(IRC nick: mercutioviz)
Sent from my iPhone
On Feb 2, 2009, at 3:35 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Hi Guys,
As a long time Asterisk user, I'm looking into freeswitch as an
alternative mainly due to (list multiple reasons here
Newbie with FS, currently have Asterisk servers front ended by Openser
Question: I have around 400 sip remote clients, if I were to deploy FS,
do I need Openser? Is there any advantage in retaining Openser?
Regards
___
Freeswitch-users mailing list
channel)
Are you using TDM cards for this? Just curious.
-MC (IRC nick: mercutioviz)
Sent from my iPhone
On Feb 2, 2009, at 3:35 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Hi Guys,
As a long time Asterisk user, I'm looking into freeswitch
...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 03 February 2009 17:08
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] OPenser - FS Do I need this?
On Tue, Feb 3, 2009 at 8:20 AM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Newbie with FS, currently have Asterisk
level do you call 'high volume'... What I call high
volume
is a telemarketer running at 2500 calls/sec and peak concurrent channel
usage in the 10,000 to 15,000 channel range
K
From: Nik Middleton nik.middle...@noblesolutions.co.uk
Subject: Re: [Freeswitch-users] OPenser - FS Do I need
Hi Guys,
Excuse my ignorance, but I'm just starting with FS.
I've loaded FS onto one of our servers in a datacenter. I'm registering
with our PSTN breakout provider just fine, but I'm a little confused
about internal/external.
Given that we have no internal clients, as they're all
Hi Guys,
Need a little help here; I connect to my PSTN provider via the LAN,
Question: As the provider authenticates on IP, how do I not send a
password? In the .xml file if I remove the password entry it complains
Secondly, the contact should be my local address, not the public one.
, at 4:30 PM, Nik Middleton wrote:
Hi Guys,
Excuse my ignorance, but I'm just starting with FS.
I've loaded FS onto one of our servers in a datacenter. I'm
registering with our PSTN breakout provider just fine, but I'm a
little confused about internal/external.
Given
Hi Guys,
Need a little help here; I connect to my PSTN provider via the LAN,
Question: As the provider authenticates on IP, how do I not send a
password? In the .xml file if I remove the password entry it complains
Secondly, the contact should be my local address, not the public one.
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Gateway setting
Its ok ;) We'll get you taken care of.. you should join us on IRC...
#freenode its a faster way to get help. irc.freenode.net
/b
On Feb 4, 2009, at 4:54 PM, Nik Middleton wrote:
Sorry, 2
I'm looking at this can you post your full gateway and
dialplan for us to see?
/b
On Feb 5, 2009, at 4:43 PM, Nik Middleton wrote:
No good, I tried
action application=set
data=effective_caller_id_number=0753960/
action application=export data=effective_caller_id_number/
action
: Re: [Freeswitch-users] Caller ID not being passed
I notice you're using 1.0.2 any way you can test this with 1.0.3 RC1
tarball?
/b
On Feb 5, 2009, at 5:12 PM, Nik Middleton wrote:
Dial plan is as per default setup with the addition of the following. To
be honest, and I'm no SIP guru
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Nik
Middleton
Sent: 05 February 2009 23:26
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Caller ID not being passed
Yes, I'll report back tomorrow
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
___
Freeswitch-users mailing list
I use along with a rails app.
Remember that if you do real time, you also need to periodically scrape
the error directory and load those (mod_cdr_xml will save to error if it
can't successfully post to your script).
On 2/6/2009 10:09 AM, Nik Middleton wrote:
Hi Guys
I'm looking for some
app.
Remember that if you do real time, you also need to periodically scrape
the error directory and load those (mod_cdr_xml will save to error if it
can't successfully post to your script).
On 2/6/2009 10:09 AM, Nik Middleton wrote:
Hi Guys
I'm looking for some pointers on how to collect
the error directory and load those (mod_cdr_xml will save
to error if it
can't successfully post to your script).
On 2/6/2009 10:09 AM,
Nik Middleton wrote:
Hi Guys
I¹m looking for some pointers on
how to collect CDR¹s and store in
mysql. Is there anything built in yet?
I can rate
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
Great, thanks for that.
One of the big issues with Asterisk's way of billing is that if let's
say a remote phone diverts a call to another number, say a mobile,
because a local channel is created for the redirect, Asterisk loses
critical information such as the account code and therefore cannot
Hi Guys,
Is there any form of Answer phone detection in FS? A search hasn't
really brought up anything
Regards,
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
Hi Guys,
I'm placing calls ok by using the event socket. However, I need to
modify the To: Sip header prior to the call going out for routing
purposes. Is it possible to do this in the Originate action?
