similar to: AXVoice Server Hacked.. accounts info leaked

Dear R-help subscribers Would you tell me how to apply SSfpl with binary data as below? Unfortunately, there is not the EXAMPLE in help(SSfpl) for binary data but for quantitative data(Chick). V1: dose V2: log-transformed dose V3: response (rate) V1 V2 V3 1 0.775 -0.2548922 0.1666667 2 5.000 1.6094379 0.8148148 3 10.000 2.3025851 0.5000000 4 20.000 2.9957323

Hi all, I am using an asterisk as a client to connect to another asterisk server by registering with the register string. Registration is done without any hassel, but after sometime my asterisk loses the registration with the server and the server starts displaying the following msgs repeatedly: [Aug 9 06:37:59] NOTICE[8380]: chan_sip.c:8151 check_auth: Correct auth, but based on stale nonce

Hi all, Some of my asterisk users have used their maximum call limit for incoming calls (peers). There incoming call limit should automatically reset to zero after hangup but its not happening and they no longer can recieve any calls as their allowed limit is already full. So is there any way to reset the call limit on peers by commands or do i have to restart my asterisk server? -- Best Regards

Hi All, I am in desperate need of this feature. I want to play background music during a call while the 2 parties are having some lovely conversation (or maybe give them a sort of cursing background if they are cursing each other). I found this post which talks about creating a ghost call with the help of queues and putting that queue in a meetme room where queue will play the song/curse and the

Dear R-ers Could you tell me how to build DLL from Fortran source and use the DLL in R? Environments: Windows 98 Fortran compiler : DIGITAL Visual Fortran Optimizing Compiler Version: V5.0 R v1.2.3 EXAMPLE SOURCE : FOO.FOR ------------------------------------- SUBROUTINE FOO(I,J) INTEGER I,J I=0 J=I+1 END ------------------------------------- I compiled FOO.FOR with DLL option.

Hi all, when a user dials any number, asterisk automatically generates ringing which caller can hear, and after 2 - 3 rings asterisk detects that the called user is busy, then caller hears busy tone. for example user hears--- tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing at the start so that user hears only beep beep beep if the called user is busy. I have used the R

Hi all, I need a fully developed web based remote provisioning system. I cant find anything reliable on the internet. Have already checked ataconfig.com and voxilla-ays.com. have tried to contact them but got no response. So if anybody knows a good provisioning system then plz tell me about it. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part

Full_Name: Leon C. Banza Version: 2.8.1 OS: Windows Vista Submission from: (NULL) (195.69.243.253) I believe there is a serious bug in the code for sub and gsub. I do as following: gsub("\\", "-", "some\\backslashed\\string") This ought return "some-backslashed-string", but prints warnings instead: Error in sub("\\", "",

Can anyone (David?) shed some light on what exactly this warning is complaining about? I started seeing it when I upgraded to Rails 2.1 RC1 and the latest RSpec from git. I''m getting it in some helper specs that I''m writing. And yes, I am including a module there, to reuse some utility methods I originally wrote for my controller tests (I''m switching to specs

Hi all, i need sample xml configuration files for linksys pap2, linksys pap-2t, sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are linksys/sipura products. So if anyone has these sample files then plz share. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, This is the first time i am using stun with asterisk for nat problems. I have read the rfc which describes how stun works. i didnt have any problems understanding it. I have also intalled the stun server called stund which i downloaded from sourceforge. I have seen on the list that most people use stund here. I have started the stun server and its running silently. Now i dont know what to

Hi all, Is there anyway i can prevent multiple sip registrations from different IPs using single username in asterisk. Does asterisk provide any aid in this respect? As far as my knowledge is concerned i dont think there is any support for this in asterisk, so i think i'll have to makeup a script which sniffs sip packets coming for asterisk and detect for multiple register requests coming from

Hello list, i have one question related to call forwarding. i have 2 number for the inbound and i want to configure asterisk like that. When the customer call the first number 0522XXXXXX the call will be forwarding automatically to anther number 0520xxxxxx Does anybody have a solution to this problem. Thanks and Regards. -------------- next part -------------- An HTML attachment was

Dear R-users I would like to apply classification and regression tree(CART) to the following data. I have some question on using 'tree' package. The data contains one response variable Y and five explanatory variables. The explanatory variable "x2" is categorical and not ordinal. But, the result obtained after running following R code has indicated that x2 is regard as

I'd like to grab the SIP response code that comes back from an INVITE. The HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get the SIP response code instead? Doug. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML

Hi all, i need an XML file format which is used in remote provisioning of different spa devices. Please somebody tell me the format or tell me where can i find it on the internet. I also need a list of parameters which are configured using auto-provisioning. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was

im having a strange problem related to call-limit for peers. well im not sure if its related to call-limmit or not. Bottom line is: I call a user A, from user B. user B hears silence, untill it goes to voicemail. when user B hangsup. user B's call limit is reset to 0 but user A's call limit is not reset.strange thing is user A's status on cli is shown as NOANSWER, while user B did not

Hi, I was trying out call file just to see how they worked and my system does not seem to do anything with them, although asterisk *is* deleting the files that I put into /var/spool/asterisk/outgoing. 1. I nano'd a quick call file like so: Channel: SIP/axVoice/9105555555 CallerID : Leebo <5555555555> MaxRetries: 2 RetryTime: 30 WaitTime: 10 Context: main_menu Extension: s Priority:

Hi list, yesterday I converted my voicemail.conf to realtime voicemail and also configured to store the voicemessages in a database using odbc as described here <http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail> and here <http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage>. I am using asterisk 1.4.2 with mysql. I also installed the proper odbc driver for

Hello there! We are planning to use Asterisk on our VoIP platform, and we are spending some brains on a way to provide the following facility: let some SIP user (extension) registrate with more than one client (ATA, SoftPhone, VoipCelular, etc) - what isn't a problem at all -, initiate calls from any of this devices that are registrated with the same user - no problems on tests too -,