date:20081030

It just hit me, this is 2005, not 2008, thats why.  Can we take a  
poll, who is still using 2005 and why?


Mike

On Oct 30, 2008, at 5:24 AM, Jonas Gauffin wrote:


Hi Mike,

This is a problem for me too. I did a fresh checkout yesterday.

Regards,
  Jonas

On Tue, Oct 28, 2008 at 10:06 AM, Tamas Cseke [EMAIL PROTECTED] 
 wrote:

Hello,

I still have problem after a fresh checkout.
I tried with MSVC++ 2008 express and I got the same errors too.

Tamas

Michael Jerris írta:
 This should work on a fresh checkout.

 Mike

 On Oct 27, 2008, at 10:23 AM, Tamas Cseke wrote:


 Hello,

 I have a problem with windows build:

 Microsoft Visual C++ 2005
 Revision: 10158

   Creating library Debug/FreeSwitchCore.lib and object
 Debug/FreeSwitchCore.exp
 switch_apr.obj : error LNK2019: unresolved external symbol
 [EMAIL PROTECTED] referenced in function [EMAIL PROTECTED]
 C:\freeswitch\freeswitch-trunk\Debug/FreeSwitch.dll : fatal error
 LNK1120: 1 unresolved externals
 Build log was saved at
 file://c:\freeswitch\freeswitch-trunk\w32\Library\Debug\BuildLog
 FreeSwitchCoreLib.htm
 FreeSwitchCoreLib - 2 error(s), 0 warning(s)
 == Build: 0 succeeded, 1 failed, 0 up-to-date, 0 skipped
 ==

 Thank you,
 Tamas

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Re: [Freeswitch-users] mod_cdr revival (or new module maybe)

2008-10-30 Thread [EMAIL PROTECTED]

I'll try some tests with various combinations first and then decide 
what's best.


I have to say that I was astonished to find out that freeswitch had so 
many event handlers from the very beginning.
However it would be great if freeswitch had the options for extra 
functionality (auto log rotation, db cdrs etc)
so that it meets the peculiarities of every different project. That 
would make a big difference compared to
other softswitch solutions where the lack of such features prohibits 
people from using them (especially in
the carrier grade level). User feedback (and wish lists) is the key for 
the success of an open source (or not)

project...

Thank you all for your replies. You' ve been very helpful.

David Knell wrote:
The sleep's done each time the directory's empty, not each time a 
file's written.  File open and close
are trivial (it's probably still cached), and the contents are going 
to have to be parsed wherever you

process it.

We've used exactly this to process deliver CDRs on boxes handling in 
excess of 500K mins/day
without issue.  And, looking at one now, the CDR processor's used 
about 4% of the CPU time
of FreeSWITCH, and about half that of the MySQL database which it 
writes the records to, also
on the local machine, from which they're simply copied to the main CDR 
processor.


It performance simply isn't worth worrying about.

--Dave

sleep for a couple of seconds

But then you could only insert 1800 cdrs per hour...
If I was to insert 36000 cdrs per hour this means that I have to
open
parse
close
10 files per second. Imagine the I/O penalty just for opening - closing the 
file.
(the persing is the same for both situations)

  



David Knell wrote:

[EMAIL PROTECTED] wrote:
  
Yes, the xml files give you tons of info... but isn't it a little 
insufficient - performance wise -
to open and close so many files in such a little time. In a PBX 
environment that wouldn't be an
issue but if we get to the small-voip-carrier level (some thousand cdrs 
per hour)

that could slow things down considerably, wouldn't it?
  

Not that you'd notice.  We run XML CDR to database scripting on each box 
that we use
for switching, and it's a pretty trivial task compared with switching 
all that media.  Doing it

this way is:-
(a) distributed - one process per box scales nicely;
(b) robust - script down, DB down, no problem: files just queue up;
(c) simple - the script logic is trivial:
- while 1
  - for each file in the XML CDR directory
- open it
- parse it (XML::Simple for us)
- insert it in to the DB
- delete it
  - sleep for a couple of seconds
Two error cases: can't parse or can't find data which should be there: 
move the file in to
another directory to be examined by real eyes; DB insert fails: break 
out of inner loop and

it'll be retried after a short pause.

--Dave

  




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--
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T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
http://www.3c.co.uk 



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Re: [Freeswitch-users] apr_md5 windows build problem

2008-10-30 Thread Tamas Cseke

Hi,

The answer is pretty simply for me: I have only 2008 express.
But  I tried 2008 express too as I said earlier and got the same.
Maybe it is not an issue with 2008 professional.

Best regards,
Tamas

Michael Jerris írta:
 It just hit me, this is 2005, not 2008, thats why.  Can we take a 
 poll, who is still using 2005 and why?

 Mike

 On Oct 30, 2008, at 5:24 AM, Jonas Gauffin wrote:

 Hi Mike,

 This is a problem for me too. I did a fresh checkout yesterday.

 Regards,
   Jonas

 On Tue, Oct 28, 2008 at 10:06 AM, Tamas Cseke 
 [EMAIL PROTECTED] wrote:
 Hello,

 I still have problem after a fresh checkout.
 I tried with MSVC++ 2008 express and I got the same errors too.

 Tamas

 Michael Jerris írta:
  This should work on a fresh checkout.
 
  Mike
 
  On Oct 27, 2008, at 10:23 AM, Tamas Cseke wrote:
 
 
  Hello,
 
  I have a problem with windows build:
 
  Microsoft Visual C++ 2005
  Revision: 10158
 
Creating library Debug/FreeSwitchCore.lib and object
  Debug/FreeSwitchCore.exp
  switch_apr.obj : error LNK2019: unresolved external symbol
  [EMAIL PROTECTED] referenced in function [EMAIL PROTECTED]
  C:\freeswitch\freeswitch-trunk\Debug/FreeSwitch.dll : fatal error
  LNK1120: 1 unresolved externals
  Build log was saved at
  file://c:\freeswitch\freeswitch-trunk\w32\Library\Debug\BuildLog
  FreeSwitchCoreLib.htm
  FreeSwitchCoreLib - 2 error(s), 0 warning(s)
  == Build: 0 succeeded, 1 failed, 0 up-to-date, 0 skipped
  ==
 
  Thank you,
  Tamas
 
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Re: [Freeswitch-users] mod_cdr revival (or new module maybe)

Actually,

The goal is to not limit the functionality by over thinking how things will
be used but to provide building blocks to make much
more possible. Our analogy for this is that if you take the common lego
reference, If you have something cool built out of legos but they are super
glued together it limits what you *could* have done had then not been.

The system that David described is indeed ideal.  I have mentioned it more
than once and its no coincidence that FreeSWITCH plays into that model to a
tee.  I do not mention it very often because I think understanding that
concept alone is valuable advice that I'd prefer not to waste on deaf ears
so I recommend you heed his suggestion.

Logging direct to DB is not a sin or anything but it's just not very
reliable.  We have so many ways to process call records as it is.
you can use the cdr_csv if you want legacy support from asterisk that you
can eventually tweak with templatable output, you can even make a template
that generates the log as ready to roll sql statements you can just cat into
mysql.

We have event_socket + channel_hangup event which has all the same info that
the cdr_csv uses to generate the file.
We have the XML cdr that is a 3 dimensional view of the call with an account
of nearly everything that happend including
each application executed, branching records when the call is transferred
complete with a timestamp for each occurance.

--However it would be great if freeswitch had the options for extra
functionality (auto log rotation, db cdrs etc)
so that it meets the peculiarities of every different project.

A single cron entry that does kill -HUP `cat freeswitch.pid` and you have
automatic rotation in unix.
In windows there are several cron equivs and instead of kill you can use an
xml-rpc script to call the FSAPI call fsctl send_sighup
also you can schedule it to happen automaticly with the FSAPI

sched_api @300 mygroup fsctl send_sighup

A script connected to event socket subscribed to channel_hangup can act as
your DB cdr engine and has the bonus that it could
connect to N boxes at once as well as be able to tell if the system was in
trouble by listening for system heartbeat events and or socket disconnect.

Moral of the story, there is more already there than first appears.  I've
always hated the M$ paper clip so I strive not to bring him back.
It looks like you are trying to run a VoIP termination company! Shall I
assume how you want your call details?



On Thu, Oct 30, 2008 at 7:43 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:

  I'll try some tests with various combinations first and then decide what's
 best.

 I have to say that I was astonished to find out that freeswitch had so many
 event handlers from the very beginning.
 However it would be great if freeswitch had the options for extra
 functionality (auto log rotation, db cdrs etc)
 so that it meets the peculiarities of every different project. That would
 make a big difference compared to
 other softswitch solutions where the lack of such features prohibits people
 from using them (especially in
 the carrier grade level). User feedback (and wish lists) is the key for the
 success of an open source (or not)
 project...

 Thank you all for your replies. You' ve been very helpful.


 David Knell wrote:

 The sleep's done each time the directory's empty, not each time a file's
 written.  File open and close
 are trivial (it's probably still cached), and the contents are going to
 have to be parsed wherever you
 process it.

 We've used exactly this to process deliver CDRs on boxes handling in excess
 of 500K mins/day
 without issue.  And, looking at one now, the CDR processor's used about 4%
 of the CPU time
 of FreeSWITCH, and about half that of the MySQL database which it writes
 the records to, also
 on the local machine, from which they're simply copied to the main CDR
 processor.

 It performance simply isn't worth worrying about.

 --Dave

 sleep for a couple of seconds

 But then you could only insert 1800 cdrs per hour...
 If I was to insert 36000 cdrs per hour this means that I have to
 open
 parse
 close
 10 files per second. Imagine the I/O penalty just for opening - closing the 
 file.
 (the persing is the same for both situations)





 David Knell wrote:

 [EMAIL PROTECTED] wrote:


  Yes, the xml files give you tons of info... but isn't it a little
 insufficient - performance wise -
 to open and close so many files in such a little time. In a PBX
 environment that wouldn't be an
 issue but if we get to the small-voip-carrier level (some thousand cdrs
 per hour)
 that could slow things down considerably, wouldn't it?



