPHONE) Ended
>> > 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1308 Close
>> > Channel
>> > sofia/external/1MYPHONE [CS_DESTROY]
>> > send 887 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.593750:
>> >
>> > ------------------------------------------------------------------------
>> > ?? SIP/2.0 503 Service Unavailable
>> > ?? Via: SIP/2.0/TCP
>> >
>> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180
>> > ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>> > ?? To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg
>> > ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
>> > ?? CSeq: 5244504 INVITE
>> > ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
>> > 18-04-05
>> > -0600
>> > ?? Accept: application/sdp
>> > ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> > ?? Supported: timer, precondition, path, replaces
>> > ?? Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>> > sla,
>> > include-session-description, presence.winfo, message-summary, refer
>> > ?? Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>> > 2011-01-28 16:15:59.593750 [NOTICE] switch_core_session.c:1306 Session 1
>> > (sofia/internal/1001 at 192.168.1.100) Ended
>> > ?? Content-Length: 02011-01-28 16:15:59.593750 [NOTICE]
>> > switch_core_session.c:1308 Close Channel
>> > sofia/internal/1001 at 192.168.1.100
>> > [CS_DESTROY]
>> >
>> > ?? Remote-Party-ID: "MYPHONE"
>> > <sip:MYPHONE at 192.168.1.100>;party=calling;privacy=off;screen=no
>> >
>> >
>> > ------------------------------------------------------------------------
>> > recv 650 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.953125:
>> >
>> > ------------------------------------------------------------------------
>> > ?? ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
>> > ?? Via: SIP/2.0/TCP
>> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport
>> > ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>> > ?? To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg
>> > ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
>> > ?? CSeq: 5244504 ACK
>> > ?? Supported: sec-agree
>> > ?? Max-Forwards: 70
>> > ?? Proxy-Authorization: Digest
>> >
>> > qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"
>> > ?? Content-Length: 0
>> >
>> >
>> > ------------------------------------------------------------------------
>> >
>> > Thank you in advance.
>> >
>> > _______________________________________________ FreeSWITCH-users mailing
>> > list FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>> http://www.freeswitch.org
>
> _______________________________________________
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>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Mon Jan 31 19:46:11 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 31 Jan 2011 10:46:11 -0600
Subject: [Freeswitch-users] Problem with mod_shout on FS git
In-Reply-To: <1296491863148-5977722.post@n2.nabble.com>
References: <1296491863148-5977722.post@n2.nabble.com>
Message-ID: <AANLkTikvNTa0OueJHE5qLwnknXDJim100DJJqttt1txF@mail.gmail.com>

I would be looking at the format of the file produced by google more
than at FS.  We have not touched mod_shout in eons.
Maybe they put in a fake error to stop people from borrowing their audio engine.


On Mon, Jan 31, 2011 at 10:37 AM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
> Last time, when I ran into problem with mod_shout, I posted
> http://freeswitch-users.2379917.n2.nabble.com/Playing-Google-translation-tts-tp5916138p5920422.html
> here . Since then, I did an update with FreeSWITCH Version 1.0.head
> (git-49a5eff 2011-01-29 03-09-06 -0500). My same diialplan
> (conf/dialplan/default/2115_Google_TTS.xml), copied from
> http://freeswitch-users.2379917.n2.nabble.com/Playing-Google-translation-tts-tp5916138p5916461.html
> here , is shown below:
> <extension name="Free_Google_Text_To_Speech">
> ?<condition field="destination_number" expression="^2115$">
> ? ?<action application="answer" data=""/>
> ? ?<action application="playback"
> data="shout://translate.google.com/translate_tts?tl=en&q=FreeSWI
> ?</condition>
> </extension>
> The fs_cli dumped the following error messages when I called 2115#:
>
> 2011-01-31 11:13:16.647405 [WARNING] sofia_reg.c:1247 SIP auth challenge
> (INVITE) on sofia profile 'internal' for [2115 at 192.168.1.123] from ip
> 192.168.1.15
> 2011-01-31 11:13:18.656994 [NOTICE] switch_channel.c:808 New Channel
> sofia/internal/1003 at 192.168.1.15 [18fd8311-60b0-4c93-b022-51204ae6dd17]
> 2011-01-31 11:13:20.091366 [INFO] mod_dialplan_xml.c:331 Processing 1003
> <1003>->2115 in context default
> 2011-01-31 11:13:22.339858 [NOTICE] mod_dptools.c:920 Channel
> [sofia/internal/1003 at 192.168.1.15] has been answered
> 2011-01-31 11:13:22.354824 [ERR] mod_shout.c:804 Error: MPG123 Error at
> /opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.31_eabi/freeswitch_git/src/mod/formats/mod_shout/mod_shout.c:627.
> 2011-01-31 11:13:22.354824 [ERR] mod_shout.c:807 Error from mpg123: Invalid
> mpg123 handle. (code 10)
> 2011-01-31 11:13:22.360467 [NOTICE] switch_core_state_machine.c:189
> sofia/internal/1003 at 192.168.1.15 has executed the last dialplan instruction,
> hanging up.
> 2011-01-31 11:13:22.367752 [NOTICE] switch_core_state_machine.c:191 Hangup
> sofia/internal/1003 at 192.168.1.15 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-01-31 11:13:24.215896 [NOTICE] switch_core_session.c:1306 Session 7
> (sofia/internal/1003 at 192.168.1.15) Ended
> 2011-01-31 11:13:24.215896 [NOTICE] switch_core_session.c:1308 Close Channel
> sofia/internal/1003 at 192.168.1.15 [CS_DESTROY]
> The above [ERR] line shows an MPG123 error on mod_shout.c file @line 627. I
> don't suppose an MPG123 lib is needed mainly because the mpeg123 codes are
> statically linked to mod_shout.so. Can anyone confirm this?
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-with-mod-shout-on-FS-git-tp5977722p5977722.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From patrick.plattes at niemann-frey.info  Mon Jan 31 19:47:16 2011
From: patrick.plattes at niemann-frey.info (Patrick Plattes)
Date: Mon, 31 Jan 2011 17:47:16 +0100
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTikVDKS40dVsf8x=+5=qnW59LqJbSv1H=vtROFaM@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
	<AANLkTi=in3CLd-n0mT3SRKYA0kxLMGVWk9KeBx3_jhY6@mail.gmail.com>
	<AANLkTin3Bu+8EEr_v_GaRGNPzOjfcnfzsV2Av2cgnWgO@mail.gmail.com>
	<AANLkTi=OC_jx0+nzK405VZURLDJPD25R4ZjyJ3RXJ1nk@mail.gmail.com>
	<AANLkTi=Sm7CjoayiE8nHmL0Eehqknj1jdpbx0YBQJQGH@mail.gmail.com>
	<AANLkTinr0RdgBs2UiS3=b3i+k6uCwkMihDy_FbAOFPc_@mail.gmail.com>
	<AANLkTi=+aObW3kVBZq5ox4b4E1uFTSgtXUL72tHLOzPi@mail.gmail.com>
	<AANLkTikVDKS40dVsf8x=+5=qnW59LqJbSv1H=vtROFaM@mail.gmail.com>
Message-ID: <AANLkTi=gJsVKCJifVaBL_H6ZnPHTD__aQPhPeVfdx-ju@mail.gmail.com>

2011/1/31 Anthony Minessale <anthony.minessale at gmail.com>:
> You need to have it working on the latest GIT and you need to
> eliminate presence_in and presence_out 100% and use presence_id
> variable as explained. ?This is how you do it and it will work the way
> you want.

I've eleminated presence in and out. There was just the two lines in
the dialplan.

> If you think presence does not work on latest, you need to report what
> phone and model you are using. ?I will guess since you say you are in
> Germany and you have presence that is not working that it's a SNOM.
> Is that correct?

That's correct. I tried it with a Snom 300. Isn't it compliance with the RFC?

Bye


From Nabble at slickdeals.endjunk.com  Mon Jan 31 19:48:29 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 31 Jan 2011 08:48:29 -0800 (PST)
Subject: [Freeswitch-users] open g729
In-Reply-To: <AANLkTikng7_w0XcLb+B8AQ=3Xo1xsHDWkbmTz=8krWOt@mail.gmail.com>
References: <1296343422.2615.4.camel@gustavo-laptop>
	<0610594E3D2241C2B0BDDEC28EDBF605@e1705>
	<AANLkTikng7_w0XcLb+B8AQ=3Xo1xsHDWkbmTz=8krWOt@mail.gmail.com>
Message-ID: <1296492509261-5977760.post@n2.nabble.com>



Steven Ayre wrote:
> Funny you mention it, but MP3 does have several patents covering aspects
> of it that expire between 2007 (fine) and 2017 (not so fine). There have
> been various companies at various times that have claimed licenses are
> required to encode/decode it. Fraunhofer Institute being a good example -
> they earned 100million euros in 2005 from mp3 licenses. In1998 they sent a
> lot of letters out to developers of software using mp3 stating that they
> needed a licence.
Does that mean it is illegal for us to use FS mainly because it includes
lame-3.97 because some of the FS modules are also linked to it?

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/open-g729-tp5973641p5977760.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveu at coppice.org  Mon Jan 31 19:50:41 2011
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 01 Feb 2011 00:50:41 +0800
Subject: [Freeswitch-users] open g729
In-Reply-To: <AANLkTikNcipSGxfofujQwSNn4E16F-97u51xLB8a=cLB@mail.gmail.com>
References: <1296343422.2615.4.camel@gustavo-laptop>	<AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com>	<1296477877.3589.24.camel@gustavo-laptop>	<AANLkTinq9AmN=ZqjGLNUxucw2nJXBLpNHKLTA5jLWfgj@mail.gmail.com>
	<AANLkTikNcipSGxfofujQwSNn4E16F-97u51xLB8a=cLB@mail.gmail.com>
Message-ID: <4D46E861.4070009@coppice.org>

On 01/31/2011 11:54 PM, Anton VG wrote:
> IPP has a fairly good performance, at least on asterisk it performs
> as good as proprietary one, works perfectly fine, has no stability
> problems, and I highly doubt that FS situation differs, unless someone
> can prove it by posting real performance test numbers.
The last time I tested, the code we use for the Freeswitch G.729 took 
about 50% as much CPU time as the codec based on the IPP code. I don't 
know much about the Asterisk codec, but people have told me the 
Freeswitch it quite a bit faster than that one, too.

Steve



From infos at madovsky.org  Mon Jan 31 19:52:28 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 31 Jan 2011 11:52:28 -0500
Subject: [Freeswitch-users] open g729
References: <1296343422.2615.4.camel@gustavo-laptop><AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com><1296477877.3589.24.camel@gustavo-laptop><AANLkTinq9AmN=ZqjGLNUxucw2nJXBLpNHKLTA5jLWfgj@mail.gmail.com><9D7F812E93DC47A99F6A30638FB8092F@e1705><AANLkTikkLnf4BHYgL1k8th=VWk44vpH5gRYSAFW5YfW1@mail.gmail.com><A64FADA67C6D4C1C8B041EEFBB8C1DBB@e1705>
	<AANLkTi=1Fr2e-sz-n5Ds=q6KVntRzfzgrAQOgc0rCYgF@mail.gmail.com>
Message-ID: <F4AB52F0A27D46418900E2841CB5524B@e1705>

ok, but if I can make a parallel
that's not because 95% of computer use $MS
that you have to use $MS.
so the choice is :

- use more CPU to transcode G729 to XXX
- buy a licence $10 and not make $10 donation to FS 


  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 31, 2011 11:32 AM
  Subject: Re: [Freeswitch-users] open g729


  Depends on who your customer/provider is.

  90% of our customers only support G711 and G729. Which is just fine since most of our gateways only support G711, G729 and G723.1 so we stick with G729 and don't need to transcode. :)

  -Steve



  On 31 January 2011 16:00, Madovsky <infos at madovsky.org> wrote:

    wholesale offer also GSM and G726
      ----- Original Message ----- 
      From: Steven Ayre 
      To: FreeSWITCH Users Help 
      Sent: Monday, January 31, 2011 10:54 AM
      Subject: Re: [Freeswitch-users] open g729


      Not always possible, for instance in wholesale it's pretty much the standard codec used.

      -Steve



      On 31 January 2011 15:48, Madovsky <infos at madovsky.org> wrote:

        so the best is,
        don't use G729 and go to buy food with your $10 ;)
          ----- Original Message ----- 
          From: Steven Ayre 
          To: FreeSWITCH Users Help 
          Sent: Monday, January 31, 2011 7:53 AM
          Subject: Re: [Freeswitch-users] open g729


          The price is set by the patent holders, there's not much that can be done about that.

          If  you want to evaluate the g729 transcoding performance you *must* test with the one you plan to recommend - the mod_com_g729, mod_sangoma_codec and ipp g729 codecs have very difference performance characteristics. It is pointless to evaluate one and then recommend another.

          - mod_com_g729 is the official $10/channel license. It has good performance. To evaluate it you should try it for yourself.
          - The unofficial illegal FS module based ipp g729 which I shall not name has stability problems and uses far more CPU than mod_com_g729. Do not use it.
          - mod_sangoma_codec uses almost no CPU, but you do have to purchase a transcoding card.

          -Steve




          On 31 January 2011 12:44, Gustavo Espeche <gustavo.espeche at upper-soft.com> wrote:

            I just need test the performance of trans-coding before to recommend to
            may customer what server it need and buy the codec, because of then i
            need testing in my lab how much cpu and memory require the trans-coding
            in freeswitch for g729.
            And searching i found the open-g729/g723 for asterisk.
            I relay don't wish to start a discussion about the g729 license i read
            and understated it. But i think that the license price per channel is to
            hight taking care that this codec has more than 10 years but the way
            this is only my opinion.


            Best Regards.
            Gustavo Espeche
            www.easyipcall.com



            On Sat, 2011-01-29 at 17:27 -0600, Rupa Schomaker wrote:
            > How do you intend to pay the license fee?
            >
            > On Sat, Jan 29, 2011 at 5:23 PM, Gustavo Espeche
            > <gustavo.espeche at upper-soft.com> wrote:
            > > Hello,
            > >    some one can compile open g729 to work with freeswitch?
            > >    http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
            > >
            > >    i appreciate a lot if some one has some experience in it.
            > >    Best Regards.
            > >
            > >    Gustavo Espeche
            > >    www.easyipcall.com
            > >
            > >
            > >
            > >
            > >
            > > _______________________________________________
            > > FreeSWITCH-users mailing list
            > > FreeSWITCH-users at lists.freeswitch.org
            > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
            > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
            > > http://www.freeswitch.org
            > >
            >
            >
            >



            _______________________________________________
            FreeSWITCH-users mailing list
            FreeSWITCH-users at lists.freeswitch.org
            http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
            UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
            http://www.freeswitch.org





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          http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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        _______________________________________________
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        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From oa at estation.dk  Mon Jan 31 11:02:54 2011
From: oa at estation.dk (=?ISO-8859-1?Q?=D8yvind_Albrigtsen?=)
Date: Mon, 31 Jan 2011 09:02:54 +0100
Subject: [Freeswitch-users] Choppy VM when using loopback (on
	Debian	lenny)
In-Reply-To: <AANLkTiktckovGv4Ksnn_56EzpvTTSnfG+iD1vPX1_4RM@mail.gmail.com>
References: <4D42B8EE.7030208@estation.dk>	<7C6F3FC7-445F-4157-BD59-D76A705E4CE7@carmickle.com>	<9894FEC6-E451-4993-A85A-2CD910E916B1@ipeva.fr>	<AANLkTim+aBjRiAvxDUF1O1vut_CJpqgct2OqCNVxi0q4@mail.gmail.com>
	<AANLkTiktckovGv4Ksnn_56EzpvTTSnfG+iD1vPX1_4RM@mail.gmail.com>
Message-ID: <4D466CAE.7030402@estation.dk>

Thanks for all your feedback. I'll keep on trying and inform you what 
worked for me.


Regards
Oyvind

On 2011-01-29 21:48, Anthony Minessale wrote:
> Everyone should try latest GIT before pondering any further because I
> added a patch like 2 days ago to adress this issue.
>
>
> On Sat, Jan 29, 2011 at 2:12 PM, Frank Park<frank at telonium.com>  wrote:
>    
>> Yeah. I, too, don't see the correlation between the NTP and hw timer..
>> I am not familiar with the timer_test command and what it's measuring, but
>> of the 50 tests it ran, min is 19089 and max is 20713.
>> Frank
>>
>>
>> On Fri, Jan 28, 2011 at 2:33 PM, David Ponzone<david.ponzone at ipeva.fr>
>> wrote:
>>      
>>> Frank,
>>> I fail to see the relationship between the hw timer and NTP.
>>> Can you please elaborate ?
>>> David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>
>>>
>>>
>>> Le 28/01/2011 ? 19:49, Frank Carmickle a ?crit :
>>>
>>> Hi
>>>
>>> On Jan 28, 2011, at 7:39 AM, ?yvind Albrigtsen wrote:
>>>
>>> Hi,
>>>
>>> I'm using latest git-version of Freeswitch, and when I go to voicemail
>>>
>>> when calling a number the sound playback is choppy and it skips some of
>>>
>>> the digits in the number I called.
>>>
>>> What kind of results do you get from timer_test at the fs_cli?  Are you
>>> running on hardware or are you virtualized?  What is your clock source set
>>> to and what are your available clock source options?  See
>>> /sys/devices/system/clocksource/clocksource0/available_clocksource and
>>> /sys/devices/system/clocksource/clocksource0/current_clocksource.  I am
>>> running virtualized with the 2.6.26-2-xen-amd64 and I can get timer_test to
>>> hang at 19998/19999 which works very well for me.  When I was having problem
>>> it was reporting numbers all over the map from 17400 to 22600 with lots of
>>> randomness in between.  I have my clocksource set to jiffies and xen
>>> independent wallclock set to 1.  Of course at that point you need to have
>>> ntp running against a bunch of servers to drive your clock nice and steady.
>>>   I know my set up is probably a lot different than yours but I thought I'd
>>> toss it out there to show that some of the harshest conditions can be dealt
>>> with and don't give up trying.  If you are running on hardware with a cpu
>>> that doesn't have constant_tsc then you might have some problems.  Just play
>>> with the different timer options until you find the one that works.
>>>
>>> HTH
>>> --FC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>        
>>
>>
>> --
>>
>> ----=======================----
>> Frank Park
>> Telonium Communications, LLC
>> frank at telonium.com
>> http://www.telonium.com
>> Follow Us on Twitter: @GetTelonium
>> 404-566-8888 x1001 Office
>> 404-939-4242 Cell
>> ----=======================----
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>      
>
>
>    



From tim.compnetwork at gmail.com  Mon Jan 31 17:52:56 2011
From: tim.compnetwork at gmail.com (Tim King)
Date: Mon, 31 Jan 2011 09:52:56 -0500
Subject: [Freeswitch-users] CDR Field Definitions
Message-ID: <AANLkTiknJVSx5QgG9NoxYR-iYLRYUyiTym9-JzCagwxr@mail.gmail.com>

Sorry for the duplicate the first one I posted came form an account that for
some reason I can not retrieve messages from the list on.