If not, can someone explain if it's possible to trigger part of the dial
plan
Hi Guys,
I'm having some issues passing an argument to an lua script.
Basically in an originate command I pass the name of a .wav file
If I hard code the file session:streamFile(myfile.wav]);
It works,
However, using this:
session:streamFile(argv[1]);
causes this error
Hi Guys
I want to access Mysql 5 from lua. The wiki is not too clear on this.
Do I need to install lua and lua mysql?
Regards
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
.
-MC
Sent from my iPhone
On Feb 8, 2009, at 2:31 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Hi Guys,
I'm having some issues passing an argument to an lua script.
Basically in an originate command I pass the name of a .wav file
@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problems passing arguments to lua
Looks like you put a , instead of a space when calling the script.
/b
On Feb 8, 2009, at 6:21 PM, Nik Middleton wrote:
cannot open /usr/local/freeswitch/scripts/helloworld.lua,myfile.wav
In the absence of any directives on lua/mysql, is it possible to launch
a PHP script from lua? All I need to do is pass some data to update a
db record. I don't need to have any links to the call as I intend to
call is in the hang-up callback
Regards,
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Making a system call with LUA
On Mon, 2009-02-09 at 13:30 +, Nik Middleton wrote:
In the absence of any directives on lua/mysql, is it possible to
launch a PHP script from lua? All I need to do is pass some data to
update
Hi Guys,
I have an IVR that's working fine on internal extensions, but when a
call is via my sip GW, they're not being trapped.
I have tried the following in the gw profile
param name=dtmf-type value=rfc2833/
param name=rfc2833-pt value=101/
param name=pass-rfc2833 value=false/
I
Further to this message, DTMF works with PMCU but not with PMCA which is
the native format for this sip provider.
Regards
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Nik
Collins
Sent: 09 February 2009 21:27
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] DTMF not being recognised
On Mon, Feb 9, 2009 at 12:21 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Further to this message, DTMF works with PMCU but not with PMCA which
Hi Guys,
I'm baffled by this error. I'm updating the db on call hang-up If I
comment out curs:close() no error, but I'm concerned about memory leaks.
Can anyone tell me what FS is complaining about?
The db gets updated in both cases
Regards
require luasql.mysql
function
update is going to return an
integer (rows affected) or boolean depending on the which server you use
since no recordset is actually requested.
--- On Tue, 2/10/09, Nik Middleton nik.middle...@noblesolutions.co.uk
wrote:
From: Nik Middleton nik.middle...@noblesolutions.co.uk
Subject
I have a situation where FS aborts
I'm running an lua script with mysql statements
First time it runs, on hangup I get
[CONSOLE] switch_core_memory.c:374 switch_core_memory_reclaim()
Returning 4 recycled memory pool(s)
If I run it again, FS exits.
Should there be an error log
you're doing?
/b
On Feb 11, 2009, at 1:15 PM, Nik Middleton wrote:
I have a situation where FS aborts
I'm running an lua script with mysql statements
First time it runs, on hangup I get
[CONSOLE] switch_core_memory.c:374 switch_core_memory_reclaim()
Returning 4 recycled
-users] FS 1.0.2 Crash and burn
How about getting a backtrace of the core dump and opening a jira?
http://wiki.freeswitch.org/wiki/Reporting_Bugs
/b
On Feb 11, 2009, at 1:35 PM, Nik Middleton wrote:
I was running in a screen session, so going back to the console it shows
it's
: [Freeswitch-users] FS 1.0.2 Crash and burn
How about getting a backtrace of the core dump and opening a jira?
http://wiki.freeswitch.org/wiki/Reporting_Bugs
/b
On Feb 11, 2009, at 1:35 PM, Nik Middleton wrote:
I was running in a screen session, so going back to the console
Hi Guys
I'm trying to set the outbound caller-id in js. The params seem to be
acceptable, except I'm getting the default +0 caller-ID sent.
Should the below work with js?
session.originate(session,'{accountcode=54321,ignore_early_media=true,or
I'm having an issue with call accounting
If I initiate a call, and it is then transferred to an IVR menu.
Person selects 1 to talk to someone.
In js
else if (data.digit == 5) {
if (session.ready()) {
var new_session = new Session();
Bang on,
Thanks
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 12 February 2009 01:10
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Call
HI,
Is there an equivalent function in FS to waitforexten ? Closest I've
seen is collectInput?