  Not that you'd notice.  We run XML CDR to database scripting on each box
 that we use
 for switching, and it's a pretty trivial task compared with switching
 all that media.  Doing it
 this way is:-
 (a) distributed - one process per box scales nicely;
 (b) robust - script down, DB down, no problem: files just queue up;
 (c) simple - the

Re: [Freeswitch-users] apr_md5 windows build problem

did you try 2008 with the freeswitch.sln file?


On Oct 30, 2008, at 9:10 AM, Tamas Cseke wrote:

 Hi,

 The answer is pretty simply for me: I have only 2008 express.
 But  I tried 2008 express too as I said earlier and got the same.
 Maybe it is not an issue with 2008 professional.

 Best regards,
 Tamas


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Re: [Freeswitch-users] apr_md5 windows build problem

2008-10-30 Thread Tamas Cseke

no, freeswitch.2008.sln or whatever is it
Michael Jerris írta:
 did you try 2008 with the freeswitch.sln file?


 On Oct 30, 2008, at 9:10 AM, Tamas Cseke wrote:

   
 Hi,

 The answer is pretty simply for me: I have only 2008 express.
 But  I tried 2008 express too as I said earlier and got the same.
 Maybe it is not an issue with 2008 professional.

 Best regards,
 Tamas

 

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Re: [Freeswitch-users] Freeswitch wedges in voicemail?

2008-10-30 Thread Marc Lewis

Here is the gdb output from the commands. Its actually wedged up twice
since I first sent my email.

Anthony Minessale wrote:

next time it happens
go into another shell

and enter

gcore -o dump `cat /usr/local/freeswitch/log/freeswitch.pid`

then

gdb /usr/local/bin/freeswitch/bin/freeswitch dump.*

then enter these 3 commands:

set pagination off
thread apply all bt
thread apply all pt full

and send me the entire output

also are you updating with make current to avoid build skew?

On Mon, Oct 27, 2008 at 12:32 PM, Marc Lewis [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:

I have a strange problem that I've been dealing with for the last week
or two. Once or twice a day, Freeswitch will wedge, in what appears
to be voicemail. The switch is sort of still responding, meaning
I can
issue various commands at the console, but no new sofia channels
will be
handled, registrations start failing on the internal profile and
no new
calls can come in from the external profile.

When it happens, existing channels stay up, but once they are
completed,
no new channels can be created. There is always one channel that
remains that looks something like (DID obfuscated):

api show channels

Content-Type: api/response
Content-Length: 394

uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate
ea758bcf-25f5-4cde-9f6c-5e3f1226130f,2008-10-27
10:02:27,1225126947,sofia/external/[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED],CS_EXECUTE,253941,253941,199.245.214.150
http://199.245.214.150,328,voicemail,default
pgm.cisvp.com http://pgm.cisvp.com 246,XML,pgm,PCMU,8000,PCMU,8000

1 total.

I've been updating every weekend with the latest code, this is
currently
trunk rev 10146.

Anyone have any thoughts on this? Its driving me crazy.

- Marc

--
Marc Lewis
Avvatel Corporation

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--
Anthony Minessale II

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

AIM: anthm
MSN:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]

IRC: irc.freenode.net http://irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
iax:[EMAIL PROTECTED]/888
http://iax:[EMAIL PROTECTED]/888
googletalk:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]

pstn:213-799-1400

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[EMAIL PROTECTED] ~]# gdb /opt/freeswitch/bin/freeswitch dump.1279
GNU gdb Red Hat Linux (6.5-37.el5_2.2rh)
Copyright (C) 2006 Free Software Foundation, Inc.
GDB is free software, covered by the GNU General Public License, and you are
welcome to change it and/or distribute copies of it under certain conditions.
Type show copying to see the conditions.
There is absolutely no warranty for GDB. Type show warranty for details.
This GDB was configured as i386-redhat-linux-gnu...Using host libthread_db
library /lib/i686/nosegneg/libthread_db.so.1.

warning: Can't read pathname for load map: Input/output error.
Reading symbols from /lib/i686/nosegneg/libm.so.6...done.
Loaded symbols for /lib/i686/nosegneg/libm.so.6
Reading symbols from /opt/freeswitch/lib/libfreeswitch.so.1...Reading symbols
from /usr/lib/debug/opt/freeswitch/lib/libfreeswitch.so.1.0.0.debug...done.
done.
Loaded symbols for /opt/freeswitch/lib/libfreeswitch.so.1
Reading symbols from /lib/libuuid.so.1...done.
Loaded symbols for /lib/libuuid.so.1
Reading symbols from /lib/i686/nosegneg/librt.so.1...done.
Loaded symbols for /lib/i686/nosegneg/librt.so.1
Reading symbols from /lib/libdl.so.2...done.
Loaded symbols for /lib/libdl.so.2
Reading symbols from /lib/libcrypt.so.1...done.
Loaded symbols for /lib/libcrypt.so.1
Reading symbols from /lib/i686/nosegneg/libpthread.so.0...done.
Loaded symbols for /lib/i686/nosegneg/libpthread.so.0
Reading symbols from /usr/lib/libncurses.so.5...done.
Loaded symbols for /usr/lib/libncurses.so.5
Reading symbols from /lib/i686/nosegneg/libc.so.6...done.
Loaded symbols for /lib/i686/nosegneg/libc.so.6
Reading symbols from /lib/ld-linux.so.2...done.
Loaded symbols for /lib/ld-linux.so.2
Reading symbols from

Re: [Freeswitch-users] apr_md5 windows build problem

2008-10-30 Thread Jonas Gauffin

Ok, the 2008 version worked.

But in the 2008 version, the setup project is not included (in the solution
file).
I added it and compiled it, and got these errros:

-- Starting pre-build validation for project 'FreeSwitchSetup' --
ERROR: Unable to find source file
'C:\projects\cpp\freeswitch\conf\sip_profiles\default.xml' for file
'default.xml', located in '[TARGETDIR]\conf\sip_profiles', the file may be
absent or locked.
ERROR: Unable to find source file
'C:\projects\cpp\freeswitch\conf\sip_profiles\nat\example.xml' for file
'example.xml', located in '[TARGETDIR]\conf\sip_profiles\nat', the file may
be absent or locked.
ERROR: Unable to find source file
'C:\projects\cpp\freeswitch\conf\dialplan\extensions\sipphone.com.noload'
for file 'sipphone.com.noload', located in
'[TARGETDIR]\conf\dialplan\extensions', the file may be absent or locked.
ERROR: Unable to find source file
'C:\projects\cpp\freeswitch\conf\autoload_configs\wanpipe.conf.xml' for file
'wanpipe.conf.xml', located in '[TARGETDIR]\conf\autoload_configs', the file
may be absent or locked.
ERROR: Unable to find source file
'C:\projects\cpp\freeswitch\conf\dialplan\extensions\sipbroker.com.noload'
for file 'sipbroker.com.noload', located in
'[TARGETDIR]\conf\dialplan\extensions', the file may be absent or locked.
ERROR: Unable to find source file
'C:\projects\cpp\freeswitch\conf\sip_profiles\outbound\example.xml' for file
'example.xml', located in '[TARGETDIR]\conf\sip_profiles\outbound', the file
may be absent or locked.
ERROR: Unable to find source file
'C:\projects\cpp\freeswitch\conf\dialplan\extensions\pulver.com.noload' for
file 'pulver.com.noload', located in '[TARGETDIR]\conf\dialplan\extensions',
the file may be absent or locked.
ERROR: Unable to find source file
'C:\projects\cpp\freeswitch\conf\dialplan\extensions\ideasip.com.noload' for
file 'ideasip.com.noload', located in
'[TARGETDIR]\conf\dialplan\extensions', the file may be absent or locked.
ERROR: Unable to find source file
'C:\projects\cpp\freeswitch\conf\sip_profiles\nat.xml' for file 'nat.xml',
located in '[TARGETDIR]\conf\sip_profiles', the file may be absent or
locked.
ERROR: Unable to find source file
'C:\projects\cpp\freeswitch\conf\sip_profiles\default\example.xml' for file
'example.xml', located in '[TARGETDIR]\conf\sip_profiles\default', the file
may be absent or locked.
ERROR: Unable to find source file
'C:\projects\cpp\freeswitch\conf\sip_profiles\outbound.xml' for file
'outbound.xml', located in '[TARGETDIR]\conf\sip_profiles', the file may be
absent or locked.
-- Pre-build validation for project 'FreeSwitchSetup' completed --
51-- Build started: Project: FreeSwitchSetup, Configuration: Release
with MySql --


On Thu, Oct 30, 2008 at 2:27 PM, Tamas Cseke [EMAIL PROTECTED]wrote:

 no, freeswitch.2008.sln or whatever is it
 Michael Jerris írta:
  did you try 2008 with the freeswitch.sln file?
 
 
  On Oct 30, 2008, at 9:10 AM, Tamas Cseke wrote:
 
 
  Hi,
 
  The answer is pretty simply for me: I have only 2008 express.
  But  I tried 2008 express too as I said earlier and got the same.
  Maybe it is not an issue with 2008 professional.
 
  Best regards,
  Tamas
 
 
 
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Re: [Freeswitch-users] apr_md5 windows build problem

The built in setup has turned out to be too limiting.  In tree there  
is an advanced installer config file at :

http://svn.freeswitch.org/svn/freeswitch/trunk/w32/Setup/freeswitch.aip

That you can use to build a msi.

I am proposing to remove the 2005 support from tree completely. I need  
to confirm that we currently build with 2008 express edition, can  
anyone confirm that?  Are there any good reasons to continue to  
support 2005.  We have been adding stuff to 2008 and not the 2005  
build for quite some time and no one has been complaining or  
contributing projects for the 2005 so it would appear there is not  
enough interest by anyone to do any work to make sure it is up to date.

