similar to: meetme-icecast2-ice2

hi, I installed icecast-2.2.0.tar.gz and ices-2.0.1.tar.gz on Fedora3 linux-2.6.12-1.1372_FC3). It works fine for playlist.ogg from the other CPU, such as 'xmms http://192.168.0.3:8000/listplay.ogg'. But when I use 'stdinpcm' like 'asterisk-ices.xml' which send voip's voice udp packets to 'asterisk-ices.xml' such as; .......(snip)...... <stream>

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Build environment: Solaris 7, JKP 106541-18 gcc 2.95.3 Current (as of a couple hours ago) CVS ice2 After a long bit of downtime, I'm throwing myself back into icecast with a fervor. I'm attempting to build ice2 on my Solaris box, and everything goes fine (well, went fine once I got libxml...) until the

i have been playing around trying to get ice2 to work now for quite some time and frankly i am a bit frustrated by now... the anoying thing is how early in the whole proccess my problems actually start! running ./autogen.sh in the newest cvs version of ice2 results in <p>idoru:/usr/src/ices# ./autogen.sh I am going to run ./configure with no arguments - if you wish to pass any to it,

Hi: If you want to use icecast 2 for streaming vorbis audio then don't get it from the CVS repository at icecast.org. That's ancient developer stuff in there. Instead get it from the xiph.org CVS repository (see http://www.xiph.org/cvs.html which I see now lists the icecast stuff (yay!). You'll need the icecast module plus the avl, httpp, log, net, thread and timing modules (check

Hi, If there are 5 asterisk servers on the local net and each server runs meetme, eg. 3311,3321,3331,3341,3351 respectively. Can I connect these 5 meetme conferences to one meetme using IAX2? Regards, Zen

I have a confbridge setup that feeds the conference from the ALSA microphone input (this is the conference leader) and then uses app_ices to send the conference audio to icecast. I start the conference leader like this: console dial 1000_admin at conferences I join the ices user to the confbridge with a call file: Channel: Local/1000 at conferences MaxRetries: 2 RetryTime: 60 WaitTime: 30

Hello Folks, What could be producing the following warnings on console, after an installation from source (Asterisk 1.4.41): [May 12 21:36:54] WARNING[15344]: loader.c:434 load_dynamic_module: Error loading module 'res_musiconhold.so': /usr/lib/asterisk/modules/res_musiconhold.so: undefined symbol: cap_set_proc [May 12 21:36:54] WARNING[15344]: loader.c:434 load_dynamic_module:

I have read more about asterisk and have succeeded in using it's app_ices function and a sample conference . I would like to learn more about lowering the latency between the Speaker on the SIPphone->Meetme Conference->ICES->Listener stream. Thank you Flash > > Let me know if there's more info you need and I'll ask my friends some > specific questions. > >

I am using CVS-v1-0-04/20/05-12:31:26. Windows Messenger5.1 works MOH fine. After I stop MOH on Windows Messenger, if the hungup signal could not send to *, the sip channel(e.g.,SIP/52001-08ca) for this MOH remains. Then the user trys again MOH, a new sip channel starts. And again the hugup signal can not send to *,......... When I 'stop now' from CLI> , * cleanups the remaining sip

Hi, I'd like to know if there's an "easy way" of doing the following: SIP phone dials a custom feature code in Asterisk, call gets answered within a custom context (Answer()), anything that the caller says should be redirected/piped to an external application. Something like "monitor" except audio should be sent live. More like "app_ices" (or

Hello, Our 500+ company is slowly moving away from our hosted conferencing solution to one I built a few weeks ago with Asterisk and MeetMe. When our Q3 conference call comes around, we will have the need to have approximately 300-400 users in this call. Obviously, all would be 'listen only' mode and only 1 or 2 two would be speaking as marked/admin users. Our conference hosting

Hey Guys, I am trying stream live music via icecast streaming server into a conference room, this will allow persons joining the conference to hear the music. I have been googling and i have come across a few tutorials, that give instructions as to how to get it done. But they all mention the use of a ices application module. It appears that asterisk 1.4 is not shipped with app_ices.0 by

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I just noticed that the Skype API for linux seems to be available. I've read before a number of posts where people were talking about implementing a chan_skype with the skype API. I wonder if there is any progress in that direction, and if anyone is working on it. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to

When I try to connect to * using a Cisco ATA 188 configured with a MGPC firmware (v3.1.1), I just keep getting this message every 30 seconds or so : May 10 10:08:21 NOTICE[7913]: chan_mgcp.c:1509 find_subchannel: Gateway '192.168.1.27' (and thus its endpoint '*') does not exist Using tcpdump, I have checked that the ATA188 (with IP 192.168.1.27 and port 2427) sends UDP packets to

Good evening, I am just getting started with Asterisk. I have it installed, and I believe I am on the right track, overall, to get it working, but I can't get the linejack to answer any calls. At this point, all I'm trying to do is have Asterisk answer an inbound call on my linejack, /dev/phone0, and play a greeting or tone. I figure, once I am able to get asterisk to actually answer the

Hi, I've been having a little problem with my asterisk servers, I have 4 identical asterisk servers setup (same hardware, same OS, same config). Once in a while (once or twice a day) one of the server crashes on the cron job reload. But I realized this only happens on 3 of the 4 servers. Tried to spot the difference between that one server that wasn't crashing. The difference I found was

I tried to checkout ice2 from CVS. But I can't. Are there any trouble? -- $ cvs -z3 -d:pserver:anonymous@cvs.icecast.org:/cvs/ice co ice2 cvs server: Updating ice2 U ice2/Makefile.am U ice2/autogen.sh : U ice2/src/os.h U ice2/src/sighandler.c U ice2/src/sighandler.h (no response) <p><p>--- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage:

I have built Asterisk 1.4.4 on my Solaris 10 x86 box: LDFLAGS='-R/usr/sfw/lib -R/opt/csw/lib -L/opt/csw/lib -L/usr/sfw/lib' CPPFLAGS=-I/opt/csw/include ./configure -with-curl=/opt/csw --without-oss --without-vpb --prefix=/opt/asterisk-1.4 The build and install go fine but the asterisk executable reproducibly dumps core with a segmentation violation. If I start it as: asterisk -gc and

> Message: 1 > Date: Tue, 15 May 2007 23:01:24 -0400 > From: Frank Tarczynski <ftarz at mindspring.com> > Subject: [asterisk-users] Asterisk 1.4.4 reproducibly dumps core on > Solaris 10 > To: asterisk-users at lists.digium.com > Message-ID: <464A7404.5000706 at mindspring.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > I have

I'm still having problems getting this to work. I cannot get anything to display on my 7914 other than blank lines. I have SIP/5920-5930 in [main] that I'd like to add to the 7914 and indicate hook status. The 7960 is registering okay as SCCP/5000. What exactly should my sccp.conf file look like? When I make changes to this, how do I enact them? Do I reload Asterisk and reboot the phone