similar to: a basic question about source streaming

Hello! I use icecast2 and deefuzzer for my streaming. do you know how can I obtain an automatic fade of a song when it ends with a silence? if the mp3 finish with 3 4 seconds of silence, is it possible to remove it automatically? deefuzzer is not able to do it, can icecast2 make this? some ideas? thank you! -- Simone Dal Maso www.nvapple.it www.nvda.it

Use liquidsoap. You can do fadein and fadeouts with it Jonnyboy! Iphones rock! > On 5 Oct 2014, at 13:33, TheDarkener <thedarkener at logicalnetworking.net> wrote: > > Hi Simone, that would be a feature of the source client (deefuzzer). I > am unfamiliar with that client, I'm sure you could search the docs to > find out if it has the ability. > > > Cheers,

Hello, in the first listen socket, I put 8001 to make a test, generally it is always set to 8000. should I change also the second listen socket? Il giorno 30/lug/2014, alle ore 22:15, Jos? Luis Artuch <artuch at speedy.com.ar> ha scritto: > El mi?, 30-07-2014 a las 21:41 +0200, Simone Dal Maso escribi?: >> Hello, >> first of all thank you for your patience and having

Il 23/02/2015 17:06, "Thomas B. R?cker" ha scritto: > OK, that's from the Icecast side an easy situation. really thank you very much! I will make this operation tomorrow, but it seems easy. Still thanks.

Hello, first of all thank you for your patience and having answered you. I tried to change port from 8000 to 8001 but nothing happened. I am using a vps server so I don?t need to open port from router. I paste here all my icecast.xml configuration file. can you please give a llok and tell me if all seems ok to you? still thank you! *** root at mail2:/etc/icecast2# nano icecast.xml GNU nano

I have seen a number of people in this newsgroup asking for information regarding asterisk interworking with Vocal. I was able to configure Vocal and Asterisk so that calls originating from vocal can land on an extension in Asterisk. I would like to share this info with the group The scenario that I tested was as follows. A call was originated from extn. 1001 on Vocal and the call was made to

Hello, I am configuring a webradio and I'm using icecast2 with deefuzzer. the music streaming work, in mp3 format. I want to make a second streaming with my iphone, and the app that I'm using, Koalasan, can stream only in OGG, AAC or Opus. I want that when I start speaking, music stops and my voice is online. Deefuzzer is able to do this with "relay", but the original stream in

Thomas B. R?cker (il 30/07/2014 23:26) ha scritto: You could run "tcpdump -ni eth0 port 8000" and try to access the > icecast web interface. If you don't see any output, then traffic to port > 8000 is blocked externally. good point of view. Ok, let's try: root at mail2:/home/spaziouser# tcpdump -ni eth0 port 8000 tcpdump: verbose output suppressed, use -v or -vv

Hi folks, I want to fix hole in my asterisk set up. I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN, Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go 'other' places. This senario works fine. Now the issue is someone else running a vocal or another SIP proxy can redirect his calls to my * as well. Those calls two will come through general

Hello, sorry for all these questions, but there is a great news, this is the last! ... of the day :-) So: I run nvRadio on port 8000, that is a problem for all people that are at work, behind firewall or protected environment. So, I thought that a proxy can be the solution for streaming on port 80. I enabled the apache2 proxy and, yes, if I type: www.nvapple.it/stream the stream is ok also

I was trying to do a little searching to see if there has even been a comparison between Asterisk and VOCAL or any of the other OSS packages? "Practical Voice Over IP using VOCAL" published by O'Reilly and Associates, attempts to make a strong case about how scalable VOCAL. Of course, considering that the book is written by the makers of VOCAL, it tends to have a one sided slant.

Hello! I solved the problem, probably in a terrible manner. but one thing is absolutely clear, the problem was the firewall. I decided simply to delete all sort of rules, reading a little tutorial on the net. I simply issued the following commands: iptables -F iptables -X iptables -t nat -F iptables -t nat -X iptables -t mangle -F iptables -t mangle -X iptables -P INPUT ACCEPT iptables -P

I want to start a community based voip network projcet and am thinkimg of using VOCAL and asterisk gateways..... my question is, has anyone bench marked asterisk vs VOCAL? is it a wise idea to use VOCAL + Asterisk or Asterisk all the way.........am expecting 1000 -> 5000 users.. your thoughts would be appreciated. _________________________________________________________________ Don't

Interesting notes on the 79xx series. The 7920 is the wireless phone; not mentioned here. For a more complete guide to Cisco's phones, see: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheets_list.html The 7902 is the "very inexpensive" Cisco phone, and it looks like it will be SCCP (Skinny) only. Twiddling my thumbs here waiting for the chan_sccp to

I'm trying to get Asterisk to talk SIP to Vocal and so far have only managed to get it partially working. Calls in from Vocal are working fine but outbound calls aren't. In sip.conf I have: [ivv] secret=SECRET username=08452416761 host=sip.intervivo.net fromuser=08452416761 externip=mt104.dyndns.org nat=yes canreinvite=no reinvite=no notransfer=yes In extensions.conf I

I am trying to connect to a vocal server from an asterisk server. A call is received via iax2 to my asterisk server. I then initiate a SIP connection to the vocal server. everything works great except dtmf doesnt work. A cisco 5300 can connect to this vocal server and do dtmf without a problem. I have my dtmf set to rfc2833 in the general section of the sip.conf . I can confirm that the

Hi If I have a HTB class with 128kbit, and I want to put "N" users in that class ( in order to share bandwidth fairly ) , which is better for me ? esfq (hash dst) or wrr ? I would attach esfq or wrr to HTB parent class. Also I''ve readed on Jim script that over WRR put a RED qdisc, but I don''t understand it. bests andres

Hello, I really don?t know if this question is already been discussed, what I say is that is about 5 days that I can?t find any solution. the problem is that icecast2 works only locally, I am not able to connect remotely. Now I?m going to explain: on ubuntu 12.04, I installed icecast2 with: apt-get install icecast2 it is not the latest version, since I see here there is the 2.4. anyway, I

On 3 Jun 2015, at 14:10, Jos? Luis Artuch wrote: > El mar, 02-06-2015 a las 21:48 -0400, Kyle Blizzard escribi?: >> Hello. >> >> I've been using Icecast for a few days now, and am really enjoying >> it. One thing I can't figure out how to do is have a specific file >> play if someone were to click on my stream url, and I wasn't actively >>

--7ZAtKRhVyVSsbBD2 Content-Type: text/plain; charset=us-ascii Content-Disposition: inline Content-Transfer-Encoding: quoted-printable Netfilter Core Team Security Advisory =20 CVE: CAN-2003-0467 Subject: Netfilter / NAT Remote DoS Released: 01 Aug 2003 Effects: Under limited circumstances, a remote user may be able to crash