Mike



On Oct 30, 2008, at 10:51 AM, Jonas Gauffin wrote:

 Ok, the 2008 version worked.

 But in the 2008 version, the setup project is not included (in the  
 solution file).
 I added it and compiled it, and got these errros:

 -- Starting pre-build validation for project 'FreeSwitchSetup'  
 --
 ERROR: Unable to find source file 'C:\projects\cpp\freeswitch\conf 
 \sip_profiles\default.xml' for file 'default.xml', located in  
 '[TARGETDIR]\conf\sip_profiles', the file may be absent or locked.
 ERROR: Unable to find source file 'C:\projects\cpp\freeswitch\conf 
 \sip_profiles\nat\example.xml' for file 'example.xml', located in  
 '[TARGETDIR]\conf\sip_profiles\nat', the file may be absent or locked.
 ERROR: Unable to find source file 'C:\projects\cpp\freeswitch\conf 
 \dialplan\extensions\sipphone.com.noload' for file  
 'sipphone.com.noload', located in '[TARGETDIR]\conf\dialplan 
 \extensions', the file may be absent or locked.
 ERROR: Unable to find source file 'C:\projects\cpp\freeswitch\conf 
 \autoload_configs\wanpipe.conf.xml' for file 'wanpipe.conf.xml',  
 located in '[TARGETDIR]\conf\autoload_configs', the file may be  
 absent or locked.
 ERROR: Unable to find source file 'C:\projects\cpp\freeswitch\conf 
 \dialplan\extensions\sipbroker.com.noload' for file  
 'sipbroker.com.noload', located in '[TARGETDIR]\conf\dialplan 
 \extensions', the file may be absent or locked.
 ERROR: Unable to find source file 'C:\projects\cpp\freeswitch\conf 
 \sip_profiles\outbound\example.xml' for file 'example.xml', located  
 in '[TARGETDIR]\conf\sip_profiles\outbound', the file may be absent  
 or locked.
 ERROR: Unable to find source file 'C:\projects\cpp\freeswitch\conf 
 \dialplan\extensions\pulver.com.noload' for file  
 'pulver.com.noload', located in '[TARGETDIR]\conf\dialplan 
 \extensions', the file may be absent or locked.
 ERROR: Unable to find source file 'C:\projects\cpp\freeswitch\conf 
 \dialplan\extensions\ideasip.com.noload' for file  
 'ideasip.com.noload', located in '[TARGETDIR]\conf\dialplan 
 \extensions', the file may be absent or locked.
 ERROR: Unable to find source file 'C:\projects\cpp\freeswitch\conf 
 \sip_profiles\nat.xml' for file 'nat.xml', located in '[TARGETDIR] 
 \conf\sip_profiles', the file may be absent or locked.
 ERROR: Unable to find source file 'C:\projects\cpp\freeswitch\conf 
 \sip_profiles\default\example.xml' for file 'example.xml', located  
 in '[TARGETDIR]\conf\sip_profiles\default', the file may be absent  
 or locked.
 ERROR: Unable to find source file 'C:\projects\cpp\freeswitch\conf 
 \sip_profiles\outbound.xml' for file 'outbound.xml', located in  
 '[TARGETDIR]\conf\sip_profiles', the file may be absent or locked.
 -- Pre-build validation for project 'FreeSwitchSetup' completed  
 --
 51-- Build started: Project: FreeSwitchSetup, Configuration:  
 Release with MySql --


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Re: [Freeswitch-users] Freeswitch wedges in voicemail?

2008-10-30 Thread Marc Lewis

Got my second wedge of the day. Last time the channel state showed in
my lua autoattendant script. This time the channel state showed in
voicemail -- which is where it usually shows when its wedged.

Here is the gdb output from this one. Hopefully this will help track
this down. I'm anxious to get this particular problem behind me.

Thanks.

- Marc

Anthony Minessale wrote:

next time it happens
go into another shell

and enter

gcore -o dump `cat /usr/local/freeswitch/log/freeswitch.pid`

then

gdb /usr/local/bin/freeswitch/bin/freeswitch dump.*

then enter these 3 commands:

set pagination off
thread apply all bt
thread apply all pt full

and send me the entire output

also are you updating with make current to avoid build skew?

On Mon, Oct 27, 2008 at 12:32 PM, Marc Lewis [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:

When it happens, existing channels stay up, but once they are
completed,
no new channels can be created. There is always one channel that
remains that looks something like (DID obfuscated):

api show channels

Content-Type: api/response
Content-Length: 394

1 total.

I've been updating every weekend with the latest code, this is
currently
trunk rev 10146.

Anyone have any thoughts on this? Its driving me crazy.

- Marc

--
Marc Lewis
Avvatel Corporation

--
Anthony Minessale II

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

AIM: anthm
MSN:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]

IRC: irc.freenode.net http://irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
iax:[EMAIL PROTECTED]/888
http://iax:[EMAIL PROTECTED]/888
googletalk:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]

pstn:213-799-1400

--
Marc Lewis
Avvatel Corporation

[EMAIL PROTECTED] ~]# gdb /opt/freeswitch/bin/freeswitch dump.23597
GNU gdb Red Hat Linux (6.5-37.el5_2.2rh)
Copyright (C) 2006 Free Software Foundation, Inc.
GDB is free software, covered by the GNU General Public License, and you are
welcome to change it and/or distribute copies of it under certain conditions.
Type show copying to see the conditions.
There is absolutely no warranty for GDB. Type show warranty for details.
This GDB was configured as i386-redhat-linux-gnu...Using host libthread_db
library /lib/i686/nosegneg/libthread_db.so.1.

Re: [Freeswitch-users] Freeswitch wedges in voicemail?

What svn revision was this?

Mike

On Oct 30, 2008, at 2:00 PM, Marc Lewis wrote:

 Got my second wedge of the day.  Last time the channel state showed  
 in my lua autoattendant script.  This time the channel state showed  
 in voicemail -- which is where it usually shows when its wedged.

 Here is the gdb output from this one.  Hopefully this will help  
 track this down.  I'm anxious to get this particular problem behind  
 me.

 Thanks.

  - Marc


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Re: [Freeswitch-users] Freeswitch wedges in voicemail?

2008-10-30 Thread Marc Lewis


Its rev 10146, but it also happened under 10067.

Its now happened to me 4 times today.  Twice in the last half hour.  I 
have dumps from those, too.


- Marc



Michael Jerris wrote:

What svn revision was this?

Mike

On Oct 30, 2008, at 2:00 PM, Marc Lewis wrote:

  
Got my second wedge of the day.  Last time the channel state showed  
in my lua autoattendant script.  This time the channel state showed  
in voicemail -- which is where it usually shows when its wedged.


Here is the gdb output from this one.  Hopefully this will help  
track this down.  I'm anxious to get this particular problem behind  
me.


Thanks.

 - Marc




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--
Marc Lewis
Avvatel Corporation

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Re: [Freeswitch-users] mod_cdr revival (or new module maybe)

2008-10-30 Thread Michael Collins

/me sends Anthony's post to the printer to be laminated and framed...
:-)

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Anthony Minessale
Sent: Thursday, October 30, 2008 6:10 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_cdr revival (or new module maybe)

 

Actually,

The goal is to not limit the functionality by over thinking how things
will be used but to provide building blocks to make much
more possible. Our analogy for this is that if you take the common lego
reference, If you have something cool built out of legos but they are
super glued together it limits what you *could* have done had then not
been.

The system that David described is indeed ideal.  I have mentioned it
more than once and its no coincidence that FreeSWITCH plays into that
model to a tee.  I do not mention it very often because I think
understanding that concept alone is valuable advice that I'd prefer not
to waste on deaf ears so I recommend you heed his suggestion.   

Logging direct to DB is not a sin or anything but it's just not very
reliable.  We have so many ways to process call records as it is.
you can use the cdr_csv if you want legacy support from asterisk that
you can eventually tweak with templatable output, you can even make a
template that generates the log as ready to roll sql statements you can
just cat into mysql.

We have event_socket + channel_hangup event which has all the same info
that the cdr_csv uses to generate the file.
We have the XML cdr that is a 3 dimensional view of the call with an
account of nearly everything that happend including
each application executed, branching records when the call is
transferred complete with a timestamp for each occurance.

--However it would be great if freeswitch had the options for extra
functionality (auto log rotation, db cdrs etc)
so that it meets the peculiarities of every different project.

A single cron entry that does kill -HUP `cat freeswitch.pid` and you
have automatic rotation in unix.
In windows there are several cron equivs and instead of kill you can use
an xml-rpc script to call the FSAPI call fsctl send_sighup
also you can schedule it to happen automaticly with the FSAPI

sched_api @300 mygroup fsctl send_sighup

A script connected to event socket subscribed to channel_hangup can act
as your DB cdr engine and has the bonus that it could
connect to N boxes at once as well as be able to tell if the system was
in trouble by listening for system heartbeat events and or socket
disconnect.

Moral of the story, there is more already there than first appears.
I've always hated the M$ paper clip so I strive not to bring him back.
It looks like you are trying to run a VoIP termination company! Shall I
assume how you want your call details?




On Thu, Oct 30, 2008 at 7:43 AM, [EMAIL PROTECTED] [EMAIL PROTECTED]
wrote:

I'll try some tests with various combinations first and then decide
what's best.

I have to say that I was astonished to find out that freeswitch had so
many event handlers from the very beginning.
However it would be great if freeswitch had the options for extra
functionality (auto log rotation, db cdrs etc)
so that it meets the peculiarities of every different project. That
would make a big difference compared to
other softswitch solutions where the lack of such features prohibits
people from using them (especially in
the carrier grade level). User feedback (and wish lists) is the key for
the success of an open source (or not) 
project...

Thank you all for your replies. You' ve been very helpful.



David Knell wrote: 

The sleep's done each time the directory's empty, not each time a file's
written.  File open and close
are trivial (it's probably still cached), and the contents are going to
have to be parsed wherever you 
process it.

