Hi Guys,
I'm using an outbound socket to control calls, and it works a charm.
However, what I'd like to do is send a custom event regarding the call
on hang-up. The way I see things happening at the moment, and I could
be wrong, is that the socket is closed when a hang-up occurs, so am I
takin
I'd heard rumours that this was going to happen and it's great news and
good news for FS as well. With a user friendly front end, FS is sure to
fly. I have no doubt that this will be the first of many.
Regards,
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mail
Hi Guys,
I'm getting a core dump when running an lua script that's been fine for
months
In Freeswitch_lua.cpp line 92 is being called, but it's not clear what
exactly this is doing
lua_State *Session::getLUA()
{
if (!L) {
switch_log_printf(SWITC
build issue with an older mod_lua with a
newer FreeSWITCH
did you update via make current?
On Thu, Sep 10, 2009 at 11:11 AM, Nik Middleton
wrote:
Hi Guys,
I'm getting a core dump when running an lua script that's been fine for
months
In Freeswitch_lua.cpp line 92 is being c
Check out this range
http://www.noblesolutions.co.uk/shop/index.php?main_page=index&manufactu
rers_id=16
You should be able to find a local supplier
We've used them for a couple of years now. They just plug into your
network.
Regards,
-Original Message-
From: freeswitch-users-boun..
Hi Guys,
This one has me stumped.
I'm originating a call, playing audio, trapping on DTMF and bridging to
another endpoint (read phone number)
If the A leg hangs up, then the call is cleared down and all is well.
However if the B Leg attempts to hang-up, the LUA script that is
handling the br
Hi,
Is there an option to hang-up both call legs in a bridge when one leg
hangs up?
In my lua script I only ever see the hang-up for the call I'm in, not
for the bridged b leg. That said, I can see both a hang-up and un
bridge event being fired for the B leg. However my issue is that the
ply to the other thread?
set the channel variable hangup_after_bridge=true on the a leg
your script must not be checking for the case when b leg hangs up that A
leg does not hangup unless that var is set.
On Fri, Dec 4, 2009 at 2:03 PM, Nik Middleton
wrote:
Hi,
Is there an option to hang-up
Hi, Is it possible to disable being able to put a call on hold using
hook flash?
Regards
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UNSUBSCRIBE:http://list
-users] how to disable hook flash hold
It can be done from the phone itself; for example on a Grandstream
phone it is done with the option "Onhook Threshold:" setting it to
"hookflash OFF"
2009/12/5 Nik Middleton
>
> Hi, Is it possible to disable being able to put a call o
n ATA or a
gateway). It is there you configure this behaviour.
T.
On Sat, Dec 5, 2009 at 6:20 PM, Nik Middleton
wrote:
Sorry, I meant from a POTS phone
Regards
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org]
Hi all,
I'll slowly pulling my hair out on this one. I had FS successfully
hanging up both legs on a bridge, now today, with nothing changed, I'm
not seeing a hangup of the b leg at all.
FS is behind a PIX, so it might be a weird NAT issue, but A leg calls
hangup just fine. Before when I
g
On Mon, Dec 7, 2009 at 11:01 AM, Nik Middleton
wrote:
Hi all,
I'll slowly pulling my hair out on this one. I had FS successfully
hanging up both legs on a bridge, now today, with nothing changed, I'm
not seeing a hangup of the b leg at all.
FS is behind a PIX, so it mig
Hi
Is it possible to trap on DTMF on a bridged call within an LUA script?
I've tried setting the gateway to use inband, but no joy. It looks like
I could use start_dtmf, but I can't see how to launch this within LUA
Regards,
___
FreeSWITCH-users
Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call
On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton
wrote:
Hi
Is it possible to trap on DTMF on a bridged call within an LUA script?
I've tried setting the gateway to use inband, but no joy. It looks like
I could use start_dtmf,
and interprets them aka calls
your callback etc.
On Mon, Dec 7, 2009 at 4:02 PM, Nik Middleton
wrote:
Hi
Is it possible to trap on DTMF on a bridged call within an LUA script?