 I have been digging for a while and I am struggling to find a page the
lists all of the possible fields (preferably with brief descriptions) that
are available with the various CDR records aka(XML and CSv).

I am also curious if anyone as found a way to record quality information
aka(Jitter, Latency, etc..) in the CDR record for the call.

Thanks

Tim
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From marcin321 at hotmail.com  Mon Jan 31 20:16:50 2011
From: marcin321 at hotmail.com (Marcin Wojtowicz)
Date: Mon, 31 Jan 2011 12:16:50 -0500
Subject: [Freeswitch-users] Outbound only calls don't connect when
 bypass_media is true.
In-Reply-To: <AANLkTi=t0ixs6L3SgUMQzEPtgvFowbFYXCVOtmvzLTEW@mail.gmail.com>
References: <COL118-W52E5226FC3337085AB6AFAE5E00@phx.gbl>,
	<COL118-W38F26048D441DAA2777942E5E30@phx.gbl>,
	<AANLkTinbiU4zNc-ZM+u+iST7E2aZ3zisgh5Lu2FTuBLt@mail.gmail.com>,
	<COL118-W196844C55E71088CAD1FC1E5E20@phx.gbl>,
	<AANLkTi=t0ixs6L3SgUMQzEPtgvFowbFYXCVOtmvzLTEW@mail.gmail.com>
Message-ID: <COL118-W34303A74089852D47F0E63E5E20@phx.gbl>


Yes, I had it set up to iLBC at 30i. It's not my cell phone (configured to ilbc, ptime=30 and mode=30), because when I call my freeswitch voicemail number, the sound is fine. I suspect it is something on the voip.ms <-> freeswitch leg because I created a sample ringback (8khz, mono, 16bit) wave file and directed my dialplan to it, but when I call from my home number to my cell, instead of hearing the ringer, I get choppy squeal.

?

> Date: Mon, 31 Jan 2011 10:40:10 -0600
> From: anthony.minessale at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Outbound only calls don't connect when bypass_media is true.
> 
> Many things have problems doing iLBC right.
> I recommend you define it in your configs as iLBC at 30i or it will try
> using the 20ms version which is not compatible with many other
> platforms.  Also make sure you are on the latest version of FS since
> we have tweaked iLBC behavior to compensate for problems like this.
> 
> 
> 
> On Sun, Jan 30, 2011 at 10:40 PM, Marcin Wojtowicz
> <marcin321 at hotmail.com> wrote:
> > OK, so I gave up on bypass media, but now I have another problem. This time
> > I set up freeswitch to communicate with voip.ms using PCMU codec (configured
> > in my external profile), and use iLBC on my phone (codec configured in my
> > internal profile, where the phone registers). When I call my mobile it
> > rings, but when I pick up all I hear is a high pitched squeal. Am I missing
> > something here?
> >
> >> Date: Sun, 30 Jan 2011 16:34:09 -0600
> >> From: anthony.minessale at gmail.com
> >> To: freeswitch-users at lists.freeswitch.org
> >> Subject: Re: [Freeswitch-users] Outbound only calls don't connect when
> >> bypass_media is true.
> >>
> >> Just do not use bypass media.
> >> That is all you can do in that situation.
> >>
> >>
> >> On Sun, Jan 30, 2011 at 3:44 PM, Marcin Wojtowicz <marcin321 at hotmail.com>
> >> wrote:
> >> > I just want to add that I enabled STUN on my cell so now the SDP message
> >> > in
> >> > the INVITE to voip.ms contains the public IP of my phone, but it still
> >> > doesn't work.
> >> >
> >> > ________________________________
> >> > From: marcin321 at hotmail.com
> >> > To: freeswitch-users at lists.freeswitch.org
> >> > Date: Fri, 28 Jan 2011 19:54:19 -0500
> >> > Subject: [Freeswitch-users] Outbound only calls don't connect when
> >> > bypass_media is true.
> >> >
> >> > Hello,
> >> >
> >> > I'm a new user of freeswitch, so please bear with me. I have the
> >> > following setup:
> >> > voip.ms <- SIP over UDP -> my desktop running freeswitch <- SIP over TCP
> >> > ->
> >> > my nokia cellphone on AT&T wireless. This setup is intended to conserve
> >> > the
> >> > battery usage.
> >> > I've managed to make everything work well when I'm calling in over any
> >> > phone
> >> > to my cell phone, and freeswitch is enabled to work in bypass_media =
> >> > true,
> >> > even though by cell is behind NAT on at&t's network. Things break when I
> >> > pick up my cell and try to call my home phone (or any phone for that
> >> > matter). This is the relevant snippet from my dialplan:
> >> > <extension name="outbound">
> >> >   <condition field="destination_number"
> >> > expression="^1?([2-9]\d{2}[2-9]\d{6})$">
> >> >     <!--<action application="set" data="bypass_media=true"/>-->
> >> >     <action application="bridge" data="sofia/gateway/voip.ms/1$1"/>
> >> >   </condition>
> >> > </extension>
> >> >
> >> > Like shown above, my call will go to my home phone. When I uncomment the
> >> > bypass_media tag, my call will not connect. Here are the siptraces
> >> > I replaced my real home phone number in the with "MYPHONE".
> >> >
> >> > recv 940 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.406250:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> >> >    Via: SIP/2.0/TCP
> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport
> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >    To: <sip:MYPHONE at 192.168.1.100>
> >> >    Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>
> >> >    Supported: 100rel,timer
> >> >    CSeq: 5244503 INVITE
> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >    Allow:
> >> > UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE
> >> >    User-Agent: S60 RM-624 v 20.2.042 (en)
> >> >    Expires: 120
> >> >    Privacy: None
> >> >    Session-Expires: 1800
> >> >    Max-Forwards: 70
> >> >    Content-Type: application/sdp
> >> >    Accept-Language: en
> >> >    Content-Length: 292
> >> >
> >> >    v=0
> >> >    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> >> >    s=-
> >> >    c=IN IP4 10.153.174.6
> >> >    t=0 0
> >> >    m=audio 49152 RTP/AVP 18 97 98
> >> >    a=sendrecv
> >> >    a=rtpmap:18 G729/8000
> >> >    a=ptime:20
> >> >    a=maxptime:40
> >> >    a=fmtp:18 annexb=no
> >> >    a=rtpmap:97 iLBC/8000
> >> >    a=rtpmap:98 telephone-event/8000
> >> >    a=fmtp:98 0-15
> >> >
> >> > ------------------------------------------------------------------------
> >> > send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    SIP/2.0 100 Trying
> >> >    Via: SIP/2.0/TCP
> >> >
> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180
> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >    To: <sip:MYPHONE at 192.168.1.100>
> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >    CSeq: 5244503 INVITE
> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> > 18-04-05
> >> > -0600
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > send 875 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    SIP/2.0 407 Proxy Authentication Required
> >> >    Via: SIP/2.0/TCP
> >> >
> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180
> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj2011-01-28
> >> > 16:15:58.406250 [WARNING] sofia_reg.c:1241 SIP auth challenge (INVITE)
> >> > on
> >> > sofia profile 'internal' for [MYPHONE at 192.168.1.100] from ip
> >> > 32.136.78.180
> >> >
> >> >    To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m
> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >    CSeq: 5244503 INVITE
> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> > 18-04-05
> >> > -0600
> >> >    Accept: application/sdp
> >> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >> >    Supported: timer, precondition, path, replaces
> >> >    Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
> >> > sla,
> >> > include-session-description, presence.winfo, message-summary, refer
> >> >    Proxy-Authenticate: Digest realm="192.168.1.100",
> >> > nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28", algorithm=MD5, qop="auth"
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > recv 368 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.765625:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> >> >    Via: SIP/2.0/TCP
> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport
> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >    To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m
> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >    CSeq: 5244503 ACK
> >> >    Supported: sec-agree
> >> >    Max-Forwards: 70
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > recv 1222 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.406250:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> >> >    Via: SIP/2.0/TCP
> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport
> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >    To: <sip:MYPHONE at 192.168.1.100>
> >> >    Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>
> >> >    Supported: 100rel,timer
> >> >    CSeq: 5244504 INVITE
> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >    Allow:
> >> > UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE
> >> >    User-Agent: S60 RM-624 v 20.2.042 (en)
> >> >    Expires: 120
> >> >    Privacy: None
> >> >    Session-Expires: 1800
> >> >    Max-Forwards: 70
> >> >    Proxy-Authorization: Digest
> >> >
> >> > qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"
> >> >    Content-Type: application/sdp
> >> >    Accept-Language: en
> >> >    Content-Length: 292
> >> >
> >> >    v=0
> >> >    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> >> >    s=-
> >> >    c=IN IP4 10.153.174.6
> >> >    t=0 0
> >> >    m=audio 49152 RTP/AVP 18 97 98
> >> >    a=sendrecv
> >> >    a=rtpmap:18 G729/8000
> >> >    a=ptime:20
> >> >    a=maxptime:40
> >> >    a=fmtp:18 annexb=no
> >> >    a=rtpmap:97 iLBC/8000
> >> >    a=rtpmap:98 telephone-event/8000
> >> >    a=fmtp:98 0-15
> >> >
> >> > ------------------------------------------------------------------------
> >> > send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.406250:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    SIP/2.0 100 Trying
> >> >    Via: SIP/2.0/TCP
> >> >
> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180
> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >    To: <sip:MYPHONE at 192.168.1.100>
> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >    CSeq: 5244504 INVITE
> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> > 18-04-05
> >> > -0600
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > 2011-01-28 16:15:59.406250 [NOTICE] switch_channel.c:808 New Channel
> >> > sofia/internal/1001 at 192.168.1.100 [e5841001-04bd-4e16-9519-64ff2c7a8c2f]
> >> > 2011-01-28 16:15:59.421875 [INFO] mod_dialplan_xml.c:331 Processing 1001
> >> > <1001>->MYPHONE in context default
> >> > 2011-01-28 16:15:59.437500 [NOTICE] switch_channel.c:808 New Channel
> >> > sofia/external/1MYPHONE [60940227-9ae0-4c0c-abdb-f3988852fae0]
> >> > send 1141 bytes to udp/[74.63.41.218]:5060 at 21:15:59.453125:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0
> >> >    Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS
> >> >    Max-Forwards: 69
> >> >    From: "Extension 1001"
> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >    To: <sip:1MYPHONE at newyork.voip.ms>
> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >    CSeq: 7788615 INVITE
> >> >    Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> > 18-04-05
> >> > -0600
> >> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> > REGISTER, REFER, NOTIFY
> >> >    Supported: timer, precondition, path, replaces
> >> >    Allow-Events: talk, hold, refer
> >> >    Content-Type: application/sdp
> >> >    Content-Disposition: session
> >> >    Content-Length: 280
> >> >    X-FS-Support: update_display
> >> >    Remote-Party-ID: "Extension 1001"
> >> > <sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off
> >> >
> >> >    v=0
> >> >    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> >> >    s=-
> >> >    c=IN IP4 10.153.174.6
> >> >    t=0 0
> >> >    m=audio 49152 RTP/AVP 18 97 98
> >> >    a=rtpmap:18 G729/8000
> >> >    a=fmtp:18 annexb=no
> >> >    a=rtpmap:97 iLBC/8000
> >> >    a=rtpmap:98 telephone-event/8000
> >> >    a=fmtp:98 0-15
> >> >    a=ptime:20
> >> >    a=maxptime:40
> >> >
> >> > ------------------------------------------------------------------------
> >> > recv 570 bytes from udp/[74.63.41.218]:5060 at 21:15:59.468750:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    SIP/2.0 407 Proxy Authentication Required
> >> >    Via: SIP/2.0/UDP
> >> >
> >> > 69.125.20.15:5080;branch=z9hG4bKNp4ZFeKSD43tS;received=69.125.20.15;rport=5080
> >> >    From: "Extension 1001"
> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843
> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >    CSeq: 7788615 INVITE
> >> >    User-Agent: VoIPMS/SERAST
> >> >    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> >> >    Supported: replaces
> >> >    Proxy-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms",
> >> > nonce="2d534dd6"
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0
> >> >    Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS
> >> >    Max-Forwards: 69
> >> >    From: "Extension 1001"
> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843
> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >    CSeq: 7788615 ACK
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > send 1344 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0
> >> >    Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN
> >> >    Max-Forwards: 69
> >> >    From: "Extension 1001"
> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >    To: <sip:1MYPHONE at newyork.voip.ms>
> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >    CSeq: 7788616 INVITE
> >> >    Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
> >> >    Expires: 300
> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> > 18-04-05
> >> > -0600
> >> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> > REGISTER, REFER, NOTIFY
> >> >    Supported: timer, precondition, path, replaces
> >> >    Allow-Events: talk, hold, refer
> >> >    Proxy-Authorization: Digest username="121628",
> >> > realm="newyork.voip.ms",
> >> > nonce="2d534dd6", algorithm=MD5, uri="sip:1MYPHONE at newyork.voip.ms",
> >> > response="16f3301efae13df926da7550f709d28a"
> >> >    Content-Type: application/sdp
> >> >    Content-Disposition: session
> >> >    Content-Length: 280
> >> >    X-FS-Support: update_display
> >> >    Remote-Party-ID: "Extension 1001"
> >> > <sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off
> >> >
> >> >    v=0
> >> >    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> >> >    s=-
> >> >    c=IN IP4 10.153.174.6
> >> >    t=0 0
> >> >    m=audio 49152 RTP/AVP 18 97 98
> >> >    a=rtpmap:18 G729/8000
> >> >    a=fmtp:18 annexb=no
> >> >    a=rtpmap:97 iLBC/8000
> >> >    a=rtpmap:98 telephone-event/8000
> >> >    a=fmtp:98 0-15
> >> >    a=ptime:20
> >> >    a=maxptime:40
> >> >
> >> > ------------------------------------------------------------------------
> >> > recv 488 bytes from udp/[74.63.41.218]:5060 at 21:15:59.484375:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    SIP/2.0 100 Trying
> >> >    Via: SIP/2.0/UDP
> >> >
> >> > 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080
> >> >    From: "Extension 1001"
> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >    To: <sip:1MYPHONE at newyork.voip.ms>
> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >    CSeq: 7788616 INVITE
> >> >    User-Agent: VoIPMS/SERAST
> >> >    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> >> >    Supported: replaces
> >> >    Contact: <sip:1MYPHONE at 74.63.41.218>
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > recv 516 bytes from udp/[74.63.41.218]:5060 at 21:15:59.562500:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    SIP/2.0 503 Service Unavailable
> >> >    Via: SIP/2.0/UDP
> >> >
> >> > 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080
> >> >    From: "Extension 1001"
> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9
> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >    CSeq: 7788616 INVITE
> >> >    User-Agent: VoIPMS/SERAST
> >> >    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> >> >    Supported: replaces
> >> >    Contact: <sip:1MYPHONE at 74.63.41.218>
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.562500:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0
> >> >    Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN
> >> >    Max-Forwards: 69
> >> >    From: "Extension 1001"
> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9
> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >    CSeq: 7788616 ACK
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > 2011-01-28 16:15:59.562500 [INFO] mod_dptools.c:2612 Originate Failed.
> >> > Cause: NO_ANSWER
> >> > 2011-01-28 16:15:59.562500 [NOTICE] sofia.c:5286 Hangup
> >> > sofia/external/1MYPHONE [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
> >> > 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:189
> >> > sofia/internal/1001 at 192.168.1.100 has executed the last dialplan
> >> > instruction, hanging up.
> >> > 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:191
> >> > Hangup
> >> > sofia/internal/1001 at 192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING]
> >> > 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1306 Session 2
> >> > (sofia/external/1MYPHONE) Ended
> >> > 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1308 Close
> >> > Channel
> >> > sofia/external/1MYPHONE [CS_DESTROY]
> >> > send 887 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.593750:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    SIP/2.0 503 Service Unavailable
> >> >    Via: SIP/2.0/TCP
> >> >
> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180
> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >    To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg
> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >    CSeq: 5244504 INVITE
> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> > 18-04-05
> >> > -0600
> >> >    Accept: application/sdp
> >> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >> >    Supported: timer, precondition, path, replaces
> >> >    Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
> >> > sla,
> >> > include-session-description, presence.winfo, message-summary, refer
> >> >    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
> >> > 2011-01-28 16:15:59.593750 [NOTICE] switch_core_session.c:1306 Session 1
> >> > (sofia/internal/1001 at 192.168.1.100) Ended
> >> >    Content-Length: 02011-01-28 16:15:59.593750 [NOTICE]
> >> > switch_core_session.c:1308 Close Channel
> >> > sofia/internal/1001 at 192.168.1.100
> >> > [CS_DESTROY]
> >> >
> >> >    Remote-Party-ID: "MYPHONE"
> >> > <sip:MYPHONE at 192.168.1.100>;party=calling;privacy=off;screen=no
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > recv 650 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.953125:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> >> >    Via: SIP/2.0/TCP
> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport
> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >    To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg
> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >    CSeq: 5244504 ACK
> >> >    Supported: sec-agree
> >> >    Max-Forwards: 70
> >> >    Proxy-Authorization: Digest
> >> >
> >> > qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> >
> >> > Thank you in advance.
> >> >
> >> > _______________________________________________ FreeSWITCH-users mailing
> >> > list FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
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From infos at madovsky.org  Mon Jan 31 20:29:12 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 31 Jan 2011 12:29:12 -0500
Subject: [Freeswitch-users] minimum UDP packet size
Message-ID: <65064C9F3B344E4A9CAE4A9B63C318D3@e1705>

is it useful to set minimum UDP  pakcet to the
MTU rate like 1500 and avoid any fragmentation ?

Thanks
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From robert.hadley at teotech.com  Mon Jan 31 20:33:48 2011
From: robert.hadley at teotech.com (Robert Hadley)
Date: Mon, 31 Jan 2011 09:33:48 -0800
Subject: [Freeswitch-users] Outbound only calls don't connect
	when	bypass_media is true.
In-Reply-To: <COL118-W34303A74089852D47F0E63E5E20@phx.gbl>
References: <COL118-W52E5226FC3337085AB6AFAE5E00@phx.gbl>,
	<COL118-W38F26048D441DAA2777942E5E30@phx.gbl>,
	<AANLkTinbiU4zNc-ZM+u+iST7E2aZ3zisgh5Lu2FTuBLt@mail.gmail.com>,
	<COL118-W196844C55E71088CAD1FC1E5E20@phx.gbl>,
	<AANLkTi=t0ixs6L3SgUMQzEPtgvFowbFYXCVOtmvzLTEW@mail.gmail.com>
	<COL118-W34303A74089852D47F0E63E5E20@phx.gbl>
Message-ID: <E826C94A3C6BE44198FAF4D66B5AB4D80FEC3E5B24@VA3DIAXVS351.RED001.local>


Check the codecs in the SDP or try manual hardcoding the codecs presented for both legs, we had a squeal problem going to a softphone that turned out to be the BV32 codec was being selected instead of SPEEX16.

Robert

From: Marcin Wojtowicz [mailto:marcin321 at hotmail.com]
Sent: Monday, January 31, 2011 9:17 AM
To: freeswitch
Subject: Re: [Freeswitch-users] Outbound only calls don't connect when bypass_media is true.

Yes, I had it set up to iLBC at 30i. It's not my cell phone (configured to ilbc, ptime=30 and mode=30), because when I call my freeswitch voicemail number, the sound is fine. I suspect it is something on the voip.ms <-> freeswitch leg because I created a sample ringback (8khz, mono, 16bit) wave file and directed my dialplan to it, but when I call from my home number to my cell, instead of hearing the ringer, I get choppy squeal.

?

> Date: Mon, 31 Jan 2011 10:40:10 -0600
> From: anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>
> To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Outbound only calls don't connect when bypass_media is true.
>
> Many things have problems doing iLBC right.
> I recommend you define it in your configs as iLBC at 30i or it will try
> using the 20ms version which is not compatible with many other
> platforms. Also make sure you are on the latest version of FS since
> we have tweaked iLBC behavior to compensate for problems like this.
>
>
>
> On Sun, Jan 30, 2011 at 10:40 PM, Marcin Wojtowicz
> <marcin321 at hotmail.com<mailto:marcin321 at hotmail.com>> wrote:
> > OK, so I gave up on bypass media, but now I have another problem. This time
> > I set up freeswitch to communicate with voip.ms using PCMU codec (configured
> > in my external profile), and use iLBC on my phone (codec configured in my
> > internal profile, where the phone registers). When I call my mobile it
> > rings, but when I pick up all I hear is a high pitched squeal. Am I missing
> > something here?
> >
> >> Date: Sun, 30 Jan 2011 16:34:09 -0600
> >> From: anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>
> >> To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
> >> Subject: Re: [Freeswitch-users] Outbound only calls don't connect when
> >> bypass_media is true.
> >>
> >> Just do not use bypass media.
> >> That is all you can do in that situation.
> >>
> >>
> >> On Sun, Jan 30, 2011 at 3:44 PM, Marcin Wojtowicz <marcin321 at hotmail.com<mailto:marcin321 at hotmail.com>>
> >> wrote:
> >> > I just want to add that I enabled STUN on my cell so now the SDP message
> >> > in
> >> > the INVITE to voip.ms contains the public IP of my phone, but it still
> >> > doesn't work.
> >> >
> >> > ________________________________
> >> > From: marcin321 at hotmail.com<mailto:marcin321 at hotmail.com>
> >> > To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
> >> > Date: Fri, 28 Jan 2011 19:54:19 -0500
> >> > Subject: [Freeswitch-users] Outbound only calls don't connect when
> >> > bypass_media is true.
> >> >
> >> > Hello,
> >> >
> >> > I'm a new user of freeswitch, so please bear with me. I have the
> >> > following setup:
> >> > voip.ms <- SIP over UDP -> my desktop running freeswitch <- SIP over TCP
> >> > ->
> >> > my nokia cellphone on AT&T wireless. This setup is intended to conserve
> >> > the
> >> > battery usage.
> >> > I've managed to make everything work well when I'm calling in over any
> >> > phone
> >> > to my cell phone, and freeswitch is enabled to work in bypass_media =
> >> > true,
> >> > even though by cell is behind NAT on at&t's network. Things break when I
> >> > pick up my cell and try to call my home phone (or any phone for that
> >> > matter). This is the relevant snippet from my dialplan:
> >> > <extension name="outbound">
> >> >   <condition field="destination_number"
> >> > expression="^1?([2-9]\d{2}[2-9]\d{6})$">
> >> >     <!--<action application="set" data="bypass_media=true"/>-->
> >> >     <action application="bridge" data="sofia/gateway/voip.ms/1$1"/>
> >> >   </condition>
> >> > </extension>
> >> >
> >> > Like shown above, my call will go to my home phone. When I uncomment the
> >> > bypass_media tag, my call will not connect. Here are the siptraces
> >> > I replaced my real home phone number in the with "MYPHONE".
> >> >
> >> > recv 940 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.406250:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> >> >    Via: SIP/2.0/TCP
> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport
> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >    To: <sip:MYPHONE at 192.168.1.100>
> >> >    Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>
> >> >    Supported: 100rel,timer
> >> >    CSeq: 5244503 INVITE
> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >    Allow:
> >> > UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE
> >> >    User-Agent: S60 RM-624 v 20.2.042 (en)
> >> >    Expires: 120
> >> >    Privacy: None
> >> >    Session-Expires: 1800
> >> >    Max-Forwards: 70
> >> >    Content-Type: application/sdp
> >> >    Accept-Language: en
> >> >    Content-Length: 292
> >> >
> >> >    v=0
> >> >    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> >> >    s=-
> >> >    c=IN IP4 10.153.174.6
> >> >    t=0 0
> >> >    m=audio 49152 RTP/AVP 18 97 98
> >> >    a=sendrecv
> >> >    a=rtpmap:18 G729/8000
> >> >    a=ptime:20
> >> >    a=maxptime:40
> >> >    a=fmtp:18 annexb=no
> >> >    a=rtpmap:97 iLBC/8000
> >> >    a=rtpmap:98 telephone-event/8000
> >> >    a=fmtp:98 0-15
> >> >
> >> > ------------------------------------------------------------------------
> >> > send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    SIP/2.0 100 Trying
> >> >    Via: SIP/2.0/TCP
> >> >
> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180
> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >    To: <sip:MYPHONE at 192.168.1.100>
> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >    CSeq: 5244503 INVITE
> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> > 18-04-05
> >> > -0600
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > send 875 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    SIP/2.0 407 Proxy Authentication Required
> >> >    Via: SIP/2.0/TCP
> >> >
> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180
> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj2011-01-28
> >> > 16:15:58.406250 [WARNING] sofia_reg.c:1241 SIP auth challenge (INVITE)
> >> > on
> >> > sofia profile 'internal' for [MYPHONE at 192.168.1.100] from ip
> >> > 32.136.78.180
> >> >
> >> >    To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m
> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >    CSeq: 5244503 INVITE
> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> > 18-04-05
> >> > -0600
> >> >    Accept: application/sdp
> >> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >> >    Supported: timer, precondition, path, replaces
> >> >    Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
> >> > sla,
> >> > include-session-description, presence.winfo, message-summary, refer
> >> >    Proxy-Authenticate: Digest realm="192.168.1.100",
> >> > nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28", algorithm=MD5, qop="auth"
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > recv 368 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.765625:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> >> >    Via: SIP/2.0/TCP
> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport
> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >    To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m
> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >    CSeq: 5244503 ACK
> >> >    Supported: sec-agree
> >> >    Max-Forwards: 70
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > recv 1222 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.406250:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> >> >    Via: SIP/2.0/TCP
> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport
> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >    To: <sip:MYPHONE at 192.168.1.100>
> >> >    Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>
> >> >    Supported: 100rel,timer
> >> >    CSeq: 5244504 INVITE
> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >    Allow:
> >> > UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE
> >> >    User-Agent: S60 RM-624 v 20.2.042 (en)
> >> >    Expires: 120
> >> >    Privacy: None
> >> >    Session-Expires: 1800
> >> >    Max-Forwards: 70
> >> >    Proxy-Authorization: Digest
> >> >
> >> > qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"
> >> >    Content-Type: application/sdp
> >> >    Accept-Language: en
> >> >    Content-Length: 292
> >> >
> >> >    v=0
> >> >    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> >> >    s=-
> >> >    c=IN IP4 10.153.174.6
> >> >    t=0 0
> >> >    m=audio 49152 RTP/AVP 18 97 98
> >> >    a=sendrecv
> >> >    a=rtpmap:18 G729/8000
> >> >    a=ptime:20
> >> >    a=maxptime:40
> >> >    a=fmtp:18 annexb=no
> >> >    a=rtpmap:97 iLBC/8000
> >> >    a=rtpmap:98 telephone-event/8000
> >> >    a=fmtp:98 0-15
> >> >
> >> > ------------------------------------------------------------------------
> >> > send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.406250:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    SIP/2.0 100 Trying
> >> >    Via: SIP/2.0/TCP
> >> >
> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180
> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >    To: <sip:MYPHONE at 192.168.1.100>
> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >    CSeq: 5244504 INVITE
> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> > 18-04-05
> >> > -0600
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > 2011-01-28 16:15:59.406250 [NOTICE] switch_channel.c:808 New Channel
> >> > sofia/internal/1001 at 192.168.1.100<mailto:sofia/internal/1001 at 192.168.1.100> [e5841001-04bd-4e16-9519-64ff2c7a8c2f]
> >> > 2011-01-28 16:15:59.421875 [INFO] mod_dialplan_xml.c:331 Processing 1001
> >> > <1001>->MYPHONE in context default
> >> > 2011-01-28 16:15:59.437500 [NOTICE] switch_channel.c:808 New Channel
> >> > sofia/external/1MYPHONE [60940227-9ae0-4c0c-abdb-f3988852fae0]
> >> > send 1141 bytes to udp/[74.63.41.218]:5060 at 21:15:59.453125:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0
> >> >    Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS
> >> >    Max-Forwards: 69
> >> >    From: "Extension 1001"
> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >    To: <sip:1MYPHONE at newyork.voip.ms>
> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >    CSeq: 7788615 INVITE
> >> >    Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> > 18-04-05
> >> > -0600
> >> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> > REGISTER, REFER, NOTIFY
> >> >    Supported: timer, precondition, path, replaces
> >> >    Allow-Events: talk, hold, refer
> >> >    Content-Type: application/sdp
> >> >    Content-Disposition: session
> >> >    Content-Length: 280
> >> >    X-FS-Support: update_display
> >> >    Remote-Party-ID: "Extension 1001"
> >> > <sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off
> >> >
> >> >    v=0
> >> >    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> >> >    s=-
> >> >    c=IN IP4 10.153.174.6
> >> >    t=0 0
> >> >    m=audio 49152 RTP/AVP 18 97 98
> >> >    a=rtpmap:18 G729/8000
> >> >    a=fmtp:18 annexb=no
> >> >    a=rtpmap:97 iLBC/8000
> >> >    a=rtpmap:98 telephone-event/8000
> >> >    a=fmtp:98 0-15
> >> >    a=ptime:20
> >> >    a=maxptime:40
> >> >
> >> > ------------------------------------------------------------------------
> >> > recv 570 bytes from udp/[74.63.41.218]:5060 at 21:15:59.468750:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    SIP/2.0 407 Proxy Authentication Required
> >> >    Via: SIP/2.0/UDP
> >> >
> >> > 69.125.20.15:5080;branch=z9hG4bKNp4ZFeKSD43tS;received=69.125.20.15;rport=5080
> >> >    From: "Extension 1001"
> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843
> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >    CSeq: 7788615 INVITE
> >> >    User-Agent: VoIPMS/SERAST
> >> >    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> >> >    Supported: replaces
> >> >    Proxy-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms",
> >> > nonce="2d534dd6"
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0
> >> >    Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS
> >> >    Max-Forwards: 69
> >> >    From: "Extension 1001"
> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843
> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >    CSeq: 7788615 ACK
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > send 1344 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0
> >> >    Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN
> >> >    Max-Forwards: 69
> >> >    From: "Extension 1001"
> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >    To: <sip:1MYPHONE at newyork.voip.ms>
> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >    CSeq: 7788616 INVITE
> >> >    Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
> >> >    Expires: 300
> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> > 18-04-05
> >> > -0600
> >> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> > REGISTER, REFER, NOTIFY
> >> >    Supported: timer, precondition, path, replaces
> >> >    Allow-Events: talk, hold, refer
> >> >    Proxy-Authorization: Digest username="121628",
> >> > realm="newyork.voip.ms",
> >> > nonce="2d534dd6", algorithm=MD5, uri="sip:1MYPHONE at newyork.voip.ms",
> >> > response="16f3301efae13df926da7550f709d28a"
> >> >    Content-Type: application/sdp
> >> >    Content-Disposition: session
> >> >    Content-Length: 280
> >> >    X-FS-Support: update_display
> >> >    Remote-Party-ID: "Extension 1001"
> >> > <sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off
> >> >
> >> >    v=0
> >> >    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> >> >    s=-
> >> >    c=IN IP4 10.153.174.6
> >> >    t=0 0
> >> >    m=audio 49152 RTP/AVP 18 97 98
> >> >    a=rtpmap:18 G729/8000
> >> >    a=fmtp:18 annexb=no
> >> >    a=rtpmap:97 iLBC/8000
> >> >    a=rtpmap:98 telephone-event/8000
> >> >    a=fmtp:98 0-15
> >> >    a=ptime:20
> >> >    a=maxptime:40
> >> >
> >> > ------------------------------------------------------------------------
> >> > recv 488 bytes from udp/[74.63.41.218]:5060 at 21:15:59.484375:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    SIP/2.0 100 Trying
> >> >    Via: SIP/2.0/UDP
> >> >
> >> > 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080
> >> >    From: "Extension 1001"
> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >    To: <sip:1MYPHONE at newyork.voip.ms>
> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >    CSeq: 7788616 INVITE
> >> >    User-Agent: VoIPMS/SERAST
> >> >    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> >> >    Supported: replaces
> >> >    Contact: <sip:1MYPHONE at 74.63.41.218>
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > recv 516 bytes from udp/[74.63.41.218]:5060 at 21:15:59.562500:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    SIP/2.0 503 Service Unavailable
> >> >    Via: SIP/2.0/UDP
> >> >
> >> > 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080
> >> >    From: "Extension 1001"
> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9
> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >    CSeq: 7788616 INVITE
> >> >    User-Agent: VoIPMS/SERAST
> >> >    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> >> >    Supported: replaces
> >> >    Contact: <sip:1MYPHONE at 74.63.41.218>
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.562500:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0
> >> >    Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN
> >> >    Max-Forwards: 69
> >> >    From: "Extension 1001"
> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9
> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >    CSeq: 7788616 ACK
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > 2011-01-28 16:15:59.562500 [INFO] mod_dptools.c:2612 Originate Failed.
> >> > Cause: NO_ANSWER
> >> > 2011-01-28 16:15:59.562500 [NOTICE] sofia.c:5286 Hangup
> >> > sofia/external/1MYPHONE [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
> >> > 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:189
> >> > sofia/internal/1001 at 192.168.1.100<mailto:sofia/internal/1001 at 192.168.1.100> has executed the last dialplan
> >> > instruction, hanging up.
> >> > 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:191
> >> > Hangup
> >> > sofia/internal/1001 at 192.168.1.100<mailto:sofia/internal/1001 at 192.168.1.100> [CS_EXECUTE] [NORMAL_CLEARING]
> >> > 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1306 Session 2
> >> > (sofia/external/1MYPHONE) Ended
> >> > 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1308 Close
> >> > Channel
> >> > sofia/external/1MYPHONE [CS_DESTROY]
> >> > send 887 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.593750:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    SIP/2.0 503 Service Unavailable
> >> >    Via: SIP/2.0/TCP
> >> >
> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180
> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >    To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg
> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >    CSeq: 5244504 INVITE
> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> > 18-04-05
> >> > -0600
> >> >    Accept: application/sdp
> >> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >> >    Supported: timer, precondition, path, replaces
> >> >    Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
> >> > sla,
> >> > include-session-description, presence.winfo, message-summary, refer
> >> >    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
> >> > 2011-01-28 16:15:59.593750 [NOTICE] switch_core_session.c:1306 Session 1
> >> > (sofia/internal/1001 at 192.168.1.100<mailto:sofia/internal/1001 at 192.168.1.100>) Ended
> >> >    Content-Length: 02011-01-28 16:15:59.593750 [NOTICE]
> >> > switch_core_session.c:1308 Close Channel
> >> > sofia/internal/1001 at 192.168.1.100<mailto:sofia/internal/1001 at 192.168.1.100>
> >> > [CS_DESTROY]
> >> >
> >> >    Remote-Party-ID: "MYPHONE"
> >> > <sip:MYPHONE at 192.168.1.100>;party=calling;privacy=off;screen=no
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > recv 650 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.953125:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> >> >    Via: SIP/2.0/TCP
> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport
> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >    To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg
> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >    CSeq: 5244504 ACK
> >> >    Supported: sec-agree
> >> >    Max-Forwards: 70
> >> >    Proxy-Authorization: Digest
> >> >
> >> > qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> >
> >> > Thank you in advance.
> >> >
> >> > _______________________________________________ FreeSWITCH-users mailing
> >> > list FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From david.ponzone at ipeva.fr  Mon Jan 31 20:37:28 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 31 Jan 2011 18:37:28 +0100
Subject: [Freeswitch-users] CDR Field Definitions
In-Reply-To: <AANLkTiknJVSx5QgG9NoxYR-iYLRYUyiTym9-JzCagwxr@mail.gmail.com>
References: <AANLkTiknJVSx5QgG9NoxYR-iYLRYUyiTym9-JzCagwxr@mail.gmail.com>
Message-ID: <95973B3D-9199-44D1-9A4A-784B771B6C59@ipeva.fr>

Tim,

enable xml_cdr (both legs), then look at one of those, and you'll see all the possibles fields you can enable in CSV.
Most of them are quite self-explanatory.

