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at 6:55 PM, Brian West br...@freeswitch.org wrote:
ding ding ding .. yep!
file /usr/local/freeswitch/bin/freeswitch will also confirm
/b
On Feb 11, 2009, at 6:37 PM, Henry Huang wrote:
Brian:
I am also running my freeswitch on my own openVZ containers. Just
how do you verify
2.6.9, not stripped
It should clearly tell you. run the file command on it.
/b
On Feb 12, 2009, at 2:13 PM, Henry Huang wrote:
I run /usr/local/freeswitch/bin/freeswitch
but I don't see a place where it says it's 32bit or 64bit.
at the end of the initial script, I do see a version
/freeswitch-users
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:(snd_ctl_open_noupdate) Invalid CTL infile
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something that people THINK exists and works well...
Reliable human/voice detection doesn't exist in ANY form.
/b
On Mar 1, 2009, at 8:20 PM, Henry Huang wrote:
Does the freeswitch VAD is able to distinguish ring tone from human
voice?
The scenario is to originate a call to a IVR system
in early media... so the best bet would be to
ignore_early_media=true
/b
On Mar 1, 2009, at 9:05 PM, Henry Huang wrote:
Well, I knew it would be some future fantasy for now..
If not human detection. I guess will try to use Dialplan Tools wait for
silence to wait till the ring tone is finished
.
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=
reg.2.outboundProxy.transport=
reg.2.acd-login-logout=0
reg.2.acd-agent-available=0
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Thanks
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Thanks
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http
that. You do have
a whole bunch of variables of the parsed sip header you can use. Use the
info application to see all the vars so you can see what you have to route
the call on.
Mike
On Aug 22, 2009, at 2:40 AM, Henry Huang wrote:
Hi:
I try to dial sip url from my softphone but seems like
to dialplan for
regex match.)
On Sat, Aug 22, 2009 at 6:30 PM, Jason White ja...@jasonjgw.net wrote:
Henry Huang red.rain.se...@gmail.com wrote:
It that case, the example of dialing sip_uri in the dialplan/default.xml
should be removed to prevent confusion. Because according to what you
said,
one
various other variables you can
condition on also... route on destination_number and you'll be fine.
/b
On Aug 22, 2009, at 9:09 AM, Henry Huang wrote:
I fully understand how the regex works in the dialplan. If you look
closely in my original email and check out the pastebin. You will
see
it to the dialplan from mod_sofia...
/b
On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:
Brian:
but why can't I pass sip: to dialplan? seems like it's being
truncated by sofia..
Can you confirm that?
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told
more than once it won't pass it to the dialplan from mod_sofia...
/b
On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:
Brian:
but why can't I pass sip: to dialplan? seems like it's being
truncated by sofia..
Can you confirm
Michael:
Thank you for making it in for dummies format. :P
These are really nice tips I can use. thanks.
On Sat, Aug 22, 2009 at 11:35 PM, Michael Collins m...@freeswitch.orgwrote:
On Sat, Aug 22, 2009 at 8:07 AM, Henry Huang red.rain.se...@gmail.comwrote:
Brian:
Oh, and again, if it's
/mailman/options/freeswitch-users
http://www.freeswitch.org
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with different combination for a week now.. Please
shed some light if you know something.
Thanks,
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the UUID to sched_hangup.
Usage: sched_hangup [+]time uuid [cause]
http://wiki.freeswitch.org/wiki/Mod_commands#sched_hangup
On Tue, Oct 13, 2009 at 10:14 AM, Henry Huang b_ball_he...@hotmail.comwrote:
Hi:
I am using mod_java. And in my script I was able to achieve using:
execute
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something to execute an api command right before or right after the
call gets bridged.
On Fri, Oct 16, 2009 at 9:27 PM, Michael Jerris m...@jerris.com wrote:
sched_api is a fsapi command not a dialplan application, I believe
sched_hangup is both.
Mike
On Oct 13, 2009, at 6:14 AM, Henry Huang
:
# http://java.sun.com/webapps/bugreport/crash.jsp
# The crash happened outside the Java Virtual Machine in native code.
# See problematic frame for where to report the bug.
On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins m...@freeswitch.org wrote:
On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang
.
Which line of code from java caused that segfault
It looks like a simple NULL string issue that we may want to hunt down.
On Wed, Oct 21, 2009 at 4:44 AM, Henry Huang red.rain.se...@gmail.comwrote:
I can't seem to find the right thing to use in mod_java to execute api
commands, only
/Variable_api_on_answer
On October 23, 2009 11:02:07 am Anthony Minessale wrote:
it's probably related to escaping the data.
I was sick of watching you suffer so i added api_on_answer variable to
trunk.
On Fri, Oct 23, 2009 at 3:54 AM, Henry Huang
red.rain.se...@gmail.comwrote:
Thanks to c6burns on IRC
and
exteranl RTP IP to the public IP in internal profile, I am still getting no
audio. Can anyone clear the concept for me here?
by the way, I am using freeswitch 1.4 stable version.
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