[Freeswitch-users] Many CS_REPORTING state Zombie session

2009-10-31 Thread Dome Charoenyost
Dear All,
  I'm disable all CDR and use nibble_bill (modify) and
odbc_query for create CDR and dedcut balance. but i found many zombie
channel and can't kill
Like a
51555b02-c5ba-11de-af77-774e15238dbe,outbound,2009-10-30
18:11:38,1256951498,sofia/external/66848087...@xxx.xxx.xx.191:7050,CS_REPORTING,FreeSWITCH,66850304266,xxx.xxx.xxx.191,7050,,,XML,public,

I'm using FreeSWITCH Version 1.0.5pre3 (hacked)

How to fix ?

Dome C.

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Re: [Freeswitch-users] Freeswitch seems to doesn't reknow dial tone after the first call using OpenZAP (analog spans)

2009-10-31 Thread Albano Daniele Salvatore - Lavoro

Hi,

i've done more tests, with svn too, but nothing to do. The strange thing 
is when freeswitch shutdown and close zaptel channel i get back the dial 
tone from my provider (i'm checking trought ztmonitor 1 -vv). Can be 
something is missing when span get closed?


PS1: with svn i can do only one call, but i can receive calls without 
problems

PS2: i joined yesterday on irc but i had few time ... hope to join monday


Best Regards,
Daniele

Michael Collins ha scritto:



On Thu, Oct 29, 2009 at 9:12 AM, Anthony Minessale 
anthony.miness...@gmail.com mailto:anthony.miness...@gmail.com wrote:


You may want to work with FreeSWITCH svn trunk for the time being.
There have been some changes and if you have a problem to fix it's
easier to debug with that version.


2009/10/28 Albano Daniele Salvatore - Lavoro i...@daccii.it
mailto:i...@daccii.it

Hi to all,

this is my first message here!

Welcome!

Like Tony said, the latest SVN trunk is probably what you want. We are 
already doing pre-releases of 1.0.5. SVN trunk is extremely stable. 
Also, you can get realtime help via IRC at irc.freenode.net 
http://irc.freenode.net, channels #freeswitch and #openzap.


-MC
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Re: [Freeswitch-users] SIP Overlap support?

2009-10-31 Thread Michael Jerris
This may be possible with tcp, how could this work on udp?  Can you  
provide an rfc reference on this?

Mike

On Oct 24, 2009, at 8:13 AM, Dennis wrote:

 ok, as written, i come back after some tests with fs and a thomson  
 cirpack.

 it did not work - at least in our tests.

 we are using socket outbound and when a call comes in, it starts the
 socket of fs. the number may be 123456. fs sends the respond 484 and
 our carrier receives this information. but fs ends the call with
 hangup_cause = invalid_number_format.
 the carrier has one more digit for the phone number and sends 1234567
 and the above mentioned behavior repeats.

 the behavior we want and expected is, that the call stays in the
 socket after response 484, so that the carrier can send the 1234567
 into the same socket.
 the management, when fs should send response 484 and when fs should be
 answered would be programmed by us.
 it also important, that fs keeps the call in the socket, so we can
 tell fs, to answer the call after x seconds anyway.

 any ideas, what we could do?


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Re: [Freeswitch-users] SIP Overlap support?

2009-10-31 Thread Patrick List
 On Oct 24, 2009, at 8:13 AM, Dennis wrote:

 ok, as written, i come back after some tests with fs and a thomson
 cirpack.

No idea if this is useful as I'm a noob with fs. If not please excuse 
the noise. In the past Asterisk to work properly with Cirpack needed the 
following patch:

diff -uNr asterisk-1.4.19.org/main/rtp.c asterisk-1.4.19/main/rtp.c
--- asterisk-1.4.19.org/main/rtp.c  2007-10-08 22:06:33.0 +0200
+++ asterisk-1.4.19/main/rtp.c  2007-11-11 13:12:28.0 +0100
@@ -1383,6 +1383,7 @@
[34] = {1, AST_FORMAT_H263},
[103] = {1, AST_FORMAT_H263_PLUS},
[97] = {1, AST_FORMAT_ILBC},
+   [96] = {0, AST_RTP_DTMF},
[99] = {1, AST_FORMAT_H264},
[101] = {0, AST_RTP_DTMF},
[110] = {1, AST_FORMAT_SPEEX},

Maybe this helps.

Regards,
Patrick

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Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

2009-10-31 Thread Rupa Schomaker
I believe this means you are hung up in CDR reporting.  You can get a
core of the running system and look at the stack traces to see where
exactly it is hung up.

Try using:

support-d/fscore_pb gcore yourname

This will put a copy of the stacktrace on the pastebin for poeple (or
you) to look at.

On Sat, Oct 31, 2009 at 6:08 AM, Dome Charoenyost d...@tel.co.th wrote:
 Dear All,
          I'm disable all CDR and use nibble_bill (modify) and
 odbc_query for create CDR and dedcut balance. but i found many zombie
 channel and can't kill
 Like a
 51555b02-c5ba-11de-af77-774e15238dbe,outbound,2009-10-30
 18:11:38,1256951498,sofia/external/66848087...@xxx.xxx.xx.191:7050,CS_REPORTING,FreeSWITCH,66850304266,xxx.xxx.xxx.191,7050,,,XML,public,

 I'm using FreeSWITCH Version 1.0.5pre3 (hacked)

 How to fix ?

 Dome C.

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-- 
-Rupa

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Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

2009-10-31 Thread Brian West
So you're doing your billing INLINE with the session using the OSBC  
Query app?  Not the most optimal way!


/b

On Oct 31, 2009, at 10:59 AM, Rupa Schomaker wrote:


I believe this means you are hung up in CDR reporting.  You can get a
core of the running system and look at the stack traces to see where
exactly it is hung up.

Try using:

support-d/fscore_pb gcore yourname

This will put a copy of the stacktrace on the pastebin for poeple (or
you) to look at.

On Sat, Oct 31, 2009 at 6:08 AM, Dome Charoenyost d...@tel.co.th  
wrote:

Dear All,
 I'm disable all CDR and use nibble_bill (modify) and
odbc_query for create CDR and dedcut balance. but i found many zombie
channel and can't kill
Like a
51555b02-c5ba-11de-af77-774e15238dbe,outbound,2009-10-30
18:11:38,1256951498,sofia/external/66848087...@xxx.xxx.xx. 
191:7050,CS_REPORTING,FreeSWITCH,66850304266,xxx.xxx.xxx. 
191,7050,,,XML,public,


I'm using FreeSWITCH Version 1.0.5pre3 (hacked)

How to fix ?

Dome C.


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Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

2009-10-31 Thread Dome Charoenyost
I use odbc_query for retrive balance and get LCR from my billing DB.
and use nibble_bill

Dome C.


2009/10/31 Brian West br...@freeswitch.org:
 So you're doing your billing INLINE with the session using the OSBC Query
 app?  Not the most optimal way!
 /b
 On Oct 31, 2009, at 10:59 AM, Rupa Schomaker wrote:

 I believe this means you are hung up in CDR reporting.  You can get a
 core of the running system and look at the stack traces to see where
 exactly it is hung up.

 Try using:

 support-d/fscore_pb gcore yourname

 This will put a copy of the stacktrace on the pastebin for poeple (or
 you) to look at.

 On Sat, Oct 31, 2009 at 6:08 AM, Dome Charoenyost d...@tel.co.th wrote:

 Dear All,

          I'm disable all CDR and use nibble_bill (modify) and

 odbc_query for create CDR and dedcut balance. but i found many zombie

 channel and can't kill

 Like a

 51555b02-c5ba-11de-af77-774e15238dbe,outbound,2009-10-30

 18:11:38,1256951498,sofia/external/66848087...@xxx.xxx.xx.191:7050,CS_REPORTING,FreeSWITCH,66850304266,xxx.xxx.xxx.191,7050,,,XML,public,

 I'm using FreeSWITCH Version 1.0.5pre3 (hacked)

 How to fix ?

 Dome C.

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Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

2009-10-31 Thread Brian West
You should never do billing inline with the session thread is all I'm  
saying.

/b

On Oct 31, 2009, at 11:32 AM, Dome Charoenyost wrote:

 I use odbc_query for retrive balance and get LCR from my billing DB.
 and use nibble_bill

 Dome C.


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Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

2009-10-31 Thread João Mesquita
Dome, just to explain what Brian is saying:

Doing billing inline in this case means that the session thread (what you
see in show channels). If, for whatever reason, something goes wrong on the
DB connection or task for the billing, this session thread will be stuck
with it leaving it dangling around the system. This is what most likely is
happening to you. The right way to do it is to let the session thread go
(which mod_cdr_csv does) and then process the billing.

None of the methods you have described do that. You might consider
post-processing your CDR/billing information to avoid coming up with this
kind of problem that are very hard to predict.

Hope that helps,

JM

PS: Nonetheless, I still think it is valid to get a core dump like Rupa
metioned.

On Sat, Oct 31, 2009 at 2:44 PM, Brian West br...@freeswitch.org wrote:

 You should never do billing inline with the session thread is all I'm
 saying.

 /b

 On Oct 31, 2009, at 11:32 AM, Dome Charoenyost wrote:

  I use odbc_query for retrive balance and get LCR from my billing DB.
  and use nibble_bill
 
  Dome C.


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Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

2009-10-31 Thread Dome Charoenyost
2009/10/31 Brian West br...@freeswitch.org:
 You should never do billing inline with the session thread is all I'm
 saying.
Sorry I don't understand.  This is my example dialplan
   action application=set
data=dialoption=nibble_account=${caller_id_number},nibble_rate=0.001,ignore_early_media=false,originate_timeout=20/
action application=odbc_query data=select carrier_name As
provider_id,balance::text As lcr_balance,(balance/rate)::int as
lcr_timeout,(balance/rate*60)::int as lcr_timeout_sec  from
lcr('${caller_id_number}','$1','external');/
action application=log data=ALERT Balance ${lcr_balance}/
action application=log data=ALERT timeout ${lcr_timeout}/
action application=execute_extension
data=ck_zerobalance_${lcr_balance} XML public/
action application=execute_extension
data=tz_say_min_${lcr_timeout} XML public/
action application=sched_hangup data=+${lcr_timeout_sec}
alloted_timeout/
action application=log data=ALERT timeout_sec ${lcr_timeout_sec}/
action application=odbc_query data=select
'route_'||route_id::text As name,gw_prefix||prefix||gw_suffix As value
from lcr('${caller_id_number}','$1','external') ;/
action application=bridge data={${dialoption}},${route_1}/
action application=bridge data={${dialoption}},${route_2}/
action application=bridge data={${dialoption}},${route_3}/


Dome C.


