similar to: Playing a certain mp3 on a schedule

Hello, Regarding accounting for track lengths, could sox (soxi) be getting it wrong? Here's my approach atm: $lengths = array(); foreach($files as $val) { //Note that $files has already been populated by glob() $temp = explode(":", shell_exec('soxi -d "'.$val.'"')); $length = $temp[0] * 60 * 60;//convert the hours to seconds. $length += ($temp[1] *

Hi All I am developing a client that uses libjingle to do xmpp stuff with ejabberd. I can also make audio calls between those clients. What I am trying to archive now is to send calls to pstn using jingle. I was told in the jingle-dev community that asterisk can do that. Is there any way to send jingle audio calls to asterisk and will it understand them ? If yes..can I forward those calls to PSTN

Hi! I just tried to call a friend using jingle, but I got refused. Errorcode was 502, he tried to call me, heard it ringing once and then it stopped. I used: originate jingle/gtalk_account/friend at jabber.linuxlovers.at [application] I'm registered to googletalk, but this should mean no harm, or should it. Once I was able to receive a text-message from him, but couldn't

Hello, Icecast and Ices are running on the same machine and it is a dedicated server. It is a Xeon 1270 V2 CPU with 16 GB ram running Centos 64-bit. Any thoughts on which settings I could try next? I have tried everything that appears to relate to the speed at which the stream occurs to no avail. One point I forgot to mention is that some tracks bring on more deviation than others. Shorter tracks,

Hi all, For one of my inverstigations it looks like i'm back to "square one" I'm trying to accept an incoming xmpp call and forward it conditionally to a sip, isdn, or voicemail. No google is involved as i use a local xmpp server (ejabberd) I was experimenting on 1.8.15.1 (with jabber.conf, jingle.conf), but some suggested me to have a look at asterisk11,so i did... I

Use Jingle, anyway Jingle kicks SIP on almost every aspect. Especially on the way the standards are made. Diana David Hogan wrote: >> I understand, but CELT would be useless for SIP if one can't >> > read/guess > >> correctly decoder configuration from the RTP data. >> >> One possible way to cope with this would be to have several CELT >>

The server doesn't control the rate, the source does. Provided you aren't getting corrupt stream data (and it doesn't sound like you are), you don't have a server bug in this case. At a basic level, the server level takes the data that comes in from the source, buffers it, and sends it right back out the door at the rate at which it came in. It isn't entirely clear to me

Hello, I'm looking for a SIP to XMPP Jingle voice gateway. I see that Asterisk has Jabber and Jingle support, but it looks like Asterisk acts as a Jabber client. Are there any Jabber server solutions, where Jabber users can call SIP users by using the SIP URI and vice versa? -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/

Just a heads-up, I received confirmation that Speex is now the default codec for the Jabber's Jingle VoIP protocol. While not the default in Google's Jabber, Speex has been reported to work on Google Talk as well as of last year. This information is not news breaking, but many people aren't aware of it yet, so spread the word. -Ivo

Peter Saint-Andre a ?crit : > Ivo Emanuel Gon?alves wrote: >> Just a heads-up, I received confirmation that Speex is now the default >> codec for the Jabber's Jingle VoIP protocol. > > Which we hope to finalize soon for broader adoption. :) That's good to hear. Are you supporting wideband or just narrowband? Jean-Marc

On 05/25/2014 01:13 PM, Jordan Verner wrote: > Icecast and Ices are running on the same machine and it is a dedicated > server. > It is a Xeon 1270 V2 CPU with 16 GB ram running Centos 64-bit. > Any thoughts on which settings I could try next? I have tried > everything that appears to relate to the speed at which the stream > occurs to no avail. > One point I forgot to mention

Hello, In conclusion I'm pretty sure this issue is the symptom of a bug, either within Ices or Icecast itself. I decided to try running the same setup on Shoutcast today, and it's working perfectly. An hour of streaming at the time of this message, and absolutely no schedule deviation. It appears as though Shoutcast has better rate control than Icecast does at the moment or something.

Hi, Are there any thoughts about how "cpu-expensive" motif is? Does it only translate SIP <--> jingle (during call-setup) if so, impact will probably neglectible. or does asterisk remains constantly in between the data-stream? In that case, it might be something to pay serious attention to, when doing multiple call conversions simultaneously... hw

I have configured support for ice in sip.conf, and made a connection with motif to jingle, but does not work for me [Jul 15 12:03:32] ERROR[21758]: chan_motif.c:1955 jingle_interpret_ice_udp_transport: Received ICE-UDP transport information on session '8b4hdffbt37vg' but ICE support not available -- Executing [s at xmpp-in:1] NoOp("Motif/allan-ce76", " llamada de

Hi, we would like to build IM-Voice community for our students around Asterisk, Jingle, Jabber. Can we already test those features ? Anyone already running such setup? Any more info ? Thanks in advance, regards, Rob.

On Mon, Jun 04, 2007 at 03:20:05PM -0600, Peter Saint-Andre wrote: > >>Hi Luca, > >> > >>Thank you for working on the Theora and Vorbis drafts in the IETF's AVT > >>WG. Do you also plan to work on the Speex draft? > > > >other people are working on the speex I-D, Ralph should be able to tell > >you which to contact ^^; > > Hi Ralph,

Hello, Are you saying I should be rounding to the nearest second rather than measuring in milliseconds? My current approach is to calculate the track lengths by dividing the number of samples by the samplerate, resulting in a value measured in seconds with at least 8 decimal places. Nevertheless, for some reason a shoutcast + sc_trans setup handles this flawlessly. With Icecast + Ices, data is

OK, im pulling my hair out, everything looks configured right, deleted, and started over, etc, etc. but can't seem to get this to work Gtalk.conf [general] context=google-in allowguest=yes bindaddr=192.168.xxx.xxx extenip=96.254.xxx.xxx [guest] context=google-in disallow=all allow=ulaw allow=g729 connection=jp_jabber jabber.conf [general] debug=yes

Hi There, There are 300 boy students and 100 girl students in a class. One interesting question is whether boy is smarter than girl or not. first given the exam with a difficulty level 1, the number of the student who got A is below 31 for boy, 10 for girl. Then we increase the difficulty level of the exam to level 2, the number of the student who got A is below 32 for boy, 10 for girl. We

Hello, I'm using Icecast, Ices, and the script module configuration to serve the playlist to the server. Playlists are scheduled once per day with regularely placed events (like spots) which run at specific times. Unfortunately it's being served too fast, so it's always getting ahead of schedule (gaining about 10 seconds on every hour). I have tried adjusting the flush-samples setting