We've used exactly this to process deliver CDRs on boxes handling in
excess of 500K mins/day
without issue.  And, looking at one now, the CDR processor's used about
4% of the CPU time
of FreeSWITCH, and about half that of the MySQL database which it writes
the records to, also
on the local machine, from which they're simply copied to the main CDR
processor.

It performance simply isn't worth worrying about.

--Dave



sleep for a couple of seconds
 
But then you could only insert 1800 cdrs per hour...
If I was to insert 36000 cdrs per hour this means that I have to
open
parse
close
10 files per second. Imagine the I/O penalty just for opening - closing
the file.
(the persing is the same for both situations)
 
  



David Knell wrote: 

[EMAIL PROTECTED] wrote:
  

Yes, the xml files give you tons of info... but isn't it a
little 
insufficient - performance wise -
to open and close so many files in such a little time. In a PBX 
environment that wouldn't be an
issue but if we get to the small-voip-carrier level (some
thousand cdrs 
per hour)
that could slow things down

Re: [Freeswitch-users] Clustering FreeSWITCH

2008-10-30 Thread Thomas Mangin

Hi,

Sorry for jumping in without reading the whole thread correctly but I  
will most likely not have the time to do so before Saturday but would  
still provide some feedback.

When using Openser as a customer phone proxy, it is my understanding  
that unless you use the PATH extension you will have problem with  
firewalls as the packet from Freeswitch will not match the opened IP  
address for the UDP port.
In the scenario where the phone send a REGISTER this way : Phone -  
Firewall (create an association) - OpenSer - Freeswitch.
If Freeswitch answers directly to the packet, it will not match the  
association on the firewall for the return packet (different IP) and  
the connection may/will be blocked, so the reply to the REGISTER must  
come back via OpenSER.
Same thing for the INVITE message, if FS tries to connect the firewall  
directly the packet will be blocked.
As well, the firewall mapping need to be kept open with some kind of  
SIP pinging using OPTIONS - for example - and going through OpenSER as  
well from the FS server

When quickly looking into it a few weeks ago - before I was side  
tracked, I found that the way OpenSER and FS way of using SIP PATH are  
not working together very well (I was planning a mail to the list with  
more info once I checked this more extensively).

I can see the point with trying to not use OpenSER and use DNS SRV but  
you may not always be able to do without.

Regards,

Thomas


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[Freeswitch-users] Status of 1.02?

2008-10-30 Thread Andy Spitzer

Woof!

Is there any updated projection of when 1.0.2 will be released?  Last I  
can find mention of it is over a month ago (http://freeswitch.org/node/143)

Thanks,

--Woof!

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Re: [Freeswitch-users] mod_cdr revival (or new module maybe)

2008-10-30 Thread [EMAIL PROTECTED]

Don't get me wrong. I am not trying to undermine the excellent work that 
has already been done on the event handling
modules of FS.

I myself am a big fun of modular constructs in order to achieve 
something complicated (e.g. the UNIX way).

But there are some situations that we will always need some extra 
features in order to accomplish our goal.

As I wrote before, my current system is using the cdr-push (from the 
gateway side) method to gather my cdrs.
Many billing systems and many terminating providers are satisfied with 
that model alone. I am not. In the end of each hour
I am gathering all the cdrs and checking them one by one against my 
database in order to verify that ALL of my cdrs were
handled by my radius servers.

In my future system (where a more batch-like mode is preferable) I am 
forced to use a cdr-pull method. My billing system will
be responisble for pulling the cdrs from the gateways and then process 
them. So if we have a look at the event handling weaponry of FS the
following modules cannot meet me needs :

mod_event_multicast
mod_event socket
mod_radius_cdr
mod_xmpp_event

because they all rely on a mechanism where the gateway is pushing my 
cdrs to my billing system (and you need a checking mechanism to verify
that all the cdrs were handled).

So now, lets have a look at the alternatives :

mod_xml_cdr : this was my first choice as it had all the info needed 
(even more). But the cdr per file approach proved out to be
inadequate in terms of performance. I wrote a few lines of code (in 
perl) that did a listing of the directory, parsed the cdrs using
XML::Simple (without even doing anything meaningful, e.g. checks, 
inserts to a database etc.) and I was not able to make my system
parse more than a 30 cdrs/sec (reference 1000 USD average system, not 
drawing any conclusions, don't get me wrong).
Also while the parser was running I had a high cpu utilization (as 
expected) that was competing with my FS service. Wouldn't that make
a bad impact of my freeswitch performance?

so I went to a different module

mod_cdr_csv : the text .csv format with the plethora of attributes was 
perfect for my application but it would be a daunting task
for me to make a script tha purges the Master csv while it was being 
used by FS. I googled the issue and found about the HUP method
and the rotate directive in the config. So far so good. But it lacked 
some functionality.

All I was trying to state in my previous messages was that it would be 
nice if the rotation was being initiated by FS, and maybe have different
behavior depending on another directive or something. Let me explain 
what I mean.

There are some people who don't care about what cdrs exist in a given 
rotated log file, as long as a) no cdrs get lost in the process of rotation
, b) no duplicates exist. I can see that the HUP method satisfied this need.

There is another group of people that DO care about what cdrs are in a 
rotated log file. E.g. : The file with the name 
Master.csv.2008-01-01-09-00-00
would only contain cdrs that were terminated from 2008-01-01 09:00:00 
to 2008-01-01 09:59:59 if an hourly setup was desirable or from
2008-01-01 09:00:00 to 2008-01-01 09:04:59 if a five-minutes setup 
was desirable. That need is not covered by the current HUP method
because some cdrs might escape from the next to the previous file due 
to the fact that there is a delay between cron executing the HUP and FS 
doing
the file log rotation.

In simple words : I am getting my job done with FS they way it already 
is (no question about that). BUT, I (or someone with different needs) 
could do
his/her job better if some minor features were present.

I am not saying that you should should embrace modules that meet the 
average needs yet sacrifice flexibility. I am merely suggesting 
extending the flexibility
of the already existing ones. Put some more lego tiles in your box set :)


Michael Collins wrote:

 /me sends Anthony’s post to the printer to be laminated and framed… J

 

 *From:* [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] *On Behalf Of 
 *Anthony Minessale
 *Sent:* Thursday, October 30, 2008 6:10 AM
 *To:* freeswitch-users@lists.freeswitch.org
 *Subject:* Re: [Freeswitch-users] mod_cdr revival (or new module maybe)

 Actually,

 The goal is to not limit the functionality by over thinking how things 
 will be used but to provide building blocks to make much
 more possible. Our analogy for this is that if you take the common 
 lego reference, If you have something cool built out of legos but they 
 are super glued together it limits what you *could* have done had then 
 not been.

 The system that David described is indeed ideal. I have mentioned it 
 more than once and its no coincidence that FreeSWITCH plays into that 
 model to a tee. I do not mention it very often because I think 
 understanding that concept alone is valuable advice that I'd prefer 
 not to

Re: [Freeswitch-users] Freeswitch as SBC ?

2008-10-30 Thread Kristian Kielhofner

On Thu, Oct 30, 2008 at 12:32 PM, Andreas Dimitriou
[EMAIL PROTECTED] wrote:
 Hello,



 Has anyone used Freeswitch as SBC ?



 BR

 Andreas D


Andreas,

  An SBC is a configured B2BUA (which FreeSWITCH is).  FreeSWITCH can
probably do what you are looking for.  What specific functionality do
you require?



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

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[Freeswitch-users] Limit number of outbound channels

Hi, Everybody,
i was wondering if there is an option or command that allows to limit 
the number of outbound channels per gateway under external profile.
Regards
Chav

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Re: [Freeswitch-users] Limit number of outbound channels

http://wiki.freeswitch.org/wiki/Mod_limit

That should get you on the right track.

/b

On Oct 30, 2008, at 3:56 PM, Chav Paskov wrote:

 Hi, Everybody,
 i was wondering if there is an option or command that allows to limit
 the number of outbound channels per gateway under external profile.
 Regards
 Chav


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Re: [Freeswitch-users] Limit number of outbound channels

Thanks a lot for lightning fast response.
Regards
Chav

Brian West wrote:
 http://wiki.freeswitch.org/wiki/Mod_limit

 That should get you on the right track.

 /b

 On Oct 30, 2008, at 3:56 PM, Chav Paskov wrote:

   
 Hi, Everybody,
 i was wondering if there is an option or command that allows to limit
 the number of outbound channels per gateway under external profile.
 Regards
 Chav
 


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Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication

just setup a gateway in fs that has reg=false and the proper credentials to
pass the challenge.


On Thu, Oct 30, 2008 at 2:49 PM, Wellie Chao [EMAIL PROTECTED] wrote:

 Hangups do not work for me under certain circumstances. Here is the
 background information:

 * Our carrier uses a Metaswitch server with Acme Packet in front as a
   proxy/SBC. Only the Acme Packet machine is publicly visible
   (64.115.128.6).

 * Our Freeswitch server is at 216.57.23.143.

 * For calls originating from Freeswitch and terminating on Metaswitch
   (216.57.23.143 - 64.115.128.6), Freeswitch authenticates with
   Metaswitch and everything works hunky-dorey. Either side can hang up and
   the other side will automatically hang up without requiring a manual
   hang-up.

 Now the problem:

 For calls originating from Metaswitch to Freeswitch (64.115.128.6 -
 216.57.23.143), Metaswitch does not authenticate with Freeswitch.
 Metaswitch also does not use the existing authenticated registration that
 our Freeswitch server initiates with Metaswitch upon startup of
 Freeswitch. Metaswitch just begins a new (unauthenticated) session and we
 have configured Freeswitch to allow any inbound calls from 64.115.128.6
 without requiring authentication.

 We receive inbound calls (Metaswitch to Freeswitch, 64.115.128.6 -
 216.57.23.143) just fine. The phone rings and we can have a normal
 conversation. If the caller (the endpoint attached to Metaswitch) hangs
 up, both sides hang up. If I hang up (remember, I'm at the endpoint
 attached to Freeswitch), the caller's line remains attached forever.