 /b

 On Oct 31, 2009, at 11:32 AM, Dome Charoenyost wrote:

 I use odbc_query for retrive balance and get LCR from my billing DB.
 and use nibble_bill

 Dome C.


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Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

2009-10-31 Thread Dome Charoenyost
Thanks JM.
   So mod_nibblebill not good for running with high traffic. so we
can't do realtime billing ?

Dome C.


2009/10/31 João Mesquita jmesqu...@freeswitch.org:
 Dome, just to explain what Brian is saying:

 Doing billing inline in this case means that the session thread (what you
 see in show channels). If, for whatever reason, something goes wrong on the
 DB connection or task for the billing, this session thread will be stuck
 with it leaving it dangling around the system. This is what most likely is
 happening to you. The right way to do it is to let the session thread go
 (which mod_cdr_csv does) and then process the billing.

 None of the methods you have described do that. You might consider
 post-processing your CDR/billing information to avoid coming up with this
 kind of problem that are very hard to predict.

 Hope that helps,

 JM

 PS: Nonetheless, I still think it is valid to get a core dump like Rupa
 metioned.

 On Sat, Oct 31, 2009 at 2:44 PM, Brian West br...@freeswitch.org wrote:

 You should never do billing inline with the session thread is all I'm
 saying.

 /b

 On Oct 31, 2009, at 11:32 AM, Dome Charoenyost wrote:

  I use odbc_query for retrive balance and get LCR from my billing DB.
  and use nibble_bill
 
  Dome C.


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Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

2009-10-31 Thread Brian West
I think once you get the backtrace like rupa said we can see that  
maybe odbc_query is really hanging or something similar.


/b

On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:


2009/10/31 Brian West br...@freeswitch.org:

You should never do billing inline with the session thread is all I'm
saying.


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Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

2009-10-31 Thread João Mesquita
No, mod_nibblebill definetely needs to be enhanced but it is not the problem
and it can be used with high load traffic.

The one I am not sure about is odbc_query since it was not developed for
that.

Do what Rupa said, please.

Regards,

JM

On Sat, Oct 31, 2009 at 3:11 PM, Brian West br...@freeswitch.org wrote:

 I think once you get the backtrace like rupa said we can see that maybe
 odbc_query is really hanging or something similar.

 /b

 On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:

 2009/10/31 Brian West br...@freeswitch.org:

 You should never do billing inline with the session thread is all I'm

 saying.



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Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

2009-10-31 Thread Dome Charoenyost
i try
support-d/fscore_pb gcore root
but only get

[New Thread 0xb7656b90 (LWP 28445)]
[New Thread 0xb77bbb90 (LWP 28444)]
0xb7a5e007 in select () from /lib/libc.so.6
Saved corefile core.28443

i got core file but no more file

Dome C.

2009/11/1 Brian West br...@freeswitch.org:
 I think once you get the backtrace like rupa said we can see that maybe
 odbc_query is really hanging or something similar.
 /b
 On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:

 2009/10/31 Brian West br...@freeswitch.org:

 You should never do billing inline with the session thread is all I'm

 saying.

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Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

2009-10-31 Thread Dome Charoenyost
I found bug in fscore_pb   $pwd should be $mypwd

Now i post already. please help me check.

Dome C.


2009/11/1 João Mesquita jmesqu...@freeswitch.org:
 No, mod_nibblebill definetely needs to be enhanced but it is not the problem
 and it can be used with high load traffic.

 The one I am not sure about is odbc_query since it was not developed for
 that.

 Do what Rupa said, please.

 Regards,

 JM

 On Sat, Oct 31, 2009 at 3:11 PM, Brian West br...@freeswitch.org wrote:

 I think once you get the backtrace like rupa said we can see that maybe
 odbc_query is really hanging or something similar.
 /b
 On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:

 2009/10/31 Brian West br...@freeswitch.org:

 You should never do billing inline with the session thread is all I'm

 saying.

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179

2009-10-31 Thread Michael Jerris
You actually can use these in conditions.  Just need to be careful  
that the var you are conditioning on is already set.


Mike

On Oct 22, 2009, at 1:54 PM, Michael Collins wrote:




On Thu, Oct 22, 2009 at 5:51 AM, Rupa Schomaker r...@rupa.com wrote:
cond would be helpful here?  I updated the wiki on this one just now
with a bit more detail.  It is a api call. so, you'd use it like:

${cond(eval ? trueval : falseval)}

so to get a value of ERR if the var my myvar is  15 you could:

${cond(${myvar}  15 ? ERR : OK)}

If both sides of the comparison operator are numeric then it does
numeric comparison otherwise it does lexical string comparison.

Rupa,

Yes, you can do the set/cond API trick but you can only do it in the  
action or anti-action tags, not in the condition tags. I'm sure you  
know that but I want all those reading this thread to make the  
connection.

-MC



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[Freeswitch-users] New FreeSwitch User - WAPSERVER Question

2009-10-31 Thread Dave Stevenson
Hi,

I am new to FreeSwitch  - and VOIP too (as the following questions will 
indicate). 

I am trying to setup a VOIP system at home and am just getting started, I have 
been trawling through the Docs/Wiki and realise that I have a lot to learn !

Anyway, here's the first (of what I'm sure will be many !) questions, hopefully 
someone will take the type to help me out..

OK, I have already worked out that FreeSwitch seems to be much better supported 
under Linux than Windows (is that a fair comment ?), but while I have a long 
term intention to give Linux a go, it's something that is on the back burner 
(and has been for some time). At the moment, the current plan is to use 
FreeSwitch under WindowsXP and I have loaded the latest Windows Install. 
Everything seems to be working fine so far - well, in test mode anyway, my VOIP 
phones and teleco interface won't be here until next week, so at the moment, 
I'm just trying out the system with X-Lite.

The first question that I have is about WAPSERVER. This was installed as part 
of the FreeSwitch install. At the moment, I am manually starting FreeSwitch and 
WAPSERVER does not seem to start automatically. When I manually start it, it 
says that it is OFFLINE until I manually put it ONLINE. So :-

1. What does WAPSERVER actually do ?
2. How should it be started ?
3. My next step is to configure the Server PC to auto-logon to my Windows 2003 
Server domain and put FreeSwitch in the Startup Group - do I need to add 
WAPSERVER there too ?

The next question is about the ServerIP address. Not thinking far enough ahead, 
the machine that I installed FreeSwitch on uses DHCP to get its IP Address. I 
guess that the FreeSwitch server should really have a static IP Address. If I 
change the PC to have a static IP Address, then 

1. Do I need to re-install FreeSwitch
2. If not, is there a or is there command to get FreeSwitch to updated itself 
with a new IP Address ?
3. Maybe the Server does not need to know the change - maybe if only affect the 
clients (Phones/SoftPhones ?)

Thanks  Best Regards
Dave___
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Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question

2009-10-31 Thread Shelby Ramsey
Dave,

You should check out the settings in $FS_DIR/conf/sofia_profiles/ and 
$FS_DIR/conf/vars.xml.  To make the changes take effect just restart FS 
or use the API commands for reloading the sofia profiles.

You'll definitely want to set a static IP address on the FS server (or 
make sure the DHCP server always assigns the same IP address to the FS 
server).  Otherwise you'll constantly be changing the clients that have 
to connect to FS.

SDR


Dave Stevenson wrote:
 Hi,
  
 I am new to FreeSwitch  - and VOIP too (as the following questions 
 will indicate).
  
 I am trying to setup a VOIP system at home and am just getting 
 started, I have been trawling through the Docs/Wiki and realise that I 
 have a lot to learn !
  
 Anyway, here's the first (of what I'm sure will be many !) questions, 
 hopefully someone will take the type to help me out..
  
 OK, I have already worked out that FreeSwitch seems to be much better 
 supported under Linux than Windows (is that a fair comment ?), but 
 while I have a long term intention to give Linux a go, it's something 
 that is on the back burner (and has been for some time). At the 
 moment, the current plan is to use FreeSwitch under WindowsXP and I 
 have loaded the latest Windows Install. Everything seems to be working 
 fine so far - well, in test mode anyway, my VOIP phones and teleco 
 interface won't be here until next week, so at the moment, I'm just 
 trying out the system with X-Lite.
  
 The first question that I have is about WAPSERVER. This was installed 
 as part of the FreeSwitch install. At the moment, I am manually 
 starting FreeSwitch and WAPSERVER does not seem to start 
 automatically. When I manually start it, it says that it is OFFLINE 
 until I manually put it ONLINE. So :-
  
 1. What does WAPSERVER actually do ?
 2. How should it be started ?
 3. My next step is to configure the Server PC to auto-logon to my 
 Windows 2003 Server domain and put FreeSwitch in the Startup Group - 
 do I need to add WAPSERVER there too ?
  
 The next question is about the ServerIP address. Not thinking far 
 enough ahead, the machine that I installed FreeSwitch on uses DHCP to 
 get its IP Address. I guess that the FreeSwitch server should really 
 have a static IP Address. If I change the PC to have a static IP 
 Address, then
  
 1. Do I need to re-install FreeSwitch
 2. If not, is there a or is there command to get FreeSwitch to updated 
 itself with a new IP Address ?
 3. Maybe the Server does not need to know the change - maybe if only 
 affect the clients (Phones/SoftPhones ?)
  
 Thanks  Best Regards
 Dave
 

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Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

2009-10-31 Thread Rupa Schomaker
fscore_pb has been updated.

Next time put in the pastebin url.  This one was 10911.

Ok, a suspicious thread looks below.  If I read it right, you are
doing an originate from within the hangup handler.  That doesn't look
normal or sane to me.

