similar to: Why doesn't Asterisk project document certain important features of Asterisk officially?

1. Minimum Mean Square Estimation (for Multi-Dimensional Linear Regression) 2. Two-Sample Kolmogorov-Smirnov Tests 3. Empirical ANOVA Methods All this to be done on empirical data. I am new to the R world, any help would be appreciated. Thank You, Bill Gilliland Lead Sensor Systems Engineer E303, Tactical Communications, Transport, and Sensors Group: Tactical

My Asterisk is Asterisk 1.2.30.2 currently running on viciserver the problem is : Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x861f6d8', 9 retries! Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x85a6420', 9 retries! Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided

TLDR: Heads up for out of tree backends - you're going to need to update your *DAGToDAGISel::Select method to unconditionally replace nodes directly instead of returning the desired replacement. So I'm working on fixing the undefined behaviour I described in llvm.org/PR26808. As part of this, we need to stop looking into deleted SDNodes to check if they were, in fact, deleted. A big place

Hi Everyone, For some reason a few phones connected to a pfSense box can't make or allow in OpenVPN in port 1194 UDP. So, I established the VPN tunnel on 1194 TCP and it works fine. I would like to know if there is any disadvantages to using TCP over UDP for the tunnel when using Asterisk or is just as reliable and solid as a UDP tunnel? Thanks -------------- next part -------------- An HTML

Hello, I'm trying to play a game (Star Wars: Rebellion) and it mostly works except for the tactical battle mode. There have been workarounds suggested on the appdb page (http://appdb.winehq.org/objectManager.php?sClass=version&iId=181&iTestingId=33818), which include getting a native d3drm.dll, which I did. This and one other native DLL (d3dxof.dll) reduced the number of errors and

Update: All in tree backends now implement `void Select`. I'll be removing the SelectImpl path on Monday. Justin Bogner <mail at justinbogner.com> writes: > TLDR: Heads up for out of tree backends - you're going to need to update > your *DAGToDAGISel::Select method to unconditionally replace nodes > directly instead of returning the desired replacement. > > So I'm

I have run a few more test at different bitrates and 1.1 is looking even worse in terms of speed compared to previous versions. I have shared a google sheet which has the raw data and charts for 6,16 and 32 kbps. Unfortunately you cannot show proper error bars on Google sheets but the standard deviation is in the data if you want to look. You can see that the profile for 1.1 is a lot different

Yes, I have tried that option with no difference in behavior. It seems it ignores that option when provided. Just for reference, I am building it on RedHat 5. I have never had this issue on any previous version of OpenSSH. I use the default configuration with only the changes specified in the RHEL 5 STIG applied. I appreciate the security advice. The root account was indicated simply as an

Hi Everyone, I am testing Asterisk 1.8 AMI events. The voipinfo page on AMI events is specific to 1.6. I am wondering if the developers cared to write about the new events that are spit out in Asterisk 1.8 somewhere on the web? I checked the tar ball for asterisk 1.8 and documentation doesn't include this event: *Event: Unlink* Privilege: call,all Channel1: SIP/9999-00000029 Channel2:

Dear All, Just when I thought I had the plague of gfortran-9 under control, I made the tactical error of allowing my mac mini to ?upgrade? to macOS 10.14.6 which apparently also upgraded Xcode to 10.3. In consequence I?m having difficulty building my packages. The current symptom is: /usr/local/clang7/bin/clang -I"/Library/Frameworks/R.framework/Resources/include" -DNDEBUG

Can you put something in the release notes when this happens? Thanks, Hans On Sat, May 21, 2016 at 10:57 AM, Justin Bogner via llvm-dev <llvm-dev at lists.llvm.org> wrote: > Update: All in tree backends now implement `void Select`. I'll be > removing the SelectImpl path on Monday. > > Justin Bogner <mail at justinbogner.com> writes: >> TLDR: Heads up for out of

Hi, Just tried upgrading to 1.8 and ran into two problem immediately; 1. Caller-ID behavior is different -- now when I set the caller-id name to something with special characters (?, for example), the SIP INVITE now has %C3%96 instead of the ? character. I've tried doing Set(CALLERID(name-charset)=utf8) as well as iso8859-1, but it's always the same behavior. 2. My AMI scripts have

I wanted to roughly benchmark how the different version of libopus performed at each complexity level for a 6kbit/s output opus file. This was conducted on a Raspberry Pi so it is a constant hardware platform. This was done on an early Pi so only 256MB RAM but it was never used up so should not make a difference. I compiled the three final versions of each major release of libopus so that was

Does the application PGSQL has been removed from Asterisk? Couldn't find it on Asterisk source and addons. Atenciosamente, Vin?cius Fontes Gerente de Seguran?a da Informa??o Canall Tecnologia em Comunica??es Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunica??es Passo Fundo - RS - Brazil +55 54 2104-7000

Hi All, What would be interesting would be a plot of complexity versus subjective or object audio quality. I've not had a chance to look at the new analysis code in 1.1 so maybe in the case of a 6kbps compression you could clarify what decisions would it be making that would justify the extra complexity? Best Regards Cliff Parris -----Original Message----- From: opus-request at

Hi folks, Is there anywhere I can possibly get a model of the exam itself, maybe possible scenarios for the prac, etc? To people who have done the exam....any helpful hints ? Thanks,

I have my list view filtering rows shown within the table <tr> tags like most of the introductory tutorials do. As I''m adding a few dozen entries I''m noticing the standard scaffolding pagination doesn''t pick up the number of rows from the filter since it is not being passed the number of rows returned within the scope of the filter statement. >From a

Hi all, Here is the script i want to make - Caller call to a number to record a message - Asterisk answer and start recording message as following + User press * to start recording + Record is finished if: + User press # + OR message duration reach 60 second + Hangup How do you counting down 60s, and how to detect # (i make a test using Read() but it cant read #) Thanks in advance

I am using asterisk 1.6.1.6 and have been setting up a system to use a Postgresql database as the realtime DB via the ODBC route. I have got extensions and voicemail working but am having trouble with SIP The problem seems to be with using a schema. If I put the table "sip" in the schema "foo" then I add this entry to extconfig.conf sippeers => odbc,psqldb,foo.sip Restart

Hi guys I seem to be having a problem, I don't know if it's a bug or whether I'm just doing it incorrectly. I want to set about 3 channel variables when I originate a call via AMI. All the documentation I have found says to do it like this: Variable: variable1=value|variable2=value|variable3=value However when I do this it runs them all together and I end up with: