similar to: Fwd: Tunneling server's waste of resources

I'm trying to make a ices/icecast streaming radio with encoding from soundcard. I did a lot of googling and couldn't find any docs other then icecast/ices docs, that are included in the source packages. Are there any other docs that i should read? Is there a howto outthere somewhere? <p>--- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage:

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

Hi all, i''m new at LARTC, and after reading the docs I found no solution to my problem ... On one side I have eth0 conected to the LAN, on the other side I have eth1 conected to a switch and to 3 DSL routers with 3 diferent providers, and also eth2 conected to a cisco 2600 conected to a LDMS line. I have readed the larct docs about multihomed conections to internet, but I''m

Hi there. Anybody knows how can I send metadata updates to a running mountpoint in Icecast? Thanks -- __________________________________________________________ | , , | | / \ | | ((__-^^-,-^^-__)) Octavio Rossell Tabet | | `-_---' `---_-' octavio at

The problem is actually very simple to describe. I manage a Linux fileserver. Most of the clients use Win98 or NT4 and that seems to work just fine - but w2k is an entirely different story. What happens is that no w2k can see the shares in the linux machine an when trying to access it w2k comes with something like "Service not avaible - connection closed". Now the really fun part starts.

Dear R users, I have a problem in the configuration of R: I just changed university, and my conection to the net is via a password, which permits me to access the packages with no problem via the internet explorer (version 6). I just updated R to R 1.81, and I cannot download nether upgrade packages (and many are not working with the update!!!). I have been looking in the help and emails, and

Regards I am installing NUT for the UPS Skate UPS-2200D, the UPS have conection with USB to PC and i have problems to configure nut the ups is conected and Linux report this config Aug 19 23:34:56 dakota kernel: [22393.099595] usb 2-2: new low speed USB device using uhci_hcd and address 7 Aug 19 23:35:06 dakota kernel: [22403.186326] usb 2-2: Product: MEC0002 Aug 19 23:35:06 dakota kernel:

Hi all, I have a isue with a Siemens Hicom conected to my asterisk, here is the scheme: Telco ---- Asterisk --- Legacy PBX --- Legacy phones The asterisk box has a TE210 (one PRI conected to Telco another PRI conected to Siemens) Everything works ok, but when I make an international call from legacy phones to the telco, for example: 0034934452740, the Siemens only sends to Asterisk

Hi all, i have a problem with 2 peers conecting to an asterisk machine, both are conected behind nat without any port mapping in the router, and the * is conected behind other nat with the port 4569 mapped to it address, the problem is: when a peer register to the asterisk the other cant register and viceversa, only gets registration the first one, im using firefly and a hardphone from wuchuan,

Hi eveybody again! I don't want to be annoying, but if nobody can help me with this, I'll have to desist of working with SIP.I have some questions about SIP, as I wrote in another mail. I have a SIP Gateway and I have two phones (an analog one and a DECT one) conected to it.Also, I have two Dlink dg102s with four phones conected to them. The main problems are two. Calls between the

Hello, I need a cheap office SSL server. I think in Via C7 proccessor with AES accel and 2 rng, but seems too closed platform. Anyone know cheap pci card with is supported under CentOS? Any alternative? Thanks in advance. -- -- Devel in Precio http://www.pas-world.com

hello friends, i have to configures asterisk n my hardware details are O.S - Ubuntu 8.04 Lts Memory - 1 GB Proccessor- core 2 duo is any one having a good link or how to related asterisk. any help,support will be higly appreciated thx -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everyone: I have a problem with the shorewall configuration. Let''s me tell you. I have installed shorewall 2.0.4 into a machine with 2.6.8 kernel. This machine works like a software-router: it has 2 netcard eth0 goes to the local network 192.168.0.0/24 eth1 is an interface for ppp0 (there is an ADSL conected) I have defined the Network Zones (net, loc); The Network Interfaces

Dunno why your phone isn't allowing you do negotiate g711u but I can tell you how to upgrade the firmware. I called them on Thursday for myself and they gave me the following tftp server address for which to program my phone. 4.3.153.50 Load this into your phone's tftp area and reboot it. It'll go out to the net and check the firmware revision and change it if required. I've done

Somone in europe have had succes getting Callir ID showed on a phone screen conected to an Handytone 286 ? Adri? Vidal -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: text/enriched Size: 235 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041124/e5514052/attachment.bin

I have a E1 conected to asterisk all zap channels are ok, but when calls come into Asterisk caller don't hear none ring, the call goes straight into the menu, how can i simulate 2 or 3 rings? here it is my conf. exten => s,1,Answer exten => s,2,Wait,2 exten => s,3,NoOp(${CALLERID}) exten => s,4,ResponseTimeout,45 exten => s,5,DigitTimeout,3 exten =>

Hi everyone, I''m controling the uplink traffic bandwidth between two ethernet linux computers, but it seems like even the download bandwidth gets limited then i''ve got trouble on that because my intention is  shaping the conection on an asymetric way (ADSL).  Lately, I''m going crazy, this problem has got me pretty worried and i''m not figuring out at all, the

Hi All. Can anyone tell what is sip tunneling ahd how its works want a make a sip tunnel between to Asterisk Server.... Please share your Kind Toughts -- Best Regards Shazi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090820/091e1ed5/attachment-0001.htm

Can someone please tell me if this is how call waiting should work? ZAP/1 and ZAP/2 are X100P on phone lines ZAP/3,4,5 have phones conected. If a call is in progress between ZAP/2 and ZAP/3 and ZAP/2 gets a callwaiting signal which ZAP/3 hears the Dial command has SIP/2002&ZAP/3&ZAP/4&ZAP/5 If ZAP/4 is off hook, I hear the call waiting signal The CLI says it is ring the 4

Hello: I would like to know if anybody has test various usb phones with an usb hub conected to an asterisk server. The idea is to build a public telephony site with one or 2 analog cards 4 or 8 FXO, least cost routing and billing and usb handsets with dialpad for users, so I really don't need a graphical interface for the soft phone ( the users just dial from dialpad). Any has done this