Thread 41 (process 30732):
#0  0xb7a5e007 in select () from /lib/libc.so.6
#1  0xb7dde39f in apr_sleep (t=1000) at time/unix/time.c:246
#2  0xb7db12c0 in do_sleep (t=1000) at src/switch_time.c:109
#3  0xb7db15ac in switch_cond_next () at src/switch_time.c:196
#4  0xb7d862b8 in switch_ivr_originate (session=0x0, bleg=0xb48c2f00,
cause=0xb48c2ed4, bridgeto=0xb1133340
{origination_caller_id_number=0857151465,callback_uuid=c32f54cc-c6b8-11de-af77-774e15238dbe,ignore_early_media=true}sofia/external/66857151...@115.178.62.191:7050|sofia/external/66857151...@115.178.62...,
timelimit_sec=60, table=0x0, cid_name_override=0xb7e8c976
FreeSWITCH, cid_num_override=0x90ec0800 0857151465,
caller_profile_override=0x0, ovars=0x0, flags=0) at
src/switch_ivr_originate.c:2132
#5  0xb6f0def1 in originate_function (cmd=0x90164ce2
{origination_caller_id_number=0857151465,callback_uuid=c32f54cc-c6b8-11de-af77-774e15238dbe,ignore_early_media=true}sofia/external/66857151...@115.178.62.191:7050|sofia/external/66857151...@115.178.62...,
session=0x0, stream=0xb48c3004) at mod_commands.c:2413
#6  0xb7d66fb1 in switch_api_execute (cmd=0x90164cd8 originate,
arg=0x90164ce2 
{origination_caller_id_number=0857151465,callback_uuid=c32f54cc-c6b8-11de-af77-774e15238dbe,ignore_early_media=true}sofia/external/66857151...@115.178.62.191:7050|sofia/external/66857151...@115.178.62...,
session=0x0, stream=0xb48c3004) at src/switch_loadable_module.c:1567
#7  0xb7d53c4a in switch_core_session_hangup_state
(session=0x958ab678) at src/switch_core_state_machine.c:506
#8  0xb7d385e6 in switch_channel_perform_hangup (channel=0x958afd70,
file=0xb6af8788 mod_dptools.c, func=0xb6af8ba8 hangup_function,
line=637, hangup_cause=SWITCH_CAUSE_NORMAL_CLEARING) at
src/switch_channel.c:1898
#9  0xb6aee906 in hangup_function (session=0x958ab678, data=0xb7e82ea5
) at mod_dptools.c:637
#10 0xb7d4fe1f in switch_core_session_exec (session=0x958ab678,
application_interface=0xb6e10ec0, arg=0xb7e82ea5 ) at
src/switch_core_session.c:1552
#11 0xb7d4f73e in switch_core_session_execute_application
(session=0x958ab678, app=0x90163a80 hangup, arg=0xb7e82ea5 ) at
src/switch_core_session.c:1474
#12 0xb7d51131 in switch_core_standard_on_execute (session=0x958ab678)
at src/switch_core_state_machine.c:166
#13 0xb7d5215a in switch_core_session_run (session=0x958ab678) at
src/switch_core_state_machine.c:340
#14 0xb7d4e199 in switch_core_session_thread (thread=0x9020bd98,
obj=0x958ab678) at src/switch_core_session.c:1135
#15 0xb7ddc2fc in dummy_worker (opaque=0x9020bd98) at
threadproc/unix/thread.c:138
#16 0xb7caaf3b in start_thread () from /lib/libpthread.so.0
#17 0xb7a64bee in clone () from /lib/libc.so.6


On Sat, Oct 31, 2009 at 12:34 PM, Dome Charoenyost d...@tel.co.th wrote:
 I found bug in fscore_pb   $pwd should be $mypwd

 Now i post already. please help me check.

 Dome C.


 2009/11/1 João Mesquita jmesqu...@freeswitch.org:
 No, mod_nibblebill definetely needs to be enhanced but it is not the problem
 and it can be used with high load traffic.

 The one I am not sure about is odbc_query since it was not developed for
 that.

 Do what Rupa said, please.

 Regards,

 JM

 On Sat, Oct 31, 2009 at 3:11 PM, Brian West br...@freeswitch.org wrote:

 I think once you get the backtrace like rupa said we can see that maybe
 odbc_query is really hanging or something similar.
 /b
 On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:

 2009/10/31 Brian West br...@freeswitch.org:

 You should never do billing inline with the session thread is all I'm

 saying.

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-- 
-Rupa

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Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

2009-10-31 Thread Seven Du
I also got some zombie channels, if someone can help me take a look that's
really nice.

http://pastebin.freeswitch.org/10912

I only loaded mod_cdr_csv
Is it ok to use mod_xml_cdr?

Thanks.


2009/11/1 Dome Charoenyost d...@tel.co.th

 I found bug in fscore_pb   $pwd should be $mypwd

 Now i post already. please help me check.

 Dome C.


 2009/11/1 João Mesquita jmesqu...@freeswitch.org:
  No, mod_nibblebill definetely needs to be enhanced but it is not the
 problem
  and it can be used with high load traffic.
 
  The one I am not sure about is odbc_query since it was not developed for
  that.
 
  Do what Rupa said, please.
 
  Regards,
 
  JM
 
  On Sat, Oct 31, 2009 at 3:11 PM, Brian West br...@freeswitch.org
 wrote:
 
  I think once you get the backtrace like rupa said we can see that maybe
  odbc_query is really hanging or something similar.
  /b
  On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:
 
  2009/10/31 Brian West br...@freeswitch.org:
 
  You should never do billing inline with the session thread is all I'm
 
  saying.
 
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Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question

2009-10-31 Thread Dave Stevenson
Hi Shelby,

thanks for the reply on the IP Address Question.

I do not have a $FS_DIR/conf/sofia_profiles/ directory ?

$FS_DIR/conf/vars.xml has a line :-
X-PRE-PROCESS cmd=set data=bind_server_ip=auto/

I'd guess that this means that the Server uses the machines IP address 
automatically, so, I can change the IP address to fixed and don't need to 
make any changes at the Server end - just need to update the Clients.

regards
Dave


- Original Message - 
From: Shelby Ramsey sicfsl...@gmail.com
To: freeswitch-users@lists.freeswitch.org
Sent: Saturday, October 31, 2009 6:13 PM
Subject: Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question


 Dave,

 You should check out the settings in $FS_DIR/conf/sofia_profiles/ and
 $FS_DIR/conf/vars.xml.  To make the changes take effect just restart FS
 or use the API commands for reloading the sofia profiles.

 You'll definitely want to set a static IP address on the FS server (or
 make sure the DHCP server always assigns the same IP address to the FS
 server).  Otherwise you'll constantly be changing the clients that have
 to connect to FS.

 SDR


 Dave Stevenson wrote:
 Hi,

 I am new to FreeSwitch  - and VOIP too (as the following questions
 will indicate).

 I am trying to setup a VOIP system at home and am just getting
 started, I have been trawling through the Docs/Wiki and realise that I
 have a lot to learn !

 Anyway, here's the first (of what I'm sure will be many !) questions,
 hopefully someone will take the type to help me out..

 OK, I have already worked out that FreeSwitch seems to be much better
 supported under Linux than Windows (is that a fair comment ?), but
 while I have a long term intention to give Linux a go, it's something
 that is on the back burner (and has been for some time). At the
 moment, the current plan is to use FreeSwitch under WindowsXP and I
 have loaded the latest Windows Install. Everything seems to be working
 fine so far - well, in test mode anyway, my VOIP phones and teleco
 interface won't be here until next week, so at the moment, I'm just
 trying out the system with X-Lite.

 The first question that I have is about WAPSERVER. This was installed
 as part of the FreeSwitch install. At the moment, I am manually
 starting FreeSwitch and WAPSERVER does not seem to start
 automatically. When I manually start it, it says that it is OFFLINE
 until I manually put it ONLINE. So :-

 1. What does WAPSERVER actually do ?
 2. How should it be started ?
 3. My next step is to configure the Server PC to auto-logon to my
 Windows 2003 Server domain and put FreeSwitch in the Startup Group -
 do I need to add WAPSERVER there too ?

 The next question is about the ServerIP address. Not thinking far
 enough ahead, the machine that I installed FreeSwitch on uses DHCP to
 get its IP Address. I guess that the FreeSwitch server should really
 have a static IP Address. If I change the PC to have a static IP
 Address, then

 1. Do I need to re-install FreeSwitch
 2. If not, is there a or is there command to get FreeSwitch to updated
 itself with a new IP Address ?
 3. Maybe the Server does not need to know the change - maybe if only
 affect the clients (Phones/SoftPhones ?)

 Thanks  Best Regards
 Dave
 

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Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question

2009-10-31 Thread Shelby Ramsey
Dave,

I'm not sure how the file structure works on Windows.  But sofia is the 
SIP endpoint (and what you're wanting to confgure).

The bind_server setting only relates to mod_dingaling (I think).  Yes 
you can change the IP address ... but you'll need to restart FS or 
reload the sofia profiles.  The conf settings for external_sip_ip, etc. 
are the ones that matter. 

Good luck!  You can also hop on freenode @ #freeswitch and everyone 
there is quite helpful.

SDR




Dave Stevenson wrote:
 Hi Shelby,

 thanks for the reply on the IP Address Question.

 I do not have a $FS_DIR/conf/sofia_profiles/ directory ?

 $FS_DIR/conf/vars.xml has a line :-
 X-PRE-PROCESS cmd=set data=bind_server_ip=auto/

 I'd guess that this means that the Server uses the machines IP address 
 automatically, so, I can change the IP address to fixed and don't need to 
 make any changes at the Server end - just need to update the Clients.

 regards
 Dave


 - Original Message - 
 From: Shelby Ramsey sicfsl...@gmail.com
 To: freeswitch-users@lists.freeswitch.org
 Sent: Saturday, October 31, 2009 6:13 PM
 Subject: Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question


   
 Dave,

 You should check out the settings in $FS_DIR/conf/sofia_profiles/ and
 $FS_DIR/conf/vars.xml.  To make the changes take effect just restart FS
 or use the API commands for reloading the sofia profiles.

