Ken,
Sign me up as a beta tester. If you don't want beta testers then sign
me up as a scumbag leech who wants to check out your cool software
without paying anything! :D
Seriously, I'm available if you'd like to get some alternate viewpoints
on how your stuff works.
-MC
Gang,
I'm working on my outbound IVR and I'm wondering if you have any
thoughts on this topic: I'd like to make a record of the digits dialed
by the called party. For example, I have an IVR menu that says, to
repeat these options, please press star. I'd like to record how many
times the
Hi all,
I build a new test boxes for freeswitch. and trying to simulate high load
condition.
I have one freeswitch box that host conference apps, and another box with
simple js to originate call to the conference room.
To monitor the sound quality, I have one phones that dial into the
conference
Hi all,
I'd like to know if there is a way how to fork call to several
destinations in the dialplan.
Something like DisAsterisk(TM) Dial(Local/xLocal/y) or OpenSER
append_branch and branch_route...
Or - if it is not possible - could someone point to any workaround how
to achieve this?
the originate string can contain a list of , and | sep urls
, means () forked simo dial
| means one at a time
both at once:
sofia/default/[EMAIL PROTECTED],sofia/default/[EMAIL PROTECTED]
one at a time
sofia/default/[EMAIL PROTECTED]|sofia/default/[EMAIL PROTECTED]
you can put , sep list
We don't do much testing on the conference but i can give you a few
pointers.
is it a 32 or 64 bit box?
if it's 32 you can try this as root before you start up.
ulimit -s 244
you can also change the conference interval to higher number of ms between
packets to give it more time to mux the
Wiki updated: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall
On Thu, May 29, 2008 at 11:43 AM, Anthony Minessale
[EMAIL PROTECTED] wrote:
the originate string can contain a list of , and | sep urls
, means () forked simo dial
| means one at a time
both at once:
Thanks Anthony,
It's a 32bits Ubuntu server version, and I had blindly follow and run the
test with following settings :)
ulimit -c unlimited
ulimit -d unlimited
ulimit -f unlimited
ulimit -i unlimited
ulimit -n 99
ulimit -q unlimited
ulimit -u unlimited
ulimit -v unlimited
ulimit -x
Thank you very much, Anthony, for your answer!
I'm affraid I didn't describe my needs clearly :-)
I want to fork call back to the dialplan, not to the endpoints - ie
parallelize walking through the dialplan from some point. (Nearly same
thing Asterisks Local channel does.)
Any sugegstions
you can call your own box over sip, that's the best you can do.
We do not have such a hack as chan_loco
On Thu, May 29, 2008 at 10:43 AM, kokoska rokoska [EMAIL PROTECTED]
wrote:
Thank you very much, Anthony, for your answer!
I'm affraid I didn't describe my needs clearly :-)
I want to
Woof!
Using an event socket on an answered parked call, I am trying to use deflect
to transfer the call off of FreeSwitch to a SIP destination, like this:
sendmsg
call-command: execute
execute-app-name: deflect
execute-app-arg: sip:[EMAIL PROTECTED]
Alas, our proxy will challenge the REFER
Anthony Minessale napsal(a):
you can call your own box over sip, that's the best you can do.
Thank you very much, Anthony, for such a hint!
I found it works few minutes ago, but not sure if it is the best way a
should go...
We do not have such a hack as chan_loco
:-)))
Thanks once
Yea, seems like a pretty old machine. and I thought it was at least a dual
core thou :(
I couldn't find a way turn hyperthreading off during boot time yet, If I do,
will report back if it does some improvment.
btw, How do you know if this is hyperthread processor from cpuinfo?
Rgds,
Johny K.
the 'noht' argument to the kernel params.
/b
On May 29, 2008, at 12:24 PM, Johny Kadarisman wrote:
Yea, seems like a pretty old machine. and I thought it was at least
a dual core thou :(
I couldn't find a way turn hyperthreading off during boot time yet,
If I do, will report back if it
Hmm... pretty strange, I change grub menu.lst, and pass 'noht'
but, it still showing the 2 'fake' cpu's.
On Thu, May 29, 2008 at 1:30 PM, Brian West [EMAIL PROTECTED] wrote:
the 'noht' argument to the kernel params.
/b
On May 29, 2008, at 12:24 PM, Johny Kadarisman wrote:
Yea, seems like
The conference is more resource intensive than normal bridging but the
general rule for media channels is about 190 channels (95 bridges) per 1
gigahertz of CPU on a 64 bit platform. If you don't need to run media into
FS you can bet on a lot more.
On Thu, May 29, 2008 at 11:54 AM, Nicolas
With FreeSwitch there are a couple of ways to accomplish what you are doing
with 3 distinct levels of performance
Way 1) Full Media Interaction/Transcoding. This is very similar to the way
asterisk works and should on modern several give you atleast 2 to 3 times
the performance you see on
Anthony and Ken (specially), thank you very much for your explanations
and figures. About what Ken said, how could I initiate a call in media
mode and then switch it to no_media when the second leg is
bridged/answered? Also, is this something my VoIP provider should be
able to support specially,
you can do originate
{ignore_early_media=true,bypass_media=true}sofia/default/[EMAIL
PROTECTED]/default/
[EMAIL PROTECTED] inline
and hairpin 2 calls between the provider
On Thu, May 29, 2008 at 2:55 PM, Nicolas Brenner [EMAIL PROTECTED]
wrote:
Anthony and Ken (specially), thank you very
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