Hi all,
I have opened JIRA for the same.
http://jira.freeswitch.org/browse/FSCORE-285
One system is Fedora, and another one is Ubuntu. Although fs 1.0.1 was also
get crashed in Ubuntu many times.
Now 1.0.3.RC1 is loaded on Ubuntu, so this happen again with my Ubuntu I
will surely post it.
And
Hello,
I want my caller to detect hangup cause as 34 so that he can try next
provider according to lcr.
Here is my setup.
Caller - switch (fs) - Terminator.
Now when terminator sends 503 Service Unavailable, I want to override this
cause, so that the caller gets hangup case 34. (According to the
Hi all
My set up is : When i originate the call from CLI, using originate command,
It comes in a dialplan and from there it goes to javascript for handling
simple IVR.
In IVR I tested both streamFile and dialplan read application to get the
DTMF from user, but it was not working, it just play
Thanks Moises. It looks like good work. When is Sangoma coming out with a
similar product ... Doug told me it was in the works, then not in the works,
then back in the works ...
The problem is this particular card is PCI only and it will only do 120
channels
Thanks!
SDR
Ok,
It's now working as expected, looks like I had something odd set in the
phone's peer definition. I'll try and back track to see what I was
doing wrong, but what ever it was, was preventing the caller id from
being sent in the INVITE.
Regards
Shelby Ramsey wrote:
Thanks Moises. It looks like good work. When is Sangoma coming out
with a similar product ... Doug told me it was in the works, then not
in the works, then back in the works ...
The problem is this particular card is PCI only and it will only do
120 channels
If
I think we have some trouble surviving issues.
So when everything is ok we do fine but if something goes wrong we don't
recover.
We are still missing state timers in the q931.
maybe you can use your new pcap thing to see what goes wrong =D
On Fri, Feb 6, 2009 at 6:32 AM, Helmut Kuper
Hi Guys
I'm looking for some pointers on how to collect CDR's and store in
mysql. Is there anything built in yet?
I can rate the calls as a batch process, I simply need the call data.
Regards
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Hello,
since yesterday I do a real life test with FS in a 40 sip extension
environment with TDM connection via wanpipe/sangoma A104d.
I detected two times the problem, that openzap stops working, while SIP
calls worked. Only restarting helped. Maybe reloading of mod_openzap
helps as well, but I
Steve,
You definitely have a better grasp on this topic than me. But I think it's
a tough sell on the host based processing ... when you look at products like
what audio codes can do on a card (3 DS-3's worth of transcoding) ... but I
have had a couple of soft switch vendors claim though that
Nik,
There are a bunch of ways to do this ... mod_xml_cdr posts to a url then you
can parse and dump ... or you can use mod_cdr_csv which allows you to
dictate exactly what you want to collect and then parse the file and dump
into mysql.
There are also a couple of examples here --
Thanks,
I was confused because I saw that mod cdr had been dropped. Due
anticipated call volumes, batch processing is ideal, it keeps any MySql
load issues away from FS
Regards,
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
Is mod_cdr_xml asynchronous ... by that I mean .. if there is a communication
failure how
does this effect load on the system or the call for that matter...
it wouldn’t hold up the progress or delay the turn up of the call or anything
would it?
I understand it would write to the error directory
Mod_xml_cdr will drop a file to the file system on failure to post. You can
also leverage this drop a file to the file system and run a CDR processor
locally.
We handle call rates in the 500+ range using the local file system as a
caching mechanism and a simple PHP script to rate the CDRs and
Yes this will be normal due to buffering. Have you tested svn trunk?
/b
On Feb 6, 2009, at 12:59 PM, Dan wrote:
Hi,
With the 1.0.2 release i was able to to stream a call using
mod_shout to an icecast server with only a 1 or two second delay to
clients. With the current trunk that delay
Hi,
I came across a company who was selling hardware which could do like
5000 G729 conversion simultanoeusly, I was like this sounds cool, they
have support for asterisk, I haven't enquirer yet how they do this,
but anyone wishes to buy it cost £15000/year for the hrdware + support
Any one
PER YEAR? ARE THEY DAFT?
/b
On Feb 6, 2009, at 1:27 PM, Mitul Limbani wrote:
it cost £15000/year for the hrdware + support
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I have, do you know what would have changed between 1.0.2 and trunk that would
cause the buffer to change? Also if its not in mod_shout.c (which I copied from
1.0.2 to trunk for testing with no luck), where else would fs be buffering? One
thing I have noticed is that in 1.0.2 as soon as the
Let me clarify.. yes this is normal file buffering was added so we
wouldn't thrash your hard drive with tiny bits of data when recording
calls so now it buffers and writes larger chunks to disk. This is why
you have this delay which is 100% normal is realtime a critical
thing? It is
For me it is. For what I'm using it for I can tolerate around a second or two
delay. I have the icecast server setup to only buffer 1K for their on-connect
burst as well as my flash/flex player to only buffer 1k (yes I might as well
not buffer at all, which I may end up doing). In 1.0.2 this
edit switch_ivr_play_say.c line 423
comment the line out and recompile.
Tell me if it helps you and i will consider making it configurable.
On Fri, Feb 6, 2009 at 2:01 PM, freeswitch-us...@digitaldan.com wrote:
For me it is. For what I'm using it for I can tolerate around a second or
two
Nik Middleton wrote:
Hi Guys
I’m looking for some pointers on how to collect CDR’s and store in
mysql. Is there anything built in yet?
I can rate the calls as a batch process, I simply need the call data.
Regards
On Sat, Feb 7, 2009 at 1:30 AM, Michael Giagnocavo m...@giagnocavo.netwrote:
$22K would buy quite a few machines with many core Xeons. I just don't see
how it'd be effective at that price. Not to mention a yearly figure.
G729 is roughly 25 MIPS (encode+decode), coppice, please correct as
What about using a radius server, would that be more resilient?
Regards
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Adam
Long
Sent: 06 February 2009 17:31
To:
Thanks,
We appreciate the positive feedback!
if you revert the change I suggested and update i added a new variable
enable_file_write_buffering=false
set this variable on the channel before you start recording it with the set
application or in the dialstring in {}
on outbound calls and it
So you're simply posting this file to a web server? How do you find the load on
it at this rate of calls?
BTW can anyone point me to resources discussing how to do this? (Using a web
server to post data to a db) I've not this sort of thing before, and I'm not
too sure what I should be goggling
BODY { font-family:Arial, Helvetica, sans-serif;font-size:12px;
}Hello guys,
Sorry it was my mistake, i re-read the entire proposal and it looks
like their specialized 1U Hardware with custom CPU can handle close to
1500 simultaneous G729 encoding n transmission, also this hardware
offers from
On line 424 I think it needs to be changed from
if (!vval || !switch_true(vval)) {
to
if (!vval || switch_true(vval)) {
Other wise it works, thanks!
- Original Message -
From: Anthony Minessale anthony.miness...@gmail.com
To: freeswitch-users@lists.freeswitch.org
Sent: Friday,
P.S. - Yes, yes I know AstLinux isn't the best name for a distro
with FreeSWITCH. Depending on my success here I have some other
ideas...
How about KickAstLinux? ;)
-MC
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This is built into mod_xml_cdr and should be covered on its wiki page...
I personally don't post the records to a web server as I think that's too
much over head at this load... I use a script to scrape the directory and
process the CDRs...
This gives me the FileSystem as a buffer, and allows me
I'm doing this on low volume pbx setup, so posting to the web server is
fine with my load. If you are doing high load, then definitely write to
files and batch process them.
On 2/6/2009 3:07 PM, Nik Middleton wrote:
So you're simply posting this file to a web server? How do you find
the load
Just out of curiosity ... You actually set the values in xml_cdr_conf.xml to
an invalid value ... and then FS tries it and then dumps it into the
err_dir?
