similar to: Play sound to callee

Hello ! I want to call a line and play a sound from the callee before putting it in connection with the caller. Is this possible? Example: Dial(SIP/111111, m) // Ring or Music... if(call==ANSWERED) Play(announce) // Play 'announce' to the called // To connect caller and called ? Best regards, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I'm having problems with LIMIT_PLAYAUDIO_CALLEE in the Dial application. I want to play the limit file to both caller and callee at the same time, but it plays the limit file first to the caller and then to the callee. I searched the list and found someone with the same problem back in '06, but couldn't find any solution for the problem :( Anyone knows? Thanks, Best regards,

Hi . I already set-up sendmail on another network server thus it would be nice to use that one or is sendmail on * server required!? I had a look in the archive but couldn't find any information where to set the mail server from localhost to my network server . Cheers Ralf -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am using EQTDDataStructures (from the poolalloc project) to resolve indirect function calls to over-approximated sets of possible callees. Unfortunately I find that it yields incorrect results even on a very simple test input. My LLVM and poolalloc sources are Subversion trunk checkouts, no more than a day older than the current trunk head. My test input is the following C source,

On Tue, Aug 9, 2011 at 6:19 PM, Ben Liblit <liblit at cs.wisc.edu> wrote: > I am using EQTDDataStructures (from the poolalloc project) to resolve > indirect function calls to over-approximated sets of possible callees. If I remember correctly, it only tries to resolve indirect calls. The analysis doesn't track direct calls because you can do it just as well yourself. Andrew

You can send both paties to a meetme conference with Manager Redirect. Or if you are feeling more adventurous you could load the Manager Bridge patch that I posted to the bugtracker two months ago. It allows bridging of any two existing channels together through a manager action: http://bugs.digium.com/view.php?id=4297 MATT--- -----Original Message----- From: Roland Zagler

sorry for the misunderstanding, robert! the point is: during the caller is listening to the soundfile played to him the dialplan should continue to dial the sip phone 100 and after sip phone 100 is answered and the announcement file is played the caller should be connected to the sip phone 100. the behaviour is just like MoH, but the problem is, that the caller has to hear a soundfile from the

try this one exten => 999,1,Answer() exten => 999,2,playback(~.mp3) exten => 999,3,dial (sip/100) exten => 999,4,playbackground(~.mp3) exten => 999,h,Hangup() not sure abt playbackground should be before the dial command or after ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Roland Zagler Sent: Sat 7/2/2005 8:23 PM To:

yes, robert, but how do i "join" the two legs inside a call file or in the dialplan? i have experienced that call files can do a call to a channel and if this call is answered it can either be connected to an extension inside a context or call an application with parameters. roland -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Hi, Is there a way to "announce" a call to the callee? For instance, I've answered an incoming call, collected some info and now want to ring an extension, and make an accouncement to that extension before connecting the caller through. Thanks, Steve

Hello, Let say Alice and Bob both have a sip phone connected to the same asterisk 11 box. Alice has T.38 enabled softphone. When Alice sends a fax to Bob extension, the following happens on my system: - Bob phone starts to ring - Bob answers - asterisk sends the incoming call to appropriate fax extension - Bob is hearing nothing at all: no tone, no sound at all. I want to play an audio file

Thank you, Robert! The announcementfile plays well, but at Dial-option "m" i have to specify a MoH class, that is something i cannot use (as i wrote in my post). Background command waits for a user input, but the caller should be connected to SIP Phone 100 after it has answered and the announcement has been played. Before connecting to SIP Phone 100 the caller should hear a

Thanks for the suggestion, C F, but the problem is there is a rather big database application behind with many users, so a static configuration is not suitable for my needs. i am working mostly with realtime and agi. regards, roland -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F Sent: Sunday, July 03,

Hello, I want that, when call is answered , callee and calling would hear different prompts and after promts the calls would be bridged. I've tried this situation: exten => s,1,Set(LIMIT_CONNECT_FILE=hello-world) exten => s,2,Dial(SIP/trunk-out/37052390920|60|rL(10000000000000)A(conf-enteringno)) But these prompts play not in the same time: just after conf-enteringno prompt

Just nitpicking: 1) DI(F) should include a component that estimate the epi/prologue cost (frameSetupCost) which InlinedDF does not have 2) The speedup should include callsite cost associated with 'C' (call instr, argument passing): Speedup(F,C) = (DI(F) + CallCost(C) - InlinedDF(F,C))/DI(F). Otherwise the proposal looks reasonable to me. David On Thu, Jul 30, 2015 at 2:25 PM,

If you have been wondering why the verifier doesn't reject calls and callees with mismatched calling convs, any why the optimizer mercilessly deletes them, there is a new FAQ here: http://llvm.org/docs/FAQ.html#callconvwrong -Chris

2013/8/13 Administrator TOOTAI <admin at tootai.net> > Le 13/08/2013 16:41, Olivier a ?crit : > >> Hello, >> > > Hi > > >> [...] >> >> >> How can I work around this ? >> Suggestions ? >> > > Answer the call, wait few seconds and then ring Bobs extension. If > asterisk detect fax it already sended to fax extension so

Hi. I have a beginner conceptual question about Asterisk: Let's suppose that there are 4 softphones registered in my Asterisk and all of them are currently online. In addiction , there is no call. Suddenly, one of these softphones sends a SIP message to the Asterisk. In this case the dialplan will execute the instruction ' exten => 2005,1,Dial(SIP/2000&SIP/2001&SIP/2002,

Hi. I am using *1 in features to initiate a mix monitor recording. However, when I hit *1, the callee hears the courtesy tone which I have, so I know when the recording is started or stopped. This is a problem, particularly in automated system where the beep is mistaken for a tone or other problems. Should I file a bug, or is this going to be fixed? -- Your life is like a penny. You're

TLDR - The proposal below is intended to allow inlining of larger callees when such inlining is expected to reduce the dynamic instructions count. Proposal ------------- LLVM inlines a function if the size growth (in the given context) is less than a threshold. The threshold is increased based on certain characteristics of the called function (inline keyword and the fraction of vector