 I have recorded a packet trace. Look at freeswitch_2.cap in the ZIP file,
 and you want to graph the first call starting at 21.202 and ending 53.798.
 If you go to time 53.087, you can see that my Freeswitch server sends a
 BYE to Metaswitch. This is a result of me hanging up my phone. At time
 53.089, you see Metaswitch responding with 401 Unauthorized. Later at time
 53.777, you see a BYE from Metaswitch to Freeswitch, but you should ignore
 this because that was a result of the caller (the guy hooked up to
 Metaswitch) manually hanging up. If he had not hung up his phone, the BYE
 from Metaswitch to Freeswitch would not have been issued and his phone
 would just stay on the line forever. Also, when I hang up my phone, I see
 the following at the Freeswitch console:

 2008-10-29 23:03:28 [ERR] sofia_reg.c:1089
 sofia_reg_handle_sip_r_challenge() No Matching gateway found

 I presume that Freeswitch emits this error because it got the 401
 Unauthorized from Metaswitch.

 I also asked our carrier for a packet trace of a successful hangup on the
 Aastra platform (the engineer at the carrier says it is an Asterisk
 derivative -- I'm not sure about that). Look at
 Aastra_authentication_test.cap in the ZIP file. Graph the first call
 starting at 43.633 and ending 93.156. If you go to 93.118, you'll see that
 the Aastra server sends a BYE. Just like our Freeswitch scenario,
 Metaswitch sends back a 401 Unauthorized, but in response to the 401
 Unauthorized, Aastra then sends back another BYE with the difference that
 the second BYE is authenticated. Metaswitch gets the second BYE and
 responds with 200 OK.

 I am pretty sure that if Freeswitch were to send back a second BYE (but
 with authentication), it would work fine. Now my question is how can I do
 this? I am not sure if this divergence of behavior is caused by: (a) my
 own error in configuring Freeswitch, (b) Metaswitch lacking standard SIP
 support (maybe it's not supposed to send the 401 Unauthorized), or (c)
 Freeswitch lacking standard SIP support (maybe it's supposed to send back
 a second BYE with authentication automatically). I don't know the SIP
 standards (or Freeswitch) well enough to know whether this problem is
 caused by me or by a deficiency in one of the two products (Metaswitch or
 Freeswitch).

 Can you provide some pointers?

 The ZIP file with the packet traces can be downloaded here:
 http://216.57.23.143/hangup_problem.zip

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[Freeswitch-users] Call Forwarding from phone

Hello,

I have a configuration (dialplan?) question.

One of my users used the call forward button on his sip phone.  The  
phone is directly registered to FS.

The forwarded calls fail.  It looks like the phone is sending a  
redirect message to FS, but then the call is then not getting routed  
correctly through FS.  So, the call is dropped.

Is there something specific I need to configure to account for  
possible call forwarding from phones??

Thanks,

-Noah

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Re: [Freeswitch-users] Call Forwarding from phone

Turn off the aggressive nat detection on the profile... if its on that  
would explain what is going on.

/b

On Oct 30, 2008, at 4:24 PM, Noah Silverman wrote:

 Hello,

 I have a configuration (dialplan?) question.

 One of my users used the call forward button on his sip phone.  The
 phone is directly registered to FS.

 The forwarded calls fail.  It looks like the phone is sending a
 redirect message to FS, but then the call is then not getting routed
 correctly through FS.  So, the call is dropped.

 Is there something specific I need to configure to account for
 possible call forwarding from phones??

 Thanks,

 -Noah


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Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication


Here is what I have:

include
  gateway name=broadview
param name=username value=MY_USERNAME/
param name=password value=MY_PASSWORD/
param name=realm value=64.115.128.6/
param name=proxy value=64.115.128.6/
param name=register value=false/
  /gateway
/include

Whether register is true or false doesn't seem to make a difference 
(except that Freeswitch then comes up with broadview in NOREG state). On 
calls from Metaswitch to Freeswitch, it's the same problem, and I get the 
same message in the Freeswitch logs:


2008-10-30 17:39:04 [ERR] sofia_reg.c:1089 
sofia_reg_handle_sip_r_challenge() No Matching gateway found


I presume this is the same thing with the 401 Unauthorized packet being 
sent by Metaswitch in response to Freeswitch's BYE message. Note that the 
call itself goes just fine. I pick up, both sides can hear each other. 
Just the hangup gets messed up and for some reason Metaswitch expects an 
authenticated BYE message even though the connection was not authenticated 
in the beginning when Metaswitch initiated it. The packet trace shows this 
and it's very odd.


Is that what you meant when you said set up a gateway in Freeswitch that 
has reg=false and the proper credentials?


On Thu, 30 Oct 2008, Anthony Minessale wrote:


Date: Thu, 30 Oct 2008 16:10:58 -0500
From: Anthony Minessale [EMAIL PROTECTED]
Reply-To: freeswitch-users@lists.freeswitch.org
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication

just setup a gateway in fs that has reg=false and the proper credentials to 
pass the challenge.


On Thu, Oct 30, 2008 at 2:49 PM, Wellie Chao [EMAIL PROTECTED] wrote:
  Hangups do not work for me under certain circumstances. Here is the
  background information:

  * Our carrier uses a Metaswitch server with Acme Packet in front as a
    proxy/SBC. Only the Acme Packet machine is publicly visible
    (64.115.128.6).

  * Our Freeswitch server is at 216.57.23.143.

  * For calls originating from Freeswitch and terminating on Metaswitch
    (216.57.23.143 - 64.115.128.6), Freeswitch authenticates with
    Metaswitch and everything works hunky-dorey. Either side can hang up and
    the other side will automatically hang up without requiring a manual
    hang-up.

  Now the problem:

  For calls originating from Metaswitch to Freeswitch (64.115.128.6 -
  216.57.23.143), Metaswitch does not authenticate with Freeswitch.
  Metaswitch also does not use the existing authenticated registration that
  our Freeswitch server initiates with Metaswitch upon startup of
  Freeswitch. Metaswitch just begins a new (unauthenticated) session and we
  have configured Freeswitch to allow any inbound calls from 64.115.128.6
  without requiring authentication.

  We receive inbound calls (Metaswitch to Freeswitch, 64.115.128.6 -
  216.57.23.143) just fine. The phone rings and we can have a normal
  conversation. If the caller (the endpoint attached to Metaswitch) hangs
  up, both sides hang up. If I hang up (remember, I'm at the endpoint
  attached to Freeswitch), the caller's line remains attached forever.

  I have recorded a packet trace. Look at freeswitch_2.cap in the ZIP file,
  and you want to graph the first call starting at 21.202 and ending 53.798.
  If you go to time 53.087, you can see that my Freeswitch server sends a
  BYE to Metaswitch. This is a result of me hanging up my phone. At time
  53.089, you see Metaswitch responding with 401 Unauthorized. Later at time
  53.777, you see a BYE from Metaswitch to Freeswitch, but you should ignore
  this because that was a result of the caller (the guy hooked up to
  Metaswitch) manually hanging up. If he had not hung up his phone, the BYE
  from Metaswitch to Freeswitch would not have been issued and his phone
  would just stay on the line forever. Also, when I hang up my phone, I see
  the following at the Freeswitch console:

  2008-10-29 23:03:28 [ERR] sofia_reg.c:1089
  sofia_reg_handle_sip_r_challenge() No Matching gateway found

  I presume that Freeswitch emits this error because it got the 401
  Unauthorized from Metaswitch.

  I also asked our carrier for a packet trace of a successful hangup on the
  Aastra platform (the engineer at the carrier says it is an Asterisk
  derivative -- I'm not sure about that). Look at
  Aastra_authentication_test.cap in the ZIP file. Graph the first call
  starting at 43.633 and ending 93.156. If you go to 93.118, you'll see that
  the Aastra server sends a BYE. Just like our Freeswitch scenario,
  Metaswitch sends back a 401 Unauthorized, but in response to the 401
  Unauthorized, Aastra then sends back another BYE with the difference that
  the second BYE is authenticated. Metaswitch gets the second BYE and
  responds with 200 OK.

Re: [Freeswitch-users] Call Forwarding from phone

Ok,

But don't I need the aggressive nat detection since most of my clients  
will be behind nat??

-N



On Oct 30, 2008, at 2:29 PM, Brian West wrote:

 Turn off the aggressive nat detection on the profile... if its on that
 would explain what is going on.

 /b

 On Oct 30, 2008, at 4:24 PM, Noah Silverman wrote:

 Hello,

 I have a configuration (dialplan?) question.

 One of my users used the call forward button on his sip phone.  The
 phone is directly registered to FS.

 The forwarded calls fail.  It looks like the phone is sending a
 redirect message to FS, but then the call is then not getting  
 routed
 correctly through FS.  So, the call is dropped.

 Is there something specific I need to configure to account for
 possible call forwarding from phones??

 Thanks,

 -Noah


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Re: [Freeswitch-users] Call Forwarding from phone

Well not if they are configured properly and use stun... It shouldn't  
ever be the responsibility of the registrar to overcome broken  
clients... thats how we got into this SIP mess in the first place.

/b

On Oct 30, 2008, at 5:03 PM, Noah Silverman wrote:

 Ok,

 But don't I need the aggressive nat detection since most of my clients
 will be behind nat??

 -N


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Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication

Turn on the TPORT_LOG=1 ./freeswitch and let me see the challenge  
packet.