Right now I'm using stream file, which is ok if they hit a digit before
stream ends, but I want them to have a certain period after the file is
played to hit a button.
Regards,
Sorry, should have said this was in js
Regards,
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Brian West
Sent: 12 February 2009 18:08
To: freeswitch-users@lists.freeswitch.org
Subject: Re:
Hi Guys,
I'm trying to get VMD running in js, does anyone have an example of how
it's called?
If I try
session:execute(vmd);
I get an error
Regards
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
That makes sense, though could it not have a call back mechanism similar
to DTMF detect?
I'm still not sure how I could use it even in an event socket. I plan
to call my js IVR script using a socket, but that has the originate call
in it which is nice and simple, but I'm unsure how I could abort
Of
Michael Collins
Sent: 12 February 2009 21:45
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] js and VMD
On Thu, Feb 12, 2009 at 12:49 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
That makes sense, though could it not have a call back mechanism
similar
to DTMF
Hi,
Not sure who updates the WIKI, but it's wrong on collectinput for the
example. In the call, dtmf needs quotes, ie dtmf
Correction is session.collectInput( mycb, dtmf, 8000 );
Without it you get
[ERR] voice.js:70 mod_spidermonkey() ReferenceError: dtmf is not
defined
if ( session.ready( )
equiv for waitforextension
YOU DO! ;) Its a user edited content portal.
/b
On Feb 12, 2009, at 4:58 PM, Nik Middleton wrote:
Not sure who updates the WIKI, but it's wrong on collectinput for the
example. In the call, dtmf needs quotes, ie dtmf
Use this method in js
var session = new
Session('{absolute_codec_string=PCMA,accountcode=54321,ignore_early_medi
a=true,origination_caller_id_number=4071122,originate_timeout=25}sof
ia/gateway/myprovider/87304071122);
-Original Message-
From:
I think this page (external) is the source
http://alexn.org/docs/dialer.html
Regards,
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 13 February 2009 14:06
Can't figure this one out.
I've enabled a hang-up hook in js to do some cleanup.
I've followed the example on the wiki, but it would appear it's never
called.
http://wiki.freeswitch.org/wiki/Example_Hangup_hook
Is the code in error?
Regards
11, 2009, at 5:36 PM, Nik Middleton wrote:
I've abandoned LUA.
All sorts of problems (DTMF etc). Also reports of memory leaks when
using MYSQL driver.
Looking on the WIKI, JavaScript seems very well supported; PLUS DTMF
works just fine (pulling my hair out on LUA)
Guess I'm going
I'm trying to capture the hang-up reason and write it to the db (Was it
busy etc). I also close the db in that function. That way I know I
don't have any open connections. This is in JavaScript BTW
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
@lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS 1.0.2 Crash and burn
Nik Middleton nik.middle...@noblesolutions.co.uk wrote:
Code
I've looked at so far is very neat, but boy is there a lack of in-line
comments. Haven't looked at the main source yet though. I always
used
to work on 3 lines
The JS hook does indeed work.
New to js, I hadn't declared the function prior calling it. I can only
guess that java scripts are processed sequentially and do not throw up
errors if a call is made to a function that hasn't been processed yet
Regards,
-Original Message-
From:
Understood.
However, using the second method, how can I trap on call failure?
If I originate a call and the user is busy, the console reports this
fact, but then the script continues to execute
if (session.ready()) {
console_log(notice,Session result=[ +
using an alternate name for your new session like my_session
etc?
this works for me, try it yourself.
var my_session = new
Session(sofia/external/7...@conference.freeswitch.org);
consoleLog(err, ready: + my_session.ready() + \n);
On Sat, Feb 14, 2009 at 7:17 AM, Nik Middleton
nik.middle
of you perhaps learning how they actually work.
On Sat, Feb 14, 2009 at 1:47 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Nope,
Still not working. Here's my little test javascript
var new_session = new
Session('{ignore_early_media=true,}sofia/internal/1...@192.168.3.206
Hi guys,
I'd like to get the number of calls on the system so that I can manage
the load.
From the cli, I've tried the following:
Show channels
This along with the call detail shows me the correct number of calls
Show calls count
This delivers a value of zero.
I should
I'm in the same boat, finding the transition from Asterisk to FS very
frustrating. Something I can do in Asterisk in 10 minutes is taking me
a day with FS.
Do I think it's worth it? Absolutely, but it's incredibly painful at
times.
What I've done is to create some WIKI pages to help
For what it's worth, using Asterisk recordings, I found FS to be better
than when played on an Asterisk system.
