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...fairly standard Pentium 4 running Windows XP. On occasion I am required to calculate up to a quarter of a million individual correlations, so any help would be very much appreciated. Best wishes, Peter James Lee _________________________ Peter James Lee Assistant Professor Psikoloji Bölümü Bilkent University Bilkent Ankara Turkey 06800 e-mail: peterjl@bilkent.edu.tr office: (90) 312 290 1807 home: (90) 312 290 3447 website: http://www.bilkent.edu.tr/~peterjl/index.html _________________________ [[alternative HTML version deleted]]

Hello, after reading the FAQ, I decided to compile icecast (1.3.11) with --with-crypt option, but, there's no mkpasswd anywhere in the source tarball. Am I missing something here? Thanks, Norberto <p>--- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a message to

Hi, i managed to connect to Ericsson MD110 with PRI at last. And made a successful call thru asterisk to ericsson. But when i try to call from ericsson to asterisk i got an error on asterisk side. And i couldnt figure out why. Here's my extensions.conf about incoming calls. [DID_span_1] include = DID_span_1_timeinterval_all,${timeinterval_all} DID_span_1_timeinterval_all] exten =

Hi, i know i am asking a lot of questions lately in this forum..sorry about that first of all. :) Ok, here is the deal.. I am trying to make a hybrid system with an ericsson MD110 and asterisk. Internally we have 4 digit phone extensions on ericsson.. and so in asterisk. So, what i want to do is to call pbx side without adding 9 or etc to the begining of the number from asterisk clients.. For

Hello, As i check the call logs, some of my clients seem to make successful calls but, in logfiles, Source field seems empty..Still I can see who is the source from Channel tab as SIP/XXXX, and the called number and the time etc but.. nothing on Source and the Called ID tab. Just some clients has this problem. But as i check nothing special in their settings. What might cause this problem. Using

Hello, I just tried to upgrade to 1.6.1.0 from 1.6.0.9 and i had problems in registering users. As i see from debug it successfully reads from users.conf but later,when a user tries to logon it say peer not found.... And there were an error msg about mysql about the username field..Smthing changed in mysql tables??? Now i downgraded to 1.6.0.9 again and everything is working..

Hello, I have a default installation of asterisk 1.6.1.9-2 When i create a user in users.conf via asterisk-gui, calls, voicemail etc works. But if i create a user realtime (and my realtime caching is available too) i can see the realtime user with sip show peers. But, my local dial rules does not work. I can call from realtime user to static users(the ones in users.conf) and if they are not

Hello, for a long time i am using asterisk 1.6 with astgui. but for production system i intend to use asterisk 1.4 which i think might be more robust. And for a more developed service options i preferd to install with freepbx. But still there are big plusses and minusses for both system. My complain about astgui+1.6 was.. For example there were no backup trunk config running on that version.Even

Hi guys, I am trying to perform edge profiling using on hello.bc file by using following command opt -insert-edge-profiling hello.bc -o hello-edge.bc but I get the error that option "-insert-edge-profiling" is unknown. Can you please help me to solve the issue. Please note that I am following the paper available at this link http://llvm.org/pubs/2010-12-Preuss-PathProfiling.pdf

Hi John, I have looked into the EmitAutoVarAlloca() in CGDecl.cpp. However, I could not figure out how to employ my custom attribute for code generation. For example, my custom attribute is visible in CGDecl.cpp but how can I generate based on my custom attribute if (D.hasAttr<myCustomAttri>()) { //What to do here? } What I wan in IR is something like below. Without Custom Attribute:

Dear All, I am using the release 33 of LLVM. This release supports -insert-edge-profiling option for opt. But it doesnt provide/supports the -dot-edge-numbers option. Where can I find the file implementing this option (e.g, the source file for edge profiling is located in lib/transforms/instrumentation)? Please help me out. Thanks Naveed Ul Mustafa

Hello, with asterisk 1.6 i am trying to make a dialplan Which i have such entry in extensions.conf exten => _8XXX,1,Dial(SIP/${EXTEN}) But some of my clients have both IAX and SIP accounts, to use iax clients while outside of my Local Area, and SIP clients (or hardware phones) in local area. But with such rule, i can only dial SIP accounts. Is there a parameter to find how the user connected?

Hello, In our previous PBX we have an option to turn off or on outside calls with a pincode.. Like, user is able to get calls or dial local lines by default, but when he/she uses a password entrance via dtmf, he can dial long distance calls etc.And at anytime he can logoff from outside call permit.. So is it possible to do smthing like this on asterisk.. A limited profile which needs sip password

Hello, I want to enable call forwarding for asterisk 1.6.0.6 I couldnt seen any config or option on gui or extensions.conf about it. I found some dialing plans to enable it on web as follows: [apps] ; Unconditional Call Forward exten => _*21*X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4}) exten => _*21*X.,2,Hangup exten => #21#,1,DBdel(CFIM/${CALLERIDNUM}) exten => #21#,2,Hangup ;

Starting from today i am receiving the following errors on asterisk.. What can be the reason for it? [May 7 11:45:16] ERROR[14885]: chan_dahdi.c:10515 dahdi_pri_error: ACK received for '0' outside of window of '3' to '4', restarting [May 7 11:45:16] WARNING[14885]: chan_dahdi.c:3347 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel

Hello, I was testing my asterisk for a while with 1.6 without much problem. Now i am trying to install a new system with asterisk 1.4 but now i am using a dual pri card instead of single pri.(TE220P) What i want is to use both PRI ports as group. Now i have zaptel.conf file created as follows -------------------------zaptel.conf-------------------------- # Span 1: TE2/0/1 "T2XXP (PCI)

Hello, I wanted to add a failover trunk to my asterisk configuration. I got 2 gateways for my calls.. one is a pri other is voip trunk. I want to keep my trunk for failover. I am using ast 1.6 with asterisk-gui. But when i add a failover trunk for test purposes asterisk-gui adds the following line to my extensions.conf. where superonline is my voip provider and span_1 is failover trunk. exten =

Hello, i am trying to enable call forwarding on asterisk 1.6 with asterisk-gui If i set my stdexten as follows (with the lines i marked) everything seems like working. But if i make any change on asterisk-gui and apply it.. it recreates the macro-stdexten and deletes my configuration regarding to it. So where should i add my call-forward configuration??? Where am i making a mistake??

Hi, I want to activate DND on ast 1.6.0.9 with asterisk-gui. Is there special commands that i need to use during such script or simply writing a code in extensions.conf that checks if the user has a DND=yes value on ast. database and act according to that like forwarding call to voicemail or sending back a busy tone or playing a DND msg. And is there a way to notify a GPX_2000 for example for a

Hi, I suddenly started to receive an error like [Aug 4 14:26:57] ERROR[2477]: chan_dahdi.c:10515 dahdi_pri_error: No more room in scheduler [Aug 4 14:26:57] ERROR[2477]: chan_dahdi.c:10515 dahdi_pri_error: Asked to delete sched id -1??? and it went on till i reboot asterisk and dahdi services.. I wonder what caused this error, because there were no physical problem between pbx and asterisk