/b

On Oct 30, 2008, at 4:45 PM, Wellie Chao wrote:

 Here is what I have:

 include
  gateway name=broadview
param name=username value=MY_USERNAME/
param name=password value=MY_PASSWORD/
param name=realm value=64.115.128.6/
param name=proxy value=64.115.128.6/
param name=register value=false/
  /gateway
 /include

 Whether register is true or false doesn't seem to make a difference  
 (except that Freeswitch then comes up with broadview in NOREG  
 state). On calls from Metaswitch to Freeswitch, it's the same  
 problem, and I get the same message in the Freeswitch logs:

 2008-10-30 17:39:04 [ERR] sofia_reg.c:1089  
 sofia_reg_handle_sip_r_challenge() No Matching gateway found

 I presume this is the same thing with the 401 Unauthorized packet  
 being sent by Metaswitch in response to Freeswitch's BYE message.  
 Note that the call itself goes just fine. I pick up, both sides can  
 hear each other. Just the hangup gets messed up and for some reason  
 Metaswitch expects an authenticated BYE message even though the  
 connection was not authenticated in the beginning when Metaswitch  
 initiated it. The packet trace shows this and it's very odd.

 Is that what you meant when you said set up a gateway in Freeswitch  
 that has reg=false and the proper credentials?


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Re: [Freeswitch-users] Call Forwarding from phone

Oh, wow.  I haven't even started to get into STUN yet.  So far, FS has  
just worked beautifully without STUN.

-N


On Oct 30, 2008, at 3:09 PM, Brian West wrote:

 Well not if they are configured properly and use stun... It shouldn't
 ever be the responsibility of the registrar to overcome broken
 clients... thats how we got into this SIP mess in the first place.

 /b

 On Oct 30, 2008, at 5:03 PM, Noah Silverman wrote:

 Ok,

 But don't I need the aggressive nat detection since most of my  
 clients
 will be behind nat??

 -N


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Re: [Freeswitch-users] mod_cdr revival (or new module maybe)

Don't worry, I don't think you are undermining anything.  It's just a
discussion thread.
If Yossi gets mod_cdr back up to speed we will happily accept it.

BTW, *hint*
You use the perl script to turn the current batch of xml files into a single
file or array of sql stmts.
then you pass that on to a single insert / update transaction.

Your realtime insert into the db idea will suffer the same slow effect of
single transaction per insert which is your bottleneck. But it will scale
rather far before it becomes a problem.  The bigger issue is the chicken or
the
egg on what to do when the db is down.  File system is the safest place to
put the data
cos if it goes, so does you box and there is little chance of dispare apart
from filling it up and
you have the ability to reprocess the files again if the db dies.  Single
file per record solves
file locking problems and the xml data is so big anyway you would want to
seggregate it.

Like I said I don't go on giving out hints all the time so that's the last
from me for now. ;)



On Thu, Oct 30, 2008 at 3:27 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:

 Don't get me wrong. I am not trying to undermine the excellent work that
 has already been done on the event handling
 modules of FS.

 I myself am a big fun of modular constructs in order to achieve
 something complicated (e.g. the UNIX way).

 But there are some situations that we will always need some extra
 features in order to accomplish our goal.

 As I wrote before, my current system is using the cdr-push (from the
 gateway side) method to gather my cdrs.
 Many billing systems and many terminating providers are satisfied with
 that model alone. I am not. In the end of each hour
 I am gathering all the cdrs and checking them one by one against my
 database in order to verify that ALL of my cdrs were
 handled by my radius servers.

 In my future system (where a more batch-like mode is preferable) I am
 forced to use a cdr-pull method. My billing system will
 be responisble for pulling the cdrs from the gateways and then process
 them. So if we have a look at the event handling weaponry of FS the
 following modules cannot meet me needs :

 mod_event_multicast
 mod_event socket
 mod_radius_cdr
 mod_xmpp_event

 because they all rely on a mechanism where the gateway is pushing my
 cdrs to my billing system (and you need a checking mechanism to verify
 that all the cdrs were handled).

 So now, lets have a look at the alternatives :

 mod_xml_cdr : this was my first choice as it had all the info needed
 (even more). But the cdr per file approach proved out to be
 inadequate in terms of performance. I wrote a few lines of code (in
 perl) that did a listing of the directory, parsed the cdrs using
 XML::Simple (without even doing anything meaningful, e.g. checks,
 inserts to a database etc.) and I was not able to make my system
 parse more than a 30 cdrs/sec (reference 1000 USD average system, not
 drawing any conclusions, don't get me wrong).
 Also while the parser was running I had a high cpu utilization (as
 expected) that was competing with my FS service. Wouldn't that make
 a bad impact of my freeswitch performance?

 so I went to a different module

 mod_cdr_csv : the text .csv format with the plethora of attributes was
 perfect for my application but it would be a daunting task
 for me to make a script tha purges the Master csv while it was being
 used by FS. I googled the issue and found about the HUP method
 and the rotate directive in the config. So far so good. But it lacked
 some functionality.

 All I was trying to state in my previous messages was that it would be
 nice if the rotation was being initiated by FS, and maybe have different
 behavior depending on another directive or something. Let me explain
 what I mean.

 There are some people who don't care about what cdrs exist in a given
 rotated log file, as long as a) no cdrs get lost in the process of rotation
 , b) no duplicates exist. I can see that the HUP method satisfied this
 need.

 There is another group of people that DO care about what cdrs are in a
 rotated log file. E.g. : The file with the name
 Master.csv.2008-01-01-09-00-00
 would only contain cdrs that were terminated from 2008-01-01 09:00:00
 to 2008-01-01 09:59:59 if an hourly setup was desirable or from
 2008-01-01 09:00:00 to 2008-01-01 09:04:59 if a five-minutes setup
 was desirable. That need is not covered by the current HUP method
 because some cdrs might escape from the next to the previous file due
 to the fact that there is a delay between cron executing the HUP and FS
 doing
 the file log rotation.

 In simple words : I am getting my job done with FS they way it already
 is (no question about that). BUT, I (or someone with different needs)
 could do
 his/her job better if some minor features were present.

 I am not saying that you should should embrace modules that meet the
 average needs yet sacrifice flexibility. I am merely suggesting
 extending the flexibility

Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication

make sure realm matches the realm in the challenge packet from the other
device.

On Thu, Oct 30, 2008 at 5:11 PM, Brian West [EMAIL PROTECTED] wrote:

 Turn on the TPORT_LOG=1 ./freeswitch and let me see the challenge
 packet.

 /b

 On Oct 30, 2008, at 4:45 PM, Wellie Chao wrote:

  Here is what I have:
 
  include
   gateway name=broadview
 param name=username value=MY_USERNAME/
 param name=password value=MY_PASSWORD/
 param name=realm value=64.115.128.6/
 param name=proxy value=64.115.128.6/
 param name=register value=false/
   /gateway
  /include
 
  Whether register is true or false doesn't seem to make a difference
  (except that Freeswitch then comes up with broadview in NOREG
  state). On calls from Metaswitch to Freeswitch, it's the same
  problem, and I get the same message in the Freeswitch logs:
 
  2008-10-30 17:39:04 [ERR] sofia_reg.c:1089
  sofia_reg_handle_sip_r_challenge() No Matching gateway found
 
  I presume this is the same thing with the 401 Unauthorized packet
  being sent by Metaswitch in response to Freeswitch's BYE message.
  Note that the call itself goes just fine. I pick up, both sides can
  hear each other. Just the hangup gets messed up and for some reason
  Metaswitch expects an authenticated BYE message even though the
  connection was not authenticated in the beginning when Metaswitch
  initiated it. The packet trace shows this and it's very odd.
 
  Is that what you meant when you said set up a gateway in Freeswitch
  that has reg=false and the proper credentials?


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[Freeswitch-users] XML cdr postgres

Hi Everybody,
i have written  few scripts to process the xml.cdrs  into postgres  from 
there basically you can do whatever you want with them.
Pls let me know if you're interested in it.
I'll be happy  to submit them.
Chav

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Re: [Freeswitch-users] XML cdr postgres

sure post them to
http://jira.freeswitch.org

would you be interested in commit access to the contrib section with your
own folder to maintain them?


On Thu, Oct 30, 2008 at 7:02 PM, Chav Paskov [EMAIL PROTECTED] wrote:

 Hi Everybody,
 i have written  few scripts to process the xml.cdrs  into postgres  from
 there basically you can do whatever you want with them.
 Pls let me know if you're interested in it.
 I'll be happy  to submit them.
 Chav

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[Freeswitch-users] OSP Interop w/ Trans Nexus

2008-10-30 Thread Gregory Boehnlein

Anyone know if Freeswitch can work w/ OSP and use something like Transnexus
for CDR/Rating?


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Re: [Freeswitch-users] XML cdr postgres

i would not mind.
the scripts + sql schema were done  in a quick and dirty manor so 
probably the interested users will have to just clean them up if they do 
not want to deal  with  all the data that XMLcdr module produces.
i can also write step by step instructions  how to set it up / it's not 
complicated /.
Chav

Anthony Minessale wrote:
 sure post them to
 http://jira.freeswitch.org

 would you be interested in commit access to the contrib section with 
 your own folder to maintain them?


 On Thu, Oct 30, 2008 at 7:02 PM, Chav Paskov [EMAIL PROTECTED] 
 mailto:[EMAIL PROTECTED] wrote:

 Hi Everybody,
 i have written  few scripts to process the xml.cdrs  into postgres
  from
 there basically you can do whatever you want with them.
 Pls let me know if you're interested in it.
 I'll be happy  to submit them.
 Chav

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Re: [Freeswitch-users] XML cdr postgres

ok good,
come into irc or drop one of the devs an email to get it setup for you.


On Thu, Oct 30, 2008 at 7:27 PM, Chav Paskov [EMAIL PROTECTED] wrote:

 i would not mind.
 the scripts + sql schema were done  in a quick and dirty manor so
 probably the interested users will have to just clean them up if they do
 not want to deal  with  all the data that XMLcdr module produces.
 i can also write step by step instructions  how to set it up / it's not
 complicated /.
 Chav

 Anthony Minessale wrote:
  sure post them to
  http://jira.freeswitch.org
 
  would you be interested in commit access to the contrib section with
  your own folder to maintain them?
 