 You'll definitely want to set a static IP address on the FS server (or
 make sure the DHCP server always assigns the same IP address to the FS
 server).  Otherwise you'll constantly be changing the clients that have
 to connect to FS.

 SDR


 Dave Stevenson wrote:
 
 Hi,

 I am new to FreeSwitch  - and VOIP too (as the following questions
 will indicate).

 I am trying to setup a VOIP system at home and am just getting
 started, I have been trawling through the Docs/Wiki and realise that I
 have a lot to learn !

 Anyway, here's the first (of what I'm sure will be many !) questions,
 hopefully someone will take the type to help me out..

 OK, I have already worked out that FreeSwitch seems to be much better
 supported under Linux than Windows (is that a fair comment ?), but
 while I have a long term intention to give Linux a go, it's something
 that is on the back burner (and has been for some time). At the
 moment, the current plan is to use FreeSwitch under WindowsXP and I
 have loaded the latest Windows Install. Everything seems to be working
 fine so far - well, in test mode anyway, my VOIP phones and teleco
 interface won't be here until next week, so at the moment, I'm just
 trying out the system with X-Lite.

 The first question that I have is about WAPSERVER. This was installed
 as part of the FreeSwitch install. At the moment, I am manually
 starting FreeSwitch and WAPSERVER does not seem to start
 automatically. When I manually start it, it says that it is OFFLINE
 until I manually put it ONLINE. So :-

 1. What does WAPSERVER actually do ?
 2. How should it be started ?
 3. My next step is to configure the Server PC to auto-logon to my
 Windows 2003 Server domain and put FreeSwitch in the Startup Group -
 do I need to add WAPSERVER there too ?

 The next question is about the ServerIP address. Not thinking far
 enough ahead, the machine that I installed FreeSwitch on uses DHCP to
 get its IP Address. I guess that the FreeSwitch server should really
 have a static IP Address. If I change the PC to have a static IP
 Address, then

 1. Do I need to re-install FreeSwitch
 2. If not, is there a or is there command to get FreeSwitch to updated
 itself with a new IP Address ?
 3. Maybe the Server does not need to know the change - maybe if only
 affect the clients (Phones/SoftPhones ?)

 Thanks  Best Regards
 Dave
 

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[Freeswitch-users] Mod_pjsip

2009-10-31 Thread Meftah Tayeb
Hi my FreeSWITCH friends
Today we talked about FreeSWITCH libraries and we found pjsip, which is 
a lot better than sofia, it can do everything sofia can do and more, 
it's thread-safe and performs well, I would love to see a mod_pjsip.
The library is GPL but they also offer alternative licensing, don't get 
me wrong, Sofia is great but I think FreeSWITCH will benefit from a 
thread-safe SIP library.
don't say that i need some stuf for free, but we need to make fs a very
also we love freeswitch to run in a Embedded PC like the blackfin 
CPU/boards, Pjsip is very good for Embedded SIP Devicesawesome Open 
Source project
With much love,
DelphiWorld Information from ESET NOD32 Antivirus, version of virus 
signature database 4539 (20091024) __

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com




__ Information from ESET NOD32 Antivirus, version of virus 
signature database 4539 (20091024) __

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com




__ Information from ESET NOD32 Antivirus, version of virus signature 
database 4539 (20091024) __

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com



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Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

2009-10-31 Thread Rupa Schomaker
Nothing jumps out at me on this one.

mod_cdr_csv and/or mod_xml_cdr should be fine.

On Sat, Oct 31, 2009 at 1:24 PM, Seven Du dujinf...@gmail.com wrote:
 I also got some zombie channels, if someone can help me take a look that's
 really nice.

 http://pastebin.freeswitch.org/10912

 I only loaded mod_cdr_csv
 Is it ok to use mod_xml_cdr?

 Thanks.


 2009/11/1 Dome Charoenyost d...@tel.co.th

 I found bug in fscore_pb   $pwd should be $mypwd

 Now i post already. please help me check.

 Dome C.


 2009/11/1 João Mesquita jmesqu...@freeswitch.org:
  No, mod_nibblebill definetely needs to be enhanced but it is not the
  problem
  and it can be used with high load traffic.
 
  The one I am not sure about is odbc_query since it was not developed for
  that.
 
  Do what Rupa said, please.
 
  Regards,
 
  JM
 
  On Sat, Oct 31, 2009 at 3:11 PM, Brian West br...@freeswitch.org
  wrote:
 
  I think once you get the backtrace like rupa said we can see that maybe
  odbc_query is really hanging or something similar.
  /b
  On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:
 
  2009/10/31 Brian West br...@freeswitch.org:
 
  You should never do billing inline with the session thread is all I'm
 
  saying.
 
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-- 
-Rupa

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Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question

2009-10-31 Thread Dave Stevenson
Thanks Shelby,

the vars.xml file tells me that if external_sip_ip is unspecified (which I 
think it is), then the bind_server_ip is used, i.e., auto

Getting off topic, I stumbled on freenode today, I wasn't sure what I needed 
to do to get IRC running on Windows - yet, another to do !

(The link on http://wiki.freeswitch.org/wiki/Main_Page to IRC Chat 
(irc://irc.freenode.net/freeswitch) appears to be an invalid link)


regards
Dave


- Original Message - 
From: Shelby Ramsey sicfsl...@gmail.com
To: freeswitch-users@lists.freeswitch.org
Sent: Saturday, October 31, 2009 7:07 PM
Subject: Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question


 Dave,

 I'm not sure how the file structure works on Windows.  But sofia is the
 SIP endpoint (and what you're wanting to confgure).

 The bind_server setting only relates to mod_dingaling (I think).  Yes
 you can change the IP address ... but you'll need to restart FS or
 reload the sofia profiles.  The conf settings for external_sip_ip, etc.
 are the ones that matter.

 Good luck!  You can also hop on freenode @ #freeswitch and everyone
 there is quite helpful.

 SDR




 Dave Stevenson wrote:
 Hi Shelby,

 thanks for the reply on the IP Address Question.

 I do not have a $FS_DIR/conf/sofia_profiles/ directory ?

 $FS_DIR/conf/vars.xml has a line :-
 X-PRE-PROCESS cmd=set data=bind_server_ip=auto/

 I'd guess that this means that the Server uses the machines IP address
 automatically, so, I can change the IP address to fixed and don't need to
 make any changes at the Server end - just need to update the Clients.

 regards
 Dave


 - Original Message - 
 From: Shelby Ramsey sicfsl...@gmail.com
 To: freeswitch-users@lists.freeswitch.org
 Sent: Saturday, October 31, 2009 6:13 PM
 Subject: Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question



 Dave,

 You should check out the settings in $FS_DIR/conf/sofia_profiles/ and
 $FS_DIR/conf/vars.xml.  To make the changes take effect just restart FS
 or use the API commands for reloading the sofia profiles.

 You'll definitely want to set a static IP address on the FS server (or
 make sure the DHCP server always assigns the same IP address to the FS
 server).  Otherwise you'll constantly be changing the clients that have
 to connect to FS.

 SDR


 Dave Stevenson wrote:

 Hi,

 I am new to FreeSwitch  - and VOIP too (as the following questions
 will indicate).

 I am trying to setup a VOIP system at home and am just getting
 started, I have been trawling through the Docs/Wiki and realise that I
 have a lot to learn !

 Anyway, here's the first (of what I'm sure will be many !) questions,
 hopefully someone will take the type to help me out..

 OK, I have already worked out that FreeSwitch seems to be much better
 supported under Linux than Windows (is that a fair comment ?), but
 while I have a long term intention to give Linux a go, it's something
 that is on the back burner (and has been for some time). At the
 moment, the current plan is to use FreeSwitch under WindowsXP and I
 have loaded the latest Windows Install. Everything seems to be working
 fine so far - well, in test mode anyway, my VOIP phones and teleco
 interface won't be here until next week, so at the moment, I'm just
 trying out the system with X-Lite.

 The first question that I have is about WAPSERVER. This was installed
 as part of the FreeSwitch install. At the moment, I am manually
 starting FreeSwitch and WAPSERVER does not seem to start
 automatically. When I manually start it, it says that it is OFFLINE
 until I manually put it ONLINE. So :-

 1. What does WAPSERVER actually do ?
 2. How should it be started ?
 3. My next step is to configure the Server PC to auto-logon to my
 Windows 2003 Server domain and put FreeSwitch in the Startup Group -
 do I need to add WAPSERVER there too ?

 The next question is about the ServerIP address. Not thinking far
 enough ahead, the machine that I installed FreeSwitch on uses DHCP to
 get its IP Address. I guess that the FreeSwitch server should really
 have a static IP Address. If I change the PC to have a static IP
 Address, then

 1. Do I need to re-install FreeSwitch
 2. If not, is there a or is there command to get FreeSwitch to updated
 itself with a new IP Address ?
 3. Maybe the Server does not need to know the change - maybe if only
 affect the clients (Phones/SoftPhones ?)

 Thanks  Best Regards
 Dave
 

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Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

2009-10-31 Thread Michael Giagnocavo
Which revision? Wasn't there some read-write lock issues on sessions checked in 
during the last week?

FWIW, we do inline billing, but the message queue timeouts are very low. 
Upgrading to current SVN kills us (~15025 works fine) -- some channels stay in 
CS_EXECUTE, some in CS_REPORTING. We know it's not our code blocking, because 
there'd be transaction timeouts within 30 seconds, and these are hanging around 
for a long time.

I suspect it's something with how we do our originates and state handlers. If 
we _overwrite_ the default originate handlers AND patch originate to blank out 
dialplan/context after duplicating the caller profile, it works fine. If we do 
the right thing and let the default handler set CS_CONSUME_MEDIA, then we get 
the calls hanging around forever. As I said, I suspect this is mismatch in what 
FS does and what our app assumes it does.

-Michael

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rupa 
Schomaker
Sent: Saturday, October 31, 2009 1:28 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

Nothing jumps out at me on this one.

mod_cdr_csv and/or mod_xml_cdr should be fine.