About quality, that depends on RTCP.
I don't think RTCP is completely implemented in FreeSWITCH yet, but I remember some people were talking about it recently.
You should search in the archives.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 31/01/2011 ? 15:52, Tim King a ?crit :

> Sorry for the duplicate the first one I posted came form an account that for some reason I can not retrieve messages from the list on.
> 
>  I have been digging for a while and I am struggling to find a page the lists all of the possible fields (preferably with brief descriptions) that are available with the various CDR records aka(XML and CSv). 
> 
> I am also curious if anyone as found a way to record quality information aka(Jitter, Latency, etc..) in the CDR record for the call.
> 
> Thanks
> 
> Tim _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at ipeva.fr  Mon Jan 31 20:46:00 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 31 Jan 2011 18:46:00 +0100
Subject: [Freeswitch-users] minimum UDP packet size
In-Reply-To: <65064C9F3B344E4A9CAE4A9B63C318D3@e1705>
References: <65064C9F3B344E4A9CAE4A9B63C318D3@e1705>
Message-ID: <5DDC294B-BCB6-42FA-8243-885625ADAF2E@ipeva.fr>

Sir,

I am not really sure what you are trying to accomplish.
How do you want to avoid fragmentation by setting a minimum packet size ?
You can't really avoid fragmentation.
If a SIP packet is larger than 1500, than it is larger.
If you want to avoid fragmentation, you will need to suppress unnecessary SIP headers, or eventually you may try to enable short headers but that's not quite supported by all vendors.
Supressing unnecessary headers is quite easy with a B2BUA like FreeSWITCH, as most headers should already not be forwarded.
SIP INVITEs over 1500 is generally a problem when you proxy, adding Route information, etc...

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 31/01/2011 ? 18:29, Madovsky a ?crit :

> is it useful to set minimum UDP  pakcet to the
> MTU rate like 1500 and avoid any fragmentation ?
>  
> Thanks
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Mon Jan 31 20:47:13 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 31 Jan 2011 11:47:13 -0600
Subject: [Freeswitch-users] open g729
In-Reply-To: <F4AB52F0A27D46418900E2841CB5524B@e1705>
References: <1296343422.2615.4.camel@gustavo-laptop>
	<AANLkTikQtZQSPA49JQPcztpBPr7a7yuQm8yUsGZtcs9F@mail.gmail.com>
	<1296477877.3589.24.camel@gustavo-laptop>
	<AANLkTinq9AmN=ZqjGLNUxucw2nJXBLpNHKLTA5jLWfgj@mail.gmail.com>
	<9D7F812E93DC47A99F6A30638FB8092F@e1705>
	<AANLkTikkLnf4BHYgL1k8th=VWk44vpH5gRYSAFW5YfW1@mail.gmail.com>
	<A64FADA67C6D4C1C8B041EEFBB8C1DBB@e1705>
	<AANLkTi=1Fr2e-sz-n5Ds=q6KVntRzfzgrAQOgc0rCYgF@mail.gmail.com>
	<F4AB52F0A27D46418900E2841CB5524B@e1705>
Message-ID: <AANLkTi=icT=Zw9xrrEcOyyozy5JFm7sXQjxTDxN3psZH@mail.gmail.com>

Its too bad this type of thread is always the only one anyone carries
past like 4 exchanges.
I would love for the other aspects of the project to get this much attention.

I would like to say for the record.  I personally spent the last many
years working with others to sign up, pay a large sum of money, and go
on the hook to provide these licenses to people.  The only people who
truly need G729, need it for business reasons.  It's patented because
someone wants to make money when you do.  This is not something I can
do anything about.  I do my part by giving you the whole soft-switch
for free.  If you have to pay $10 for a channel, the bright side is
you do not have to pay 100k for a commercial soft-switch and if you
have to actually buy a lot of G729 licenses you must be doing
something right and making a lot of money.

I'd prefer if we not waste our time bickering about moot political
issues and move on.  There is a new codec being developed that is open
and free and may some day eliminate G729.  Until then let's focus on
something we can actually do something about.

Carry on if you wish, I just think we have better things to do.




On Mon, Jan 31, 2011 at 10:52 AM, Madovsky <infos at madovsky.org> wrote:
> ok, but if I can make a parallel
> that's not because 95% of computer use $MS
> that you have to use $MS.
> so the choice is :
>
> - use more CPU to transcode G729 to XXX
> - buy a licence $10 and not make $10 donation to FS
>
>
>
> ----- Original Message -----
> From: Steven Ayre
> To: FreeSWITCH Users Help
> Sent: Monday, January 31, 2011 11:32 AM
> Subject: Re: [Freeswitch-users] open g729
> Depends on who your customer/provider is.
>
> 90% of our customers only support G711 and G729. Which is just fine since
> most of our gateways only support G711, G729 and G723.1 so we stick with
> G729 and don't need to transcode. :)
>
> -Steve
>
>
> On 31 January 2011 16:00, Madovsky <infos at madovsky.org> wrote:
>>
>> wholesale offer also GSM and G726
>>
>> ----- Original Message -----
>> From: Steven Ayre
>> To: FreeSWITCH Users Help
>> Sent: Monday, January 31, 2011 10:54 AM
>> Subject: Re: [Freeswitch-users] open g729
>> Not always possible, for instance in wholesale it's pretty much the
>> standard codec used.
>>
>> -Steve
>>
>>
>> On 31 January 2011 15:48, Madovsky <infos at madovsky.org> wrote:
>>>
>>> so the best is,
>>> don't use G729 and go to buy food with your $10 ;)
>>>
>>> ----- Original Message -----
>>> From: Steven Ayre
>>> To: FreeSWITCH Users Help
>>> Sent: Monday, January 31, 2011 7:53 AM
>>> Subject: Re: [Freeswitch-users] open g729
>>> The price is set by the patent holders, there's not much that can be done
>>> about that.
>>>
>>> If? you want to evaluate the g729 transcoding performance you *must* test
>>> with the one you plan to recommend - the mod_com_g729, mod_sangoma_codec and
>>> ipp g729 codecs have very difference performance characteristics. It is
>>> pointless to evaluate one and then recommend another.
>>>
>>> - mod_com_g729 is the official $10/channel license. It has good
>>> performance. To evaluate it you should try it for yourself.
>>> - The unofficial illegal FS module based ipp g729 which I shall not name
>>> has stability problems and uses far more CPU than mod_com_g729. Do not use
>>> it.
>>> - mod_sangoma_codec uses almost no CPU, but you do have to purchase a
>>> transcoding card.
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 31 January 2011 12:44, Gustavo Espeche
>>> <gustavo.espeche at upper-soft.com> wrote:
>>>>
>>>> I just need test the performance of trans-coding before to recommend to
>>>> may customer what server it need and buy the codec, because of then i
>>>> need testing in my lab how much cpu and memory require the trans-coding
>>>> in freeswitch for g729.
>>>> And searching i found the open-g729/g723 for asterisk.
>>>> I relay don't wish to start a discussion about the g729 license i read
>>>> and understated it. But i think that the license price per channel is to
>>>> hight taking care that this codec has more than 10 years but the way
>>>> this is only my opinion.
>>>>
>>>> Best Regards.
>>>> Gustavo Espeche
>>>> www.easyipcall.com
>>>>
>>>>
>>>> On Sat, 2011-01-29 at 17:27 -0600, Rupa Schomaker wrote:
>>>> > How do you intend to pay the license fee?
>>>> >
>>>> > On Sat, Jan 29, 2011 at 5:23 PM, Gustavo Espeche
>>>> > <gustavo.espeche at upper-soft.com> wrote:
>>>> > > Hello,
>>>> > > ? ?some one can compile open g729 to work with freeswitch?
>>>> > > ? ?http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
>>>> > >
>>>> > > ? ?i appreciate a lot if some one has some experience in it.
>>>> > > ? ?Best Regards.
>>>> > >
>>>> > > ? ?Gustavo Espeche
>>>> > > ? ?www.easyipcall.com
>>>> > >
>>>> > >
>>>> > >
>>>> > >
>>>> > >
>>>> > > _______________________________________________
>>>> > > FreeSWITCH-users mailing list
>>>> > > FreeSWITCH-users at lists.freeswitch.org
>>>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > >
>>>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > > http://www.freeswitch.org
>>>> > >
>>>> >
>>>> >
>>>> >
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> ________________________________
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> ________________________________
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> ________________________________
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From infos at madovsky.org  Mon Jan 31 21:06:50 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 31 Jan 2011 13:06:50 -0500
Subject: [Freeswitch-users] minimum UDP packet size
References: <65064C9F3B344E4A9CAE4A9B63C318D3@e1705>
	<5DDC294B-BCB6-42FA-8243-885625ADAF2E@ipeva.fr>
Message-ID: <DD30C2D3ADBD47EBB03DDD75BB61AA98@e1705>

I talk about rtp media packet in UDP..
actually what is the minimum rtp packet size sent by FS ?

thanks
  ----- Original Message ----- 
  From: David Ponzone 
  To: FreeSWITCH Users Help 
  Sent: Monday, January 31, 2011 12:46 PM
  Subject: Re: [Freeswitch-users] minimum UDP packet size


  Sir,


  I am not really sure what you are trying to accomplish.
  How do you want to avoid fragmentation by setting a minimum packet size ?
  You can't really avoid fragmentation.
  If a SIP packet is larger than 1500, than it is larger.
  If you want to avoid fragmentation, you will need to suppress unnecessary SIP headers, or eventually you may try to enable short headers but that's not quite supported by all vendors.
  Supressing unnecessary headers is quite easy with a B2BUA like FreeSWITCH, as most headers should already not be forwarded.
  SIP INVITEs over 1500 is generally a problem when you proxy, adding Route information, etc...


  David Ponzone  Direction Technique
  email: david.ponzone at ipeva.fr
  tel:      01 74 03 18 97
  gsm:   06 66 98 76 34


  Service Client IPeva
  tel:      0811 46 26 26
  www.ipeva.fr  -   www.ipeva-studio.com


  Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.







  Le 31/01/2011 ? 18:29, Madovsky a ?crit :


    is it useful to set minimum UDP  pakcet to the
    MTU rate like 1500 and avoid any fragmentation ?

    Thanks
    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From david.ponzone at ipeva.fr  Mon Jan 31 21:22:23 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 31 Jan 2011 19:22:23 +0100
Subject: [Freeswitch-users] minimum UDP packet size
In-Reply-To: <DD30C2D3ADBD47EBB03DDD75BB61AA98@e1705>
References: <65064C9F3B344E4A9CAE4A9B63C318D3@e1705>
	<5DDC294B-BCB6-42FA-8243-885625ADAF2E@ipeva.fr>
	<DD30C2D3ADBD47EBB03DDD75BB61AA98@e1705>
Message-ID: <72DE7141-3090-419A-B4C1-7386E97CC369@ipeva.fr>

Well, if you do voice and your RTP packets are bigger than 1500 bytes, you have an issue.
You would need to use a large ptime (awful quality), even with g711, to send such packets.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 31/01/2011 ? 19:06, Madovsky a ?crit :

> I talk about rtp media packet in UDP..
> actually what is the minimum rtp packet size sent by FS ?
>  
> thanks
> ----- Original Message -----
> From: David Ponzone
> To: FreeSWITCH Users Help
> Sent: Monday, January 31, 2011 12:46 PM
> Subject: Re: [Freeswitch-users] minimum UDP packet size
> 
> Sir,
> 
> I am not really sure what you are trying to accomplish.
> How do you want to avoid fragmentation by setting a minimum packet size ?
> You can't really avoid fragmentation.
> If a SIP packet is larger than 1500, than it is larger.
> If you want to avoid fragmentation, you will need to suppress unnecessary SIP headers, or eventually you may try to enable short headers but that's not quite supported by all vendors.
> Supressing unnecessary headers is quite easy with a B2BUA like FreeSWITCH, as most headers should already not be forwarded.
> SIP INVITEs over 1500 is generally a problem when you proxy, adding Route information, etc...
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 31/01/2011 ? 18:29, Madovsky a ?crit :
> 
>> is it useful to set minimum UDP  pakcet to the
>> MTU rate like 1500 and avoid any fragmentation ?
>>  
>> Thanks
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Mon Jan 31 21:46:22 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 31 Jan 2011 12:46:22 -0600
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTi=gJsVKCJifVaBL_H6ZnPHTD__aQPhPeVfdx-ju@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
	<AANLkTi=in3CLd-n0mT3SRKYA0kxLMGVWk9KeBx3_jhY6@mail.gmail.com>
	<AANLkTin3Bu+8EEr_v_GaRGNPzOjfcnfzsV2Av2cgnWgO@mail.gmail.com>
	<AANLkTi=OC_jx0+nzK405VZURLDJPD25R4ZjyJ3RXJ1nk@mail.gmail.com>
	<AANLkTi=Sm7CjoayiE8nHmL0Eehqknj1jdpbx0YBQJQGH@mail.gmail.com>
	<AANLkTinr0RdgBs2UiS3=b3i+k6uCwkMihDy_FbAOFPc_@mail.gmail.com>
	<AANLkTi=+aObW3kVBZq5ox4b4E1uFTSgtXUL72tHLOzPi@mail.gmail.com>
	<AANLkTikVDKS40dVsf8x=+5=qnW59LqJbSv1H=vtROFaM@mail.gmail.com>
	<AANLkTi=gJsVKCJifVaBL_H6ZnPHTD__aQPhPeVfdx-ju@mail.gmail.com>
Message-ID: <AANLkTimeuCHf_7Y1TErPwrJbdGZ3DoQmeqQbJ1_3YNpT@mail.gmail.com>

There were several bugs identified in the latest firmware of SNOM that
made some things not work in FS.
In older revisions of FS we were trying harder to ignore those
behaviors but now there are some issues we need to work around or get
SNOM to fix the code but we have been waiting almost a year now for
any interaction with them and this is with me sitting in person with
their CEO at one point.



On Mon, Jan 31, 2011 at 10:47 AM, Patrick Plattes
<patrick.plattes at niemann-frey.info> wrote:
> 2011/1/31 Anthony Minessale <anthony.minessale at gmail.com>:
>> You need to have it working on the latest GIT and you need to
>> eliminate presence_in and presence_out 100% and use presence_id
>> variable as explained. ?This is how you do it and it will work the way
>> you want.
>
> I've eleminated presence in and out. There was just the two lines in
> the dialplan.
>
>> If you think presence does not work on latest, you need to report what
>> phone and model you are using. ?I will guess since you say you are in
>> Germany and you have presence that is not working that it's a SNOM.
>> Is that correct?
>
> That's correct. I tried it with a Snom 300. Isn't it compliance with the RFC?
>
> Bye
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From patrick.plattes at niemann-frey.info  Mon Jan 31 22:19:44 2011
From: patrick.plattes at niemann-frey.info (Patrick Plattes)
Date: Mon, 31 Jan 2011 20:19:44 +0100
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTimeuCHf_7Y1TErPwrJbdGZ3DoQmeqQbJ1_3YNpT@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
	<AANLkTi=in3CLd-n0mT3SRKYA0kxLMGVWk9KeBx3_jhY6@mail.gmail.com>
	<AANLkTin3Bu+8EEr_v_GaRGNPzOjfcnfzsV2Av2cgnWgO@mail.gmail.com>
	<AANLkTi=OC_jx0+nzK405VZURLDJPD25R4ZjyJ3RXJ1nk@mail.gmail.com>
	<AANLkTi=Sm7CjoayiE8nHmL0Eehqknj1jdpbx0YBQJQGH@mail.gmail.com>
	<AANLkTinr0RdgBs2UiS3=b3i+k6uCwkMihDy_FbAOFPc_@mail.gmail.com>
	<AANLkTi=+aObW3kVBZq5ox4b4E1uFTSgtXUL72tHLOzPi@mail.gmail.com>
	<AANLkTikVDKS40dVsf8x=+5=qnW59LqJbSv1H=vtROFaM@mail.gmail.com>
	<AANLkTi=gJsVKCJifVaBL_H6ZnPHTD__aQPhPeVfdx-ju@mail.gmail.com>
	<AANLkTimeuCHf_7Y1TErPwrJbdGZ3DoQmeqQbJ1_3YNpT@mail.gmail.com>
Message-ID: <AANLkTikDxzntMFRpFhfiqJ4pzvjsTQR4w_FfyLfT5Y7z@mail.gmail.com>

Hi :-)

I think SNOM has a problem with the quality of the firmware. maybe
they should glp it, so we can fix it ;-)

Tomorrow I will test it with an Grandstream GPX-2000 maybe it works
with this device. If it does I will also ask SNOM for a fix.

