similar to: Icecast Newbie Questions

So we need to resample them first? Any live resamplers? -----Original Message----- From: icecast-bounces@xiph.org [mailto:icecast-bounces@xiph.org] On Behalf Of Geoff Shang Sent: S?bado, 12 de Febrero de 2005 09:51 a.m. To: icecast@xiph.org Subject: Re: [Icecast] Icecast Newbie Questions Anton Krall wrote: > 1. Can icecast limit the bps of an incoming live source? for example, > limit

Nice! Im running linux... So ices for the backup streams and resampled to a lower bps.... Now I just need to figure out how to limit the bps of the live source... Maybe using 2 icecast and streamtranscoder right? Ice 1 (private ice) -> transcoder -> ice 2 (public ice) How does that sound? -----Original Message----- From: icecast-bounces@xiph.org [mailto:icecast-bounces@xiph.org] On

Guys Im having a few issues with Languages. Ive setup the english language is it came from default: /var/log/asterisk/sounds /var/log/asterisk/sounds/phonetic /var/log/asterisk/sounds/digit /var/log/asterisk/sounds/letters and then Spanish as /var/log/asterisk/sounds/sp /var/log/asterisk/sounds/sp/phonetic /var/log/asterisk/sounds/sp/digit /var/log/asterisk/sounds/sp/letters Then one of

Guys. Can you dial 800 and 888 toll free numbers using FWD? how do you dial them cause I tried using 1800xxxxx and 1888xxxxx and I simply get a "nobody can asnwer the call" signal on asterisk. Can you dial 800 toll free from FWD?

Hello All, I am looking for a sip phone that is capable of automatic nat. The Cisco ata186 for example works fine for natting with iconnecthere, but as for asterisk, both my 7960 and polycom ip600 require you to set the nat ip on the tftp. Does anyone know a good phone (or ata) that can do this automatically? For example, I want to give a phone to my brother, who is going to europe. His ICH

Guys.. Im trying to setup a fotphone using iaxcomm and when I dial that softphones extension, * complains of this: Mar 5 01:54:54 NOTICE[9962]: app_dial.c:936 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3) Any hints?

Guys, this is weird.. Today I started having some problems with calls been dropped. Im suing X100p cards (clones) and I have this setting on my zatala fle: [channels] language=sp signalling=fxs_ks usecallerid=yes cidsignalling=bell cidstart=ring hidecallerid=no callwaiting=yes usecallingpres=yes ;sendcalleridafter=1 callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes

Hello Guys. I recently contacted polycoms tech support asking if their phones supported XML pushed information to which they replied that only model 600 had a microbrwoser capable of reading dhtml files and such. My question to the community is: is somebody doing any XML info push to any brand of phones except Cisco? How are you doing it? One of the wonders of VoIP should be the means to send

Hi, Has anyone had any luck using modems on ata's other than with Cisco ATA-188's? I really don't have the money pay for the 188's as this is for my personal use. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041003/5ffde3f4/attachment.htm

Hey, For the bridge issue, take a look at 'notransfer=yes' option in your iax.conf. It'll force * to stay in the path http://www.mail-archive.com/asterisk-users@lists.digium.com/msg42262.html

Guys. Im writing this because Ive checked the wiki, Xten website and read a lot of docs and still cant figure out a way around the NAT issues. Maybe somebody else can give me some ideas from a fresh perpective. My test setup is this: Asterisk -> 2wire homeportal Firewall -> internet Computer with Xten eyebeam The asterisk box and the computer with xten beam are behind the same

Hello everyone. I haven't used Icecast yet but I'm hoping it can help me with what I want to do. I listen to podcasts a lot. Most of the time I'm at work I have something playing in the background. The problem is once one ends I have to select another to play. Or if they have all been played I can't use the auto next feature. So what I would like to do is setup a machine

Hi, I have been playing about with meetme as a conference bridge, and find it lacking in some features which I believe are out their somewhere. Viewing this wiki page http://www.voip-info.org/wiki-Asterisk+MeetMe2+Design it looks like a plan happened, but where is meetme2 at now? Things like recording a conference, allowing callers to adjust volume, allowing the conference to be locked, having

Guys.. Im new to asterisk but is it possible to simulate a dialtone for example, in other PBX when you pick up the phone you can hear a certain dialup, which is the PBX dialtone, and when you hit 9, you can hear the PSTN dialtone, is this possible? __________________________________________________________________ Anton Krall

Guys... Ive been having problems with my callerid and I have no more clues as to what I could be.. dates and times stamped on voicemail and info received on the phones display are off by +6 hours and also the date for example today is Jan 02 :) What can I do to modify this? __________________________________________________________________ Anton Krall

christophe.guerin@etud.univ-pau.fr wrote: > > * Enough bandwidth to run the server. If you want to broadcast to 100 > listeners at 24kbps, you'll need about 24kbps*100 = 2,400kbps = 2.4Mbps > of bandwidth. That's about 2 T1 lines worth of bandwidth. Trying to > push 100 128kbps listeners down your 768kbps cable modem isn't going > to work :) It

Guys. Anybody knows why this is happening? Seems every time I make an internal call, the manager shows this and I don't get the callerid on my identapop but rather the calledid.. Event: Dial Privilege: call,all Source: SIP/intruder1-85f0 Destination: SIP/test-f037 CallerID: 201 CallerIDName: Anton Krall SrcUniqueID: 1117038116.7 DestUniqueID: 1117038116.8 Event: Newchannel Privilege:

This is not specifically on-topic for Asterisk, but I have found on many occasions while working with Asterisk that it would have been very handy to be able to measure, with some precision, the bandwidth being used by a particular host, port, or combination of the two. So, I went searching for various tools, none of which were what I wanted. They either were too clever, or too limited in

On Thu, Jun 19, 2014 at 03:30:22PM +0400, lvqcl wrote: > BTW, what can you say about the following place in stream_decoder.c > in read_subframe_lpc_() function: > > /*@@@@@@ technically not pessimistic enough, should be more like > if( (FLAC__uint64)order * ((((FLAC__uint64)1)<<bps)-1) * ((1<<subframe->qlp_coeff_precision)-1) < (((FLAC__uint64)-1)

Uys, Steve Underwood I just got a Sangoma A101 card and Im using unicall 0.0.3.pre9 for R2MFC, I get the far and local end unblocked but as soon as I try to make a call I get dialing and then protocol failure.. Do you guys know if there are any issues with sangoma and unicall? Anybody has an a101 card working with unicall and r2mfc? Are you out there Steve? :)