I came to the same conclusion early on that the included prompts with FS
were of a relatively poor nature. Not volunteering to record new ones,
but they do let the product down, as
Having spent the last week developing a small js app, I ran some tests
today. With just 5 calls going on, I'm seeing huge delays from when the
call is answered to when the audio file is played. Sometimes it doesn't
even play at all!!
Example 3 calls and the matching playbacks
2009-02-17
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Big delays in playing audio files
we would need to see your script.
On Tue, Feb 17, 2009 at 12:23 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Having spent the last week developing a small js app, I ran some
: [Freeswitch-users] Big delays in playing audio files
Is this the entire script?!
-MC
On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
if (first_session.ready()) {
console_log(notice,Session state=[ +
first_session.state + ] \n
-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 17 February 2009 20:57
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Big delays in playing audio files
On Tue, Feb 17, 2009 at 12:48 PM, Nik Middleton
I've got it working now thanks
I've also added a working example to the Wiki (lua/addBody) which was
empty
Regards,
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 17
Err, that's what I just posted :)
Regards,
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 17 February 2009 23:30
To: freeswitch-users@lists.freeswitch.org
Subject: Re:
] AddBody to events in lua
Good... keep up the good work adding more docs. ;)
/b
On Feb 17, 2009, at 5:33 PM, Nik Middleton wrote:
Err, that's what I just posted :)
Regards,
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http
] On Behalf Of
Brian West
Sent: 18 February 2009 00:15
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] AddBody to events in lua
And you ran this in lua?
/b
On Feb 17, 2009, at 6:07 PM, Nik Middleton wrote:
I ran 10,000 events, which completed in around 20 seconds, all
Kristian,
You're my hero, if I hadn't come across astlinux 3 years ago, I wouldn't
be doing this stuff right now. Not too sure if that's a good thing
though ;)
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org]
] Originate and bridge with lua
Nik,
What are you building? I'm wondering if this is the correct approach
for your application. You might be better off using the even socket
and controlling your calls from a central point.
-MC
On Wed, Feb 18, 2009 at 11:26 AM, Nik Middleton
nik.middle
: [Freeswitch-users] Originate and bridge with lua
On Wed, Feb 18, 2009 at 11:53 AM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
I'm trying to build an emergency broadcasting solution.
So I place a call, and have ivr in the lua script. But I also want to
give them the option
Sorted now thanks, it needed to be in the format
session:execute(bridge, {params}sofia/gateway/Mygateway/number);
key change was ''
Now I've converted my js script to lua going to run some tests tomorrow.
I sincerely hope it'll handle more than the 10 calls js would break at.
Here's my
issue, you should have been doing
something similar there too.
BTW,
If you make another comparison to asterisk comment, I will never answer
another email from you again I don't have time for that crap.
On Wed, Feb 18, 2009 at 3:56 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote
Hi Guys
I'm having problems with seg faults about every 10 mins with call loads
200. I've processed the core dump
(http://pastebin.freeswitch.org/7436) but I'm unsure what I should be
looking for. I don't see the point where the crash occurred. Can
someone point me to where I should be
Works for me, see snippet below
var first_session = new Session(dial_string);
// Trap for call failure
if (!first_session.ready()) {
consoleLog(err, Disposition: +
first_session.cause + \n);
if
Hi Guys,
I've been running a test script written in lua which now works very well
thanks to Anthony's fix to stream file.
Right now I'm using an event socket to initiate the call and passing the
name of the script along with originate thus:
$dialstring = originate
to identify what you want to do with the failed calls.
On Sat, Feb 28, 2009 at 4:49 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Hi Guys,
I've been running a test script written in lua which now works very well
thanks to Anthony's fix to stream file.
Right now I'm using
Well if it's any consolation, I have a 4 day ish old copy of SVN and I
have around 200 of these hung calls, though after an hour or so they did
seem to clear.
That said, FS made 138,330 call attempts today, not too shabby, and
through out the call quality was as good as the first one. Not sure
Just curious here.
I've always followed the fedora route but became disillusioned with the
focus on the desktop rather than the server mode. Of late I've moved my
servers to Centos. I felt the need for stable systems.
