Displaying 11 results from an estimated 11 matches for "minsing".
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mining
2010 Mar 31
2
Generative Topographic Map
...t;- RANDU()
}
}
for(i in 1:ncol(MU)) {
for(j in 1:nrow(MU)) {
MU[j,i] <- RANDU()
}
}
sigma <-1
FI <- gtm_gbf(MU,sigma,X)
W <- gtm_ri(T,FI)
Y= FI%*%W
b = gtm_bi(Y)
lambda <- 0.001
for (m in 1:15) {
trnResult = gtm_trn(T, FI, W, lambda, 1, b, 2,quiet = TRUE, minSing = 0.01)
W = trnResult$W
b = trnResult$beta
Y = FI %*% W
}
I ran the above script on my own data representing 1969 samples of 7 dihedral angles of a folding molecule (attached.
Upon running the 1st iteration of the training function "gtm_trn" I get the following error that I...
2009 May 08
2
Override sip.conf settings in extensions.conf? Possible?
Hi all...
Does anyone know if it is possible to override sip.conf settings in extensions.conf
(for example: session-minse=90) without needing to create an overarching peer in sip.conf
and selecting it specifically in the dial plan?
I'm on the 1.4 stable code base and looking to implement session-timers on certain call
flows in a modular dial plan.
Thanks,
Josh Fuller josh.fuller at
2012 Oct 31
2
Asterisk and OpenLDAP
Hello guys,
i would like to implement authentication for my sip extension with an
openldap server.
Following this guide
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html
i see a template named [sip] to map the information of sip peers into ldap.
But i'm not interested to create a template, i would only authenticate
sip extensions using username
2009 Apr 03
1
conference calling
Greetings listers.
I'm running asterisk 1.4.21.2 on SUSE 11.0 using
Polycom 501 phones. My outgoing connections are Zapata using a TDM401P.
For the most part I can make and receive calls fine except for these 3
issues:
1. When I call an external conference, the call never bridges and
hangs up after 60-90 seconds.
2. When I call another number there is a
2009 Jan 28
1
asterisk-users Digest, Vol 54, Issue 94
> Date: Wed, 28 Jan 2009 13:11:19 -0600
> From: "Danny Nicholas" <danny at debsinc.com>
> Subject: Re: [asterisk-users] SIP Registrations broken on 1.4.22.1?
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <D32AD473FC574B41AE6A842E46549174 at db0005>
>
2011 May 02
3
out of the blue one way audio
Greetings List.
we're facing a strange case with my system where in the middle of the call .. after like 7 minutes (not necessarily ) the callee is unable to hear the caller however the caller is able to hear the called party. the scenario is the following.
1- 15 computers running Windows XP SP3 joining a Windows Domain Controller with DHCP , DNS, ISA Internet Acceleration Server.
2- Internet
2014 Jul 26
1
Rejecting secure audio stream without encryption details - when using ws clients and Kamailio integration
Greetings,
I've noticed a problem that might originate from my Asterisk configuration,
could use a hand in sorting it out. Problem is a 488 response from Asterisk
whenever it gets RTP/SAVPF profile in the SDP.
My current setup has Asterisk Kamailio realtime integration, and Kamailio
uses dispatcher to route calls for Asterisk to handle. Now I have only one
Asterisk, on the same machine as
2009 Jan 28
2
SIP Registrations broken on 1.4.22.1?
Hi,
I had a Trixbox 1.4.18 that I "yum update"d to 1.4.22.1.
Now, I seem to have a huge problem with phones not staying registered
(registrations worked perfectly at 1.4.18).
I phone will register the first time I plug it in, and then once you
make a call and hangup (or sometimes even during the call)
all the lights will go orange meaning a lost registration. Every so
often the lights
2010 Oct 28
0
Intermittent failure when placing calls - unable to create channel of type SIP
Hello community,
I've been running Asterisk on an embedded device for about six months, and
my operation has been largely trouble-free. I'm hoping I could get some help
with a minor problem:
Every week or three, my PBX gets stuck in a state where it can receive
calls, but it becomes completely unable to originate outgoing calls until I
do a "sip reload". After doing the SIP
2011 Jan 28
3
Disabling Music On Hold
Hello,
I have been trying to completely disable music on hold on my asterisk
system. When a call is put on hold I do not want any music on hold, but I
would like the remote user to get informed of this event (depending on the
technology e.g. with a SIP reinvite and an SDP indicating the call is on
hold).
I have searched and tried out various approaches, but when putting the
call on hold
2010 Apr 19
2
OpenSIPS with Asterisk Backend
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