similar to: recording in mp3

Problem with this is client needs to listen to the call recordings and my interface will only display .wav or .mp3 so they will moan if they have to wait until the next day for today's recordings Sent from Samsung Mobile <div>-------- Original message --------</div><div>From: binary <dreamer.binary at gmail.com> </div><div>Date:01/07/2014 6:09 PM

Currently using tikal crystal call recording Do you guys know of any better ones? Sent from Samsung Mobile <div>-------- Original message --------</div><div>From: binary <dreamer.binary at gmail.com> </div><div>Date:01/07/2014 6:33 PM (GMT+02:00) </div><div>To: asterisk-users at lists.digium.com </div><div>Subject: Re:

Hi, I am getting *Can't provide secure audio requested in SDP offer* with sipml5 client hosted on my local system [1060] ; This will be WebRTC client type=friend username=1060 ; The Auth user for SIP.js host=dynamic ; Allows any host to register secret=sameer ; The SIP Password for SIP.js encryption=yes ; Tell Asterisk to use encryption for this peer avpf=yes ; Tell Asterisk to use AVPF

Hi, I had tried all the steps which I used to inatall Asterisk 12.3.2 Pjsip in Asterisk 12.3.2 is working but in new release Asterisk 12.4.0 it is not working I am getting XXX in make menuselect resource_module. I tried all trouble shooting steps along with ldconfig etc. I think its a bug can any one help me on this ? -- Regards Sameer Rathod 8109413462 -------------- next part

Hi, I am new to asterisk I want to configure my asterisk server such that it only establishes the call rest the audio must bypass the server and transmitted directly to the peer In my config file I did changes which are below canreinvite=yes nat=force_rtp dirtectmedia=yes directsetup=yes I am using asterisk version 12.3 -- Regards Sameer Rathod 8109413462 -------------- next part

Dear concern, I want to configure packet2packet bridging in asterisk. How could I do this any of the tutorial or instructions will help ? I found the setting the canreinvite=yes will do the stuff but it is not working I am using asterisk 12.3 version I am very new to asterisk please help me in doing the same. Thanks in advance. -- Regards Sameer Rathod 8109413462 -- Regards Sameer

Hi, with canreinvite=no and directmedia=no I and getting the message in the logs for all calls "switching from simple_bridge technology to native_rtp" -- Executing [102 at mkg:1] Dial("SIP/101-00000017", "SIP/102") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/102 -- SIP/102-00000018 is ringing -- SIP/102-00000018 answered SIP/101-00000017

Hi, I am using chrome version 36 and opera with asterisk 11.9.0 and cent os I am getting the below error if i do call on sipml5 from blink 1. Failed to set remote offer sdp: Called with SDP without DTLS fingerprint. tsk_utils.js?svn=224:128 1. tsk_utils_log_errortsk_utils.js?svn=224:128 2. tmedia_session_jsep01.onSetRemoteDescriptionError

Hi, I want to authenticate user with a random authentication key before registration in asterisk for a click2dial feature in my website. The goal is to not to display the password to the client. The client will be provided with a authentication key and when the request comes to the server form the web browser (via webrtc) it will fetch the relevant userId and password, register the sip and the

Hi, I have updated the review here: https://reviews.llvm.org/D21723 As Sameer pointed out, the motivation is: In OpenCL 2.x, two atomic operations on the same atomic object need to have the same scope to prevent a data race. This derives from the definition of "inclusive scope" in OpenCL 2.x. Encoding OpenCL 2.x scope as metadata in LLVM IR would be a problem because there cannot be a

Hi Preston, I was wondering ... "Bottom" is a bit overloaded as far as terms go. Would SCEVNaN be a better name for this beast? Sameer. > -----Original Message----- > From: llvmdev-bounces at cs.uiuc.edu [mailto:llvmdev-bounces at cs.uiuc.edu] On > Behalf Of Sameer Sahasrabuddhe > Sent: Monday, October 08, 2012 9:16 AM > To: preston.briggs at gmail.com > Cc: LLVM

On 1/9/2015 4:14 AM, Chandler Carruth wrote: > On Wed, Jan 7, 2015 at 8:03 PM, Sahasrabuddhe, Sameer > <sameer.sahasrabuddhe at amd.com <mailto:sameer.sahasrabuddhe at amd.com>> > wrote: > > Here's what this looks like to me: > > 1. LLVM text format will use string symbols for memory scopes, > and not numbers. The set of strings is target

> On Aug 17, 2016, at 2:08 PM, Zhuravlyov, Konstantin <Konstantin.Zhuravlyov at amd.com> wrote: > > >Why not going with a metadata attachment directly and kill the "singlethread" keyword? Something like: > >Something like: > > cmpxchg i32* %addr, i32 42, i32 0 monotonic monotonic, 3, !memory.scope{!42} > > cmpxchg i32* %addr, i32 42, i32 0 monotonic

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Test A: Outside line calling in

On Tue, Jan 13, 2015 at 10:27 PM, Sameer Sahasrabuddhe < sameer.sahasrabuddhe at amd.com> wrote: > Ping! We need to close on whether everyone is convinced that symbolic > memory scopes have a significant advantage over opaque numbers. Either of > them will be examined by optimizations using a target-implemented API. I > personally don't think that readability in the LLVM

On 11/19/2014 4:05 AM, Chandler Carruth wrote: > > On Fri, Nov 14, 2014 at 1:09 PM, Sahasrabuddhe, Sameer > <sameer.sahasrabuddhe at amd.com <mailto:sameer.sahasrabuddhe at amd.com>> > wrote: > > 1. Update the synchronization scope field in atomic instructions > from a > single bit to a wider field, say 32-bit unsigned integer. > > > I

Hi, I have Asterisk set up on Fedora with a single SIP trunk, with a few handsets configured. The Asterisk box has both public and private addressing, so "canreinvite=no" is set on both the SIP trunk and handset configurations so I can get around the nasty NAT issues. One odd behaviour I am seeing is certain destinations are resulting in different SIP codes being sent back to Asterisk,

I've not had a good chance to look at the patches in detail, but just to clarify one point: I don't really care whether we number things going up or down from single threaded to "every thread". I just think it makes sense to expose them in the in-memory IR interface as an enum with a particular ordering so that code can use the obvious sorts of tests for comparing two orderings

> On Nov 18, 2014, at 2:35 PM, Chandler Carruth <chandlerc at google.com> wrote: > > > On Fri, Nov 14, 2014 at 1:09 PM, Sahasrabuddhe, Sameer <sameer.sahasrabuddhe at amd.com <mailto:sameer.sahasrabuddhe at amd.com>> wrote: > 1. Update the synchronization scope field in atomic instructions from a > single bit to a wider field, say 32-bit unsigned integer.

Hi All, I am trying to write a machine learning algorithm that would predict a good optimization to apply to a given function. I wanted to perform this by modifying the list of optimizations applied by the FPPassManager based on some dynamic data. At present I was trying to do this by modifying the FPPassManager::runOnFunction(Function) implementation. However this is causing the following error: