similar to: authentication user with custom authentication key

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

Hey guys Is it possible to record with mixmonitor straight into mp3. I am trying to reduce disk space and want my calls to be recorded in mp3 Instead of wav. Sent from Samsung Mobile <div>-------- Original message --------</div><div>From: Sameer Rathod <sameer at hostnsoft.com> </div><div>Date:30/06/2014 9:23 PM (GMT+02:00) </div><div>To:

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

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:

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

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 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

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

Had I have been smart originally I would have done this to start. Some rudimentary documentation above and beyond Asttapi 0.10's poor documentation is available along with the download at http://www.kirkhamsystems.com/asttapi. Clint -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kerry Garrison Sent:

I've finished a patched version of asttapi that will work with asterisk 1.2. There were fundamental changes to the Asterisk Management interface between 1.0 and 1.2 that broke asttapi. I think my patched version will work on 1.0 and 1.2 branches, but I have no way of testing since I don't have a 1.0 install nor do I want one :). I'm looking for testers, if anyone's willing to

Yes, I have the exact same problem. :( -----Original Message----- From: Tomislav Vojvodic [mailto:tomislav@vox-mundi.net] Sent: Thursday, May 11, 2006 5:48 AM To: xytek@hotmail.com; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial Hey, thanks for your reply.. ;) I'm also using asttapi from website you posted

Hi, I'm wondering whether or not it is achievable to build a web based click2dial application that could automatically detect that a user is connected from office or home. Another option is to directly ask user or let them change default option but having this automatically detected is a bonus. Has anyone tried to build such location based SIP services ? I've read few lines about

The password is related to a rsync server, when you communicate through the rsync port: grep rsync /etc/services rsync 873/tcp # rsync rsync 873/udp # rsync the authentication therefore is done against the rsync serverpassword. Yo wanted to "file in" the ssh password? Not possible in this way ... Rainer -----Urspr?ngliche

Hello Opus community, I'd like to ask you for advice and recommendations. WebRTC uses Opus, and I noticed https://webrtc-codereview.appspot.com/5549004 started referring to currently internal Opus headers. This is possible because for Chromium the Opus sources are just checked in, so any header can be #included. I detected this when trying to package Chromium for Linux distributions with

> On Aug 21, 2016, at 11:14 AM, Philip Reames <listmail at philipreames.com> wrote: > > On 08/17/2016 03:05 PM, Mehdi Amini wrote: >> >>> On Aug 17, 2016, at 2:08 PM, Zhuravlyov, Konstantin <Konstantin.Zhuravlyov at amd.com <mailto:Konstantin.Zhuravlyov at amd.com>> wrote: >>> >>> >Why not going with a metadata attachment directly

Hi, When implementing click2dial feature, I can trigger an Aastra phone to auto-answer using statement like : SIPAddHeader(Alert-Info: info=alert-autoanswer); This is very convenient when trying to reach a distant party (ie through PSTN) The trouble is when 2 Aastra are calling each other over the LAN, this single statement is memorized somehow and both phones (caller and callee) auto-answer.

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

Hey all, So we have been advised from this thread https://github.com/muaz-khan/WebRTC-Experiment/issues/28#issuecomment-18385702 to not use http put as it is not in real-time, instead they are suggesting the use of SDP, is that something that icecast supports? Or does anyone have other ideas on this? ~stephen On Sun 12 May 2013 01:51:31 AM CDT, Thomas Ruecker wrote: > Hi, > > On 11

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