similar to: Question over broadcasting using icecast server

Displaying 20 results from an estimated 8000 matches similar to: "Question over broadcasting using icecast server"

2006 Apr 21
2
extension match sip address
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
2008 Feb 02
3
IE, flash and icecast
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
2006 Apr 13
1
placing call with agi
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
2007 May 10
1
ices low volume
(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
2010 Mar 09
1
confbridge manager/cli
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
2006 Apr 24
3
Faster Sound Files
I'd like to increase the speed of the Asterisk sound files. Miss Alison talks a bit slow. I can use sox to increase the speed, but then the pitch changes and she starts to sound like a chipmunk. Any audio experts out there know how I can increase the speed a little bit, and change the pitch accordingly so that she sounds ... normal? Thanks Doug.
2006 Mar 14
3
Voice volume using Monitor application
I am using the Monitor() application (with soxmix for combining the audios) and the voice connected to the phone network is recorded at a lower volume then the voice connected directory to the Zap analog phone card. How can I get both the audios to be at the same volume on recording? Thanks Jeff -------------- next part -------------- An HTML attachment was scrubbed... URL:
2010 Feb 10
1
problems with 1.6
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
2010 Feb 24
2
audio glitches in conference
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
2010 Apr 05
2
call files in 1.6
I just switched from 1.4.30 to 1.6.2 I initiated a call file - same way in 1.4.30 and nothing happened. I was not aware of changes in the call file to 1.6.2? I was watching the cli and no error showed or anything. In the manager.conf I have things setup. [MyDial] secret=#### permit=127.0.0.1/255.255.255.0 read = system,call,command,agent,user write = system,call,command,agent,user Any
2010 Sep 02
3
Metadata update
Hi there. Anybody knows how can I send metadata updates to a running mountpoint in Icecast? Thanks -- __________________________________________________________ | , , | | / \ | | ((__-^^-,-^^-__)) Octavio Rossell Tabet | | `-_---' `---_-' octavio at
2010 Sep 02
4
Metadata update
This is exactly what I want. I have a continuous stream to a live mountpoint that needs to be metadata updates. Is this http request you mention any documentation to study? El 02/09/10 08:43, Johann Soukup escribi?: > Hi Octavio, > > I guess you are talking about live mount points. > > We use a http request through the admin/metadata application of the > Icecast admin
2009 Feb 03
2
icecast system requirements
Paul Martin wrote: > On Tue, Feb 03, 2009 at 09:57:19AM +0200, kosnickx wrote: > >> Hi, i am thinking of installing Icecast to some old pc of mine. The pc >> is a pentium mmx at 166 with memory of 32 MB. >> Is this ok with icecast? >> > > Are you going to use this to serve already compressed material or are you > going to be compressing on the fly
2006 Apr 24
2
outbound calls to sip urls
Hi, I wish to use the manager API to make an outbound call to a sip url,subsequently play a prompt and hangup.Any hints on how to acheive this/feasability will be much appreciated. Regards, Ajit
2019 Nov 14
1
SASL AUTH with postfix
On 2019-11-14 15:41, Benny Pedersen via dovecot wrote: > Jorge Bastos via dovecot skrev den 2019-11-14 16:25: > >> I've tried to search for the realm part in dovecot but didn't found it, >> Could you guide me to it? > > doveconf -d | grep realm > > doveconf -n if more help is needed Thanks Benny, It was in my front and didn't saw it. Now i need to
2006 May 05
5
Code parsing error?
This code executes just fine, and leaves the SIP peer's mailbox setting from sip.conf in variable target. exten => 1,1,Set(target=${CHANNEL:4}-) exten => 1,n,Set(target=${SIPPEER(${CUT(target,,1)}:mailbox}) exten => 1,n,VoiceMailMain(${target}) However, every time it runs I get an error in the CLI as follows WARNING[5629]: pbx.c:1366 ast_func_read: Can't
2008 Sep 06
4
schedule a fallback and / 2 sources streaming at 1 server ???
Dick Trump wrote: > kosnickx wrote: > >> So my question is if it is possible to schedule >> some sort of fallback to another mount point whenever i want to >> > > Yes. I have a Windows installation where my live stream comes from Simplecast with multiple mount points with the same data. > > At certain points in the day, I need to insert alternate
2006 Apr 20
1
channels change names
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
2005 May 16
4
Shutdown from PXELinux
Hi Please excuse my english ... Does somebody have a .c32 , .cbt or .exe program to shutdown an ACPI and/or APM BIOS PC from PXELinux or DOS ? Or could be shutdown added as an option to PXELinux ? School environment, remote administration. On night time the electrical power is turned off in labs and PC loose WakeOnLan(WOL) and AlertStandardFormat(ASF) capabilities. In the morning when the
2003 Mar 05
17
Call recording
Hello, How would I go ahead a record all phone calls into and out of my asterisk server. I know the legality issues behind it, but I could always play a recording to let people know they will be recorded. Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-537-2817 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017