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
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 bandwidth congestion issue. Is there anyway on
a bridged call I
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's what I know
] 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 this is largely down
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
7, 2009 at 11:01 AM, Nik Middleton
nik.middle...@noblesolutions.co.uk 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
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,
___
Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call
On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk 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
at 4:02 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk 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, but I can't see how to launch this within LUA
Regards
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
nik.middle...@noblesolutions.co.uk wrote:
Can this be done in an lua script?
Regards,
From: freeswitch-users
Hi, Is it possible to disable being able to put a call on hold using
hook flash?
Regards
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
-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 nik.middle...@noblesolutions.co.uk
Hi, Is it possible to disable being able to put a call
to something... (ZAP channel or an ATA or a
gateway). It is there you configure this behaviour.
T.
On Sat, Dec 5, 2009 at 6:20 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Sorry, I meant from a POTS phone
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
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
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
nik.middle...@noblesolutions.co.uk wrote:
Hi
Check out this range
http://www.noblesolutions.co.uk/shop/index.php?main_page=indexmanufactu
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:
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) {
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
nik.middle...@noblesolutions.co.uk 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
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
If this is Linux, there's nothing wrong with it using most of the
memory, if it starts to use the swap, then there might be an issue.
Utilizing the memory does not mean there is a memory leak
Regards,
From: freeswitch-users-boun...@lists.freeswitch.org
On Jul 11, 2009, at 4:08 PM, Nik Middleton wrote: Excellent.
Do I need to supply uuid on an out...
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting channel variables using event
socket
If you're going to do it that way you can just use set.
uuid_setvar is an api call...
/b
On Jul 12, 2009, at 5:10 PM, Nik Middleton wrote:
HI Guys,
Can't seem to get this to work
call-command
Hi Guys,
Is it possible to set a channel variable while a call is in progress
using an outbound event socket? I have a listening process that
examines the hang-up events and it would be neat if it could also get
some variables that I have set mid call as well. Note: I know it's
possible to
@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting channel variables using event
socket
uuid_setvar
/b
On Jul 11, 2009, at 2:19 PM, Nik Middleton wrote:
Hi Guys,
Is it possible to set a channel variable while a call is in progress
using an outbound event socket? I have
I'm ONLY use PCMA, so I would agree with Brian
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Brian West
Sent: 01 July 2009 20:35
To: freeswitch-users@lists.freeswitch.org
Subject: Re:
Hi Guys,
Is it possible to bridge to another destination while controlling a call
via the outbound socket?
In other words, I'm controlling a call using an outbound socket and at
some point want to originate a new call leg and bridge the two.
If it can't be done that way, I'm thinking I
] On Behalf Of
Michael Collins
Sent: 29 June 2009 21:22
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] bridge call from outbound socket
would you mind doing a pb of your script that is handling the OB event
socket connection?
-MC
On Mon, Jun 29, 2009 at 12:47 PM, Nik
Hi Guys,
I'm trying to parse events in C++ for an outbound socket. The docs are
a little contradictory, so I wonder if someone could help me out.
As I understand it and event is terminated with double LF's (\n\n)
However if there is a Content-Length header the wiki very confusingly
says
But from where? After the double LF of the header as one part of the wiki says
or after the line containing the content-length that another part of the wiki
says?
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
OK, finally figured it out.
Have updated the Wiki to remove ambiguity and posted some SUDO code
Regards,
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Nik
Middleton
Sent: 28 June 2009 20:38
...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Brian West
Sent: 29 June 2009 00:00
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Myevents in outbound socket
Are you using ESL?
/b
On Jun 28, 2009, at 5:55 PM, Nik
Hi Guys,
Scratching my head on this one, under load FS is not playing an audio
file, OR and lua script is not getting executed. Not all the time, just
some. I've changed ulimit -n to 9 but no diff, and ideas where else
I might look?
Regards,
-users-boun...@lists.freeswitch.org] On Behalf Of Nik
Middleton
Sent: 23 June 2009 17:46
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Sound file or lua script not played under
load
Hi Guys,
Scratching my head on this one, under load FS is not playing an audio
file
to a local_stream is not reading
from that stream source, such as if you paused during playback of a
local_stream.
They are only a real issue if you are getting them with no calls up.