Everyone seems to slate Centos, but to my surprise Anthony recommends
Hi Guys,
In External.xml in sip profiles I have
param name=ext-rtp-ip value=$${external_rtp_ip}/
param name=ext-sip-ip value=$${external_sip_ip}/
Can I override these for a given gateway profile? I have one gateway
that's expecting a local routed IP address due to the way that it's
, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Hi Guys,
In External.xml in sip profiles I have
param name=ext-rtp-ip value=$${external_rtp_ip}/
param name=ext-sip-ip value=$${external_sip_ip}/
Can I override these for a given gateway profile? I have one gateway
that's
We use the VIA mini ITX boards. Great for small offices and very stable
with various fan-less options
Regards
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Henry Huang
Sent: 06
Hi Guys,
I'm trying to debug some SIP messaging issues. Is there a way of doing
the Asterisk equivalent of SIP Debug so I can see what's being sent?
Regards,
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
Hi Guys,
Now that IAX has been published as an RFC
(http://www.rfc-editor.org/authors/rfc5456.txt) are there any plans to
support registrations?
Not a moan, just curious as to the road map.
A lot of my users have Asterisk PBX's using IAX and I'd love to replace
my Asterisk central
: Re: [Freeswitch-users] Getting a sip trace on the console
I use the ngrep tool on the OS console and write the output to a file:
ngrep -d any port 5060 -W byline outfile.txt
Best regards
Peter
Nik Middleton schrieb:
Hi Guys,
I'm trying to debug some SIP messaging issues. Is there a way
To be fair, most of these messages are 4-5 years old. That said to
date, I can crash * by repeatedly doing a 'show channels'. All the same
FS should be robust enough to suffer this abuse. If it's not,. the
issue needs to be investigated.
Regards,
From:
allow show channels to work and the
answer is, sorry no.
On Sun, Mar 15, 2009 at 7:01 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
To be fair, most of these messages are 4-5 years old. That said to
date, I can crash * by repeatedly doing a 'show channels'. All the same
FS
Another issue with this module is the resources it consumes. We had it
running on 50 calls yesterday and the cpu's all went to 90+%
Regards,
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On
Hmm,
Well We're connected direct to E1's and it doesn't work reliably here.
That said, DTMF detect does recognise the beeps most of the time.
Perhaps there's a regional variation. I wonder if it's country
specific. The code looks logical. When I get some time I'll have a
look at it and see how
Hi Guys,
I know this sounds an odd question, but I need to inject audio into an
outbound call. The reason for this is that for a pre-paid billing app,
I need to let the call initiator know they are running out of credit.
Is there a facility to do this? Ideally I only want to let the
Worked for me, just needed to add the missing codec for media player
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Giovanni Maruzzelli
Sent: 31 March 2009 21:09
To:
Well you almost had me there, but SIP over SMTP? That was too much.
Regards,
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 01 April 2009 16:31
To:
Hi Guys,
I'm no Linux guru, but today I inadvertently had 1000+ call attempts
going through FS, load according to TOP was 16.5. Calls were still
absolutely perfect. Can I throw out the rule book on load ? CPU was
~45% on each core. (dual)
Regards,
Hi Guys,
I'm getting a few of these errors below
sofia.c:3247 sofia_handle_sip_i_state() Reinvite RTP Error!
Are these caused by a fax machine? Or am I barking up the wrong tree?
Regards,
___
Freeswitch-users mailing list
Hi Guys,
I'm looking for the optimum audio format when using streamfile in a lua
script.
I've found CPU load increases rapidly with the number of threads playing
a .wav file. Can anyone tell me the optimum when using g711a?
Right now the the .wav files are
Audio format: PCM
them all into raw alaw
files and rename them with a .PCMA extension
to avoid the g711 transconding but g711 to PCM is pretty trivial. it's
more likely a file i/o distress you see.
On Thu, Apr 16, 2009 at 5:04 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Hi Guys,
I'm looking
Can anyone tell me what would or cause the above hang-up cause? I'm
using latest svn and get loads of these above 10 Concurrent calls
Regards
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
Do the phones and FS have a firewall between them? If so, sounds like
the pin hole in the fw is being closed. Alot only stay open for 4 mins
Regards,
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf
:50
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Phones become unreachable after some
time
They do, but all necessary ports for FS are open. If that is fw issue,
are there ways to fight with it?
Nik Middleton wrote:
Do the phones and FS have a firewall between them
...@lists.freeswitch.org] On Behalf Of
paul.degt
Sent: 30 April 2009 14:35
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Phones become unreachable after some
time
Worked for Grandstream, but not for X-Lite.
Nik Middleton wrote:
Don't know where the setting is in FS, but force
Hi Guys,
Is there an alternative to the hang-up event that doesn't send quite as
much data? This event is HUGE!
All I'm looking for this the result of the call, duration, dialed number
and the ability to pass variables. The hang-up event does all of this I
know, but I also get
1 - 100 of 156 matches
Mail list logo
