similar to: streaming recordings

I'm trying to set up a system so that I can record a conversation over SIP. Monitor and the like don't work so well for me, because I need to pipe the conversation to other programs in realtime, rather than record to a file, so I've been trying to use EAGI instead. (if anyone has any other suggestions about this, it would be greatly appreciated!) At this point, I'm a little

I'm having a problem with conferences both meetme and app_conference, though I've done most of the testing with meetme. Essentially, I get little gaps in the audio - usually fewer than a dozen or so samples, though it does vary. They seem to occur at random, but I usually get one ever few seconds, on average. They also seem to delay some buffer somewhere, so that if I start recording

Hello, I made a post to the forums (http://forums.digium.com/viewtopic.php?f=1&t=72901&sid=3d5c2717ca5ab7ad676957ae436d4b51) but haven't received any replies, so thought I'd try here. On my debian machine running asterisk 1:1.4.21.2~dfsg-3, I've been noticing that there's a problem with conferences (using both meetme and app_conference) and the audio sent out to an

Hi i got lots of this from the asterisk console what does this mean? format_wav.c:247 update_header: Unable to find our position asterisk console: Oct 24 16:39:19 WARNING[4432]: format_wav.c:247 update_header: Unable to find our position Oct 24 16:39:19 WARNING[2812]: format_wav.c:247 update_header: Unable to find our position Oct 24 16:39:19 WARNING[4430]: format_wav.c:247 update_header:

In an attempt to fix problems with EAGI delays in 1.4 (see my other message for more on that), I've tried upgrading to 1.6, in case it's a bug that's fixed in the newer version. Unfortunately, I'm having all kinds of trouble with this new install. My system relies on conferences, and whenever I add any channel to it (adding a SIP connection, playing an audio file, activating

(this was also posted to the asterisk forum, but received no replies... Maybe someone here can help) I'm using the ices command to stream a conference to an icecast server. This is working nicely, for the most part, but the volume is very low. The streamed ogg vorbis audio is much quieter than what I hear in a SIP client, for example (on the same machine with the same audio hardware, of

I have a persistent problem with a PBX I commissioned recently. After a few days it goes into a spasm, creating thousand of log files and giving the message below on the CLI. Dell PE 1600 with Sangoma A200. pbtpbx*CLI> show version Asterisk 1.2.14 built by root @ pbtpbx.local on a i686 running Linux on 2007-01-13 18:31:56 UTC Asterisk Queue Logger restarted Rotated Logs Per SIGXFSZ

Is there a way to have an extension match on a sip address? I've tried the obvious - _.@. but it seems to behave just like _. which is no good. Is there a better way? -- Jon-o Addleman - http://redowl.dyndns.org

I've just started switching my project to use confbridge instead of meetme and app_conference (because of audio glitches that kept appearing in those applications). However, I can't find any way to interact with an existing confbridge conference. Surely there's some equivalent to meetme's 'meetme list' command? Anything else I can use through the cli or manager API? I

I'm having trouble getting an IE client to hear mp3 streams through a flash player. It appears to be the same problem as described at http://icecast.imux.net/viewtopic.php?t=2039 - the flash player connects to the icecast server and begins downloading the audio data, but never actually plays it. I've tried all the suggestions from that thread - I patched icecast so that the Content-length

I'm writing a php script to dial numbers and connect them to a conference. This is fairly straightforward: Action: originate Channel: Local/conf@default Context: default Exten: $extension Priority: 1 This is pretty straightforward. However, the script then loads the list of members in the conference (using the meetme list ... command). For local extensions this works fine - the list of

Does anyone know what this means "Warning [65542]: chan_sip:c:501 retrans_plct: Maximum retries exceeded on call 7438737dc873850@172.16.0.52 for seqno102 (Non-critical Request. 172.16.0.52 is the Asterisk Server I'm guessing that I have something miss configured just not sure what it is. If you need more info just ask.

Hello, i am using well proven set of programs to stream video from firewire to icecast2, few days ago i freshly installed it again on lenny with apt-get. When i run : dvgrab --format raw - | ffmpeg2theora -a 0 -v 5 -f dv -x 320 -y 240 -o /dev/stdout - | oggfwd myserver.org 8000 password /test.ogg stream starts for and for a sec and appears on icecast2, but then it fails with : Found AV/C

On Tue, Nov 02, 2010 at 08:08:52AM +0700, Bino Oetomo wrote: > Dear Thomas > > Thomas B. Ruecker wrote: > > On Mon, Nov 01, 2010 at 05:17:12PM +0700, Bino Oetomo wrote: > > > >> Now .. I want to do it the otherway with Icecast > >> > >> I try with wget -nv -O - http://root:root at 192.168.10.234/mjpg/video.mjpg > >> | ffmpeg2theora -a 0

How can I execute an EAGI script with params from extensions.conf Example python script: InfMsg -s 1 in my extensions.conf exten => 492,1,Answer exten => 492,2,eagi,InfMsg -s 1 exten => 492,3,Hangup() It doesn?t work my * report... -- Executing [92@telpin-112:2] EAGI("Zap/4-1", "InfMsg -s 1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/InfMsg

have you tried "cat $1.opus $2.opus $3.opus | oggfwd" could be a band-aid until ices2 can work with opus files - if it works... On Sun, Feb 16, 2014 at 10:19 PM, <epicanis+icecast at dogphilosophy.net>wrote: > Figured I'd join the list since I saw someone else had just popped in > looking for exactly the same thing I am (non-transcoding opus streamer for > sending

Figured I'd join the list since I saw someone else had just popped in looking for exactly the same thing I am (non-transcoding opus streamer for sending multiple .opus files to icecast2). It looks like ices2 would do exactly what I personally need, except for not having been updated to support opus yet. (I chatted on IRC once or twice with someone who it sounds like is interested in adding

Hi, we've connected Sphinx4 through eagi script (modified eagi example) to Asterisk. Users can now say their wishes - but for gradual evolution we would like also to provide "older" way of DTMF navigation too - can we recognize DTMF while reading sound in eagi ? Any advice or examples ? Thanks in advance, regards, Rob.

Hi, we've connected Sphinx4 through eagi script (modified eagi example) to Asterisk. Users can now say their wishes - but for gradual evolution we would like to provide "older" way of DTMF navigation too - can we recognize DTMF while reading sound in eagi ? Any advice or examples ? Thanks in advance, regards, Rob.

Hello folks, I'm interested in detecting tones on specific frequencies with specific timing; for example, I'd like Asterisk to dial out and when the channel starts/call connects, listen for a 1200Hz tone that plays for 100ms. Is this doable with Asterisk using something already extant? After looking through documentation, mailing lists, and some of the source I had the idea that I might