similar to: Info required

Hi Hisham, Most likely the basic blocks are the headers of two different loops. Try running viewCFG() on the function in question to see if this is the case. Tom ----- Original Message ----- From: "Hisham Chowdhury" <hisham_chow at yahoo.com> To: llvmdev at cs.uiuc.edu Sent: Wednesday, June 16, 2010 7:22:00 PM GMT -05:00 US/Canada Eastern Subject: [LLVMdev] Loopinfo Analysis

hi all, i just finished developing some incoming call features in a macro. that macro gets executed everytime an incoming call is received and a new mysql connection is made using the MYSQL cmd in dialplan. i want to use a single mysql connection for every incoming call. my idea of doing it is like this, i want to get a mysql connection in a global variable, just to share the connection with

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

Hello, I have a question regrading the analysis pass that generates loop info from an .ll code. My previous understanding was there will be just one loop header(in the loop info) for a particular loop. But, when i use isLoopHeader() member function from the loop info class I get 'true' return value for two different basic blocks. Note both basic blocks are loop conditional block(break

Hi, I am doing kernel density plots, and am trying to make the lines thicker. I comparing three groups, in sm.density.compare. I tried changing lwd to make the line sthicker right on the density compare call, but was not able to do it. There is not an option in sm.options to specify line thickness, as well as cex.ylab or cex.xlab- I tried it and it does not change the thickness of the lines.

Hi all, There is a problem with my dialplan. here is the dialplan: exten=> 123,1,Dial(SIP/U1,,Ttg) exten=> 123,2,Hangup exten=> h,1,AGI(onhangup.pl) The problem is whenever U1 is called or calls someone, if U1 hangsup the call then the h extension is NOT executed. but if the other person hangsup the call, then the h extension is executed (assuming that the other person is calling

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

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, 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

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 There, I am configuring for 2 days now new Smaba 4.1.12 with the same annoying results. Any help is very much appreciated. SID S-1-5-21-1390067357-606747145-725345543-7571 -> getpwuid(16777216) failed [2015/04/16 17:59:36.608681, 3] ../source3/auth/token_util.c:316(create_local_nt_token_from_info3) Failed to finalize nt token I can run perfectly well the commands: wbinfo -g

Hi all! I'm new to this forum so please excuse me if I don't conform perfectly to the protocols on this board! I'm trying to get an estimate of true prevalence based upon results from an imperfect test. I have various estimates of se/sp which could inform my priors (at least upper and lower limits even if with a uniform distribution) and found the following code on this website..

Hi guys, i just came to know that CDR(dst) field is set to current extension instead of the dialed no. i need to set it to DNID because our every user has 5 dids and i want to show the caller at the end of the month which numbers he dialed for every call, along with other cdr info. Our rating depends on the dialed number also. here is my extensions.conf exten=> 1212,1,Dial(SIP/rizwan)

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, We are planning to shift our sip users from one platform to another. (basically from one asterisk server to another). the problem we are facing is that both asterisk servers are using different ports to listen for sip. and both have live customers on them. provisioning their ata's is not a good option for us coz of our settup. we cant just ask the customers to change their ports for

Hi all, Im trying dial a user according to the device s/he uses. i mean if the user is using asterisk as a peer, then i have to pass the extension in the dial application like this: Dial(SIP/${EXTEN}@user) ;so that s/he can perform routing according to the DNID. and if the user is using sipura, linksys or grandstream i dial the user like this, Dial(SIP/user) so is there a way to know what kind

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 im trying access the www.voip-info.org website since yesterday but i cant open it. google search diaplay correct search results but it doesnt open when i click the link. it displays a message about tcp error which says -->"There was a problem communicating with the server". I dont know what the problem is. I just want to ask whether their server is down or not and is everybody

  Hello, I am trying to clone a machine basic block when I ran into some issues, where I am not able to make some headway. Any of yours help is highly appreciated here: My question is about Machine Basic Block Duplication:   -          Is there a utility to clone a MachineBasicBlock in LLVM? I found utility to clone machineInstrs, but couldn’t find similar utility for MachineBasicBlock. So, I

On Wed, 2010-04-14 at 17:30 -0700, Hisham Chowdhury wrote: > > > > - Is there a utility to clone a MachineBasicBlock in LLVM > > > > > > There is CloneBasicBlock routine in ./lib/Transforms/Utils/CloneFunction.cpp - Sanjiv > _______________________________________________ > LLVM Developers mailing list > LLVMdev at cs.uiuc.edu