On Sat, Oct 31, 2009 at 1:24 PM, Seven Du dujinf...@gmail.com wrote:
 I also got some zombie channels, if someone can help me take a look that's
 really nice.

 http://pastebin.freeswitch.org/10912

 I only loaded mod_cdr_csv
 Is it ok to use mod_xml_cdr?

 Thanks.


 2009/11/1 Dome Charoenyost d...@tel.co.th

 I found bug in fscore_pb   $pwd should be $mypwd

 Now i post already. please help me check.

 Dome C.


 2009/11/1 João Mesquita jmesqu...@freeswitch.org:
  No, mod_nibblebill definetely needs to be enhanced but it is not the
  problem
  and it can be used with high load traffic.
 
  The one I am not sure about is odbc_query since it was not developed for
  that.
 
  Do what Rupa said, please.
 
  Regards,
 
  JM
 
  On Sat, Oct 31, 2009 at 3:11 PM, Brian West br...@freeswitch.org
  wrote:
 
  I think once you get the backtrace like rupa said we can see that maybe
  odbc_query is really hanging or something similar.
  /b
  On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:
 
  2009/10/31 Brian West br...@freeswitch.org:
 
  You should never do billing inline with the session thread is all I'm
 
  saying.
 
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-- 
-Rupa

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Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question

2009-10-31 Thread Shelby Ramsey
Dave,

As I mentioned before ... the bind_server_ip directive is only for 
mod_dingaling.  I don't think sofia will use it.

As for IRC ... I have no knowledge of Windows machines or apps ... but I 
know there is an IRC client for FireFox called chatzilla.

SDR



Dave Stevenson wrote:
 Thanks Shelby,

 the vars.xml file tells me that if external_sip_ip is unspecified (which I 
 think it is), then the bind_server_ip is used, i.e., auto

 Getting off topic, I stumbled on freenode today, I wasn't sure what I needed 
 to do to get IRC running on Windows - yet, another to do !

 (The link on http://wiki.freeswitch.org/wiki/Main_Page to IRC Chat 
 (irc://irc.freenode.net/freeswitch) appears to be an invalid link)


 regards
 Dave


 - Original Message - 
 From: Shelby Ramsey sicfsl...@gmail.com
 To: freeswitch-users@lists.freeswitch.org
 Sent: Saturday, October 31, 2009 7:07 PM
 Subject: Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question


   
 Dave,

 I'm not sure how the file structure works on Windows.  But sofia is the
 SIP endpoint (and what you're wanting to confgure).

 The bind_server setting only relates to mod_dingaling (I think).  Yes
 you can change the IP address ... but you'll need to restart FS or
 reload the sofia profiles.  The conf settings for external_sip_ip, etc.
 are the ones that matter.

 Good luck!  You can also hop on freenode @ #freeswitch and everyone
 there is quite helpful.

 SDR




 Dave Stevenson wrote:
 
 Hi Shelby,

 thanks for the reply on the IP Address Question.

 I do not have a $FS_DIR/conf/sofia_profiles/ directory ?

 $FS_DIR/conf/vars.xml has a line :-
 X-PRE-PROCESS cmd=set data=bind_server_ip=auto/

 I'd guess that this means that the Server uses the machines IP address
 automatically, so, I can change the IP address to fixed and don't need to
 make any changes at the Server end - just need to update the Clients.

 regards
 Dave


 - Original Message - 
 From: Shelby Ramsey sicfsl...@gmail.com
 To: freeswitch-users@lists.freeswitch.org
 Sent: Saturday, October 31, 2009 6:13 PM
 Subject: Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question



   
 Dave,

 You should check out the settings in $FS_DIR/conf/sofia_profiles/ and
 $FS_DIR/conf/vars.xml.  To make the changes take effect just restart FS
 or use the API commands for reloading the sofia profiles.

 You'll definitely want to set a static IP address on the FS server (or
 make sure the DHCP server always assigns the same IP address to the FS
 server).  Otherwise you'll constantly be changing the clients that have
 to connect to FS.

 SDR


 Dave Stevenson wrote:

 
 Hi,

 I am new to FreeSwitch  - and VOIP too (as the following questions
 will indicate).

 I am trying to setup a VOIP system at home and am just getting
 started, I have been trawling through the Docs/Wiki and realise that I
 have a lot to learn !

 Anyway, here's the first (of what I'm sure will be many !) questions,
 hopefully someone will take the type to help me out..

 OK, I have already worked out that FreeSwitch seems to be much better
 supported under Linux than Windows (is that a fair comment ?), but
 while I have a long term intention to give Linux a go, it's something
 that is on the back burner (and has been for some time). At the
 moment, the current plan is to use FreeSwitch under WindowsXP and I
 have loaded the latest Windows Install. Everything seems to be working
 fine so far - well, in test mode anyway, my VOIP phones and teleco
 interface won't be here until next week, so at the moment, I'm just
 trying out the system with X-Lite.

 The first question that I have is about WAPSERVER. This was installed
 as part of the FreeSwitch install. At the moment, I am manually
 starting FreeSwitch and WAPSERVER does not seem to start
 automatically. When I manually start it, it says that it is OFFLINE
 until I manually put it ONLINE. So :-

 1. What does WAPSERVER actually do ?
 2. How should it be started ?
 3. My next step is to configure the Server PC to auto-logon to my
 Windows 2003 Server domain and put FreeSwitch in the Startup Group -
 do I need to add WAPSERVER there too ?

 The next question is about the ServerIP address. Not thinking far
 enough ahead, the machine that I installed FreeSwitch on uses DHCP to
 get its IP Address. I guess that the FreeSwitch server should really
 have a static IP Address. If I change the PC to have a static IP
 Address, then

 1. Do I need to re-install FreeSwitch
 2. If not, is there a or is there command to get FreeSwitch to updated
 itself with a new IP Address ?
 3. Maybe the Server does not need to know the change - maybe if only
 affect the clients (Phones/SoftPhones ?)

 Thanks  Best Regards
 Dave
 

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 FreeSWITCH-users@lists.freeswitch.org
 

Re: [Freeswitch-users] Mod_pjsip

2009-10-31 Thread Diego Viola
This is pretty interesting, is sofia thread safe as well?

Diego

On Sat, Oct 31, 2009 at 8:21 PM, Meftah Tayeb tayeb.mef...@gmail.com wrote:
                Hi my FreeSWITCH friends
 Today we talked about FreeSWITCH libraries and we found pjsip, which is
 a lot better than sofia, it can do everything sofia can do and more,
 it's thread-safe and performs well, I would love to see a mod_pjsip.
 The library is GPL but they also offer alternative licensing, don't get
 me wrong, Sofia is great but I think FreeSWITCH will benefit from a
 thread-safe SIP library.
 don't say that i need some stuf for free, but we need to make fs a very
 also we love freeswitch to run in a Embedded PC like the blackfin
 CPU/boards, Pjsip is very good for Embedded SIP Devicesawesome Open
 Source project
 With much love,
 DelphiWorld Information from ESET NOD32 Antivirus, version of virus
 signature database 4539 (20091024) __

 The message was checked by ESET NOD32 Antivirus.

 http://www.eset.com




 __ Information from ESET NOD32 Antivirus, version of virus
 signature database 4539 (20091024) __

 The message was checked by ESET NOD32 Antivirus.

 http://www.eset.com




 __ Information from ESET NOD32 Antivirus, version of virus signature 
 database 4539 (20091024) __

 The message was checked by ESET NOD32 Antivirus.

 http://www.eset.com



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 FreeSWITCH-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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Re: [Freeswitch-users] Mod_pjsip

2009-10-31 Thread Michael S Collins
I can guarantee that the FS devs are well aware of pj-sip. If it was/ 
is a viable alternative then it would be considered. The fact that it  
isn't being used is a pretty good indication that it isn't suitable  
for FS at this time.

-MV

Sent from my iPhone

On Oct 31, 2009, at 1:21 PM, Meftah Tayeb tayeb.mef...@gmail.com  
wrote:

Hi my FreeSWITCH friends
 Today we talked about FreeSWITCH libraries and we found pjsip, which  
 is
 a lot better than sofia, it can do everything sofia can do and more,
 it's thread-safe and performs well, I would love to see a mod_pjsip.
 The library is GPL but they also offer alternative licensing, don't  
 get
 me wrong, Sofia is great but I think FreeSWITCH will benefit from a
 thread-safe SIP library.
 don't say that i need some stuf for free, but we need to make fs a  
 very
 also we love freeswitch to run in a Embedded PC like the blackfin
 CPU/boards, Pjsip is very good for Embedded SIP Devicesawesome Open
 Source project
 With much love,
 DelphiWorld Information from ESET NOD32 Antivirus, version of  
 virus
 signature database 4539 (20091024) __

 The message was checked by ESET NOD32 Antivirus.

 http://www.eset.com




 __ Information from ESET NOD32 Antivirus, version of virus
 signature database 4539 (20091024) __

 The message was checked by ESET NOD32 Antivirus.

 http://www.eset.com




 __ Information from ESET NOD32 Antivirus, version of virus  
 signature database 4539 (20091024) __

 The message was checked by ESET NOD32 Antivirus.

 http://www.eset.com



 ___
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 FreeSWITCH-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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 users
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Re: [Freeswitch-users] Mod_pjsip

2009-10-31 Thread Stefan Knoblich
Michael S Collins wrote:
 I can guarantee that the FS devs are well aware of pj-sip. If it was/ 
 is a viable alternative then it would be considered. The fact that it  
 isn't being used is a pretty good indication that it isn't suitable  
 for FS at this time.
 
 -MV
 
 Sent from my iPhone

We already mentioned some of the reasons why it did get dropped 3 years ago
(first two points from memory, last two from old IRC logs):


[License incompatible (GPL), but i think tony tried to negotiate on alternate 
license terms]

Not possible to have multiple SIP profiles (due to global variables being used 
in the lib).

A race-condition under high load, that couldn't be resolved back then (with the 
help of the pjsip developers).

And the sofia module just working and surviving the scalability tests, so all 
efforts were focussed on
mod_sofia and pjsip got dropped.



stkn

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[Freeswitch-users] [mod_socket] Can't set channel variable SOMETIMES

2009-10-31 Thread Artem Shiyanov
Hello!

I've got strange problem:
In my app which talks to FreeSwitch via mod_socket there is such logic:
pre conditions: channel1 - newly created, parked channell - with help of
'uuid_create' and 'originate ... park'; channel2 is talking with channel3
for (channel in {array of channel1, channel2, channel3}) {
 set value for the channel variable X
}

uuid_transfer channel1 extension Y
uuid_transfer -both channel2 extension Y


In extension Y there is a logic based on the channel variable X.
The problem itself: every second launching falls, the reason - channel
variable X hasn't been set for one of channel1..3.
Sniffer shows that commands are sent (and confirmated by FreeSwitch)
consiquently, FreeSwitch sends OK for each command in time.
Temporary sollution that I found is to wait 0.5 sec after 'set' commands
have been sent and before transfer any channel to extension Y.

Maybe some already faced with similar problem?


Thanks,
Artem
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Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

2009-10-31 Thread Anthony Minessale
it something doing a read/write lock then not clearing it or its bad build
and you have to rebuild clean.
the trace had no session threads present in it.
look in the code of any non-standard mods you are using for something that
did either switch_ivr_originate or switch_core_session_locate and never
called switch_core_session_rwunlock

#define SWITCH_DEBUG_RWLOCKS at the top of switch.h
and make clean ; make install to get rwlock debugging.


On Sat, Oct 31, 2009 at 3:00 PM, Michael Giagnocavo m...@giagnocavo.netwrote:

 Which revision? Wasn't there some read-write lock issues on sessions
 checked in during the last week?

 FWIW, we do inline billing, but the message queue timeouts are very low.
 Upgrading to current SVN kills us (~15025 works fine) -- some channels stay
 in CS_EXECUTE, some in CS_REPORTING. We know it's not our code blocking,
 because there'd be transaction timeouts within 30 seconds, and these are
 hanging around for a long time.

 I suspect it's something with how we do our originates and state handlers.
 If we _overwrite_ the default originate handlers AND patch originate to
 blank out dialplan/context after duplicating the caller profile, it works
 fine. If we do the right thing and let the default handler set
 CS_CONSUME_MEDIA, then we get the calls hanging around forever. As I said, I
 suspect this is mismatch in what FS does and what our app assumes it does.

 -Michael

 -Original Message-
 From: freeswitch-users-boun...@lists.freeswitch.org [mailto:
 freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rupa Schomaker
 Sent: Saturday, October 31, 2009 1:28 PM
 To: freeswitch-users@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

 Nothing jumps out at me on this one.

 mod_cdr_csv and/or mod_xml_cdr should be fine.

 On Sat, Oct 31, 2009 at 1:24 PM, Seven Du dujinf...@gmail.com wrote:
  I also got some zombie channels, if someone can help me take a look
 that's
  really nice.
 
  http://pastebin.freeswitch.org/10912
 
  I only loaded mod_cdr_csv
  Is it ok to use mod_xml_cdr?
 
  Thanks.
 
 
  2009/11/1 Dome Charoenyost d...@tel.co.th
 
  I found bug in fscore_pb   $pwd should be $mypwd
 
  Now i post already. please help me check.
 
  Dome C.
 
 
  2009/11/1 João Mesquita jmesqu...@freeswitch.org:
   No, mod_nibblebill definetely needs to be enhanced but it is not the
   problem
   and it can be used with high load traffic.
  
   The one I am not sure about is odbc_query since it was not developed
 for
   that.
  
   Do what Rupa said, please.
  
   Regards,
  
   JM
  
   On Sat, Oct 31, 2009 at 3:11 PM, Brian West br...@freeswitch.org
   wrote:
  
   I think once you get the backtrace like rupa said we can see that
 maybe
   odbc_query is really hanging or something similar.
   /b
   On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:
  
   2009/10/31 Brian West br...@freeswitch.org:
  
   You should never do billing inline with the session thread is all I'm
  
   saying.
  
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 --
 -Rupa

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-- 
Anthony Minessale II

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Re: [Freeswitch-users] [mod_socket] Can't set channel variable SOMETIMES

2009-10-31 Thread Anthony Minessale
wait for execute_complete events to be sure you application execute request
was received and processed or use uuid_setvar instead which is instant.



On Sat, Oct 31, 2009 at 4:09 PM, Artem Shiyanov shiya...@gmail.com wrote:

 Hello!

 I've got strange problem:
 In my app which talks to FreeSwitch via mod_socket there is such logic:
 pre conditions: channel1 - newly created, parked channell - with help of
 'uuid_create' and 'originate ... park'; channel2 is talking with channel3
 for (channel in {array of channel1, channel2, channel3}) {
  set value for the channel variable X
 }

 uuid_transfer channel1 extension Y
 uuid_transfer -both channel2 extension Y


 In extension Y there is a logic based on the channel variable X.
 The problem itself: every second launching falls, the reason - channel
 variable X hasn't been set for one of channel1..3.
 Sniffer shows that commands are sent (and confirmated by FreeSwitch)
 consiquently, FreeSwitch sends OK for each command in time.
 Temporary sollution that I found is to wait 0.5 sec after 'set' commands
 have been sent and before transfer any channel to extension Y.

 Maybe some already faced with similar problem?


 Thanks,
 Artem


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Re: [Freeswitch-users] Mod_pjsip

2009-10-31 Thread Meftah Tayeb

hi,
Pjsip support ICE, STUN and TURN!
to STKN:
if you don't pjsip, please stop talking or exit the discution
we want to kype Freeswitch Clean and universal
Stefan Knoblich a écrit :

Michael S Collins wrote:
  
I can guarantee that the FS devs are well aware of pj-sip. If it was/ 
is a viable alternative then it would be considered. The fact that it  
isn't being used is a pretty good indication that it isn't suitable  
for FS at this time.


-MV

Sent from my iPhone



We already mentioned some of the reasons why it did get dropped 3 years ago
(first two points from memory, last two from old IRC logs):


[License incompatible (GPL), but i think tony tried to negotiate on alternate 
license terms]

Not possible to have multiple SIP profiles (due to global variables being used 
in the lib).

A race-condition under high load, that couldn't be resolved back then (with the 
help of the pjsip developers).

And the sofia module just working and surviving the scalability tests, so all 
efforts were focussed on
mod_sofia and pjsip got dropped.



stkn

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Re: [Freeswitch-users] Mod_pjsip

2009-10-31 Thread Stefan Knoblich
Meftah Tayeb wrote:
 hi,
 Pjsip support ICE, STUN and TURN!
 to STKN:
 if you don't pjsip, please stop talking or exit the discution
 we want to kype Freeswitch Clean and universal

Huh?

In which part did i tell people to stop doing whatever they want with pjsip?

In my last email or on IRC?

I'm merely pointing out the reasons why mod_pjsip was abandoned by the core 
team 3 years ago
(so some of the points may have been fixed in the meantime).

If somebody feels the need to resurrect it, well, please do.
But don't expect Tony or anyone else from the core team to do that for you.

This is OpenSource after all...


stkn

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Re: [Freeswitch-users] Mod_pjsip

2009-10-31 Thread Anthony Minessale
Meftah,

He is 100% correct.  Please do not insult my volunteer developers. Without
help from him you would not have any FreeSWITCH right now so please drop
this subject we are not using pjsip.



On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb tayeb.mef...@gmail.comwrote:

  hi,
 Pjsip support ICE, STUN and TURN!
 to STKN:
 if you don't pjsip, please stop talking or exit the discution
 we want to kype Freeswitch Clean and universal
 Stefan Knoblich a écrit :

 Michael S Collins wrote:


  I can guarantee that the FS devs are well aware of pj-sip. If it was/
 is a viable alternative then it would be considered. The fact that it
 isn't being used is a pretty good indication that it isn't suitable
 for FS at this time.

 -MV

 Sent from my iPhone



 We already mentioned some of the reasons why it did get dropped 3 years ago
 (first two points from memory, last two from old IRC logs):


 [License incompatible (GPL), but i think tony tried to negotiate on alternate 
 license terms]

 Not possible to have multiple SIP profiles (due to global variables being 
 used in the lib).

 A race-condition under high load, that couldn't be resolved back then (with 
 the help of the pjsip developers).

 And the sofia module just working and surviving the scalability tests, so 
 all efforts were focussed on
 mod_sofia and pjsip got dropped.



 stkn

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Re: [Freeswitch-users] Mod_pjsip

2009-10-31 Thread Meftah Tayeb

hi Anthony
i agry
i say that because STKN hate all my suggestions.
about pjsip, i will contribute aditional module in the contrib.
thanks Anthony
Anthony Minessale a écrit :

Meftah,

He is 100% correct.  Please do not insult my volunteer developers. 
Without help from him you would not have any FreeSWITCH right now so 
please drop this subject we are not using pjsip.




On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb tayeb.mef...@gmail.com 
mailto:tayeb.mef...@gmail.com wrote:


hi,
Pjsip support ICE, STUN and TURN!
to STKN:
if you don't pjsip, please stop talking or exit the discution
we want to kype Freeswitch Clean and universal
Stefan Knoblich a écrit :

Michael S Collins wrote:
  
I can guarantee that the FS devs are well aware of pj-sip. If it was/ 
is a viable alternative then it would be considered. The fact that it  
isn't being used is a pretty good indication that it isn't suitable  
for FS at this time.


-MV

Sent from my iPhone


We already mentioned some of the reasons why it did get dropped 3
years ago (first two points from memory, last two from old IRC
logs): [License incompatible (GPL), but i think tony tried to
negotiate on alternate license terms] Not possible to have
multiple SIP profiles (due to global variables being used in the
lib). A race-condition under high load, that couldn't be resolved
back then (with the help of the pjsip developers). And the sofia
module just working and surviving the scalability tests, so all
efforts were focussed on mod_sofia and pjsip got dropped. stkn
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sip:8...@conference.freeswitch.org 
mailto:sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888 
http://iax:gu...@conference.freeswitch.org/888
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Re: [Freeswitch-users] Mod_pjsip

2009-10-31 Thread Anthony Minessale
Meftah,
Feel free.