I've tried setting the gateway to use inband, but no joy. It looks like
I could use
half Of
Anthony Minessale
Sent: 07 December 2009 23:21
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call
did you set the inputcallback too?
On Mon, Dec 7, 2009 at 4:59 PM, Nik Middleton
wrote:
Can this be done in an lua script?
Thought I'd send this little hurrah! As there seems to have been a lot
of negativity on this list lately.
>From my point of view, having looked at many solutions out there, FS is
still number one with regards to flexibility and performance. I cannot
imagine doing what I'm using FS for, with a
itch-users] no hangup on B leg
We will really need debug logs and sip traces to be able to figure out
what exactly is going on here.
Mike
On Dec 7, 2009, at 4:06 PM, Nik Middleton wrote:
Sorry no, apart from the fact that I was seeing the hangup.
I'm wondering if this a
condition when the call is in process
of tearing down.
Mike
On Dec 8, 2009, at 6:48 PM, Nik Middleton wrote:
No doubt, but that's a little difficult as this only happens
occasionally and I have 200 calls going on at the time. It's needle in
the haystack stuff.
Here
ch.org] On Behalf Of
Fred-145
Sent: 09 December 2009 19:55
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Rocks!!!!!!!!!
Nik Middleton wrote:
> I cannot imagine doing what I'm using FS for, with any other product.
Yes
> it's frustrating at times, but
Hi Guys,
As a long time Asterisk user, I'm looking into freeswitch as an
alternative mainly due to (list multiple reasons here)
Can anyone give me a pointer as to how I would achieve the following?
I need to replicate an emergency broadcast system currently running
under Asterisk.
A
annel)
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"
wrote:
Hi Guys,
As a long time Asterisk user, I'm looking into freeswitch as an
alternative mai
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
answer is "yes, FS can do that."
> > The first thing that you should do is unlearn "the Asterisk way" of
> > thinking. Usually there is an elegant way of doing things in FS that
> > wasn't possible in Ast.
> >
> > I would recommend that you st
sers-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@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
wrote:
>
can FS
hang,
but what at what 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
> Subject: Re: [Freeswitch-users] OPenser <->
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 exte
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.
WITCH.org
On Feb 4, 2009, 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 fi
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.
@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Gateway setting
You don't use gateways if they auth by IP... just dial
sofia/profile/num...@remoteip
Also please stop hijacking threads.
/b
On Feb 4, 2009, at 4:40 PM, Nik Middleton wrote:
Hi Guys,
Need a little help he
23:00
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:
around on the
profiles or turn auth to false on the internal profile if you don't
require any digest auth or phones registering.
/b
On Feb 4, 2009, at 4:50 PM, Nik Middleton wrote:
Do apologise about the hijacking,
Question: My ISP sends inbound calls via 5060, so it seems I ne
: Re: [Freeswitch-users] FS in ISP Mode
You can reverse the ports or try to get your provider to send to 5060 or
you can bind an additional IP and have the external profile listen there
K
From: Nik Middleton
Reply-To:
Date: Wed, 4 Feb 2009 23:07:58 -
To
Hi Guys,
Simple question, tried asking on IRC but no joy, they're too busy
slating other systems.
I'm trying to dial out via a remote sip gateway via the dial plan
This works fine, but I'd like to wild card the extension so it matches
on anythin
, 2009, at 2:36 PM, Nik Middleton wrote:
mailto:sofia/$%7buse_profile%7d/0773400...@21x.xxx.xxx> /XXX"/>
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswi
Try as I might, I cannot seem to get caller ID passed to the external
sip gateway
This GW happily processes caller id from Asterisk
If tried adding param name="caller-id-in-from" value="true" in gw
definition, and even
in the dial plan to no
avail
Can anyone shed some light on this
...@lists.freeswitch.org] On Behalf Of
Brian West
Sent: 05 February 2009 22:29
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Caller ID not being passed
Try application export
/b
On Feb 5, 2009, at 4:19 PM, Nik Middleton wrote:
Try as I might, I cannot seem to get caller ID
rs] Caller ID not being passed
Nik,
While 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
But surely, If I have the proper values in the sip phones x
bject: 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
TE.