Thanks,
 Patrick


From marcin321 at hotmail.com  Mon Jan 31 21:12:30 2011
From: marcin321 at hotmail.com (Marcin Wojtowicz)
Date: Mon, 31 Jan 2011 13:12:30 -0500
Subject: [Freeswitch-users] Outbound only calls don't connect when
 bypass_media is true.
In-Reply-To: <E826C94A3C6BE44198FAF4D66B5AB4D80FEC3E5B24@VA3DIAXVS351.RED001.local>
References: <COL118-W52E5226FC3337085AB6AFAE5E00@phx.gbl>, ,
	<COL118-W38F26048D441DAA2777942E5E30@phx.gbl>, ,
	<AANLkTinbiU4zNc-ZM+u+iST7E2aZ3zisgh5Lu2FTuBLt@mail.gmail.com>, ,
	<COL118-W196844C55E71088CAD1FC1E5E20@phx.gbl>, ,
	<AANLkTi=t0ixs6L3SgUMQzEPtgvFowbFYXCVOtmvzLTEW@mail.gmail.com>,
	<COL118-W34303A74089852D47F0E63E5E20@phx.gbl>,
	<E826C94A3C6BE44198FAF4D66B5AB4D80FEC3E5B24@VA3DIAXVS351.RED001.local>
Message-ID: <COL118-W33A8A74086A6BBE2E6B1B7E5E20@phx.gbl>


SDP looks ok to me, but there is one warning about ptime in iLBC below. I don't see how a wrong codec can be selected because I narrowed down my external profile inbound/outbound to PCMU only and my internal is iLBC at 30i only.


freeswitch at kuffel> recv 1206 bytes from udp/[74.63.41.218]:5060 at 17:59:32.187500:
   ------------------------------------------------------------------------
   INVITE sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms SIP/2.0
   Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK30d7505b;rport
   From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64
   To: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
   Contact: <sip:MYPHONE#@74.63.41.218>
   Call-ID: 1298959c0099d50d177b6bae689fc028 at 74.63.41.218
   CSeq: 102 INVITE
   User-Agent: VoIPMS/SERAST
   Max-Forwards: 70
   Remote-Party-ID: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;privacy=off;screen=no
   Date: Mon, 31 Jan 2011 17:59:17 GMT
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Supported: replaces
   Content-Type: application/sdp
   Content-Length: 515
   
   v=0
   o=root 2831 2831 IN IP4 74.63.41.218
   s=session
   c=IN IP4 74.63.41.218
   t=0 0
   m=audio 16884 RTP/AVP 0 4 3 8 112 5 10 7 18 111 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:4 G723/8000
   a=fmtp:4 annexa=no
   a=rtpmap:3 GSM/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:112 AAL2-G726-32/8000
   a=rtpmap:5 DVI4/8000
   a=rtpmap:10 L16/8000
   a=rtpmap:7 LPC/8000
   a=rtpmap:18 G729/8000
   a=fmtp:18 annexb=no
   a=rtpmap:111 G726-32/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20
   a=sendrecv
   ------------------------------------------------------------------------
send 396 bytes to udp/[74.63.41.218]:5060 at 17:59:32.187500:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK30d7505b;rport=5060
   From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64
   To: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
   Call-ID: 1298959c0099d50d177b6bae689fc028 at 74.63.41.218
   CSeq: 102 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05 -0600
   Content-Length: 0
   
   ------------------------------------------------------------------------
2011-01-31 12:59:32.187500 [NOTICE] switch_channel.c:808 New Channel sofia/external/MYPHONE#@74.63.41.218 [f35f408a-f863-4784-a308-8b4fb3284b70]
2011-01-31 12:59:32.187500 [INFO] mod_dialplan_xml.c:331 Processing MYPHONE# <MYPHONE#>->121628 in context public
2011-01-31 12:59:32.203125 [NOTICE] switch_ivr.c:1606 Transfer sofia/external/MYPHONE#@74.63.41.218 to XML[1001 at default]
2011-01-31 12:59:32.203125 [INFO] mod_dialplan_xml.c:331 Processing MYPHONE# <MYPHONE#>->1001 in context default
2011-01-31 12:59:32.234375 [NOTICE] switch_channel.c:808 New Channel sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060 [7230b9e8-37a7-4fc6-9b52-25740a6f7ca4]


2011-01-31 12:59:32.265625 [WARNING] sofia_glue.c:213 Codec iLBC payload 98 added to sdp wanting ptime 30 but it's already 20 (PCMU:0:20), disabling ptime.


send 1315 bytes to tcp/[32.140.14.196]:46743 at 17:59:32.265625:
   ------------------------------------------------------------------------
   INVITE sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP SIP/2.0
   Via: SIP/2.0/TCP 69.125.20.15;rport;branch=z9hG4bK797eFQ6rgQKmQ
   Route: <sip:M9jdt73ig0oOJSbt6Uyy at 32.140.14.196:46743>;transport=TCP
   Max-Forwards: 68
   From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa
   To: <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP>
   Call-ID: b281a1fa-a806-122e-f799-c1188a708e17
   CSeq: 7912322 INVITE
   Contact: <sip:mod_sofia at 69.125.20.15:5060;transport=tcp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05 -0600
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 234
   X-FS-Support: update_display
   Remote-Party-ID: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;party=calling;screen=no;privacy=off
   
   v=0
   o=FreeSWITCH 1296474878 1296474879 IN IP4 69.125.20.15
   s=FreeSWITCH
   c=IN IP4 69.125.20.15
   t=0 0
   m=audio 21894 RTP/AVP 0 98 101 13
   a=rtpmap:98 iLBC/8000
   a=fmtp:98 mode=30
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   ------------------------------------------------------------------------
recv 318 bytes from tcp/[32.140.14.196]:46743 at 17:59:37.093750:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/TCP 69.125.20.15;branch=z9hG4bK797eFQ6rgQKmQ;rport=5060;received=69.125.20.15
   To: <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155>
   From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa
   Call-ID: b281a1fa-a806-122e-f799-c1188a708e17
   CSeq: 7912322 INVITE
   Content-Length: 0
   
   ------------------------------------------------------------------------
recv 476 bytes from tcp/[32.140.14.196]:46743 at 17:59:42.296875:
   ------------------------------------------------------------------------
   SIP/2.0 180 Ringing
   Via: SIP/2.0/TCP 69.125.20.15;branch=z9hG4bK797eFQ6rgQKmQ;rport=5060;received=69.125.20.15
   Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP>
   From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa
   To: <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155>;tag=p4rl1jbfvmnbvfs1d5rktoj2
   Supported: 100rel
   Call-ID: b281a1fa-a806-122e-f799-c1188a708e17
   CSeq: 7912322 INVITE
   Allow: INVITE,ACK,CANCEL,OPTIONS,BYE
   Content-Length: 0
   
   ------------------------------------------------------------------------
2011-01-31 12:59:42.296875 [INFO] sofia.c:729 sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060 Update Callee ID to "Outbound Call" <M9jdt73ig0oOJSbt6Uyy>
2011-01-31 12:59:42.296875 [NOTICE] sofia.c:4724 Ring-Ready sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060!
2011-01-31 12:59:42.312500 [INFO] switch_ivr_originate.c:1101 Sending early media
2011-01-31 12:59:42.343750 [NOTICE] mod_sofia.c:2252 Pre-Answer sofia/external/MYPHONE#@74.63.41.218!
send 1079 bytes to udp/[74.63.41.218]:5060 at 17:59:42.343750:
   ------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK30d7505b;rport=5060
   From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64
   To: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>;tag=eK0X80BS0091S
   Call-ID: 1298959c0099d50d177b6bae689fc028 at 74.63.41.218
   CSeq: 102 INVITE
   Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05 -0600
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 247
   Remote-Party-ID: "121628" <sip:121628 at 192.168.1.100>;party=calling;privacy=off;screen=no
   
   v=0
   o=FreeSWITCH 1296476876 1296476877 IN IP4 69.125.20.15
   s=FreeSWITCH
   c=IN IP4 69.125.20.15
   t=0 0
   m=audio 19906 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20
   ------------------------------------------------------------------------
recv 772 bytes from tcp/[32.140.14.196]:46743 at 17:59:43.812500:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/TCP 69.125.20.15;branch=z9hG4bK797eFQ6rgQKmQ;rport=5060;received=69.125.20.15
   To: <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155>;tag=p4rl1jbfvmnbvfs1d5rktoj2
   Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP>
   From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa
   Supported: timer
   Call-ID: b281a1fa-a806-122e-f799-c1188a708e17
   CSeq: 7912322 INVITE
   Allow: INVITE,ACK,CANCEL,OPTIONS,BYE
   Content-Type: application/sdp
   Content-Length: 269
   
   v=0
   o=M9jdt73ig0oOJSbt6Uyy 63464734759229750 63464734759229750 IN IP4 10.208.245.155
   s=-
   c=IN IP4 10.208.245.155
   t=0 0
   m=audio 49152 RTP/AVP 98 101
   a=sendrecv
   a=rtpmap:98 iLBC/8000
   a=ptime:30
   a=maxptime:180
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   ------------------------------------------------------------------------
send 464 bytes to tcp/[32.140.14.196]:46743 at 17:59:43.828125:
   ------------------------------------------------------------------------
   ACK sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP SIP/2.0
   Via: SIP/2.0/TCP 69.125.20.15;rport;branch=z9hG4bK8j17gjQvD096j
   Max-Forwards: 70
   From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa
   To: <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP>;tag=p4rl1jbfvmnbvfs1d5rktoj2
   Call-ID: b281a1fa-a806-122e-f799-c1188a708e17
   CSeq: 7912322 ACK
   Contact: <sip:mod_sofia at 69.125.20.15:5060;transport=tcp>
   Content-Length: 0
   
   ------------------------------------------------------------------------
2011-01-31 12:59:43.828125 [NOTICE] sofia.c:5230 Channel [sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060] has been answered
send 1061 bytes to udp/[74.63.41.218]:5060 at 17:59:43.843750:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK30d7505b;rport=5060
   From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64
   To: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>;tag=eK0X80BS0091S
   Call-ID: 1298959c0099d50d177b6bae689fc028 at 74.63.41.218
   CSeq: 102 INVITE
   Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp>
2011-01-31 12:59:43.843750 [NOTICE] switch_ivr_originate.c:3328 Channel [sofia/external/MYPHONE#@74.63.41.218] has been answered
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05 -0600
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 247
   Remote-Party-ID: "Outbound Call" <sip:M9jdt73ig0oOJSbt6Uyy at 192.168.1.100>;party=calling;privacy=off;screen=no
   
   v=0
   o=FreeSWITCH 1296476876 1296476877 IN IP4 69.125.20.15
   s=FreeSWITCH
   c=IN IP4 69.125.20.15
   t=0 0
   m=audio 19906 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20
   ------------------------------------------------------------------------
recv 533 bytes from udp/[74.63.41.218]:5060 at 17:59:43.859375:
   ------------------------------------------------------------------------
   ACK sip:gw+voip.ms at 69.125.20.15:5080;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK0cbbc6ef;rport
   From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64
   To: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>;tag=eK0X80BS0091S
   Contact: <sip:MYPHONE#@74.63.41.218>
   Call-ID: 1298959c0099d50d177b6bae689fc028 at 74.63.41.218
   CSeq: 102 ACK
   User-Agent: VoIPMS/SERAST
   Max-Forwards: 70
   Remote-Party-ID: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;privacy=off;screen=no
   Content-Length: 0

From: robert.hadley at teotech.com
To: freeswitch-users at lists.freeswitch.org
Date: Mon, 31 Jan 2011 09:33:48 -0800
Subject: Re: [Freeswitch-users] Outbound only calls don't connect	when	bypass_media is true.



 Check the codecs in the SDP or try manual hardcoding the codecs presented for both legs, we had a squeal problem going to a softphone that turned out to be the BV32 codec was being selected instead of SPEEX16. Robert From: Marcin Wojtowicz [mailto:marcin321 at hotmail.com] 
Sent: Monday, January 31, 2011 9:17 AM
To: freeswitch
Subject: Re: [Freeswitch-users] Outbound only calls don't connect when bypass_media is true. Yes, I had it set up to iLBC at 30i. It's not my cell phone (configured to ilbc, ptime=30 and mode=30), because when I call my freeswitch voicemail number, the sound is fine. I suspect it is something on the voip.ms <-> freeswitch leg because I created a sample ringback (8khz, mono, 16bit) wave file and directed my dialplan to it, but when I call from my home number to my cell, instead of hearing the ringer, I get choppy squeal.

?

