similar to: Broken Call Screening

Displaying 20 results from an estimated 500 matches similar to: "Broken Call Screening"

2012 Oct 14
6
[Bug 1963] IPQoS not honoured
https://bugzilla.mindrot.org/show_bug.cgi?id=1963 --- Comment #5 from martin f. krafft <bugzilla.mindrot.org at pobox.madduck.net> --- With reference to http://bugs.debian.org/650512, which I just reopened, I am sorry to say that the bug persists in OpenSSH 6.0. -- You are receiving this mail because: You are watching the assignee of the bug. You are watching someone on the CC list of the
2005 Feb 02
3
Boxplot by factors
Dear all, I have the following data format cellnumber force 1 100 1 230 1 100 1 200 1 130 1 210 2 179 2 298 2 400 2 500 2 600 ........... I want to make a boxplot of the force according to the cellnumber. Here the cellnumber is actually a factor. It has 1, 2 two levels. How can I do that using boxplot? Thanks in advance Ming
2010 Mar 10
35
[Bug 1733] New: Enhance support for QoS (ToS) by supporting DSCP/CS and adding option
https://bugzilla.mindrot.org/show_bug.cgi?id=1733 Summary: Enhance support for QoS (ToS) by supporting DSCP/CS and adding option Product: Portable OpenSSH Version: 5.4p1 Platform: All OS/Version: Linux Status: NEW Severity: enhancement Priority: P2 Component: ssh AssignedTo:
2007 Sep 25
1
Configure one call per line on Cisco 7941/7961
Anyone aware of how to configure one call per line on a Cisco 7941/7961? The default behaviour is to have two calls per line button, and this is confusing for some of my users so I'd like to be able to have the 2nd call ring the second line button, rather than being shared with the first. I'm hoping this is something that is configurable in the XML or on the phone UI. Thanks Gary
2018 Nov 09
3
Proposed new min and max intrinsics
On Thu, Nov 8, 2018 at 11:35 PM Fabian Giesen via llvm-dev < llvm-dev at lists.llvm.org> wrote: > What is so complicated about these? Shouldn't they just correspond to > two compares + selects? > > To give a concrete example, x86 MIN[SP][SD] and MAX[SP][SD], > respectively, correspond exactly to > > MIN*: select(a < b, a, b) (i.e. "a < b ? a : b")
2018 Jul 02
2
Rotates, once again
On 7/2/2018 3:16 PM, Sanjay Patel wrote: > I also agree that the per-element rotate for vectors is what we want for > this intrinsic. > > So I have this so far: > > declare i32 @llvm.catshift.i32(i32 %a, i32 %b, i32 %shift_amount) > declare <2 x i32> @llvm.catshift.v2i32(<2 x i32> %a, <2 x i32> %b, <2 x i32> %shift_amount) > > For
2019 Sep 02
2
AVX2 codegen - question reg. FMA generation
On Mon, 2 Sep 2019 at 16:59, Roman Lebedev <lebedev.ri at gmail.com> wrote: > > It appears you need 'reassoc' on fmul/fadd: > https://godbolt.org/z/nuTzx2 Thanks very much, that was it. Either that or providing -enable-unsafe-fp-math to llc yielded FMAs. I didn't expect this since using FMAs here instead of mul/add appears to be safer (the reverse is unsafe). ~ Uday
2018 May 14
5
Rotates, once again
Hi everyone! I recently ran into some interesting issues with generation of rotate instructions - the details are in the bug tracker (https://bugs.llvm.org/show_bug.cgi?id=37387 and related bugs) for those interested - and it brought up the issue of rotates in the IR again. Now this is a proposal that has been made (and been rejected) several times, but I've been told that this time round we
2001 Aug 26
4
On the &quot;broken&quot; .WAV files issue
A friend of mine recently had a problem with a "broken" .WAV file (as you call them) because oggenc first printed out a warning and then didn't accept the file because of a "unexpected EOF error". Because I was interested in the issue, I decided to take a look at the oggenc source, and in fact, it is your .WAV reader that's wrong. More precisely: there are two version
2018 Nov 08
2
Proposed new min and max intrinsics
Alex, After looking into this a bit, it looks to me like the best thing to do for targets that do not natively support ISD::MINIMUM and ISD::MAXIMUM would be to fall back to a libcall, since implementing these operations in terms of existing operations is actually rather complicated. Do you think it would make sense to add builtin functions to compiler-rt to implement these operations, or is
2019 Oct 20
2
Matrix Multiplication not Vectorized using double pointers
Hello, My matrix multiplication code has variables allocated via double pointers on heap. The code is not getting vectorized. Following is the code. int **buffer_A = (int **)malloc(vecsize * sizeof(int *)); int **buffer_B = (int **)malloc(vecsize * sizeof(int *)); for(p = 0; p < vecsize; p++) { buffer_A[p] = (int *)malloc(vecsize * sizeof(int)); } for(p = 0; p < vecsize;
2007 Feb 02
1
WARNING[4218]: res_features.c:1385 ast_bridge_call: Bridge failed on channels ( when I use asyncgoto)
Hi All, I download the app_asyncgoto.c, compile the app_asyncgoto.so. Then according to this page http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO ; when I dial ,there have this warning: -- Executing AsyncGoto("SIP/111-086497c8", "SIP/113-08674628|dynamic-nway|111|1") in new stack Feb 2 16:53:10 DEBUG[4218]: app_asyncgoto.c:95 asyncgoto_exec: Attempting
2018 May 15
0
Rotates, once again
Thanks for writing this up. I'd like to have this intrinsic too. Another argument for having the intrinsic is shown in PR37426: https://bugs.llvm.org/show_bug.cgi?id=37426 Vectorization goes overboard because the throughput cost model used by the vectorizers doesn't match the 6 IR instructions that correspond to 1 x86 rotate instruction. Instead, we have: $ opt 37426prevectorize.ll -S
2011 Sep 27
1
Screening Mode Ghost
Hi, It seems there is random behavior that causes screening mode to be activated when a user calls and the line answered and then forwarded using a dial command such as: EXEC "Dial" "SIP/13365551212 at 8x8|60&SIP/13365541212 at 8x8 |60&SIP/13365531212 at 8x8 |60|dgF(callFlo-in^3^1)M(record^39ff6274-c0f0-453d-aa05-402a7bd6d567^)" -- AGI Script Executing
2008 Mar 18
2
call screening feature
Hi, I have our software with SIP running on it.I configured asterisk server as proxy. How do I implement the call screening features(incoming and outgoing) using asterisk server.Please suggest me how to proceed on this. Thanks & Regards, Jahnavi. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2007 May 21
0
AGI: Festival & Ringing on Screening not working properly
I am running into two problems: 1) The ringing stops during call screening once the extension picks up (but has not yet approved call) When a person calls and choose an extension, the Dial link is called and the person hears the ring -- but as soon as the receiving caller picks up (even though they have to approved the call), the ringing stops... the person calling hears silence. This is
2003 Oct 21
0
CallerID Screening Prohibit
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, How can I check if (i.e.) my provider is requesting me to hide the callerid? I.e. (Telco)E1/PRI---Zap(Asterisk1)IAX---IAX(Asterisk2)SIP---EP Now, if a call comes from the Telco with CLI screening prohobited to Asterisk1, where the call is forwarded using Dial() via IAX to Asterisk2 and then (also using Dial()) on to a SIP endpoint, how do I
2007 Jul 05
2
Call Screening Not Working
I am using the Find-me/Follow-me example below with screening: http://www.voip-info.org/wiki/view/Asterisk+tips+findme Here is my actual config: [macro-screen] exten => s,1,Wait(1) exten => s,n,Background(press-1-to-be-connected-to-the-caller) exten => s,n,Set(TIMEOUT(response=5)) exten => 1,1,NoOp(Caller accepted) exten => i,1,Set(MACRO_RESULT=CONTINUE) exten =>
2005 Aug 08
0
Screening Sip Calls - Record()
I've posted about this before, but it's been so long I thought I'd see if there is a new solution (can't find anything on google or wiki) I use the Record() app on my incoming zap calls to record a persons name if their caller id is not in the db. After the name is recorded, the call is parked and ParkAndAnnounce announces that a call is holding from ${SCREENNAME}. Works fine
2006 Jan 05
2
Screening incoming calls.
The PBX I'm getting ready to replace has a really nifty feature -- one that I'm not even sure Asterisk -could- do -- though I'm hoping to be proven wrong. When a call goes to voicemail, the end-user can listen to the VM as it's being recorded, and can interrupt and answer the call if it's someone they want to talk to. Is there any way to implement this? Thanks, -Ken