Nik,
Just configure the xml_conf_cdr and it will post all of the channel
variables to your web server ... you can look at the variables and
Nope you comment out that line and it wont even attempt to post and will
drop it into the log/xml_cdr directory
From: Shelby Ramsey sicfsl...@gmail.com
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Fri, 6 Feb 2009 15:30:13 -0600
To: freeswitch-users@lists.freeswitch.org
Subject: Re:
Hello,
could this option be used to lower I/O load - to rather write more bytes
at once rather than one by one - on file recording (record_session)?
Regards,
Tamas
Anthony Minessale írta:
Thanks,
We appreciate the positive feedback!
if you revert the change I suggested and update i
if you revert the change I suggested and update i added a new variable
enable_file_write_buffering=false
set this variable on the channel before you start recording it with the set
application or in the dialstring in {}
on outbound calls and it should skip the buffering.
Could you test it
Sure “Custom CPU” isn’t just a 4 socket Intel setup?
-Michael
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Mitul
Limbani
Sent: Friday, February 06, 2009 4:02 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re:
Even better ... Thanks Ken!
SDR
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Done!
- Original Message -
From: Michael Collins m...@freeswitch.org
To: freeswitch-users@lists.freeswitch.org
Sent: Friday, February 6, 2009 2:36:39 PM GMT -07:00 US/Canada Mountain
Subject: Re: [Freeswitch-users] mod_shout delay in trunk
if you revert the change I suggested and
Done!
Many thanks!
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Guy's,
Thanks for all the responses; it's truly refreshing to get so much
valuable input. I'm reading the docs furiously, but I still don't know
what I don't know yet. But given time I will return the favor to those
that come later.
Regards
Thanks for all the responses; it's truly refreshing to get so much valuable
input. I'm reading the docs furiously, but I still don't know what I don't
know yet. But given time I will return the favor to those that come later.
Sounds good! If you feel up to doing any wiki documentation
Hey Folks:
For a FS box that's generally handling higher amounts of
inbound/outbound call traffic (say 500 - 700 calls) and registrations
(30 - 50 per second), is it recommended to split off the signaling and
media traffic onto separate NICs for performance reasons? Or is it
better to split all
If the nic's have their own bus you could do that to improve network
performance of sip signaling and media... now the neat thing would be
to have an option to have say three network interfaces and have
FreeSWITCH round robin them per call. I smell a bounty.
/b
On Feb 6, 2009, at 5:19 PM,
Question regarding the xml cdr's
Let's say I have a cron job looking at these files and processing them.
How does FS create them. Does a MV occur from some other DIR, as
otherwise it's possible I might try and open an in progress record.
No worries - the file isn't available until it's
Bonding? Intel ANS?
On Fri, Feb 6, 2009 at 6:28 PM, Brian West br...@freeswitch.org wrote:
If the nic's have their own bus you could do that to improve network
performance of sip signaling and media... now the neat thing would be
to have an option to have say three network interfaces and have
That would work too I suspect.
/b
On Feb 6, 2009, at 6:00 PM, Kristian Kielhofner wrote:
Bonding? Intel ANS?
On Fri, Feb 6, 2009 at 6:28 PM, Brian West br...@freeswitch.org
wrote:
If the nic's have their own bus you could do that to improve network
performance of sip signaling and
Wouldn't be a huge deal if each card has a dedicated bus then it
wouldn't be fighting for bandwidth... its still going to be hitting a
limit at some point but you might get more milage out of it.
/b
On Feb 6, 2009, at 6:06 PM, Kristian Kielhofner wrote:
I think the big problem is still
Hello all,
Can anyone please let me know how I might be able to configure the voice
mail prompts and their playback speed?
Thanks,
Maxim.
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Not sure what you mean by playback speed. All the prompts for
voicemail are defined in the phrase macros in the configuration.
Mike
On Feb 6, 2009, at 7:32 PM, Maxim Karp mk...@securesilence.com
wrote:
Hello all,
Can anyone please let me know how I might be able to configure the
On Feb 6, 2009, at 6:27 PM, Michael Jerris m...@jerris.com wrote:
Not sure what you mean by playback speed. All the prompts for
voicemail are defined in the phrase macros in the configuration.
Check out conf/lang/en/vm/sounds.xml
-MC
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