similar to: ICECAST ADMINistration

I would just like to say -- GO FOR IT. Your ideas sound good to me. Ross. Mickael Floc'hlay wrote: >Hi everyone ! > >We are a group of 8 students working (as a scolar work) on ICECAST and >we have thought that it would be a good thing to provide an >administration web interface implemented via SSL and are actually >working on it. > >We have thought of several things

On Thursday 10 April 2003 00:07, Mickael Floc'hlay wrote: > Hi everyone ! > > We are a group of 8 students working (as a scolar work) on ICECAST and > we have thought that it would be a good thing to provide an > administration web interface implemented via SSL and are actually > working on it. > > We have thought of several things this interface could do : > -

Hi, here is a group of 8 students from ENSEIRB (FRANCE, Bordeaux) and we'll work for 6 weeks on Icecast. We'll try to do several things: - make a technical doc. - reorganize the source code. - administrate icecast via web. - make a doc for administration. - make a user doc. <p>We saw the Vorbisfile documentation on www.xiph.org and we'd like to do the same for the icecast

Hello, 3 months ago, we (a group of 8 students of the ENSEIRB engineering school (France)) started to work on icecast. We decided to work on another CVS server. The aim _was not to fork a new version_ of Icecast : the idea was to clean the code as fast as possible in order for new developers to enter the sources in an easier way. We mainly worked on format, sources, and tried to clearly

Hi, Where is the mount option 'limit-rate' in the current version? I checked in cfgfile.c and in the documentation, no mention. Yet this option did exist at one time: http://lists.xiph.org/pipermail/icecast/2010-October/011703.html http://lists.xiph.org/pipermail/icecast/2009-January/011391.html I try to limit the bitrate of a mount-point, is there another solution? Do you know why this

Hello, after the reply of Jerome Alet, I forward our questions to you. I hope it is now the right place to put it. Could you please send your answers, questions, comments to interopcast-general@nongnu.org so everyone in the group can get a copy of it. Thank you. ----- Original Message ----- From: "Jerome Alet" <alet@librelogiciel.com> To: <interopcast-general@nongnu.org>

Hello, Thank you for your response. It is on the kh version.. https://github.com/karlheyes/icecast-kh Le sam. 3 nov. 2018 à 21:47, Thomas B. Rücker <thomas at ruecker.fi> a écrit : > Hi, > > On 11/03/2018 07:33 PM, Mickael MONSIEUR wrote: > > Hi, > > Where is the mount option 'limit-rate' in the current version? > > I checked in cfgfile.c and in the

Hello, I have Asterisk 1.6.2.9-2, the directive "sendrpid" does not work! I placed this in my peer: (sip.conf) sendrpid=yes trustrpid=yes or sendrpid=yes trustrpid=no (and restarted Asterisk) and the line "Remote-Party-ID" does not appear in my sip debug! Please help me, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I am running an AGI script in a command dial, or call a SIP trunk. I want to execute after 10 minutes a voice message (stream file) on the channel to warn the person that the call is about to end. How to do that? Thank you, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello ! 1) I have this error logged by bdrb: undefined method `send_request'' for nil:NilClass framework/packet_master.rb:58:in `ask_worker'' backgroundrb/server/master_worker.rb:59:in `process_work'' backgroundrb/server/master_worker.rb:16:in `receive_data'' backgroundrb/framework/bin_parser.rb:29:in `call'' backgroundrb/framework/bin_parser.rb:29:in

Michael Smith wrote: >On Thursday 31 July 2003 10:56, Pierre Jarillon wrote: > > >>First, can you test interopcast ? On the CVS >>http://savannah.nongnu.org/cgi-bin/viewcvs/interopcast/ENSEIRB/icecast/icec >>ast/ See http://savannah.nongnu.org/cvs/?group=interopcast >> >>interopcast is an improvement of icecast2. We wrote to the author but he

Good morning, or Good afternoon! It depends :-) I have a standard Asterisk configuration: SIP friends (phones) <-----> Asterisk <-----> SIP gateway to PSTN converter 80.236.215.61 109.69.217.6 internal IP ( 10.4.0.10/255.255.255.0) When analyzing traffic on a SIP friend/phone I see this: INVITE sip:xxxx at 80.236.215.61:64946;ob

All, According to my readings CDRs are stored at the end of the call. My concerns is when asterisk goes down (I know that it's never happen but it's just in case) or when the is a power shutdown of the server. then CDRs are not stored in mysql. is there a way to store periodially CDR during a call, and set the periodical timer regarding the context. if no is there a way to retreive CDR,

Hello I just coding a AGI script for billing. - For external calls, I pass the call directly on a trunk. I do : Dial(trunk1/extension) -> OK ! - For internal calls (shortcode, others users ...) I am Dial(Local/extension at context/n) The problem is that through chan_local.so, I sound as it cut! Example if I call the voicemail ... "You have No messa ..." or "You have

Hello, When I get a SIP INVITE as follows: INVITE sip:s at 10.1.0.191:5060 SIP/2.0 Max-Forwards: 69 From: "0475XXXXXX" <sip:1053212 at sip.domain.com>;tag=as7df9ab18 To: <sip:02XXXXXX at IP:5060> Contact: <sip:1053212 at IP:5060> Call-ID: 344d42bd16975a54141d11f635bdfc71 at sip.domain.com CSeq: 102 INVITE Date: Wed, 26 Mar 2014 15:06:01 GMT Allow: INVITE, ACK, CANCEL,

Hi, I am working on Trixbox. I want to create my own dial() function (named specificdial()) and I want to know how I can create a module and integrate the module in the trixbox plateform. thanks a lot Mickael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091006/b69d4103/attachment.htm

Hello, I look ARI (Asterisk Recording Interface) the publisher site is closed... http://www.littlejohnconsulting.com/ari Thank you, Mickael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100623/a8d923ae/attachment.htm

Hello, I have a Cisco AS5300 connected to Asterisk (1.6.2.9) Between 15-16 minutes, the call is disconnected without reason. Here is what is displayed in the debug: Received an SDES from 10.4.0.10:17399 -- Got SIP response 420 "Bad Extension" back from 10.4.0.10 -- Stopped music on hold on SIP/as5300-1-0000004d == Spawn extension (dialin, 065939191, 2) exited non-zero on

Hi, I need your help to use dovecot with sendmail I use a fresh debian 8 jessie and i installed sendmail and dovecot 2. Sendmail works fine. All mails go to /var/mail/xxx this is problem for dovecot. dovecot cannot get it. # 2.2.13: /etc/dovecot/dovecot.conf # OS: Linux 3.16.0-4-amd64 x86_64 Debian 8.1 auth_mechanisms = plain login auth_verbose = yes auth_verbose_passwords = yes

Hello CentOS, I've just taken 2 old servers running CentOS 3.x, wiped it out, and installed CentOS 4.1 The systems seemed to work fine with CentOS 4.1, but when I proceeded to run yum -y update, all the RPM's (101) download and seem to install fine - however, upon reboot, partitions such as /usr/ and /var will not mount (ext3). These machines both use the Intel S875WP1 P4 Mainboards.