Regards
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
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
Freeswitch-
what 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
>
>
&
e 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
>
>
>
>
e, 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
>
>
>
> I¹m looking for some pointers on
> how to
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
___
More than happy to add my 2 cents worth when I have something useful to
say
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
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 be
Hi Guys,
Is there any form of Answer phone detection in FS? A search hasn't
really brought up anything
Regards,
___
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Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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 external
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
___
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Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinf
utomatically use the sound dir path.
-MC
Sent from my iPhone
On Feb 8, 2009, at 2:31 PM, "Nik Middleton"
wrote:
Hi Guys,
I'm having some issues passing an argument to an lua script.
Basically in an originate command I pass the
@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,
___
: 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
&g
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
___
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Freeswitch-users@lists.freeswitc
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
chael 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
wrote:
> Further to this message, DTMF works with PMCU but not with PMCA which
is the
> native format f
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"
functio
t. An SQL 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
wrote:
From: Nik Middleton
Subject: [Freeswitch-users] Strange error message
To: freesw
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 so
ary 2009 19:18
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS 1.0.2 Crash and burn
Can you show us what 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 scri
-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
: [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 it
ubject: Re: [Freeswitch-users] FS 1.0.2 Crash and burn
Try starting it from your /usr/local/freeswitch/bin... ./freeswitch
it'll dump in the same folder.
/b
On Feb 11, 2009, at 2:20 PM, Nik Middleton wrote:
Where is the core dump written?
___
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
igination_caller_id_number=0763060
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();
new_sessio
step. Also can you point me to where on
the wiki that keeps talking about session.originate? I need to clean
them off there.
/b
On Feb 11, 2009, at 6:09 PM, Nik Middleton wrote:
else if (data.digit == "5") {
if (session.ready()) {
var ne
users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Nik
Middleton
Sent: 12 February 2009 00:48
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Call accounting not working as expected
Thanks, that cured the call
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 accountin
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: [Freesw
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
___
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Freeswitch-users@lists.freeswitch.org
http://
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
eswitch.org] On Behalf 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
wrote:
> That makes sense, though could it not have a call back mechanism
similar
> to DTMF d
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 qu
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: freeswitch-users-boun...@lists.freeswi
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
__
On Feb 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)
Gue
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
[mailto:freeswitch
3:24
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS 1.0.2 Crash and burn
Nik Middleton 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
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: freeswitc
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=[" +
se
d variable name for the session you executed the app
on.
are you 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", "
g how they actually work.
On Sat, Feb 14, 2009 at 1:47 PM, Nik Middleton
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');
//set the on_h
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 those
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 th
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 15
exit();
}
}
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 17 February 2009 18:34
To: freeswitch-users@lists.freeswitch.org
Subject: Re:
l Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 17 February 2009 19:25
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Big delays in playing audio files
Is this th
s.freeswitch.org] On Behalf Of Nik
Middleton
Sent: 17 February 2009 20:11
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Big delays in playing audio files
Pretty much
I haven't included the on-event hooks as it never gets to the point
where they're called.
Onl
28
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Big delays in playing audio files
On Tue, Feb 17, 2009 at 12:11 PM, Nik Middleton
wrote:
> Pretty much
>
> I haven't included the on-event hooks as it never gets to the point
> where they're called.
2009 at 2:30 PM, Nik Middleton
wrote:
I'm starting to think it's a thread/DTMF issue. Ran 15 lines to my
office number (using latest trunk)
2009-02-17 20:19:38 [CRIT] switch_core_state_machine.c:259
handle_fatality() Caught signal 11 for unmapped thread!Aborted (core
dumped)
Also th
From: freeswitch-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
Hi Guys,
I'm having real problems doing something trivial, and there doesn't seem
to be any docs on this issue
In js I do this
//Disposition = disp;
//Create Custom event
custom_msg =
"call_dispos
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