 
  On Thu, Oct 30, 2008 at 7:02 PM, Chav Paskov [EMAIL PROTECTED]
  mailto:[EMAIL PROTECTED] wrote:
 
  Hi Everybody,
  i have written  few scripts to process the xml.cdrs  into postgres
   from
  there basically you can do whatever you want with them.
  Pls let me know if you're interested in it.
  I'll be happy  to submit them.
  Chav
 
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  ClueCon http://www.cluecon.com/
 
  AIM: anthm
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Re: [Freeswitch-users] XML cdr postgres

i'll do it  later tonight.
Thank you
Chav

Anthony Minessale wrote:
 ok good,
 come into irc or drop one of the devs an email to get it setup for you.


 On Thu, Oct 30, 2008 at 7:27 PM, Chav Paskov [EMAIL PROTECTED] 
 mailto:[EMAIL PROTECTED] wrote:

 i would not mind.
 the scripts + sql schema were done  in a quick and dirty manor so
 probably the interested users will have to just clean them up if
 they do
 not want to deal  with  all the data that XMLcdr module produces.
 i can also write step by step instructions  how to set it up /
 it's not
 complicated /.
 Chav

 Anthony Minessale wrote:
  sure post them to
  http://jira.freeswitch.org
 
  would you be interested in commit access to the contrib section with
  your own folder to maintain them?
 
 
  On Thu, Oct 30, 2008 at 7:02 PM, Chav Paskov [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]
  mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
 
  Hi Everybody,
  i have written  few scripts to process the xml.cdrs  into
 postgres
   from
  there basically you can do whatever you want with them.
  Pls let me know if you're interested in it.
  I'll be happy  to submit them.
  Chav
 
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  AIM: anthm
  MSN:[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]
  mailto:[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]
  GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]
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 FreeSWITCH http://www.freeswitch.org/
 ClueCon http://www.cluecon.com/

 AIM: anthm
 MSN:[EMAIL PROTECTED] 
 mailto:[EMAIL PROTECTED]
 GTALK/JABBER/PAYPAL:[EMAIL PROTECTED] 
 mailto:[EMAIL PROTECTED]
 IRC: irc.freenode.net http://irc.freenode.net #freeswitch

 FreeSWITCH Developer Conference
 sip:[EMAIL PROTECTED] 
 mailto:[EMAIL PROTECTED]
 iax:[EMAIL PROTECTED]/888 
 http://iax:[EMAIL PROTECTED]/888
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 mailto:[EMAIL PROTECTED]
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Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication

Here is the BYE from Freeswitch to Metaswitch:

send 683 bytes to udp/[64.115.128.6]:5060 at 00:44:46.607025:

BYE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 216.57.23.143;rport;branch=z9hG4bKNa693jZ8SD54D
Max-Forwards: 70
From: sip:[EMAIL PROTECTED];tag=r4yBmtX3U0Hrr
To: 
sip:[EMAIL PROTECTED]:5060;transport=udp;tag=Broadview1+1+25f76f+cc3ba534
Call-ID: [EMAIL PROTECTED]
CSeq: 106588607 BYE
Contact: sip:[EMAIL PROTECTED]:5060;transport=udp
User-Agent: FreeSWITCH-mod_sofia/1.0.1-9171
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, 
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: 100rel, timer, precondition, path, replaces
Reason: Q.850;cause=16;text=NORMAL_CLEARING
Content-Length: 0

Metaswitch is not so happy with the BYE message:

recv 491 bytes from udp/[64.115.128.6]:5060 at 00:44:46.630445:

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
216.57.23.143;received=216.57.23.143;branch=z9hG4bKNa693jZ8SD54D;rport=5060
From: sip:[EMAIL PROTECTED];tag=r4yBmtX3U0Hrr
To: 
sip:[EMAIL PROTECTED]:5060;transport=udp;tag=Broadview1+1+25f76f+cc3ba534
Call-ID: [EMAIL PROTECTED]
CSeq: 106588607 BYE
WWW-Authenticate: Digest 
realm=SipLocal,nonce=3e952db60fb8,stale=false,algorithm=MD5,qop=auth
Server: DC-SIP/2.0
Organization:
Supported: 100rel
Content-Length: 0

Right after receiving the 401 Unauthorized message from Metaswitch, 
Freeswitch emits the following error on the console:

2008-10-30 20:44:46 [ERR] sofia_reg.c:1089 
sofia_reg_handle_sip_r_challenge() No Matching gateway found

At this point, the caller (the endpoint connected to Metaswitch) just 
remains on the line, never having received the BYE.

Did you take a look at the packet traces I captured? The carrier 
gave me a packet trace for an Aastra PBX/softswitch, and it had the same 
interaction, but immediately upon receiving the 401 Unauthorized from 
Metaswitch, the Aastra machine then sent a second BYE, this time with 
authentication. Is there some way I can tell Freeswitch to do the same?

Regards,
Wellie

On Thu, 30 Oct 2008, Brian West wrote:

 Date: Thu, 30 Oct 2008 17:11:04 -0500
 From: Brian West [EMAIL PROTECTED]
 Reply-To: freeswitch-users@lists.freeswitch.org
 To: freeswitch-users@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication

 Turn on the TPORT_LOG=1 ./freeswitch and let me see the challenge
 packet.

 /b

 On Oct 30, 2008, at 4:45 PM, Wellie Chao wrote:

 Here is what I have:

 include
  gateway name=broadview
param name=username value=MY_USERNAME/
param name=password value=MY_PASSWORD/
param name=realm value=64.115.128.6/
param name=proxy value=64.115.128.6/
param name=register value=false/
  /gateway
 /include

 Whether register is true or false doesn't seem to make a difference
 (except that Freeswitch then comes up with broadview in NOREG
 state). On calls from Metaswitch to Freeswitch, it's the same
 problem, and I get the same message in the Freeswitch logs:

 2008-10-30 17:39:04 [ERR] sofia_reg.c:1089
 sofia_reg_handle_sip_r_challenge() No Matching gateway found

 I presume this is the same thing with the 401 Unauthorized packet
 being sent by Metaswitch in response to Freeswitch's BYE message.
 Note that the call itself goes just fine. I pick up, both sides can
 hear each other. Just the hangup gets messed up and for some reason
 Metaswitch expects an authenticated BYE message even though the
 connection was not authenticated in the beginning when Metaswitch
 initiated it. The packet trace shows this and it's very odd.

 Is that what you meant when you said set up a gateway in Freeswitch
 that has reg=false and the proper credentials?

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Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication

If I change the realm, I will not be able to make outgoing calls because 
the realm must be 64.115.128.6 in order to register with Metaswitch for 
outbound calls.


Are you suggesting I create two gateway entries, one for outbound and one 
for inbound? Will Freeswitch respond with an authenticated BYE message 
even if the session was initiated in an authenticated fashion by 
Metaswitch? It's a little frustrating because Metaswitch is initiating the 
call to Freeswitch without authentication, yet it expects Freeswitch to 
reply with an authenticated BYE message in order to end the call. It 
really should be smart enough to realize that since it initiated the call 
to a particular Freeswitch instance at A.B.C.D IP address, it should allow 
unauthenticated BYE messages from that IP address.


On Thu, 30 Oct 2008, Anthony Minessale wrote:


Date: Thu, 30 Oct 2008 18:54:34 -0500
From: Anthony Minessale [EMAIL PROTECTED]
Reply-To: freeswitch-users@lists.freeswitch.org
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication

make sure realm matches the realm in the challenge packet from the other device.

On Thu, Oct 30, 2008 at 5:11 PM, Brian West [EMAIL PROTECTED] wrote:
  Turn on the TPORT_LOG=1 ./freeswitch and let me see the challenge
  packet.

  /b

On Oct 30, 2008, at 4:45 PM, Wellie Chao wrote:

 Here is what I have:

 include
  gateway name=broadview
    param name=username value=MY_USERNAME/
    param name=password value=MY_PASSWORD/
    param name=realm value=64.115.128.6/
    param name=proxy value=64.115.128.6/
    param name=register value=false/
  /gateway
 /include

 Whether register is true or false doesn't seem to make a difference
 (except that Freeswitch then comes up with broadview in NOREG
 state). On calls from Metaswitch to Freeswitch, it's the same
 problem, and I get the same message in the Freeswitch logs:

 2008-10-30 17:39:04 [ERR] sofia_reg.c:1089
 sofia_reg_handle_sip_r_challenge() No Matching gateway found

 I presume this is the same thing with the 401 Unauthorized packet
 being sent by Metaswitch in response to Freeswitch's BYE message.
 Note that the call itself goes just fine. I pick up, both sides can
 hear each other. Just the hangup gets messed up and for some reason
 Metaswitch expects an authenticated BYE message even though the
 connection was not authenticated in the beginning when Metaswitch
 initiated it. The packet trace shows this and it's very odd.

 Is that what you meant when you said set up a gateway in Freeswitch
 that has reg=false and the proper credentials?


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--
Anthony Minessale II

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

AIM: anthm
MSN:[EMAIL PROTECTED]
GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:[EMAIL PROTECTED]
iax:[EMAIL PROTECTED]/888
googletalk:[EMAIL PROTECTED]
pstn:213-799-1400

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Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication

Just to correct an error, I asked below Will Freeswitch respond with an 
authenticated BYE message even if the session was initiated in an 
authenticated fashion by Metaswitch? and I meant even if the session was 
initiated in an UNauthenticated fashion.


Also, to expound on the problem, I don't see how changing the realm will 
help because the problem is not that Freeswitch is sending the wrong 
realm, but that Freeswitch is not responding with an authenticated BYE 
message at all -- it's only sending an unauthenticated BYE message. The 
realm, while possibly important later, assumes that Freeswitch is using 
authentication in the BYE message it sends to Metaswitch. Right now it's 
not. From an intuitive point of view, it makes sense: Freeswitch is 
thinking, well Metaswitch, you called me, why do I need to authenticate 
myself to you. The Aastra softswitch (in the other packet trace in the 
ZIP file I sent in a previous email to this list) deals with this by 
sending an authenticated BYE when the unauthenticated BYE fails with 401 
Unauthorized. Is there some way I can configure Freeswitch to do the same?


On Thu, 30 Oct 2008, Wellie Chao wrote:


Date: Thu, 30 Oct 2008 20:55:37 -0400 (EDT)
From: Wellie Chao [EMAIL PROTECTED]
Reply-To: freeswitch-users@lists.freeswitch.org
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication

If I change the realm, I will not be able to make outgoing calls because the 
realm must be 64.115.128.6 in order to register with Metaswitch for outbound 
calls.


Are you suggesting I create two gateway entries, one for outbound and one for 
inbound? Will Freeswitch respond with an authenticated BYE message even if 
the session was initiated in an authenticated fashion by Metaswitch? It's a 
little frustrating because Metaswitch is initiating the call to Freeswitch 
without authentication, yet it expects Freeswitch to reply with an 
authenticated BYE message in order to end the call. It really should be smart 
enough to realize that since it initiated the call to a particular Freeswitch 
instance at A.B.C.D IP address, it should allow unauthenticated BYE messages 
from that IP address.


On Thu, 30 Oct 2008, Anthony Minessale wrote:


 Date: Thu, 30 Oct 2008 18:54:34 -0500
 From: Anthony Minessale [EMAIL PROTECTED]
 Reply-To: freeswitch-users@lists.freeswitch.org
 To: freeswitch-users@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] Hangup problem/SIP BYE lacking
 authentication

 make sure realm matches the realm in the challenge packet from the other
 device.

 On Thu, Oct 30, 2008 at 5:11 PM, Brian West [EMAIL PROTECTED] wrote:
   Turn on the TPORT_LOG=1 ./freeswitch and let me see the challenge
   packet.

   /b

 On Oct 30, 2008, at 4:45 PM, Wellie Chao wrote:

  Here is what I have:
 
  include

   gateway name=broadview
     param name=username value=MY_USERNAME/
     param name=password value=MY_PASSWORD/
     param name=realm value=64.115.128.6/
     param name=proxy value=64.115.128.6/
     param name=register value=false/
   /gateway
  /include
 
  Whether register is true or false doesn't seem to make a difference

  (except that Freeswitch then comes up with broadview in NOREG
  state). On calls from Metaswitch to Freeswitch, it's the same
  problem, and I get the same message in the Freeswitch logs:
 
  2008-10-30 17:39:04 [ERR] sofia_reg.c:1089

  sofia_reg_handle_sip_r_challenge() No Matching gateway found
 
  I presume this is the same thing with the 401 Unauthorized packet

  being sent by Metaswitch in response to Freeswitch's BYE message.
  Note that the call itself goes just fine. I pick up, both sides can
  hear each other. Just the hangup gets messed up and for some reason
  Metaswitch expects an authenticated BYE message even though the
  connection was not authenticated in the beginning when Metaswitch
  initiated it. The packet trace shows this and it's very odd.
 
  Is that what you meant when you said set up a gateway in Freeswitch

  that has reg=false and the proper credentials?


 ___
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 --
 Anthony Minessale II

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

 AIM: anthm
 MSN:[EMAIL PROTECTED]
 GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]
 IRC: irc.freenode.net #freeswitch

 FreeSWITCH Developer Conference
 sip:[EMAIL PROTECTED]
 iax:[EMAIL PROTECTED]/888
 googletalk:[EMAIL PROTECTED]
 pstn:213-799-1400


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Re: [Freeswitch-users] Call Forwarding from phone

Can anyone recommend a best open source STUN server to work with FS??

Thanks,

- N


On Oct 30, 2008, at 3:09 PM, Brian West wrote:

 Well not if they are configured properly and use stun... It shouldn't
 ever be the responsibility of the registrar to overcome broken
 clients... thats how we got into this SIP mess in the first place.

 /b

 On Oct 30, 2008, at 5:03 PM, Noah Silverman wrote:

 Ok,

 But don't I need the aggressive nat detection since most of my  
 clients
 will be behind nat??

 -N


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Re: [Freeswitch-users] Call Forwarding from phone

Its not so much for FreeSWITCH its for your phones to figure out their  
public IP so they don't lie when they register.

http://sourceforge.net/project/showfiles.php?group_id=47735

/b

On Oct 30, 2008, at 8:28 PM, Noah Silverman wrote:

 Can anyone recommend a best open source STUN server to work with  
 FS??

 Thanks,

 - N


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Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication

They really shouldn't be challenging the bye like that because not  
every device can answer the challenge properly.  Either case it  
doesn't matter if you have a proper domain 206.57.23.143 in your user  
directory with the proper user and realm set to meet this challenge.
We have a setting in FreeSWITCH to auth all packets and when we did  
that we found out that some devices just do not work properly when  
doing that.  I need to see your original gateway XML.  You can email  
it off list if you like and let me try to solve this.


/b

On Oct 30, 2008, at 7:52 PM, Wellie Chao wrote:


Here is the BYE from Freeswitch to Metaswitch:

send 683 bytes to udp/[64.115.128.6]:5060 at 00:44:46.607025:


   BYE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 216.57.23.143;rport;branch=z9hG4bKNa693jZ8SD54D
   Max-Forwards: 70
   From: sip:[EMAIL PROTECTED];tag=r4yBmtX3U0Hrr
   To:
sip: 
[EMAIL PROTECTED]:5060;transport=udp;tag=Broadview1+1+25f76f 
+cc3ba534

   Call-ID: [EMAIL PROTECTED]
   CSeq: 106588607 BYE
   Contact: sip:[EMAIL PROTECTED]:5060;transport=udp
   User-Agent: FreeSWITCH-mod_sofia/1.0.1-9171
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,  
SUBSCRIBE,

NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: 100rel, timer, precondition, path, replaces
   Reason: Q.850;cause=16;text=NORMAL_CLEARING
   Content-Length: 0




Metaswitch is not so happy with the BYE message:

recv 491 bytes from udp/[64.115.128.6]:5060 at 00:44:46.630445:


   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP
216.57.23.143 
;received=216.57.23.143;branch=z9hG4bKNa693jZ8SD54D;rport=5060

   From: sip:[EMAIL PROTECTED];tag=r4yBmtX3U0Hrr
   To:
sip: 
[EMAIL PROTECTED]:5060;transport=udp;tag=Broadview1+1+25f76f 
+cc3ba534

   Call-ID: [EMAIL PROTECTED]
   CSeq: 106588607 BYE
   WWW-Authenticate: Digest
realm 
=SipLocal,nonce=3e952db60fb8,stale=false,algorithm=MD5,qop=auth

   Server: DC-SIP/2.0
   Organization:
   Supported: 100rel
   Content-Length: 0




Right after receiving the 401 Unauthorized message from Metaswitch,
Freeswitch emits the following error on the console:

2008-10-30 20:44:46 [ERR] sofia_reg.c:1089
sofia_reg_handle_sip_r_challenge() No Matching gateway found

At this point, the caller (the endpoint connected to Metaswitch) just
remains on the line, never having received the BYE.

Did you take a look at the packet traces I captured? The carrier
gave me a packet trace for an Aastra PBX/softswitch, and it had the  
same

interaction, but immediately upon receiving the 401 Unauthorized from
Metaswitch, the Aastra machine then sent a second BYE, this time with
authentication. Is there some way I can tell Freeswitch to do the  
same?


Regards,
Wellie


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[Freeswitch-users] how to response to message OPTIONS

2008-10-30 Thread mashudi

Dear Folks,
i have a softswitch and i will connect this softswitch to freeswitch as 
media server. My softswitch send OPTIONS message periodically to 
freeswitch but i don'tknow how to response to this message.
please help how to config freeswitch in order to make response from 
softswitch OPTIONS message ,and how to accept every message comming from 
softswitch ip address to freeswitch without authentication.
thanks you for helps.

best regards,

mashudi

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Re: [Freeswitch-users] OSP Interop w/ Trans Nexus

There is currently no OSP support, although it would be interesting to  
add it.

Mike

On Oct 30, 2008, at 8:16 PM, Gregory Boehnlein wrote:

 Anyone know if Freeswitch can work w/ OSP and use something like  
 Transnexus
 for CDR/Rating?


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Re: [Freeswitch-users] OSP Interop w/ Trans Nexus

2008-10-30 Thread Wasim Baig

I spoke to Jim about this back in January, to see if he would be able to add
OSP / FS integration to transnexus.
Last we left it at was that he'd try to get in touch with some of the
developers at VoN and see what could be done.
Perhaps, its a good time to re-initiate that discussion.

-wasim

On Fri, Oct 31, 2008 at 10:07 AM, Michael Jerris [EMAIL PROTECTED] wrote:

 There is currently no OSP support, although it would be interesting to
 add it.

 Mike

 On Oct 30, 2008, at 8:16 PM, Gregory Boehnlein wrote:

  Anyone know if Freeswitch can work w/ OSP and use something like
  Transnexus
  for CDR/Rating?


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-- 
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you scope creep, so shall we reap ...
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Re: [Freeswitch-users] Help with Pocketsphinx and setting up pizza demo on Windows XP

2008-10-30 Thread mszlazak


 It looks like I'll also need a file like speechTools.jm


 


 

-Original Message-
From: [EMAIL PROTECTED]
To: freeswitch-users@lists.freeswitch.org
Sent: Thu, 30 Oct 2008 8:54 pm
Subject: [Freeswitch-users] Help with Pocketsphinx and setting up pizza demo on 
Windows XP









First, what version of pocketsphinx comes with the current FS? 



I understand that the latest release from Carnagie Mellon of pocketsphinx is 
version 0.5.



Second, how do I set up the pizza demo with FS on Windows XP?? The wiki Mod 
pocketsphinx (http://wiki.freeswitch.org/wiki/Mod_pocketsphinx) seems only for 
Linux but I have unpacked and placed the sound and grammar files in similar 
folders on XP. I'm only missing the script file ps_pizza.js. Where do I get 
that?



What else do I need to do to get this demo running on my system?



Thanks.



Mark.





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Re: [Freeswitch-users] Help with Pocketsphinx and setting up pizza demo on Windows XP