On Tue, Jun 23, 2009 at 11:54 AM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Hmm,
Looking at console I'm
at 3:54 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Hi Guys,
This one has me a little baffled. If have a recent build (in the last
week) of FS installed on two near identical HP servers. One happily
runs 400 concurrent calls at around 50% CPU. The other can only run
around 50
You are indeed correct, it's the 64bit server that performs well, not
the 32bit PAE version. I'm hoping that's the cause. I need to dig
around and find out if it's possible to change the kernel remotely and
see it sorts the issue. Ultimately I'll update it to 64 bit anyway, but
that's a 500
Hi Guys,
This one has me a little baffled. If have a recent build (in the last
week) of FS installed on two near identical HP servers. One happily
runs 400 concurrent calls at around 50% CPU. The other can only run
around 50 calls without the CPU going to 98%. Identical configs and lua
Anything that's dedicated undoubtedly has less load that something
that's multifunctioned. However the lack of any conversations on front
ending a SIP server to FS would likely indicate that no one's found a
requirement for it at this time.
I would truly hate to see discussions of theoretical
I couldn't agree more. We're working with a group that are developing a
massive PHP based music application. They are experts in PHP and MySQL but not
in VOIP/Telephony. By tuning an abstraction layer that uses PHP to communicate
with the FS event socket, allows them to work on the areas
Ok, so I did a mere 86,000 calls today, but when it was all over, I had
6 sessions remaining like the one below (number and ISP changed)
Anyone have an idea why these 6 sessions remain? I also had 120 calls
that I didn't get a hang-up for, but that might be me not processing the
events
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 11 June 2009 22:20
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Orphaned calls
On Thu, Jun 11, 2009 at 2:07 PM, Nik
Not sure where enhancement requests should be posted, but here it is
anyway
I would dearly love to be able to send a status event that returns an
event style output that provides machine readable output rather than the
wordy human readable response. (I hate parsing)
Is there such an
] Orphaned calls
If they were still showing in status, can you use gcore to dump a core
next time this happens, leave it running somewhere we can get to it and
post a thread apply all bt to Jira.
Mike
On Jun 11, 2009, at 5:40 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote
Hi Guys,
Ran make current today, and am getting the following errors. I ran
bootstrap and configure, but still get these messages.
Any ideas ? Looks like I'm now missing some libraries
Regards,
configure: configuring in libs/pcre
configure: running /bin/sh './configure.gnu'
: [Freeswitch-users] Problems with make current
your svn update failed,
rm -rf libs/pcre svn update ./bootstrap.sh ./configure make
current
On Jun 10, 2009, at 2:30 PM, Nik Middleton wrote:
Hi Guys,
Ran make current today, and am getting the following errors. I ran
bootstrap
I finally got around to looking at why mod vmd didn't appear to run when
using LUA. Turned out that the example in the wiki was wrong.
It should have been session:execute(vmd,start);
And not session:execute(vmd);
I've updated the wiki
Regards,
I've put some c++ test code together to let the outbound socket control
the call, all works as expected, apart from the event subscription
Sending myevents\n\ngives the channel events
However sending event text all\n\n doesn't give me any events apart from
the channel events.
As I understand it, a new 'feature' was added over the weekend to
resolve NAT. If you're firewall is not allowing ICMP then FS waits
until it times out. At this time there is no option to disable it.
Regards
From:
Hi Guys,
I'm going some work with outbound socket, and have a few questions.
When each call is answered, I get a connection to my server socket.
Is it right to assume that this connection will remain for the duration
of the call?
If so, do I still need to pass the UUID when I call
Hi Guys,
Been running 'make current' and appropriate intervals over the last few
months and all's been well until today
Now I get the following, obviously mod_sndfile isn't happy, but I'm not
sure what to do to fix it
Regards,
making all mod_sndfile
cd . /bin/sh
@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Make current fails
Reboot strap.
/b
On Jun 1, 2009, at 2:33 PM, Nik Middleton wrote:
Hi Guys,
Been running 'make current' and appropriate intervals over the last few
months and all's been well until today
Now I get
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Nik
Middleton
Sent: 01 June 2009 21:52
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Make current fails
Thanks
Well I can only assume build 13537 is brain dead. Surely I shouldn't
have to edit a whole bunch of configs to get it working. FS now takes 3
minutes to start, with no indication as to what it's looking for in the
logs. That said, to date 'make current' has always worked well for me.
Guess I was
Hi Jay,
Have to say my DTMF works flawlessly on thousands of calls. (SVN trunk
from a couple of days ago. We handle around 100,000 calls/day via FS)
That said, I've found it depends on your SIP trunk provider.That
doesn't mean to say there isn't a problem; it's just that I haven't
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
you specify in your event.
On Sat, May 2, 2009 at 1:36 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
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
...@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
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
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
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
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,
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 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:
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
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
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:
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
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,
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
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
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
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
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'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
] 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
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,
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] 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]
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
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