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
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
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
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
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.
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
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
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.
___
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
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
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
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
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
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
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
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
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
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
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
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
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
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.
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
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
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
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
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
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
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
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
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,
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
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
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
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
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,
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
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
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
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
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
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
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
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
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
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
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
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
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
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?
___
FreeSWITCH-users mailing list
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
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