similar to: All I want to do is stream...

Hi All - First post, and something of an audio newbie, so please be gentle. I'm cross-posting this to both icecast and vorbis, albeit in separate emails. I'll try to give as much detail as possible so this might get a bit wordy. I've been trying to get a live stream running for a local college radio station. It's currently up on a Slackware system - but

Since my last query went ignored let me try a different approach... I've got a Telos hardware encoder that works under Shoutcast but gets "authentication errors" when trying to use Icecast. Here's what I see when trying to add the Telos as a relay: [110:Connection Handler] Kicking source 107 [192.168.200.200] [Error in request, relay refused entrance] [relay], connected for 0

I have a situation where I'd like to be able to have icecast reach out to another icecast server as a client. Is this possible with relaying? From what I read relaying and master relaying both have the server reach out to another server. The reason is this. I have a hardware encoder (Telos audioactive) sitting on a cablemodem. I have a remote provider (streamcheap or something) set to

I am attempting to install Gentoo on my cdless machine using the thumbdrive as a boot device. The motherboard is an Intel D815EEA, and is the latest firmware revision (at least on the BIOS). I believe I have done everything right, but it tells me I have unknown keywords in my syslinux.cfg and corrupt kernel images. The steps I have followed are thus: 1) Starting with a 4GB flash device,

Hello all! I've talked recently to the head of R&D dept. of Telkom Telos (www.telos.com.pl) - a big Polish company specialised in making phones. I gave them the idea of creating a cheap (cost-effective) hardware IP phone. The phone we discussed would include hardware support for IAX (though probably SIP/sth. else would be required too if it were to hit the market.. what do you

I also tried installing Icecast 1.31 ... the config file claims its relaying capabilities aren't very well tested and I can verify this. But now I'm concerned that even 2.0 won't work for me in relaying. I'm trying to relay multiple streams from a cluster these hardware encoding devices: http://www.audioactive.com/products/realtime_enc/rte_ovw.html ...to do so you

this link: http://www.telos-systems.com/techtalk/digiphones/digiphones_4.htm States the following: "Generic PBXs will not do for our broadcast application ? they just don?t have the features necessary. For example, while lines may certainly be shared to multiple phones, there is no way to switch groups of lines from studio to studio. There is also no way to connect computers for

I've been working with AudioActive the last couple of days to fix their behaviour... looks like they're going to update their firmware EPROMs. What are the chances that Michael's fix below could be incorporated into a future version, so that all those boxes without the Firmware upgrade will work? -Ian. <p>On 10-May-04, at 7:20 PM, Michael Smith wrote: > On Tuesday 11

On Tuesday 11 May 2004 11:28, Ian Andrew Bell wrote: > I also tried installing Icecast 1.31 ... the config file claims its > relaying capabilities aren't very well tested and I can verify this. > But now I'm concerned that even 2.0 won't work for me in relaying. > > I'm trying to relay multiple streams from a cluster these hardware > encoding devices: > >

If propagation of Ogg Vorbis is the (or a) goal of Icecast then probably the best strategy is to build an embedded Ogg Flash Player of some sort, so that people unwittingly become Ogg users without lifting a finger, and broadcasters can start using it without fear that they'll lose listeners due to their protocol choice. I've been streaming for more than 7 years now with Quicktime,

Hello, In our SIP network, Asterisk is the central PBX, and it routes calls to the PSTN thru a Cisco Router - IOS 12.2(11)T9. If a client softphone calls directly via Cisco to the PSTN, the call works successfully. If the client softphone calls via Asterisk to other SIP internal extension, it work fine too. The problem is when a client calls an Asterisk extension, and Asterisk transfers

I have a similar setup to you and get the same message regularly. I don't think it's the cause of your problem. I did some research on it a while ago: IIRC the cisco uses codec 13 for "silence suppression" whereas asterisk (correctly) uses codec 19. The router can be configured to use 19 also, but I didn't bother. I'm sure somebody will correct me if I'm wrong about

Hello, I'm trying to registrate a Nuance Server in Asterisk (using SIP) with no success. It seems that Nuance does not send any secret/password (there is no way to define it!), this is the list of parameters that Nuance provides for registration: audio.sip.UserAgentURI=sip:user@domain audio.sip.UserAgentPort=<port> audio.sip.ProxyServerURI=sip:<IP>:<port>

Hi, I installed samba4 (zentyal 3.4) and I followed the Samba4 AD DC Howto from the wiki to test the linux-side. There all seems OK. Unfortunately, when I try to register a Win7 PC, the domain is not found. So what can I do to test things on the Windows-side ? I did : C:\net view /domain:ace_domain returns : \\ZENTYAL1 which is my DC After manually "mounting" the test1-share : C:\net

Hi Is there any function to get the name of the components of a given list object? ok. [[alternative HTML version deleted]]

I have a project that I need some hardware pointers for. I need to build some Centos "appliances" (dedicated boxes to do one thing only). Target cost is under $250/box. Need: OS: Centos 5 Hardware Cost: less than $250 USD USB: at least 2 (not including keyboard) Memory: at least 128K Storage: prefer flash (USB stick OK) Network: 10 Base T Want: Height: less than 4" (fit on a

Hi, I'm working on a project to add more audio formats other than MP3 and Ogg that can be streamed by Icecast. Yes, I mean streaming those formats, not converting them to MP3. These formats I'd like to add include PCM (.wav) and ADPCM (.adp). Although they are much less advanced codecs compared to MP3, they have a valid market for low-power clients. I've been working on the source

Hello: I would like to use Asterisk as a redirect/proxy sip server to route SIP calls on a sip header/parameter basis. I've tried some things successfully: - SIP registration from clients. - On-the-fly compression for wan VoIP transfers: SIP G.711 --> GSM IAX --> (wan) --> GSM IAX --> SIP G.711 - Sending custom parameters in URI: exten => 1,1,Setvar,VXML_URL=var1=value1

The Wine development release 1.3.17 is now available. What's new in this release (see below for details): - Implementation of the new Vista file dialogs. - Initial support for patching in MSI installers. - Improvements to the calendar control. - A few fixes for file associations. - Menus cleaned up in built-in applications. - Various bug fixes. The source is available from the

Hello I have a question about incoming calls on TDM400P cards. I want to know why an incoming call appear in a sorpresive way on a phone that I pickup to call out. I am using ChanIsAvailable to check those lines ( Zap channels )that are free. I have four lines connected to my TDM400P card and when I get a free Zap channel to call I hear the voice of a people on the other side from an incomming