similar to: IceCast + Real Audio?

I'm trying to complie ices2.0.1 on a more-or-less stock Fedora 3 installation, and I'm getting the following compiler error (during linking, I think). /bin/sh ../libtool --mode=link gcc -I/usr/include/libxml2 -pthread -g -O2 -o ices input.o cfgparse.o stream.o ices.o signals.o im_playlist.o reencode.o encode.o playlist_basic.o im_stdinpcm.o stream_shared.o metadata.o playlist_script.o

I'm still having trouble (after reading the archives) geting ices2.0.0 to work with ALSA (I seem to be able to do arecord just fine. I'm using the following packages (along with ices-2.0.0.tar.gz). kernel-smp-2.4.22-1.2188.nptl kernel-smp-module-alsa-1.0.4-1_2.4.22_1.2188.nptl alsa-driver-1.0.4-1 alsa-lib-1.0.4-1.1.fc1.fr alsa-lib-devel-1.0.4-1.1.fc1.fr (and a few other alsa ones, but

Hello. I've made some progress on the issue. I found a tool called TrPlayer which is a text mode front end for real player - initially developed for the use of the visually impaired. The theory is that I can use this and pipe the live stream into vsound which then in turn is passed into ices or another source client. Trouble is, I'm having all sorts of trouble compiling Trplayer on the

I'm working with a streaming Ogg Vorbis system where I'm taking the stream output (from a darkice server) and saving it to hour-long files, then reassembling excerpts from these files (sometimes spanning two or three) into a single file for playback. I've got two problems (well, related to this, but anyway). 1) The "chunk" files I'm saving into have mangled headers. 2)

I currently utilize Icecast on my website, but I am looking for the same kind of functionality as I get with realaudio server when I stream static content - namely, the ability to seek, show the time remaining, etc, especially with Vorbis audio. I realize the intent of Icecast is to serve mainly live, streaming content (or at least radio-style content), but I utilize it mostly for streaming

I'm trying to convice a webmaster to start offering Vorbis in addition to his current RealAudio 3(GAH!) web streams. I was just wondering what the status on the RealOne/Helix DNA plugin was so I could include it in a letter I'm going to be sending him today. <p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe

Well more specifically I want to transcode a realaudio stream into mp3 on the same machine and on the command line only - reliably! Following up, I managed to compile the trplayer using a substitute __pure_virtual function compiled with extern "C" (without it the function name was being mangled by g++) by now trplayer runs and exits without warning, error or success... the funny thing

mailboxes). Are you certain that removing either 612 or 610 mailbox would keep Asterisk from complaining ? > > However, the MWI does not indicate voice mails for 610 and I keep seeing > this error message: > > ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox > 610 in context a10 > > However, mailbox 610 is clearly defined in voicemail.conf: >

Hello, all. I've been playing with MeetMe and talker optimization seemed like a great idea. I activated it as follows: exten => 201,1,MeetMe(100201,cTo) However, although I can see who is the talker on the CLI pbx01*CLI> meetme list 100201 User #: 01 1001 Denise Dion-Sullivan Channel: SIP/1001-1e1db7c8 (not talking) 00:00:33 User #: 02 1000 John A. Sullivan III

Hello all, I am a little new to Linux so I would appreciate some assistance with this issue. I have been running a Helix server streaming RealAudio encoded content live and on-demand for a while. For the live stream, I use the Real equivalent to Ices which on the Helix platform is called SLTA. All the content on the server is encoded as RealAudio. I have also successfully setup an Icecast2

Hello, all. I see there is an "o" option for the Dial() command which reverts to the previous behavior of using the original callerid throughout the call - I suppose more specifically, using the callerid from leg 1 for leg 2 in B2BUA if I understand it correctly. That seems to be highly desirable behavior; I know we are seeing some problems with call history and call forwarding because

Hello, all. After reading the README, UPGRADE.txt, and a quick tour through google, is it safe to assume to upgrade from 1.6.1.1 to 1.6.1.2, one simply compiles and installs over the old installation being careful to NOT install the sample files? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsullivan at opensourcedevel.com

Hello, all. After many pages of googling and testing in the lab, I'm still a bit perplexed about how to implement tls protection for the asterisk manager. manager.conf allows one to specify the cert file but one normally must also specify the private key file. If I simply enter the cert file: sslenable=yes sslbindport=5038 sslbindaddr=172.x.x.8 sslcert=/etc/pki/tls/certs/pbxc.pem ; path

Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=612 at a10, 610 at a10 However, the MWI does not indicate voice mails for 610 and I keep seeing this error message: ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox 610 in context a10 However,

Hi: I know this is off-topic, but figured someone here would know, particularly given Xiph's involvement in Helix. Up till now I've used TRplayer <http://linux-speakup.org/trplayer.html> to play Realmedia content. This is because, as a Blind person, I've no need or desire for X11, and TRplayer uses the Realplayer libs without all that X overhead. but now I would like to play

Hello, all. I am attempting to use IMAP voice mail storage in Asterisk 1.6.1.1 on CentOS 5.3 using Zimbra 5.1.6. I will not be using it as it has proved terribly unstable - Asterisk segfaults on every voice mail message although the message is successfully deliver to my email inbox - but I thought I should report it. Here are the errors from the Asterisk console: -- Executing [210 at

Hello, all. Since implementing an iptables firewall between the Asterisk PBX and several SIP phones, the Asterisk PBX ability to "reinvite" has been broken even when the phones are on the same network (i.e., no firewall between the phones). We've been beating our heads against the wall thinking it was the complex rule set but it appears the issue is ip_conntrack_sip. Before I drop

We are happily running xen on a server which uses a SATA driver (thank you all you xenists for producing such a great product). However, we are creating a custom kernel for a domU. When we attempt to run mkinitrd we get: No module ata_piix found for kernel 2.6.11.12-xenUIPSec, aborting. This is the SATA driver. In the past (xen 2.0.5), we simply ran make ARCH=xen menuconfig and told it to

Hello, everyone. No need to read this message. I'm posting for documentation for other poor, ignorant slobs like me who are struggling to pull together the many technologies to make converged networks happen. Hopefully, this will help save someone else the time I spent. I started the below email until I realized I had solved multiple parts of a compound problem but not all at the same time.

Hello, all. My apologies up front but I must be brain cramping on something very simple. I've tried to pare down my configuration to the absolute minimum for SIP traffic just to understand how it works. My incoming calls are not finding the "s" extension in my dial-plan. I am assuming SIP calls can do this. I am using Asterisk 1.6.1.1 sip.conf has nothing but: [general]