> Date: Mon, 31 Jan 2011 10:40:10 -0600
> From: anthony.minessale at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Outbound only calls don't connect when bypass_media is true.
> 
> Many things have problems doing iLBC right.
> I recommend you define it in your configs as iLBC at 30i or it will try
> using the 20ms version which is not compatible with many other
> platforms. Also make sure you are on the latest version of FS since
> we have tweaked iLBC behavior to compensate for problems like this.
> 
> 
> 
> On Sun, Jan 30, 2011 at 10:40 PM, Marcin Wojtowicz
> <marcin321 at hotmail.com> wrote:
> > OK, so I gave up on bypass media, but now I have another problem. This time
> > I set up freeswitch to communicate with voip.ms using PCMU codec (configured
> > in my external profile), and use iLBC on my phone (codec configured in my
> > internal profile, where the phone registers). When I call my mobile it
> > rings, but when I pick up all I hear is a high pitched squeal. Am I missing
> > something here?
> >
> >> Date: Sun, 30 Jan 2011 16:34:09 -0600
> >> From: anthony.minessale at gmail.com
> >> To: freeswitch-users at lists.freeswitch.org
> >> Subject: Re: [Freeswitch-users] Outbound only calls don't connect when
> >> bypass_media is true.
> >>
> >> Just do not use bypass media.
> >> That is all you can do in that situation.
> >>
> >>
> >> On Sun, Jan 30, 2011 at 3:44 PM, Marcin Wojtowicz <marcin321 at hotmail.com>
> >> wrote:
> >> > I just want to add that I enabled STUN on my cell so now the SDP message
> >> > in
> >> > the INVITE to voip.ms contains the public IP of my phone, but it still
> >> > doesn't work.
> >> >
> >> > ________________________________
> >> > From: marcin321 at hotmail.com
> >> > To: freeswitch-users at lists.freeswitch.org
> >> > Date: Fri, 28 Jan 2011 19:54:19 -0500
> >> > Subject: [Freeswitch-users] Outbound only calls don't connect when
> >> > bypass_media is true.
> >> >
> >> > Hello,
> >> >
> >> > I'm a new user of freeswitch, so please bear with me. I have the
> >> > following setup:
> >> > voip.ms <- SIP over UDP -> my desktop running freeswitch <- SIP over TCP
> >> > ->
> >> > my nokia cellphone on AT&T wireless. This setup is intended to conserve
> >> > the
> >> > battery usage.
> >> > I've managed to make everything work well when I'm calling in over any
> >> > phone
> >> > to my cell phone, and freeswitch is enabled to work in bypass_media =
> >> > true,
> >> > even though by cell is behind NAT on at&t's network. Things break when I
> >> > pick up my cell and try to call my home phone (or any phone for that
> >> > matter). This is the relevant snippet from my dialplan:
> >> > <extension name="outbound">
> >> >   <condition field="destination_number"
> >> > expression="^1?([2-9]\d{2}[2-9]\d{6})$">
> >> >     <!--<action application="set" data="bypass_media=true"/>-->
> >> >     <action application="bridge" data="sofia/gateway/voip.ms/1$1"/>
> >> >   </condition>
> >> > </extension>
> >> >
> >> > Like shown above, my call will go to my home phone. When I uncomment the
> >> > bypass_media tag, my call will not connect. Here are the siptraces
> >> > I replaced my real home phone number in the with "MYPHONE".
> >> >
> >> > recv 940 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.406250:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> >> >    Via: SIP/2.0/TCP
> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport
> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >    To: <sip:MYPHONE at 192.168.1.100>
> >> >    Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>
> >> >    Supported: 100rel,timer
> >> >    CSeq: 5244503 INVITE
> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >    Allow:
> >> > UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE
> >> >    User-Agent: S60 RM-624 v 20.2.042 (en)
> >> >    Expires: 120
> >> >    Privacy: None
> >> >    Session-Expires: 1800
> >> >    Max-Forwards: 70
> >> >    Content-Type: application/sdp
> >> >    Accept-Language: en
> >> >    Content-Length: 292
> >> >
> >> >    v=0
> >> >    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> >> >    s=-
> >> >    c=IN IP4 10.153.174.6
> >> >    t=0 0
> >> >    m=audio 49152 RTP/AVP 18 97 98
> >> >    a=sendrecv
> >> >    a=rtpmap:18 G729/8000
> >> >    a=ptime:20
> >> >    a=maxptime:40
> >> >    a=fmtp:18 annexb=no
> >> >    a=rtpmap:97 iLBC/8000
> >> >    a=rtpmap:98 telephone-event/8000
> >> >    a=fmtp:98 0-15
> >> >
> >> > ------------------------------------------------------------------------
> >> > send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    SIP/2.0 100 Trying
> >> >    Via: SIP/2.0/TCP
> >> >
> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180
> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >    To: <sip:MYPHONE at 192.168.1.100>
> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >    CSeq: 5244503 INVITE
> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> > 18-04-05
> >> > -0600
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > send 875 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    SIP/2.0 407 Proxy Authentication Required
> >> >    Via: SIP/2.0/TCP
> >> >
> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180
> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj2011-01-28
> >> > 16:15:58.406250 [WARNING] sofia_reg.c:1241 SIP auth challenge (INVITE)
> >> > on
> >> > sofia profile 'internal' for [MYPHONE at 192.168.1.100] from ip
> >> > 32.136.78.180
> >> >
> >> >    To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m
> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >    CSeq: 5244503 INVITE
> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> > 18-04-05
> >> > -0600
> >> >    Accept: application/sdp
> >> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >> >    Supported: timer, precondition, path, replaces
> >> >    Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
> >> > sla,
> >> > include-session-description, presence.winfo, message-summary, refer
> >> >    Proxy-Authenticate: Digest realm="192.168.1.100",
> >> > nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28", algorithm=MD5, qop="auth"
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > recv 368 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.765625:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> >> >    Via: SIP/2.0/TCP
> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport
> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >    To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m
> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >    CSeq: 5244503 ACK
> >> >    Supported: sec-agree
> >> >    Max-Forwards: 70
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > recv 1222 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.406250:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> >> >    Via: SIP/2.0/TCP
> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport
> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >    To: <sip:MYPHONE at 192.168.1.100>
> >> >    Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>
> >> >    Supported: 100rel,timer
> >> >    CSeq: 5244504 INVITE
> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >    Allow:
> >> > UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE
> >> >    User-Agent: S60 RM-624 v 20.2.042 (en)
> >> >    Expires: 120
> >> >    Privacy: None
> >> >    Session-Expires: 1800
> >> >    Max-Forwards: 70
> >> >    Proxy-Authorization: Digest
> >> >
> >> > qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"
> >> >    Content-Type: application/sdp
> >> >    Accept-Language: en
> >> >    Content-Length: 292
> >> >
> >> >    v=0
> >> >    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> >> >    s=-
> >> >    c=IN IP4 10.153.174.6
> >> >    t=0 0
> >> >    m=audio 49152 RTP/AVP 18 97 98
> >> >    a=sendrecv
> >> >    a=rtpmap:18 G729/8000
> >> >    a=ptime:20
> >> >    a=maxptime:40
> >> >    a=fmtp:18 annexb=no
> >> >    a=rtpmap:97 iLBC/8000
> >> >    a=rtpmap:98 telephone-event/8000
> >> >    a=fmtp:98 0-15
> >> >
> >> > ------------------------------------------------------------------------
> >> > send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.406250:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    SIP/2.0 100 Trying
> >> >    Via: SIP/2.0/TCP
> >> >
> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180
> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >    To: <sip:MYPHONE at 192.168.1.100>
> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >    CSeq: 5244504 INVITE
> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> > 18-04-05
> >> > -0600
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > 2011-01-28 16:15:59.406250 [NOTICE] switch_channel.c:808 New Channel
> >> > sofia/internal/1001 at 192.168.1.100 [e5841001-04bd-4e16-9519-64ff2c7a8c2f]
> >> > 2011-01-28 16:15:59.421875 [INFO] mod_dialplan_xml.c:331 Processing 1001
> >> > <1001>->MYPHONE in context default
> >> > 2011-01-28 16:15:59.437500 [NOTICE] switch_channel.c:808 New Channel
> >> > sofia/external/1MYPHONE [60940227-9ae0-4c0c-abdb-f3988852fae0]
> >> > send 1141 bytes to udp/[74.63.41.218]:5060 at 21:15:59.453125:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0
> >> >    Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS
> >> >    Max-Forwards: 69
> >> >    From: "Extension 1001"
> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >    To: <sip:1MYPHONE at newyork.voip.ms>
> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >    CSeq: 7788615 INVITE
> >> >    Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> > 18-04-05
> >> > -0600
> >> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> > REGISTER, REFER, NOTIFY
> >> >    Supported: timer, precondition, path, replaces
> >> >    Allow-Events: talk, hold, refer
> >> >    Content-Type: application/sdp
> >> >    Content-Disposition: session
> >> >    Content-Length: 280
> >> >    X-FS-Support: update_display
> >> >    Remote-Party-ID: "Extension 1001"
> >> > <sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off
> >> >
> >> >    v=0
> >> >    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> >> >    s=-
> >> >    c=IN IP4 10.153.174.6
> >> >    t=0 0
> >> >    m=audio 49152 RTP/AVP 18 97 98
> >> >    a=rtpmap:18 G729/8000
> >> >    a=fmtp:18 annexb=no
> >> >    a=rtpmap:97 iLBC/8000
> >> >    a=rtpmap:98 telephone-event/8000
> >> >    a=fmtp:98 0-15
> >> >    a=ptime:20
> >> >    a=maxptime:40
> >> >
> >> > ------------------------------------------------------------------------
> >> > recv 570 bytes from udp/[74.63.41.218]:5060 at 21:15:59.468750:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    SIP/2.0 407 Proxy Authentication Required
> >> >    Via: SIP/2.0/UDP
> >> >
> >> > 69.125.20.15:5080;branch=z9hG4bKNp4ZFeKSD43tS;received=69.125.20.15;rport=5080
> >> >    From: "Extension 1001"
> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843
> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >    CSeq: 7788615 INVITE
> >> >    User-Agent: VoIPMS/SERAST
> >> >    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> >> >    Supported: replaces
> >> >    Proxy-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms",
> >> > nonce="2d534dd6"
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0
> >> >    Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS
> >> >    Max-Forwards: 69
> >> >    From: "Extension 1001"
> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843
> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >    CSeq: 7788615 ACK
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > send 1344 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0
> >> >    Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN
> >> >    Max-Forwards: 69
> >> >    From: "Extension 1001"
> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >    To: <sip:1MYPHONE at newyork.voip.ms>
> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >    CSeq: 7788616 INVITE
> >> >    Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
> >> >    Expires: 300
> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> > 18-04-05
> >> > -0600
> >> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> > REGISTER, REFER, NOTIFY
> >> >    Supported: timer, precondition, path, replaces
> >> >    Allow-Events: talk, hold, refer
> >> >    Proxy-Authorization: Digest username="121628",
> >> > realm="newyork.voip.ms",
> >> > nonce="2d534dd6", algorithm=MD5, uri="sip:1MYPHONE at newyork.voip.ms",
> >> > response="16f3301efae13df926da7550f709d28a"
> >> >    Content-Type: application/sdp
> >> >    Content-Disposition: session
> >> >    Content-Length: 280
> >> >    X-FS-Support: update_display
> >> >    Remote-Party-ID: "Extension 1001"
> >> > <sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off
> >> >
> >> >    v=0
> >> >    o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
> >> >    s=-
> >> >    c=IN IP4 10.153.174.6
> >> >    t=0 0
> >> >    m=audio 49152 RTP/AVP 18 97 98
> >> >    a=rtpmap:18 G729/8000
> >> >    a=fmtp:18 annexb=no
> >> >    a=rtpmap:97 iLBC/8000
> >> >    a=rtpmap:98 telephone-event/8000
> >> >    a=fmtp:98 0-15
> >> >    a=ptime:20
> >> >    a=maxptime:40
> >> >
> >> > ------------------------------------------------------------------------
> >> > recv 488 bytes from udp/[74.63.41.218]:5060 at 21:15:59.484375:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    SIP/2.0 100 Trying
> >> >    Via: SIP/2.0/UDP
> >> >
> >> > 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080
> >> >    From: "Extension 1001"
> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >    To: <sip:1MYPHONE at newyork.voip.ms>
> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >    CSeq: 7788616 INVITE
> >> >    User-Agent: VoIPMS/SERAST
> >> >    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> >> >    Supported: replaces
> >> >    Contact: <sip:1MYPHONE at 74.63.41.218>
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > recv 516 bytes from udp/[74.63.41.218]:5060 at 21:15:59.562500:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    SIP/2.0 503 Service Unavailable
> >> >    Via: SIP/2.0/UDP
> >> >
> >> > 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080
> >> >    From: "Extension 1001"
> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9
> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >    CSeq: 7788616 INVITE
> >> >    User-Agent: VoIPMS/SERAST
> >> >    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> >> >    Supported: replaces
> >> >    Contact: <sip:1MYPHONE at 74.63.41.218>
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.562500:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0
> >> >    Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN
> >> >    Max-Forwards: 69
> >> >    From: "Extension 1001"
> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
> >> >    To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9
> >> >    Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
> >> >    CSeq: 7788616 ACK
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > 2011-01-28 16:15:59.562500 [INFO] mod_dptools.c:2612 Originate Failed.
> >> > Cause: NO_ANSWER
> >> > 2011-01-28 16:15:59.562500 [NOTICE] sofia.c:5286 Hangup
> >> > sofia/external/1MYPHONE [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
> >> > 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:189
> >> > sofia/internal/1001 at 192.168.1.100 has executed the last dialplan
> >> > instruction, hanging up.
> >> > 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:191
> >> > Hangup
> >> > sofia/internal/1001 at 192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING]
> >> > 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1306 Session 2
> >> > (sofia/external/1MYPHONE) Ended
> >> > 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1308 Close
> >> > Channel
> >> > sofia/external/1MYPHONE [CS_DESTROY]
> >> > send 887 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.593750:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    SIP/2.0 503 Service Unavailable
> >> >    Via: SIP/2.0/TCP
> >> >
> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180
> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >    To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg
> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >    CSeq: 5244504 INVITE
> >> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
> >> > 18-04-05
> >> > -0600
> >> >    Accept: application/sdp
> >> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >> >    Supported: timer, precondition, path, replaces
> >> >    Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
> >> > sla,
> >> > include-session-description, presence.winfo, message-summary, refer
> >> >    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
> >> > 2011-01-28 16:15:59.593750 [NOTICE] switch_core_session.c:1306 Session 1
> >> > (sofia/internal/1001 at 192.168.1.100) Ended
> >> >    Content-Length: 02011-01-28 16:15:59.593750 [NOTICE]
> >> > switch_core_session.c:1308 Close Channel
> >> > sofia/internal/1001 at 192.168.1.100
> >> > [CS_DESTROY]
> >> >
> >> >    Remote-Party-ID: "MYPHONE"
> >> > <sip:MYPHONE at 192.168.1.100>;party=calling;privacy=off;screen=no
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> > recv 650 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.953125:
> >> >
> >> > ------------------------------------------------------------------------
> >> >    ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
> >> >    Via: SIP/2.0/TCP
> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport
> >> >    From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
> >> >    To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg
> >> >    Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
> >> >    CSeq: 5244504 ACK
> >> >    Supported: sec-agree
> >> >    Max-Forwards: 70
> >> >    Proxy-Authorization: Digest
> >> >
> >> > qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"
> >> >    Content-Length: 0
> >> >
> >> >
> >> > ------------------------------------------------------------------------
> >> >
> >> > Thank you in advance.
> >> >
> >> > _______________________________________________ FreeSWITCH-users mailing
> >> > list FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
_______________________________________________
FreeSWITCH-users mailing list
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Mon Jan 31 23:00:10 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 31 Jan 2011 14:00:10 -0600
Subject: [Freeswitch-users] Outbound only calls don't connect when
 bypass_media is true.
In-Reply-To: <COL118-W33A8A74086A6BBE2E6B1B7E5E20@phx.gbl>
References: <COL118-W52E5226FC3337085AB6AFAE5E00@phx.gbl>
	<COL118-W38F26048D441DAA2777942E5E30@phx.gbl>
	<AANLkTinbiU4zNc-ZM+u+iST7E2aZ3zisgh5Lu2FTuBLt@mail.gmail.com>
	<COL118-W196844C55E71088CAD1FC1E5E20@phx.gbl>
	<AANLkTi=t0ixs6L3SgUMQzEPtgvFowbFYXCVOtmvzLTEW@mail.gmail.com>
	<COL118-W34303A74089852D47F0E63E5E20@phx.gbl>
	<E826C94A3C6BE44198FAF4D66B5AB4D80FEC3E5B24@VA3DIAXVS351.RED001.local>
	<COL118-W33A8A74086A6BBE2E6B1B7E5E20@phx.gbl>
Message-ID: <AANLkTikSs4sn-+x5G9pRpszogW7Hwng_cLJop0zrAueu@mail.gmail.com>

 a=fmtp:98 mode=30 is missing in the 200 ok from the phone.
Also did you mention the revision you are on. I had indicated that the
very latest code may have more tolerant ilbc codec code in it.
http://latest.freeswitch.org


