search for: newcall

...ontinue; SmallVector<Value*, 16> callargs(call->op_begin(), call->op_end()); callargs.insert(callargs.begin(), ConstantInt::get(Type::getInt32Ty(context), call->getNumArgOperands())); callargs.insert(callargs.begin(), callee->getName()); CallInst* newcall = CallInst::Create(launch, callargs, "", call); newcall->takeName(call); newcall->setCallingConv(call->getCallingConv()); newcall->setAttributes(call->getAttributes()); newcall->setDebugLoc(call->getDebugLoc()); call->replaceAllUsesWith(new...

...hich so far has performed admirably. In particular, it uses ls() and not its explicit source code, so only has a dependency on its name and the name of its all.names argument. Here is my solution: lsall <- function(...) { thecall <- as.call(c(as.name('ls'), list(...))) newcall <- match.call(definition=ls, call=thecall) if( !('all.names' %in% names(newcall)) ) newcall[['all.names']] <- TRUE eval(newcall, envir=parent.frame()) }#### end lsall In my hands this function has always acted exactly as I expected, identically to ls() but with...

...ConstantInt::get(Type::getInt32Ty(context), call->getNumArgOperands())); callargs.insert(callargs.begin(), callee->getName()); CallInst* newcall = CallInst::Create(launch, callargs, "", call); newcall->takeName(call); newcall->setCallingConv(call->getCallingConv()); newcall->setAttributes(call->get...

...y mobile. I'm using voiptalk.org for incoming and outgoing calls and SIP phones for extentions (so all IP based - no real phone lines). I tried this (from voip-info.org wiki)... exten => 1234,1,dial(sip/1234,20) exten => 1234,2,playback(pls-wait-connect-call) exten => 1234,3,Setvar(NewCaller=${CALLERIDNUM}) exten => 1234,4,SetCIDNum(0${CALLERIDNUM}) exten => 1234,5,dial(${TRUNK}c/9871234321,20,r) exten => 1234,6,SetCIDNum(${NewCaller}) exten => 1234,7,voicemail2(u1234@default) exten => 1234,101,voicemail2(b1234@default) exten => 1234,102,hangup Mine looks like this...

Hi Everyone, I am writing an open source application that brings desktops widgets to OS X (http://sourceforge.net/projects/astrxtools4osx/), for which I am trying to get my head around the Asterisk Manager Interface. I had been using the Event: NewCallerid to detect a new call which my Asterisk server doesn't seem to send to the socket anymore, because of which I have reverted to using Event: Newexten. Which is the most efficient way of monitoring if a new phone call is coming my way? Also my application will only monitor a single extension,...

Hi, as noone out there seems to be able to maintain chan_sccp, i'm trying to switch to chan_skinny. With the newest 1.4 svn the Softkeys are mostly wrong/non functional. I see Redial NewCall CFwdAll more (more) CFwdBu... GPickUp Confrn more NewCall works, CFwdAll seems to toggle DnD, the rest of the buttons do notting. Any ideas how to fix this? Regards, Andreas _________________________________________________________________ Live Search delivers results the way you like it....

...llow+me) but I am having a brain fart as I haven't a clue where to get started with what to do with this. From my main menu, I want the extension 300 to execute the script as follows: exten => 300,1,dial(sip/200,20) exten => 300,2,playback(pls-wait-connect-call) exten => 300,3,Setvar(NewCaller=${CALLERIDNUM}) exten => 300,4,SetCIDNum(0${CALLERIDNUM}) exten => 300,5,dial(${TRUNK}c/2831385,20,r) exten => 300,6,SetCIDNum(${NewCaller}) exten => 300,7,voicemail2(u200@default) exten => 300,101,voicemail2(b200@default) exten => 300,102,hangup Regardless of what (and where)...

if i go off hook and dial 666 from an internal sipura spa-x000 (at extn 141), it rings straight through to extn 666. using the same dialplan, from a cisco 7960 with 7.1 sip code (at extn 142), i have to go off hook hit NewCall punch 142 (or any valid extn in the dialplan) hit Dial then dial 666 wtf? sip.conf for crisco [fiji] callerid="crisco" <142> type=friend host=dynamic port=5060 secret=pfui qualify=1000 dtmfmode=rfc2833 canreinvite=yes context=in-i...

...'voicemail/default/9969/busy' (language 'en') -- Playing 'vm-intro' (language 'en') -- Channel 0/1, span 1 got hangup This is the full dialplan for this extension - exten => 9969,1,Wait(1) exten => 9969,2,Dial(SIP/9969,20) exten => 9969,3,Setvar(NewCaller=${CALLERIDNUM}) exten => 9969,4,SetCIDNum(${CALLERIDNUM}) exten => 9969,5,Dial(Zap/g1/13169321000,10,m) exten => 9969,6,SetCIDNum(${NewCaller}) exten => 9969,7,Voicemail2(u9969) exten => 9969,102,Voicemail2(b9969) exten => 9969,103,Hangup Now the CNAM is very important and I w...

I'm a day-old R newbie (but a war-weary S veteran), with couple of first-day questions: In R 0.61, this code fails. Ops.test <- function(e1,e2) { e1 <- NextMethod(.Generic) e1 } x <- 4 class(x) <- "test" y <- x < 3 The error message is "Error in NextMethod(.Generic) : negative length vectors are not allowed.". I assume it is a bug.

Hello People, I was wondering if you could take a look at this script, exten => 505,1,dial(iax2/6311${EXTEN},t,25) exten => 505,2,playback(pls-wait-connect-call) exten => 505,3,set(NewCaller=${CALLERID(num)}) exten => 505,4,Set(CALLERID(num)=0${CALLERID(num)}) exten => 505,5,dial(Zap/g1/c/0296389675,20,r) exten => 505,6,Set(CALLERID(num)=${NewCaller}) exten => 505,7,voicemail(u6311503@default) exten => 505,101,voicemail(b6311503@default) exten => 505,102,hangu...

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...srvlookup=yes disallow=all allow=ulaw [mustafa] type=friend secret=mustafa host=dynamic dtmfmode=inband mailbox=1604 extension.conf: exten => 20531604,1,Dial(SIP/mustafa,20) exten => 20531604,2,playback(pls-wait-connect-call) exten => 20531604,3,Setvar(NewCaller=${CALLERIDNUM}) exten => 20531604,4,SetCIDNum(0${CALLERIDNUM}) exten => 20531604,5,dial(${TRUNK}/0193041624,20,r) exten => 20531604,6,SetCIDNum(${NewCaller}) exten => 20531604,7,VoiceMail(u1604) exten => 20531604,8,Hangup can anyone give an advise or some idea?? please.. thanks.....

...the hopes of finding a more diverse user community. I am using a Cisco 7960 (version 6.0 SIP firmware) with Asterisk, and have been experiencing situations where the phone locks up. "Locks up" means that the bottom part of the screen ("Your current options" and the redial/newcall/cfwdall keys) disappears, and all keys on the keypad are non-functional except for the *-6-settings reboot keys. Previous software (4.4) exhibited the same symptoms. I replaced the 7960 with a brand new 7960G, thinking it was bad hardware. Same symptoms. I replaced the power supply, thinking...

...(); i != e; ++i) { Args.push_back(i); } if("sleep" == CI->getCalledFunction()->getName()) { CallInst *NewCall = CallInst::Create(hook,Args.begin(), Args.end(), "hook", (Instruction *)BI); } } } return true; } }; } char bishe_insert::ID = 0; static RegisterPass<bishe_insert> X("bishe_insert", &quo...

Well, I'm about ready to throw Asterisk across the room. Can someone tell me WHY, when you've sent a Dial command to Asterisk via AGI, if the callee hangs up the call, Asterisk sends a return code, but if the caller hangs up, it does not??? This means if an agi script services a call, and after the two parties have finished speaking, the person who initiated the call hangs up, the agi

I have need of a very simple callback function - when any call is made to a special SIP DID, the call is not answered but Asterisk then calls a pre-determined number - no need for CallerID to capture the calling number. Does anyone have a simple script to do this? Chris -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

...eed to insert a Proceeding() statement very early in your dialplan, otherwise an inbound overlap call may retry and fail. Our experience was from a PRI connected PABX, if we took too long doing database lookups, the PABX would resend the SETUP, asterisk doesn't handle these yet, treats as not newcall and rejects it. Please feedback to the bug. Currently the patch is for Asterisk 1.6.1, but the issue applies to all. Alec Davis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100210/9435ba86/atta...

Is there a way to make the Cisco 7960 SIP phone dial out automatically without having to press the dial button, once the numbers that you have entered match a specific pattern? This feature is present when the phone is working with a Cisco CallManager. For example, if all of my internal extensions begin with a '5' and are four digits long, if I dialed '5123' on the phone, the