P.S.

STKN was the guy who made the first mod_pjsip for FS that we abandoned years
ago. So you should believe him.
Both him and I agreed it was not working out.  So if you don't believe me,
find out for yourself.




On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb tayeb.mef...@gmail.comwrote:

  hi Anthony
 i agry
 i say that because STKN hate all my suggestions.
 about pjsip, i will contribute aditional module in the contrib.
 thanks Anthony
 Anthony Minessale a écrit :

 Meftah,

 He is 100% correct.  Please do not insult my volunteer developers. Without
 help from him you would not have any FreeSWITCH right now so please drop
 this subject we are not using pjsip.



 On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb tayeb.mef...@gmail.comwrote:

 hi,
 Pjsip support ICE, STUN and TURN!
 to STKN:
 if you don't pjsip, please stop talking or exit the discution
 we want to kype Freeswitch Clean and universal
 Stefan Knoblich a écrit :

  Michael S Collins wrote:


  I can guarantee that the FS devs are well aware of pj-sip. If it was/
 is a viable alternative then it would be considered. The fact that it
 isn't being used is a pretty good indication that it isn't suitable
 for FS at this time.

 -MV

 Sent from my iPhone



 We already mentioned some of the reasons why it did get dropped 3 years ago
 (first two points from memory, last two from old IRC logs):


 [License incompatible (GPL), but i think tony tried to negotiate on alternate 
 license terms]

 Not possible to have multiple SIP profiles (due to global variables being 
 used in the lib).

 A race-condition under high load, that couldn't be resolved back then (with 
 the help of the pjsip developers).

 And the sofia module just working and surviving the scalability tests, so 
 all efforts were focussed on
 mod_sofia and pjsip got dropped.



 stkn

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Re: [Freeswitch-users] Mod_pjsip

2009-10-31 Thread Meftah Tayeb

hello anthony
we are realy sory about this subject
Ok, I apologize, I really love you and respect you, and I believe you
i will drop it from now.
Anthony Minessale a écrit :

Meftah,

He is 100% correct.  Please do not insult my volunteer developers. 
Without help from him you would not have any FreeSWITCH right now so 
please drop this subject we are not using pjsip.




On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb tayeb.mef...@gmail.com 
mailto:tayeb.mef...@gmail.com wrote:


hi,
Pjsip support ICE, STUN and TURN!
to STKN:
if you don't pjsip, please stop talking or exit the discution
we want to kype Freeswitch Clean and universal
Stefan Knoblich a écrit :

Michael S Collins wrote:
  
I can guarantee that the FS devs are well aware of pj-sip. If it was/ 
is a viable alternative then it would be considered. The fact that it  
isn't being used is a pretty good indication that it isn't suitable  
for FS at this time.


-MV

Sent from my iPhone


We already mentioned some of the reasons why it did get dropped 3
years ago (first two points from memory, last two from old IRC
logs): [License incompatible (GPL), but i think tony tried to
negotiate on alternate license terms] Not possible to have
multiple SIP profiles (due to global variables being used in the
lib). A race-condition under high load, that couldn't be resolved
back then (with the help of the pjsip developers). And the sofia
module just working and surviving the scalability tests, so all
efforts were focussed on mod_sofia and pjsip got dropped. stkn
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Re: [Freeswitch-users] Mod_pjsip

2009-10-31 Thread Meftah Tayeb

Anthony Minessale a écrit :

Meftah,
Feel free.



thanks



P.S.

STKN was the guy who made the first mod_pjsip for FS that we abandoned 
years ago. So you should believe him.
Both him and I agreed it was not working out.  So if you don't believe 
me, find out for yourself.

anthony, why i don't believe  you?

never say that.
i believe you and all Freeswitch Staf and thank you and to all 
Freeswitch Staf.




On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb tayeb.mef...@gmail.com 
mailto:tayeb.mef...@gmail.com wrote:


hi Anthony
i agry
i say that because STKN hate all my suggestions.
about pjsip, i will contribute aditional module in the contrib.
thanks Anthony
Anthony Minessale a écrit :

Meftah,

He is 100% correct.  Please do not insult my volunteer
developers. Without help from him you would not have any
FreeSWITCH right now so please drop this subject we are not using
pjsip.



On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb
tayeb.mef...@gmail.com mailto:tayeb.mef...@gmail.com wrote:

hi,
Pjsip support ICE, STUN and TURN!
to STKN:
if you don't pjsip, please stop talking or exit the discution
we want to kype Freeswitch Clean and universal
Stefan Knoblich a écrit :

Michael S Collins wrote:
  
I can guarantee that the FS devs are well aware of pj-sip. If it was/ 
is a viable alternative then it would be considered. The fact that it  
isn't being used is a pretty good indication that it isn't suitable  
for FS at this time.


-MV

Sent from my iPhone


We already mentioned some of the reasons why it did get
dropped 3 years ago (first two points from memory, last two
from old IRC logs): [License incompatible (GPL), but i think
tony tried to negotiate on alternate license terms] Not
possible to have multiple SIP profiles (due to global
variables being used in the lib). A race-condition under
high load, that couldn't be resolved back then (with the
help of the pjsip developers). And the sofia module just
working and surviving the scalability tests, so all efforts
were focussed on mod_sofia and pjsip got dropped. stkn
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Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

2009-10-31 Thread Dome Charoenyost
Yes. i user api_hangup_hook for do callback.
action application=set data=api_hangup_hook=originate
{origination_caller_id_number=${caller_id_number},callback_uuid=${uuid},ignore_early_media=true}${route_1}|${route_2}|${route_3}
call_ivr${caller_id_number} XML public /

may be need originate_timeout

Dome C.


2009/11/1 Rupa Schomaker r...@rupa.com:
 fscore_pb has been updated.

 Next time put in the pastebin url.  This one was 10911.

 Ok, a suspicious thread looks below.  If I read it right, you are
 doing an originate from within the hangup handler.  That doesn't look
 normal or sane to me.

 Thread 41 (process 30732):
 #0  0xb7a5e007 in select () from /lib/libc.so.6
 #1  0xb7dde39f in apr_sleep (t=1000) at time/unix/time.c:246
 #2  0xb7db12c0 in do_sleep (t=1000) at src/switch_time.c:109
 #3  0xb7db15ac in switch_cond_next () at src/switch_time.c:196
 #4  0xb7d862b8 in switch_ivr_originate (session=0x0, bleg=0xb48c2f00,
 cause=0xb48c2ed4, bridgeto=0xb1133340
 {origination_caller_id_number=0857151465,callback_uuid=c32f54cc-c6b8-11de-af77-774e15238dbe,ignore_early_media=true}sofia/external/66857151...@115.178.62.191:7050|sofia/external/66857151...@115.178.62...,
 timelimit_sec=60, table=0x0, cid_name_override=0xb7e8c976
 FreeSWITCH, cid_num_override=0x90ec0800 0857151465,
 caller_profile_override=0x0, ovars=0x0, flags=0) at
 src/switch_ivr_originate.c:2132
 #5  0xb6f0def1 in originate_function (cmd=0x90164ce2
 {origination_caller_id_number=0857151465,callback_uuid=c32f54cc-c6b8-11de-af77-774e15238dbe,ignore_early_media=true}sofia/external/66857151...@115.178.62.191:7050|sofia/external/66857151...@115.178.62...,
 session=0x0, stream=0xb48c3004) at mod_commands.c:2413
 #6  0xb7d66fb1 in switch_api_execute (cmd=0x90164cd8 originate,
 arg=0x90164ce2 
 {origination_caller_id_number=0857151465,callback_uuid=c32f54cc-c6b8-11de-af77-774e15238dbe,ignore_early_media=true}sofia/external/66857151...@115.178.62.191:7050|sofia/external/66857151...@115.178.62...,
 session=0x0, stream=0xb48c3004) at src/switch_loadable_module.c:1567
 #7  0xb7d53c4a in switch_core_session_hangup_state
 (session=0x958ab678) at src/switch_core_state_machine.c:506
 #8  0xb7d385e6 in switch_channel_perform_hangup (channel=0x958afd70,
 file=0xb6af8788 mod_dptools.c, func=0xb6af8ba8 hangup_function,
 line=637, hangup_cause=SWITCH_CAUSE_NORMAL_CLEARING) at
 src/switch_channel.c:1898
 #9  0xb6aee906 in hangup_function (session=0x958ab678, data=0xb7e82ea5
 ) at mod_dptools.c:637
 #10 0xb7d4fe1f in switch_core_session_exec (session=0x958ab678,
 application_interface=0xb6e10ec0, arg=0xb7e82ea5 ) at
 src/switch_core_session.c:1552
 #11 0xb7d4f73e in switch_core_session_execute_application
 (session=0x958ab678, app=0x90163a80 hangup, arg=0xb7e82ea5 ) at
 src/switch_core_session.c:1474
 #12 0xb7d51131 in switch_core_standard_on_execute (session=0x958ab678)
 at src/switch_core_state_machine.c:166
 #13 0xb7d5215a in switch_core_session_run (session=0x958ab678) at
 src/switch_core_state_machine.c:340
 #14 0xb7d4e199 in switch_core_session_thread (thread=0x9020bd98,
 obj=0x958ab678) at src/switch_core_session.c:1135
 #15 0xb7ddc2fc in dummy_worker (opaque=0x9020bd98) at
 threadproc/unix/thread.c:138
 #16 0xb7caaf3b in start_thread () from /lib/libpthread.so.0
 #17 0xb7a64bee in clone () from /lib/libc.so.6


 On Sat, Oct 31, 2009 at 12:34 PM, Dome Charoenyost d...@tel.co.th wrote:
 I found bug in fscore_pb   $pwd should be $mypwd

 Now i post already. please help me check.

 Dome C.


 2009/11/1 João Mesquita jmesqu...@freeswitch.org:
 No, mod_nibblebill definetely needs to be enhanced but it is not the problem
 and it can be used with high load traffic.

 The one I am not sure about is odbc_query since it was not developed for
 that.

 Do what Rupa said, please.