On Mon, Jan 31, 2011 at 12:12 PM, Marcin Wojtowicz
<marcin321 at hotmail.com> wrote:
> SDP looks ok to me, but there is one warning about ptime in iLBC below. I
> don't see how a wrong codec can be selected because I narrowed down my
> external profile inbound/outbound to PCMU only and my internal is iLBC at 30i
> only.
>
>
> freeswitch at kuffel> recv 1206 bytes from udp/[74.63.41.218]:5060 at
> 17:59:32.187500:
> ?? ------------------------------------------------------------------------
> ?? INVITE sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms SIP/2.0
> ?? Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK30d7505b;rport
> ?? From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64
> ?? To: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
> ?? Contact: <sip:MYPHONE#@74.63.41.218>
> ?? Call-ID: 1298959c0099d50d177b6bae689fc028 at 74.63.41.218
> ?? CSeq: 102 INVITE
> ?? User-Agent: VoIPMS/SERAST
> ?? Max-Forwards: 70
> ?? Remote-Party-ID: "MYPHONE#"
> <sip:MYPHONE#@74.63.41.218>;privacy=off;screen=no
> ?? Date: Mon, 31 Jan 2011 17:59:17 GMT
> ?? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> ?? Supported: replaces
> ?? Content-Type: application/sdp
> ?? Content-Length: 515
>
> ?? v=0
> ?? o=root 2831 2831 IN IP4 74.63.41.218
> ?? s=session
> ?? c=IN IP4 74.63.41.218
> ?? t=0 0
> ?? m=audio 16884 RTP/AVP 0 4 3 8 112 5 10 7 18 111 101
> ?? a=rtpmap:0 PCMU/8000
> ?? a=rtpmap:4 G723/8000
> ?? a=fmtp:4 annexa=no
> ?? a=rtpmap:3 GSM/8000
> ?? a=rtpmap:8 PCMA/8000
> ?? a=rtpmap:112 AAL2-G726-32/8000
> ?? a=rtpmap:5 DVI4/8000
> ?? a=rtpmap:10 L16/8000
> ?? a=rtpmap:7 LPC/8000
> ?? a=rtpmap:18 G729/8000
> ?? a=fmtp:18 annexb=no
> ?? a=rtpmap:111 G726-32/8000
> ?? a=rtpmap:101 telephone-event/8000
> ?? a=fmtp:101 0-16
> ?? a=silenceSupp:off - - - -
> ?? a=ptime:20
> ?? a=sendrecv
> ?? ------------------------------------------------------------------------
> send 396 bytes to udp/[74.63.41.218]:5060 at 17:59:32.187500:
> ?? ------------------------------------------------------------------------
> ?? SIP/2.0 100 Trying
> ?? Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK30d7505b;rport=5060
> ?? From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64
> ?? To: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
> ?? Call-ID: 1298959c0099d50d177b6bae689fc028 at 74.63.41.218
> ?? CSeq: 102 INVITE
> ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> -0600
> ?? Content-Length: 0
>
> ?? ------------------------------------------------------------------------
> 2011-01-31 12:59:32.187500 [NOTICE] switch_channel.c:808 New Channel
> sofia/external/MYPHONE#@74.63.41.218 [f35f408a-f863-4784-a308-8b4fb3284b70]
> 2011-01-31 12:59:32.187500 [INFO] mod_dialplan_xml.c:331 Processing MYPHONE#
> <MYPHONE#>->121628 in context public
> 2011-01-31 12:59:32.203125 [NOTICE] switch_ivr.c:1606 Transfer
> sofia/external/MYPHONE#@74.63.41.218 to XML[1001 at default]
> 2011-01-31 12:59:32.203125 [INFO] mod_dialplan_xml.c:331 Processing MYPHONE#
> <MYPHONE#>->1001 in context default
> 2011-01-31 12:59:32.234375 [NOTICE] switch_channel.c:808 New Channel
> sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060
> [7230b9e8-37a7-4fc6-9b52-25740a6f7ca4]
>
>
> 2011-01-31 12:59:32.265625 [WARNING] sofia_glue.c:213 Codec iLBC payload 98
> added to sdp wanting ptime 30 but it's already 20 (PCMU:0:20), disabling
> ptime.
>
>
> send 1315 bytes to tcp/[32.140.14.196]:46743 at 17:59:32.265625:
> ?? ------------------------------------------------------------------------
> ?? INVITE sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP SIP/2.0
> ?? Via: SIP/2.0/TCP 69.125.20.15;rport;branch=z9hG4bK797eFQ6rgQKmQ
> ?? Route: <sip:M9jdt73ig0oOJSbt6Uyy at 32.140.14.196:46743>;transport=TCP
> ?? Max-Forwards: 68
> ?? From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa
> ?? To: <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP>
> ?? Call-ID: b281a1fa-a806-122e-f799-c1188a708e17
> ?? CSeq: 7912322 INVITE
> ?? Contact: <sip:mod_sofia at 69.125.20.15:5060;transport=tcp>
> ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> -0600
> ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> ?? Supported: timer, precondition, path, replaces
> ?? Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
> ?? Content-Type: application/sdp
> ?? Content-Disposition: session
> ?? Content-Length: 234
> ?? X-FS-Support: update_display
> ?? Remote-Party-ID: "MYPHONE#"
> <sip:MYPHONE#@192.168.1.100>;party=calling;screen=no;privacy=off
>
> ?? v=0
> ?? o=FreeSWITCH 1296474878 1296474879 IN IP4 69.125.20.15
> ?? s=FreeSWITCH
> ?? c=IN IP4 69.125.20.15
> ?? t=0 0
> ?? m=audio 21894 RTP/AVP 0 98 101 13
> ?? a=rtpmap:98 iLBC/8000
> ?? a=fmtp:98 mode=30
> ?? a=rtpmap:101 telephone-event/8000
> ?? a=fmtp:101 0-16
> ?? ------------------------------------------------------------------------
> recv 318 bytes from tcp/[32.140.14.196]:46743 at 17:59:37.093750:
> ?? ------------------------------------------------------------------------
> ?? SIP/2.0 100 Trying
> ?? Via: SIP/2.0/TCP
> 69.125.20.15;branch=z9hG4bK797eFQ6rgQKmQ;rport=5060;received=69.125.20.15
> ?? To: <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155>
> ?? From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa
> ?? Call-ID: b281a1fa-a806-122e-f799-c1188a708e17
> ?? CSeq: 7912322 INVITE
> ?? Content-Length: 0
>
> ?? ------------------------------------------------------------------------
> recv 476 bytes from tcp/[32.140.14.196]:46743 at 17:59:42.296875:
> ?? ------------------------------------------------------------------------
> ?? SIP/2.0 180 Ringing
> ?? Via: SIP/2.0/TCP
> 69.125.20.15;branch=z9hG4bK797eFQ6rgQKmQ;rport=5060;received=69.125.20.15
> ?? Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP>
> ?? From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa
> ?? To:
> <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155>;tag=p4rl1jbfvmnbvfs1d5rktoj2
> ?? Supported: 100rel
> ?? Call-ID: b281a1fa-a806-122e-f799-c1188a708e17
> ?? CSeq: 7912322 INVITE
> ?? Allow: INVITE,ACK,CANCEL,OPTIONS,BYE
> ?? Content-Length: 0
>
> ?? ------------------------------------------------------------------------
> 2011-01-31 12:59:42.296875 [INFO] sofia.c:729
> sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060 Update Callee ID
> to "Outbound Call" <M9jdt73ig0oOJSbt6Uyy>
> 2011-01-31 12:59:42.296875 [NOTICE] sofia.c:4724 Ring-Ready
> sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060!
> 2011-01-31 12:59:42.312500 [INFO] switch_ivr_originate.c:1101 Sending early
> media
> 2011-01-31 12:59:42.343750 [NOTICE] mod_sofia.c:2252 Pre-Answer
> sofia/external/MYPHONE#@74.63.41.218!
> send 1079 bytes to udp/[74.63.41.218]:5060 at 17:59:42.343750:
> ?? ------------------------------------------------------------------------
> ?? SIP/2.0 183 Session Progress
> ?? Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK30d7505b;rport=5060
> ?? From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64
> ?? To:
> <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>;tag=eK0X80BS0091S
> ?? Call-ID: 1298959c0099d50d177b6bae689fc028 at 74.63.41.218
> ?? CSeq: 102 INVITE
> ?? Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp>
> ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> -0600
> ?? Accept: application/sdp
> ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> ?? Supported: timer, precondition, path, replaces
> ?? Allow-Events: talk, hold, refer
> ?? Content-Type: application/sdp
> ?? Content-Disposition: session
> ?? Content-Length: 247
> ?? Remote-Party-ID: "121628"
> <sip:121628 at 192.168.1.100>;party=calling;privacy=off;screen=no
>
> ?? v=0
> ?? o=FreeSWITCH 1296476876 1296476877 IN IP4 69.125.20.15
> ?? s=FreeSWITCH
> ?? c=IN IP4 69.125.20.15
> ?? t=0 0
> ?? m=audio 19906 RTP/AVP 0 101
> ?? a=rtpmap:0 PCMU/8000
> ?? a=rtpmap:101 telephone-event/8000
> ?? a=fmtp:101 0-16
> ?? a=silenceSupp:off - - - -
> ?? a=ptime:20
> ?? ------------------------------------------------------------------------
> recv 772 bytes from tcp/[32.140.14.196]:46743 at 17:59:43.812500:
> ?? ------------------------------------------------------------------------
> ?? SIP/2.0 200 OK
> ?? Via: SIP/2.0/TCP
> 69.125.20.15;branch=z9hG4bK797eFQ6rgQKmQ;rport=5060;received=69.125.20.15
> ?? To:
> <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155>;tag=p4rl1jbfvmnbvfs1d5rktoj2
> ?? Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP>
> ?? From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa
> ?? Supported: timer
> ?? Call-ID: b281a1fa-a806-122e-f799-c1188a708e17
> ?? CSeq: 7912322 INVITE
> ?? Allow: INVITE,ACK,CANCEL,OPTIONS,BYE
> ?? Content-Type: application/sdp
> ?? Content-Length: 269
>
> ?? v=0
> ?? o=M9jdt73ig0oOJSbt6Uyy 63464734759229750 63464734759229750 IN IP4
> 10.208.245.155
> ?? s=-
> ?? c=IN IP4 10.208.245.155
> ?? t=0 0
> ?? m=audio 49152 RTP/AVP 98 101
> ?? a=sendrecv
> ?? a=rtpmap:98 iLBC/8000
> ?? a=ptime:30
> ?? a=maxptime:180
> ?? a=rtpmap:101 telephone-event/8000
> ?? a=fmtp:101 0-15
> ?? ------------------------------------------------------------------------
> send 464 bytes to tcp/[32.140.14.196]:46743 at 17:59:43.828125:
> ?? ------------------------------------------------------------------------
> ?? ACK sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP SIP/2.0
> ?? Via: SIP/2.0/TCP 69.125.20.15;rport;branch=z9hG4bK8j17gjQvD096j
> ?? Max-Forwards: 70
> ?? From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa
> ?? To:
> <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP>;tag=p4rl1jbfvmnbvfs1d5rktoj2
> ?? Call-ID: b281a1fa-a806-122e-f799-c1188a708e17
> ?? CSeq: 7912322 ACK
> ?? Contact: <sip:mod_sofia at 69.125.20.15:5060;transport=tcp>
> ?? Content-Length: 0
>
> ?? ------------------------------------------------------------------------
> 2011-01-31 12:59:43.828125 [NOTICE] sofia.c:5230 Channel
> [sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060] has been
> answered
> send 1061 bytes to udp/[74.63.41.218]:5060 at 17:59:43.843750:
> ?? ------------------------------------------------------------------------
> ?? SIP/2.0 200 OK
> ?? Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK30d7505b;rport=5060
> ?? From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64
> ?? To:
> <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>;tag=eK0X80BS0091S
> ?? Call-ID: 1298959c0099d50d177b6bae689fc028 at 74.63.41.218
> ?? CSeq: 102 INVITE
> ?? Contact: <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp>
> 2011-01-31 12:59:43.843750 [NOTICE] switch_ivr_originate.c:3328 Channel
> [sofia/external/MYPHONE#@74.63.41.218] has been answered
> ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05
> -0600
> ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> ?? Supported: timer, precondition, path, replaces
> ?? Allow-Events: talk, hold, refer
> ?? Content-Type: application/sdp
> ?? Content-Disposition: session
> ?? Content-Length: 247
> ?? Remote-Party-ID: "Outbound Call"
> <sip:M9jdt73ig0oOJSbt6Uyy at 192.168.1.100>;party=calling;privacy=off;screen=no
>
> ?? v=0
> ?? o=FreeSWITCH 1296476876 1296476877 IN IP4 69.125.20.15
> ?? s=FreeSWITCH
> ?? c=IN IP4 69.125.20.15
> ?? t=0 0
> ?? m=audio 19906 RTP/AVP 0 101
> ?? a=rtpmap:0 PCMU/8000
> ?? a=rtpmap:101 telephone-event/8000
> ?? a=fmtp:101 0-16
> ?? a=silenceSupp:off - - - -
> ?? a=ptime:20
> ?? ------------------------------------------------------------------------
> recv 533 bytes from udp/[74.63.41.218]:5060 at 17:59:43.859375:
> ?? ------------------------------------------------------------------------
> ?? ACK sip:gw+voip.ms at 69.125.20.15:5080;transport=udp SIP/2.0
> ?? Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK0cbbc6ef;rport
> ?? From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64
> ?? To:
> <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>;tag=eK0X80BS0091S
> ?? Contact: <sip:MYPHONE#@74.63.41.218>
> ?? Call-ID: 1298959c0099d50d177b6bae689fc028 at 74.63.41.218
> ?? CSeq: 102 ACK
> ?? User-Agent: VoIPMS/SERAST
> ?? Max-Forwards: 70
> ?? Remote-Party-ID: "MYPHONE#"
> <sip:MYPHONE#@74.63.41.218>;privacy=off;screen=no
> ?? Content-Length: 0
>
> ________________________________
> From: robert.hadley at teotech.com
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 31 Jan 2011 09:33:48 -0800
> Subject: Re: [Freeswitch-users] Outbound only calls don't connect when
> bypass_media is true.
>
>
>
> Check the codecs in the SDP or try manual hardcoding the codecs presented
> for both legs, we had a squeal problem going to a softphone that turned out
> to be the BV32 codec was being selected instead of SPEEX16.
>
>
>
> Robert
>
>
>
> From: Marcin Wojtowicz [mailto:marcin321 at hotmail.com]
> Sent: Monday, January 31, 2011 9:17 AM
> To: freeswitch
> Subject: Re: [Freeswitch-users] Outbound only calls don't connect when
> bypass_media is true.
>
>
>
> Yes, I had it set up to iLBC at 30i. It's not my cell phone (configured to
> ilbc, ptime=30 and mode=30), because when I call my freeswitch voicemail
> number, the sound is fine. I suspect it is something on the voip.ms <->
> freeswitch leg because I created a sample ringback (8khz, mono, 16bit) wave
> file and directed my dialplan to it, but when I call from my home number to
> my cell, instead of hearing the ringer, I get choppy squeal.
>
> ?
>
>> Date: Mon, 31 Jan 2011 10:40:10 -0600
>> From: anthony.minessale at gmail.com
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Outbound only calls don't connect when
>> bypass_media is true.
>>
>> Many things have problems doing iLBC right.
>> I recommend you define it in your configs as iLBC at 30i or it will try
>> using the 20ms version which is not compatible with many other
>> platforms. Also make sure you are on the latest version of FS since
>> we have tweaked iLBC behavior to compensate for problems like this.
>>
>>
>>
>> On Sun, Jan 30, 2011 at 10:40 PM, Marcin Wojtowicz
>> <marcin321 at hotmail.com> wrote:
>> > OK, so I gave up on bypass media, but now I have another problem. This
>> > time
>> > I set up freeswitch to communicate with voip.ms using PCMU codec
>> > (configured
>> > in my external profile), and use iLBC on my phone (codec configured in
>> > my
>> > internal profile, where the phone registers). When I call my mobile it
>> > rings, but when I pick up all I hear is a high pitched squeal. Am I
>> > missing
>> > something here?
>> >
>> >> Date: Sun, 30 Jan 2011 16:34:09 -0600
>> >> From: anthony.minessale at gmail.com
>> >> To: freeswitch-users at lists.freeswitch.org
>> >> Subject: Re: [Freeswitch-users] Outbound only calls don't connect when
>> >> bypass_media is true.
>> >>
>> >> Just do not use bypass media.
>> >> That is all you can do in that situation.
>> >>
>> >>
>> >> On Sun, Jan 30, 2011 at 3:44 PM, Marcin Wojtowicz
>> >> <marcin321 at hotmail.com>
>> >> wrote:
>> >> > I just want to add that I enabled STUN on my cell so now the SDP
>> >> > message
>> >> > in
>> >> > the INVITE to voip.ms contains the public IP of my phone, but it
>> >> > still
>> >> > doesn't work.
>> >> >
>> >> > ________________________________
>> >> > From: marcin321 at hotmail.com
>> >> > To: freeswitch-users at lists.freeswitch.org
>> >> > Date: Fri, 28 Jan 2011 19:54:19 -0500
>> >> > Subject: [Freeswitch-users] Outbound only calls don't connect when
>> >> > bypass_media is true.
>> >> >
>> >> > Hello,
>> >> >
>> >> > I'm a new user of freeswitch, so please bear with me. I have the
>> >> > following setup:
>> >> > voip.ms <- SIP over UDP -> my desktop running freeswitch <- SIP over
>> >> > TCP
>> >> > ->
>> >> > my nokia cellphone on AT&T wireless. This setup is intended to
>> >> > conserve
>> >> > the
>> >> > battery usage.
>> >> > I've managed to make everything work well when I'm calling in over
>> >> > any
>> >> > phone
>> >> > to my cell phone, and freeswitch is enabled to work in bypass_media =
>> >> > true,
>> >> > even though by cell is behind NAT on at&t's network. Things break
>> >> > when I
>> >> > pick up my cell and try to call my home phone (or any phone for that
>> >> > matter). This is the relevant snippet from my dialplan:
>> >> > <extension name="outbound">
>> >> > ? <condition field="destination_number"
>> >> > expression="^1?([2-9]\d{2}[2-9]\d{6})$">
>> >> > ??? <!--<action application="set" data="bypass_media=true"/>-->
>> >> > ??? <action application="bridge" data="sofia/gateway/voip.ms/1$1"/>
>> >> > ? </condition>
>> >> > </extension>
>> >> >
>> >> > Like shown above, my call will go to my home phone. When I uncomment
>> >> > the
>> >> > bypass_media tag, my call will not connect. Here are the siptraces
>> >> > I replaced my real home phone number in the with "MYPHONE".
>> >> >
>> >> > recv 940 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.406250:
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > ?? INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
>> >> > ?? Via: SIP/2.0/TCP
>> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport
>> >> > ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>> >> > ?? To: <sip:MYPHONE at 192.168.1.100>
>> >> > ?? Contact:
>> >> > <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>
>> >> > ?? Supported: 100rel,timer
>> >> > ?? CSeq: 5244503 INVITE
>> >> > ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
>> >> > ?? Allow:
>> >> > UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE
>> >> > ?? User-Agent: S60 RM-624 v 20.2.042 (en)
>> >> > ?? Expires: 120
>> >> > ?? Privacy: None
>> >> > ?? Session-Expires: 1800
>> >> > ?? Max-Forwards: 70
>> >> > ?? Content-Type: application/sdp
>> >> > ?? Accept-Language: en
>> >> > ?? Content-Length: 292
>> >> >
>> >> > ?? v=0
>> >> > ?? o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
>> >> > ?? s=-
>> >> > ?? c=IN IP4 10.153.174.6
>> >> > ?? t=0 0
>> >> > ?? m=audio 49152 RTP/AVP 18 97 98
>> >> > ?? a=sendrecv
>> >> > ?? a=rtpmap:18 G729/8000
>> >> > ?? a=ptime:20
>> >> > ?? a=maxptime:40
>> >> > ?? a=fmtp:18 annexb=no
>> >> > ?? a=rtpmap:97 iLBC/8000
>> >> > ?? a=rtpmap:98 telephone-event/8000
>> >> > ?? a=fmtp:98 0-15
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > ?? SIP/2.0 100 Trying
>> >> > ?? Via: SIP/2.0/TCP
>> >> >
>> >> >
>> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180
>> >> > ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>> >> > ?? To: <sip:MYPHONE at 192.168.1.100>
>> >> > ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
>> >> > ?? CSeq: 5244503 INVITE
>> >> > ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
>> >> > 18-04-05
>> >> > -0600
>> >> > ?? Content-Length: 0
>> >> >
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > send 875 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > ?? SIP/2.0 407 Proxy Authentication Required
>> >> > ?? Via: SIP/2.0/TCP
>> >> >
>> >> >
>> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180
>> >> > ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj2011-01-28
>> >> > 16:15:58.406250 [WARNING] sofia_reg.c:1241 SIP auth challenge
>> >> > (INVITE)
>> >> > on
>> >> > sofia profile 'internal' for [MYPHONE at 192.168.1.100] from ip
>> >> > 32.136.78.180
>> >> >
>> >> > ?? To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m
>> >> > ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
>> >> > ?? CSeq: 5244503 INVITE
>> >> > ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
>> >> > 18-04-05
>> >> > -0600
>> >> > ?? Accept: application/sdp
>> >> > ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> >> > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> >> > ?? Supported: timer, precondition, path, replaces
>> >> > ?? Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>> >> > sla,
>> >> > include-session-description, presence.winfo, message-summary, refer
>> >> > ?? Proxy-Authenticate: Digest realm="192.168.1.100",
>> >> > nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28", algorithm=MD5,
>> >> > qop="auth"
>> >> > ?? Content-Length: 0
>> >> >
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > recv 368 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.765625:
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > ?? ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
>> >> > ?? Via: SIP/2.0/TCP
>> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport
>> >> > ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>> >> > ?? To: <sip:MYPHONE at 192.168.1.100>;tag=FQy5v5emcyt1m
>> >> > ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
>> >> > ?? CSeq: 5244503 ACK
>> >> > ?? Supported: sec-agree
>> >> > ?? Max-Forwards: 70
>> >> > ?? Content-Length: 0
>> >> >
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > recv 1222 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.406250:
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > ?? INVITE sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
>> >> > ?? Via: SIP/2.0/TCP
>> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport
>> >> > ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>> >> > ?? To: <sip:MYPHONE at 192.168.1.100>
>> >> > ?? Contact:
>> >> > <sip:M9jdt73ig0oOJSbt6Uyy at 10.153.174.6:5060;transport=TCP>
>> >> > ?? Supported: 100rel,timer
>> >> > ?? CSeq: 5244504 INVITE
>> >> > ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
>> >> > ?? Allow:
>> >> > UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE
>> >> > ?? User-Agent: S60 RM-624 v 20.2.042 (en)
>> >> > ?? Expires: 120
>> >> > ?? Privacy: None
>> >> > ?? Session-Expires: 1800
>> >> > ?? Max-Forwards: 70
>> >> > ?? Proxy-Authorization: Digest
>> >> >
>> >> >
>> >> > qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"
>> >> > ?? Content-Type: application/sdp
>> >> > ?? Accept-Language: en
>> >> > ?? Content-Length: 292
>> >> >
>> >> > ?? v=0
>> >> > ?? o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
>> >> > ?? s=-
>> >> > ?? c=IN IP4 10.153.174.6
>> >> > ?? t=0 0
>> >> > ?? m=audio 49152 RTP/AVP 18 97 98
>> >> > ?? a=sendrecv
>> >> > ?? a=rtpmap:18 G729/8000
>> >> > ?? a=ptime:20
>> >> > ?? a=maxptime:40
>> >> > ?? a=fmtp:18 annexb=no
>> >> > ?? a=rtpmap:97 iLBC/8000
>> >> > ?? a=rtpmap:98 telephone-event/8000
>> >> > ?? a=fmtp:98 0-15
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.406250:
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > ?? SIP/2.0 100 Trying
>> >> > ?? Via: SIP/2.0/TCP
>> >> >
>> >> >
>> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180
>> >> > ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>> >> > ?? To: <sip:MYPHONE at 192.168.1.100>
>> >> > ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
>> >> > ?? CSeq: 5244504 INVITE
>> >> > ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
>> >> > 18-04-05
>> >> > -0600
>> >> > ?? Content-Length: 0
>> >> >
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > 2011-01-28 16:15:59.406250 [NOTICE] switch_channel.c:808 New Channel
>> >> > sofia/internal/1001 at 192.168.1.100
>> >> > [e5841001-04bd-4e16-9519-64ff2c7a8c2f]
>> >> > 2011-01-28 16:15:59.421875 [INFO] mod_dialplan_xml.c:331 Processing
>> >> > 1001
>> >> > <1001>->MYPHONE in context default
>> >> > 2011-01-28 16:15:59.437500 [NOTICE] switch_channel.c:808 New Channel
>> >> > sofia/external/1MYPHONE [60940227-9ae0-4c0c-abdb-f3988852fae0]
>> >> > send 1141 bytes to udp/[74.63.41.218]:5060 at 21:15:59.453125:
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > ?? INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0
>> >> > ?? Via: SIP/2.0/UDP
>> >> > 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS
>> >> > ?? Max-Forwards: 69
>> >> > ?? From: "Extension 1001"
>> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>> >> > ?? To: <sip:1MYPHONE at newyork.voip.ms>
>> >> > ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
>> >> > ?? CSeq: 7788615 INVITE
>> >> > ?? Contact:
>> >> > <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
>> >> > ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
>> >> > 18-04-05
>> >> > -0600
>> >> > ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> >> > REGISTER, REFER, NOTIFY
>> >> > ?? Supported: timer, precondition, path, replaces
>> >> > ?? Allow-Events: talk, hold, refer
>> >> > ?? Content-Type: application/sdp
>> >> > ?? Content-Disposition: session
>> >> > ?? Content-Length: 280
>> >> > ?? X-FS-Support: update_display
>> >> > ?? Remote-Party-ID: "Extension 1001"
>> >> > <sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off
>> >> >
>> >> > ?? v=0
>> >> > ?? o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
>> >> > ?? s=-
>> >> > ?? c=IN IP4 10.153.174.6
>> >> > ?? t=0 0
>> >> > ?? m=audio 49152 RTP/AVP 18 97 98
>> >> > ?? a=rtpmap:18 G729/8000
>> >> > ?? a=fmtp:18 annexb=no
>> >> > ?? a=rtpmap:97 iLBC/8000
>> >> > ?? a=rtpmap:98 telephone-event/8000
>> >> > ?? a=fmtp:98 0-15
>> >> > ?? a=ptime:20
>> >> > ?? a=maxptime:40
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > recv 570 bytes from udp/[74.63.41.218]:5060 at 21:15:59.468750:
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > ?? SIP/2.0 407 Proxy Authentication Required
>> >> > ?? Via: SIP/2.0/UDP
>> >> >
>> >> >
>> >> > 69.125.20.15:5080;branch=z9hG4bKNp4ZFeKSD43tS;received=69.125.20.15;rport=5080
>> >> > ?? From: "Extension 1001"
>> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>> >> > ?? To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843
>> >> > ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
>> >> > ?? CSeq: 7788615 INVITE
>> >> > ?? User-Agent: VoIPMS/SERAST
>> >> > ?? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> >> > ?? Supported: replaces
>> >> > ?? Proxy-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms",
>> >> > nonce="2d534dd6"
>> >> > ?? Content-Length: 0
>> >> >
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > ?? ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0
>> >> > ?? Via: SIP/2.0/UDP
>> >> > 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS
>> >> > ?? Max-Forwards: 69
>> >> > ?? From: "Extension 1001"
>> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>> >> > ?? To: <sip:1MYPHONE at newyork.voip.ms>;tag=as7e7ea843
>> >> > ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
>> >> > ?? CSeq: 7788615 ACK
>> >> > ?? Content-Length: 0
>> >> >
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > send 1344 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > ?? INVITE sip:1MYPHONE at newyork.voip.ms SIP/2.0
>> >> > ?? Via: SIP/2.0/UDP
>> >> > 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN
>> >> > ?? Max-Forwards: 69
>> >> > ?? From: "Extension 1001"
>> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>> >> > ?? To: <sip:1MYPHONE at newyork.voip.ms>
>> >> > ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
>> >> > ?? CSeq: 7788616 INVITE
>> >> > ?? Contact:
>> >> > <sip:gw+voip.ms at 69.125.20.15:5080;transport=udp;gw=voip.ms>
>> >> > ?? Expires: 300
>> >> > ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
>> >> > 18-04-05
>> >> > -0600
>> >> > ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> >> > REGISTER, REFER, NOTIFY
>> >> > ?? Supported: timer, precondition, path, replaces
>> >> > ?? Allow-Events: talk, hold, refer
>> >> > ?? Proxy-Authorization: Digest username="121628",
>> >> > realm="newyork.voip.ms",
>> >> > nonce="2d534dd6", algorithm=MD5, uri="sip:1MYPHONE at newyork.voip.ms",
>> >> > response="16f3301efae13df926da7550f709d28a"
>> >> > ?? Content-Type: application/sdp
>> >> > ?? Content-Disposition: session
>> >> > ?? Content-Length: 280
>> >> > ?? X-FS-Support: update_display
>> >> > ?? Remote-Party-ID: "Extension 1001"
>> >> > <sip:1001 at 69.125.20.15>;party=calling;screen=yes;privacy=off
>> >> >
>> >> > ?? v=0
>> >> > ?? o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6
>> >> > ?? s=-
>> >> > ?? c=IN IP4 10.153.174.6
>> >> > ?? t=0 0
>> >> > ?? m=audio 49152 RTP/AVP 18 97 98
>> >> > ?? a=rtpmap:18 G729/8000
>> >> > ?? a=fmtp:18 annexb=no
>> >> > ?? a=rtpmap:97 iLBC/8000
>> >> > ?? a=rtpmap:98 telephone-event/8000
>> >> > ?? a=fmtp:98 0-15
>> >> > ?? a=ptime:20
>> >> > ?? a=maxptime:40
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > recv 488 bytes from udp/[74.63.41.218]:5060 at 21:15:59.484375:
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > ?? SIP/2.0 100 Trying
>> >> > ?? Via: SIP/2.0/UDP
>> >> >
>> >> >
>> >> > 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080
>> >> > ?? From: "Extension 1001"
>> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>> >> > ?? To: <sip:1MYPHONE at newyork.voip.ms>
>> >> > ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
>> >> > ?? CSeq: 7788616 INVITE
>> >> > ?? User-Agent: VoIPMS/SERAST
>> >> > ?? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> >> > ?? Supported: replaces
>> >> > ?? Contact: <sip:1MYPHONE at 74.63.41.218>
>> >> > ?? Content-Length: 0
>> >> >
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > recv 516 bytes from udp/[74.63.41.218]:5060 at 21:15:59.562500:
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > ?? SIP/2.0 503 Service Unavailable
>> >> > ?? Via: SIP/2.0/UDP
>> >> >
>> >> >
>> >> > 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080
>> >> > ?? From: "Extension 1001"
>> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>> >> > ?? To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9
>> >> > ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
>> >> > ?? CSeq: 7788616 INVITE
>> >> > ?? User-Agent: VoIPMS/SERAST
>> >> > ?? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> >> > ?? Supported: replaces
>> >> > ?? Contact: <sip:1MYPHONE at 74.63.41.218>
>> >> > ?? Content-Length: 0
>> >> >
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.562500:
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > ?? ACK sip:1MYPHONE at newyork.voip.ms SIP/2.0
>> >> > ?? Via: SIP/2.0/UDP
>> >> > 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN
>> >> > ?? Max-Forwards: 69
>> >> > ?? From: "Extension 1001"
>> >> > <sip:121628 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>> >> > ?? To: <sip:1MYPHONE at newyork.voip.ms>;tag=as632cb7d9
>> >> > ?? Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
>> >> > ?? CSeq: 7788616 ACK
>> >> > ?? Content-Length: 0
>> >> >
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > 2011-01-28 16:15:59.562500 [INFO] mod_dptools.c:2612 Originate
>> >> > Failed.
>> >> > Cause: NO_ANSWER
>> >> > 2011-01-28 16:15:59.562500 [NOTICE] sofia.c:5286 Hangup
>> >> > sofia/external/1MYPHONE [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
>> >> > 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:189
>> >> > sofia/internal/1001 at 192.168.1.100 has executed the last dialplan
>> >> > instruction, hanging up.
>> >> > 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:191
>> >> > Hangup
>> >> > sofia/internal/1001 at 192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING]
>> >> > 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1306
>> >> > Session 2
>> >> > (sofia/external/1MYPHONE) Ended
>> >> > 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1308 Close
>> >> > Channel
>> >> > sofia/external/1MYPHONE [CS_DESTROY]
>> >> > send 887 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.593750:
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > ?? SIP/2.0 503 Service Unavailable
>> >> > ?? Via: SIP/2.0/TCP
>> >> >
>> >> >
>> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180
>> >> > ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>> >> > ?? To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg
>> >> > ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
>> >> > ?? CSeq: 5244504 INVITE
>> >> > ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13
>> >> > 18-04-05
>> >> > -0600
>> >> > ?? Accept: application/sdp
>> >> > ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> >> > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> >> > ?? Supported: timer, precondition, path, replaces
>> >> > ?? Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>> >> > sla,
>> >> > include-session-description, presence.winfo, message-summary, refer
>> >> > ?? Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>> >> > 2011-01-28 16:15:59.593750 [NOTICE] switch_core_session.c:1306
>> >> > Session 1
>> >> > (sofia/internal/1001 at 192.168.1.100) Ended
>> >> > ?? Content-Length: 02011-01-28 16:15:59.593750 [NOTICE]
>> >> > switch_core_session.c:1308 Close Channel
>> >> > sofia/internal/1001 at 192.168.1.100
>> >> > [CS_DESTROY]
>> >> >
>> >> > ?? Remote-Party-ID: "MYPHONE"
>> >> > <sip:MYPHONE at 192.168.1.100>;party=calling;privacy=off;screen=no
>> >> >
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > recv 650 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.953125:
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> > ?? ACK sip:MYPHONE at 192.168.1.100;transport=TCP SIP/2.0
>> >> > ?? Via: SIP/2.0/TCP
>> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport
>> >> > ?? From: <sip:1001 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>> >> > ?? To: <sip:MYPHONE at 192.168.1.100>;tag=g0Qyy0ZQ96gmg
>> >> > ?? Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn
>> >> > ?? CSeq: 5244504 ACK
>> >> > ?? Supported: sec-agree
>> >> > ?? Max-Forwards: 70
>> >> > ?? Proxy-Authorization: Digest
>> >> >
>> >> >
>> >> > qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"
>> >> > ?? Content-Length: 0
>> >> >
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> >
>> >> > Thank you in advance.
>> >> >
>> >> > _______________________________________________ FreeSWITCH-users
>> >> > mailing
>> >> > list FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org
>> >> googletalk:conf+888 at conference.freeswitch.org
>> >> pstn:+19193869900
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Mon Jan 31 23:04:14 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 31 Jan 2011 14:04:14 -0600
Subject: [Freeswitch-users] Persistence BLFs (or: Dear PBX please
 remember the BLF state)
In-Reply-To: <AANLkTikDxzntMFRpFhfiqJ4pzvjsTQR4w_FfyLfT5Y7z@mail.gmail.com>
References: <AANLkTikuFEof7tYtjM1rTFYzV-8jt45ns5GNdcaj+Xz5@mail.gmail.com>
	<AANLkTik15djSwmE5j8r5Von57qPu9V2XDStbT4GfFgzW@mail.gmail.com>
	<AANLkTi=in3CLd-n0mT3SRKYA0kxLMGVWk9KeBx3_jhY6@mail.gmail.com>
	<AANLkTin3Bu+8EEr_v_GaRGNPzOjfcnfzsV2Av2cgnWgO@mail.gmail.com>
	<AANLkTi=OC_jx0+nzK405VZURLDJPD25R4ZjyJ3RXJ1nk@mail.gmail.com>
	<AANLkTi=Sm7CjoayiE8nHmL0Eehqknj1jdpbx0YBQJQGH@mail.gmail.com>
	<AANLkTinr0RdgBs2UiS3=b3i+k6uCwkMihDy_FbAOFPc_@mail.gmail.com>
	<AANLkTi=+aObW3kVBZq5ox4b4E1uFTSgtXUL72tHLOzPi@mail.gmail.com>
	<AANLkTikVDKS40dVsf8x=+5=qnW59LqJbSv1H=vtROFaM@mail.gmail.com>
	<AANLkTi=gJsVKCJifVaBL_H6ZnPHTD__aQPhPeVfdx-ju@mail.gmail.com>
	<AANLkTimeuCHf_7Y1TErPwrJbdGZ3DoQmeqQbJ1_3YNpT@mail.gmail.com>
	<AANLkTikDxzntMFRpFhfiqJ4pzvjsTQR4w_FfyLfT5Y7z@mail.gmail.com>
Message-ID: <AANLkTimVrhpfX+0D2mFPwGHK1EDsALWjFGp8hh7=ecqt@mail.gmail.com>

