search for: alphanet

Hi... i would like know how to one playlist handler whith perl? where a search one "HOW TO" thanks.. -- ALPHANET INFORM?TICA LTDA www.alphanetbh.com.br Belo Horizonte MG Leandro Campos (31)30726251 (31)87883925 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/icecast/attachments/20071226/c577c111/attachment.htm

How I change the playlist handled of music... which the ices 0.4... for example... i want that 12 pm, play one specific music...... thanks... -- ALPHANET INFORM?TICA LTDA www.alphanetbh.com.br Belo Horizonte MG Leandro Campos (31)30726251 (31)87883925

...the console Mar 21 05:54:15 NOTICE[68071]: chan_iax2.c:6123 socket_read: Rejected connect at tempt from 203.13.163.245, who was trying to reach 's@' the s part i can understand by the @nothing ......?!? my iax.conf looks like this register => aaaaaa:bbbbbbb@proxy.freecall.net.au [alphanet] type=friend username=aaaaaaaa auth=plaintext ; ugh plaintext secret=bbbbbbbb host=proxy.freecall.net.au context=main disallow=all allow=ilbc my extensions.conf looks like this [default] exten => s,1,Answer exten => s,2,Dial(SIP/me,40,tr) also I have the same under [main] aswell anyone go...

...d wrong number). I tried different things like: exten => h,1,Goto(s,1) but that didn't help either. I get disconnected after a call dialed from DISA completes. Is there a way to get back to Asterisk, authenticate and get dial tone again without disconnecting ? Thanks Jake -- ----- AlphaNet - najtaniej w sieci! -------- Odnowienia domen w rewelacyjnych cenach! .pl - 65 zl, .com.pl - 50 zl, reg - 20 zl http://www.domeny.alpha.pl --------------------------------------------

Hi to all. ? am a newbie about icecast. ? just wanna know that, Can i stream my sound card input sound? I mean i want to stream the sound that comes into my sound cards input, from a microphone. is it possible? ?f it is possible how can i do that. Can u help me? Take it easy... Mesut GULNAZ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi , thanks for you help I try to explain to you my usage of icecast: I'm running icecast for streaming auvio and video in vorbis and theora , and my users can see my streaming on my webpage , where I use cortado web layer. SO my problem it's that my user can see easily the icecast address and can connect to it with mediaplayer like vlc , without use my website. So I want to manage

My housemate has formalized a sortof new attack against unix-style operating systems. He''s a BSD fan, so that''s where he developed the attack. He asked me to check Linux, which I did. It seems Linux is not vulnerable to it. This attack is going out to BUGTRAQ tonight. The attack isn''t too serious because it requires physical access to the console, but it

Hi, the proprietary FileMaker software -- which runs under an old WINE release -- requires a `mail sending program' or library. Apparently, Mozilla Thunderbird could fullfill this role. Unfortunately, the versions where Mozilla Thunderbird works are the newer versions where the proprietary FileMaker software doesn't. A simple fix could be a very simple mail sending program or library

Hi, I have even now connected to IAXtel at number 1-700-895-5211 when I am in the office, so Asterisk is great. I just found something strange, which is that if I am already in a connection with my Grandstream and talking, and a second call comes in, it rings on the Grandstream. However, if I am not talking but waiting for dialing, the caller gets a busy signal (good). How can I make sure

Hi, to test my Asterisk / IAX connection I have configured the Swiss phone number 032 841 47 74 to a IAX gateway. You can dial 1-700, 1-800 and other numbers from this number (prefix with 00: for example 0018005551212). This is a local rate number. I have not yet implemented IAXtel -> Swiss 1-800 yet because I didn't succeed in registering two IAXtel numbers yet. Feel free to test

Hi, on a configuration with one external ISDN S bus (to telco) and one internal S bus (to ISDN telephone), where Asterisk is in the middle (using HFC hardware), I noted the following: - when a GSM phone or ISDN phone calls in, the Transfer capability is Requested transfer capability: 0x00 - SPEECH - when an analog phone calls in (either from an analog line or an analog ISDN

Hi, apart from the mmap(2)ed DBM files that Samba uses, are they any cases where Samba will *modify* data files without setting the mtime ? I have issues with rsync not seeing changes to Samba exported files (md5sum don't match). The mtime is however in the very distant past (say 2004), but the content seems to have changed. I don't think it's a data integrity issue. Any idea ?

Hi, I have two HFC-s boards I configured in NT and TE mode respectively. When I connect the two boards together, I can dial extensions and I see the correct called and caller ID numbers: -- Executing SetCallerID("Zap/2-1", "7516862") in new stack == CDR updated on Zap/2-1 -- Executing Dial("Zap/2-1", "Zap/g2/0795025602|30|r") in new stack --

Hi, my setup is quite simple: an asterix CVS of 2003-11-15 on a 2.4.21-debian-5 GNU/Linux box in an internal network (192.168.1.0/24, asterisk is 192.168.1.10). - with a SIP phone configured as 192.168.1.190, and with its SIP server being 192.168.1.190 - with an ISDN AVM c4 i4l card on an ISDN connection with 2 channels. I try to: - dial-in from ISDN, then transfer to the SIP

(I have some problems with my mailing-list alias, I hope this doesn't get sent twice) On Sat, Nov 15, 2003 at 04:35:20PM +0100, Philipp von Klitzing wrote: Thank you for your comments Philipp: > > - with a SIP phone configured as 192.168.1.190, and with its SIP > > server being 192.168.1.190 > > That doesn't look right. Do you have another "SIP