search for: newcaller

Hi, I'm trying to replace function call with call to wrapper(function_name, num_args, ...), where varargs hold args of original call. Function* launch = Function::Create( TypeBuilder<int(const char*, int, ...), false>::get(context), GlobalValue::ExternalLinkage, "kernelgen_launch_", m2); { CallInst* call = dyn_cast<CallInst>(cast<Value>(I)); if

Dear useRs, Some time ago I queried the list as to an efficient way of building a function which acts as ls() but with a different default for all.names: http://tolstoy.newcastle.edu.au/R/e6/help/09/03/7588.html I have struck upon a solution which 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

Hi Dimitry, This makes sense if you think about it from the perspective that the string you want passing must be passed at runtime, and so can't use a const char * from compile time. You need to make the string visible in the compiled image, and use that as the argument. A string is an array of 8-bit integers, so you need to create a ConstantArray. Value *v = ConstantArray::get(Context,

...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, s...

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

...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) I...

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>

...'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 wou...

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,hangup...

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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.....

This is almost certainly not an Asterisk-specific posting, but due to my inability to find a VoIP-focused Cisco list, I'll post here in 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

Hello, everyone I am new to LLVM, now I got a problem I want to add a function call before sleep(int a, int b) code below #include <stdio.h> int sleep(int a, int b) { return a+b; } int main(int argc, char **argv) { sleep(1,2); } after use opt -load ../llvm-2.8/Release+Asserts/lib/bishe_insert.so -bishe_insert <1.bc> 2.bc I want get the code #include <stdio.h>

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.

Overlap receiving timeout, plus dialplan latency, causes network to retry SETUP https://issues.asterisk.org/view.php?id=16789 This patch removes the requirement that some may have found that you need 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

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