You are heading the wrong way down the ladder but its worth a try.
Grandstream is notorious for not following specs =D

I have engaged SNOM to again try to sync up for interop.  Stay tuned.
I would really prefer to have them working.

Meanwhile try firmware 7.1.35 on your snom as that was before things
started to break.

Also if you have polycom or linksys they have been known to work
better with blf.



On Mon, Jan 31, 2011 at 1:19 PM, Patrick Plattes
<patrick.plattes at niemann-frey.info> wrote:
> Hi :-)
>
> I think SNOM has a problem with the quality of the firmware. maybe
> they should glp it, so we can fix it ;-)
>
> Tomorrow I will test it with an Grandstream GPX-2000 maybe it works
> with this device. If it does I will also ask SNOM for a fix.
>
> Thanks,
> ?Patrick
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From jonas.gauffin at gmail.com  Mon Jan 31 23:08:05 2011
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Mon, 31 Jan 2011 21:08:05 +0100
Subject: [Freeswitch-users] Outbound faxes fail
In-Reply-To: <AE397829-B4F5-411B-90FB-06B8E387F098@ipeva.fr>
References: <AANLkTin7g6g9CwvvLhq_smi4BZvUEMy9OB-k34-g0qfJ@mail.gmail.com>
	<13F4AC41-3E40-43A3-B155-2A6FE638B432@ipeva.fr>
	<AANLkTimiYu1d8q5=+-o5zh71Fw9b=90DbjCpR_phPTVG@mail.gmail.com>
	<AE397829-B4F5-411B-90FB-06B8E387F098@ipeva.fr>
Message-ID: <AANLkTikk-h7UFpBd-aZdAvMpSFFriPDANW_iTqaU4gRK@mail.gmail.com>

How do I reduce the transmit speed? Can't find any suitable option in the
wiki?

I tried to set "disable-v17" to true in fax.config.xml. 14400 is still used
as speed.

On Mon, Jan 31, 2011 at 12:25 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Well you need to make sure your outbound gateway supports T38.
> You could also try to reduce your fax transmit speed to 9600.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 31/01/2011 ? 12:15, Jonas Gauffin a ?crit :
>
> I got a fax tone if I called the number manually.
>
> I've added <param name="enable-t38" value="true" /> to my fax.conf.xml.
> What else do I need to do?
>
> I'm doing this for outbound faxes:
>
> 1. Invoke an originate though eventsocket bgapi
> 2. The originate runs a mod_managed application
> 3. The managed application invokes session.execute with txfax(xxxxxx)
>
>
> On Mon, Jan 31, 2011 at 11:46 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>
>> it seems you are using Fax over G711.
>> Can you switch to T.38 ?
>> What happens if you make a voice call to the same number ?
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 31/01/2011 ? 11:34, Jonas Gauffin a ?crit :
>>
>> Hello,
>>
>> A lot of my outbound faxes fail with an error message saying "Timed out
>> waiting for initial communication".
>> Can someone help me determine the cause?
>>
>> I got a log here: http://pastebin.freeswitch.org/15187
>>
>> //Jonas
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jonas.gauffin at gmail.com  Mon Jan 31 23:55:50 2011
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Mon, 31 Jan 2011 21:55:50 +0100
Subject: [Freeswitch-users] Outbound faxes fail
In-Reply-To: <AANLkTikk-h7UFpBd-aZdAvMpSFFriPDANW_iTqaU4gRK@mail.gmail.com>
References: <AANLkTin7g6g9CwvvLhq_smi4BZvUEMy9OB-k34-g0qfJ@mail.gmail.com>
	<13F4AC41-3E40-43A3-B155-2A6FE638B432@ipeva.fr>
	<AANLkTimiYu1d8q5=+-o5zh71Fw9b=90DbjCpR_phPTVG@mail.gmail.com>
	<AE397829-B4F5-411B-90FB-06B8E387F098@ipeva.fr>
	<AANLkTikk-h7UFpBd-aZdAvMpSFFriPDANW_iTqaU4gRK@mail.gmail.com>
Message-ID: <AANLkTim12xoc6M+ktBS_Uf5HFd25Q-H3P+sKdN4dpJR0@mail.gmail.com>

Here is the reinvite when I've enabled t.38:

http://pastebin.freeswitch.org/15194

<http://pastebin.freeswitch.org/15194>Have I configured something
incorrectly or why do my gw provider respond with 488?

On Mon, Jan 31, 2011 at 9:08 PM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:

> How do I reduce the transmit speed? Can't find any suitable option in the
> wiki?
>
> I tried to set "disable-v17" to true in fax.config.xml. 14400 is still used
> as speed.
>
> On Mon, Jan 31, 2011 at 12:25 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>
>> Well you need to make sure your outbound gateway supports T38.
>> You could also try to reduce your fax transmit speed to 9600.
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 31/01/2011 ? 12:15, Jonas Gauffin a ?crit :
>>
>> I got a fax tone if I called the number manually.
>>
>> I've added <param name="enable-t38" value="true" /> to my fax.conf.xml.
>> What else do I need to do?
>>
>> I'm doing this for outbound faxes:
>>
>> 1. Invoke an originate though eventsocket bgapi
>> 2. The originate runs a mod_managed application
>> 3. The managed application invokes session.execute with txfax(xxxxxx)
>>
>>
>> On Mon, Jan 31, 2011 at 11:46 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>>
>>> it seems you are using Fax over G711.
>>> Can you switch to T.38 ?
>>> What happens if you make a voice call to the same number ?
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 31/01/2011 ? 11:34, Jonas Gauffin a ?crit :
>>>
>>> Hello,
>>>
>>> A lot of my outbound faxes fail with an error message saying "Timed out
>>> waiting for initial communication".
>>> Can someone help me determine the cause?
>>>
>>> I got a log here: http://pastebin.freeswitch.org/15187
>>>
>>> //Jonas
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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