 Regards,

 JM

 On Sat, Oct 31, 2009 at 3:11 PM, Brian West br...@freeswitch.org wrote:

 I think once you get the backtrace like rupa said we can see that maybe
 odbc_query is really hanging or something similar.
 /b
 On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:

 2009/10/31 Brian West br...@freeswitch.org:

 You should never do billing inline with the session thread is all I'm

 saying.

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Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

2009-10-31 Thread Anthony Minessale
Use bgapi originate ...

On Oct 31, 2009 9:09 PM, Dome Charoenyost d...@tel.co.th wrote:

Yes. i user api_hangup_hook for do callback.
action application=set data=api_hangup_hook=originate
{origination_caller_id_number=${caller_id_number},callback_uuid=${uuid},ignore_early_media=true}${route_1}|${route_2}|${route_3}
call_ivr${caller_id_number} XML public /

may be need originate_timeout

Dome C.


2009/11/1 Rupa Schomaker r...@rupa.com:

 fscore_pb has been updated.   Next time put in the pastebin url.  This
one was 10911.   Ok, a ...
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Re: [Freeswitch-users] Mod_pjsip

2009-10-31 Thread Chris Burns
My favorite part of this 'civilized' discussion on IRC was when DelphiWord and 
diegoviola sat around tryin to take the piss outta stkn on this issue for 
seemingly no reason. Thanks for making the channel a cool place, guys ;)

On October 31, 2009 07:32:03 pm Meftah Tayeb wrote:
 Anthony Minessale a écrit :
  Meftah,
  Feel free.
 
  thanks
 
  P.S.
 
  STKN was the guy who made the first mod_pjsip for FS that we abandoned
  years ago. So you should believe him.
  Both him and I agreed it was not working out.  So if you don't believe
  me, find out for yourself.
  anthony, why i don't believe  you?

 never say that.
 i believe you and all Freeswitch Staf and thank you and to all
 Freeswitch Staf.

  On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb tayeb.mef...@gmail.com
  mailto:tayeb.mef...@gmail.com wrote:
 
  hi Anthony
  i agry
  i say that because STKN hate all my suggestions.
  about pjsip, i will contribute aditional module in the contrib.
  thanks Anthony
 
  Anthony Minessale a écrit :
  Meftah,
 
  He is 100% correct.  Please do not insult my volunteer
  developers. Without help from him you would not have any
  FreeSWITCH right now so please drop this subject we are not using
  pjsip.
 
 
 
  On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb
  tayeb.mef...@gmail.com mailto:tayeb.mef...@gmail.com wrote:
 
  hi,
  Pjsip support ICE, STUN and TURN!
  to STKN:
  if you don't pjsip, please stop talking or exit the discution
  we want to kype Freeswitch Clean and universal
 
  Stefan Knoblich a écrit :
  Michael S Collins wrote:
  I can guarantee that the FS devs are well aware of pj-sip. If
  it was/ is a viable alternative then it would be considered. The fact
  that it isn't being used is a pretty good indication that it isn't
  suitable for FS at this time.
 
  -MV
 
  Sent from my iPhone
 
  We already mentioned some of the reasons why it did get
  dropped 3 years ago (first two points from memory, last two
  from old IRC logs): [License incompatible (GPL), but i think
  tony tried to negotiate on alternate license terms] Not
  possible to have multiple SIP profiles (due to global
  variables being used in the lib). A race-condition under
  high load, that couldn't be resolved back then (with the
  help of the pjsip developers). And the sofia module just
  working and surviving the scalability tests, so all efforts
  were focussed on mod_sofia and pjsip got dropped. stkn
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Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

2009-10-31 Thread Dome Charoenyost
How to use bgapi in my flow.
1. user call did
2. FS send ringing and check balance, LCR from DB (by mod_odbc_quey)
3. Hangup (by use or timeout)
4. FS callback to user and bridge to IVR

I'm not sure  bgapi can do after channel hangup.

Dome C.


2009/11/1 Anthony Minessale anthony.miness...@gmail.com:
 Use bgapi originate ...

 On Oct 31, 2009 9:09 PM, Dome Charoenyost d...@tel.co.th wrote:

 Yes. i user api_hangup_hook for do callback.
 action application=set data=api_hangup_hook=originate
 {origination_caller_id_number=${caller_id_number},callback_uuid=${uuid},ignore_early_media=true}${route_1}|${route_2}|${route_3}
 call_ivr${caller_id_number} XML public /

 may be need originate_timeout

 Dome C.


 2009/11/1 Rupa Schomaker r...@rupa.com:

 fscore_pb has been updated.   Next time put in the pastebin url.  This
 one was 10911.   Ok, a ...

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[Freeswitch-users] SIP Proxy with direct media path

2009-10-31 Thread Dmitry Gromov
Hi!

I was a long time user of MySipSwitch service (www.mysipswitch.com) but it
looks like they are planning to close their service. I can't use the
replacement they have and would actually prefer some locally installed
solution.

Question - is it possible to use FreeSwitch exactly like MySipSwitch - just
SIP Proxy with very flexible dial plan?
With MySipSwitch it is possible to:
- register service with multiple SIP providers as client
- register your ATA with MySipSwitch
- create smart dial plan (in Ruby) to route calls based on prefixes and/or
call source and some other logic
- accept and route calls from all registered providers

MySipSwitch does not have any media capabilities - RTP is always established
directly between call parties (no re-Invites either).

If this is possible - is there any beginner guide for this?

Also - I know that FreeSwitch could run in Windows as well as in Linux -
would it be possible to use the above functionality in both OSes?

Thanks,
Dmitry

-- 
DG
NJ
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Re: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.5pre3 is available for testing!

2009-10-31 Thread Michael Jerris
We still do plan on branching 1.0 into bugfix only.  This has not yet  
happened but may happen at some point after 1.0.5.  In the mean time,  
the vast majority of the work lately has been fixes with small feature  
improvements, most all of this would stay in a 1.0 branch even if we  
were already branched.  Trunk remains for the most part very stable  
with new features mostly coming in new modules.


Mike

On Oct 28, 2009, at 8:40 PM, Craig Askings wrote:

Is the plan to run 1.0.5 as a stable branch with bug fix updates or  
will it be the case of just follow the trunk like it is with any  
problems you encounter in 1.0.4?


2009/10/29 Michael Collins m...@freeswitch.org
Hello FreeSWITCHers,

The latest FreeSWITCH version is now available for download on the  
files site. The announcement story is in the main FreeSWITCH site.  
Please download and test, and then test some more. We need your  
feedback. The sooner we get your feedback, the more quickly we can  
roll the official 1.0.5.


--
Craig Askings

Network Engineer | Over the Wire Pty Ltd
cr...@overthewire.com.au | www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365
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Re: [Freeswitch-users] SIP Proxy with direct media path

2009-10-31 Thread Jason White
 With MySipSwitch it is possible to:
 - register service with multiple SIP providers as client

This is easy once you understand the FreeSWITCH XML configuration.
 - register your ATA with MySipSwitch

That's really a special case of the above.
 - create smart dial plan (in Ruby) to route calls based on prefixes and/or
 call source and some other logic

FreeSWITCH can do better than that: there are lots of variables which can be
matched by Perl-compatible regular expressions, including the destination
number.
 - accept and route calls from all registered providers

That's handled in the same way as the previous item.
 
 MySipSwitch does not have any media capabilities - RTP is always established
 directly between call parties (no re-Invites either).

FreeSWITCH gives you a choice: you can use proxy media or bypass media modes
if you don't want it to handle the media.
 
 If this is possible - is there any beginner guide for this?

http://wiki.freeswitch.org/ is the best there is.

It is being improved by the community over time.

You can also take advantage of the IRC channel, the FreeSWITCH conference and
of course the mailing list.


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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179

2009-10-31 Thread Michael Collins
How would you do an expression like: if $x  24 in a condition tag? Just
curious. I would like to make sure that is properly documented.
-MC

On Sat, Oct 31, 2009 at 10:42 AM, Michael Jerris m...@jerris.com wrote:

 You actually can use these in conditions.  Just need to be careful that the
 var you are conditioning on is already set.

 Mike

 On Oct 22, 2009, at 1:54 PM, Michael Collins wrote:



 On Thu, Oct 22, 2009 at 5:51 AM, Rupa Schomaker r...@rupa.com wrote:

 cond would be helpful here?  I updated the wiki on this one just now
 with a bit more detail.  It is a api call. so, you'd use it like:

 ${cond(eval ? trueval : falseval)}

 so to get a value of ERR if the var my myvar is  15 you could:

 ${cond(${myvar}  15 ? ERR : OK)}

 If both sides of the comparison operator are numeric then it does
 numeric comparison otherwise it does lexical string comparison.


 Rupa,

 Yes, you can do the set/cond API trick but you can only do it in the action
 or anti-action tags, not in the condition tags. I'm sure you know that but I
 want all those reading this thread to make the connection.
 -MC



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Re: [Freeswitch-users] Setting up Conference with Moderator

2009-10-31 Thread Michael Jerris
Have you answered the call?

On Oct 30, 2009, at 11:34 AM, Rob Forman wrote:

 Hm, strange.  I haven't seen that before.  Can you pastebin your logs
 at debug level?

 On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote:

 It's strange... a tcpdump tells me that there is no DTMF from my
 provider when using IVR, but when I call into a TN that goes
 directly into the Conference App, I see DTMF from the provider.



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Re: [Freeswitch-users] mod_t38gateway

2009-10-31 Thread Michael Jerris
This is a non working module, just a shell for development.

Mike

On Oct 30, 2009, at 5:52 PM, Tihomir Culjaga wrote:

 does anybody know how does it work and how to use it in a dialplan?



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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179

2009-10-31 Thread Michael Jerris

see rupa's explanation below.


On Nov 1, 2009, at 1:24 AM, Michael Collins wrote:

How would you do an expression like: if $x  24 in a condition tag?  
Just curious. I would like to make sure that is properly documented.

-MC





On Thu, Oct 22, 2009 at 5:51 AM, Rupa Schomaker r...@rupa.com  
wrote:

${cond(${myvar}  15 ? ERR : OK)}

If both sides of the comparison operator are numeric then it does
numeric comparison otherwise it does lexical string comparison.




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