similar to: 100 users; icecast capable?

will the new version of ices/icecast support sending live broudcast sent as mp3 as opposed to ogg. I remember su ccesfullty sending an mp3 song over the net but I dont remember what happened with a live broadcast. thanks <p> __________________________________ Do you Yahoo!? Yahoo! Domains – Claim yours for only $14.70/year

hi all, i'm only recently starting with speex coding. i've read the sample in the speex doc and it looked easy enough. but i'm having difficulty just the same. to start with, i've written a C++ wrapper to handle speex functionalities. at the moment i only have Encoding and Decoding, yet, and i'm still debugging them. what is the best way for me to check if my Encoding

First, a little background: For over a year now, we've had a 400 Mhz celeron running Samba 2.2.7 on RedHat 9, acting as a PDC for an office of about 8 Windows XP client PCs. It has done a great job, but we recently bought a new Dell Poweredge 400 sc (2.4 Ghz P-4) to replace it. I installed Gentoo on it, with Samba 3.0.2a. I also copied the entire /home tree from the old box to the new one,

I use Centos 4.2 with all service pack installed. I verified traffic on link WEB and I see port TCP 80 with many traffic. I accessed lod /var/log/httpd/access_log and show below. ca.com/members/index.php HTTP/1.0" 401 - "http://members.sapphicerotica.com/members/index.php" "Mozilla/5.0 ( compatible; MSIE 5.01; Windows XP; NetCaptor )" 68.119.110.138 - -

Message: 12 Date: Tue, 17 Oct 2006 18:07:04 -0700 (PDT) From: sdgesa gaeharth <pollux1234567890@yahoo.com> Subject: Re: [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute To: asterisk-users@lists.digium.com Message-ID: <20061018010704.55118.qmail@web50806.mail.yahoo.com> Content-Type: text/plain; charset="iso-8859-1"

I cant seem to get outgoing calls to be placed properly .. No matter what I try I get an error from the PSTN company saying that the "call can not be completed as dialed" or "you need to dial a one..." The asterisk debugging seems to show the correct number being dialed out of the zap interface... the "9" is being stripped and there is a "1" where it is

I appreciate any recomendation of a simple CallingCard Application and resources of users manual. __________________________________ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/

This is a dialpaln issue. I solved the same problem recently. For 4 digit extensions you need to append the dialplan statement in the sip.cfg configuration file as follows <digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2 -9]xxxT|1xxxT" dialplan.digitmap.timeOut="3"/> Michael > I am not sure what I did but blind transfers do not

I am using Polycom 501s with asterisk 1.2.4. When transfering to call parking wih "#1" -> 700 the user is able to hear asterisk tell him what extension the call was parked on. However, when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on. It seems like the polycom is hanging up before asterisk is able to

On 25 May 2004 at 9:50, sdgesa gaeharth wrote: > I was using windows media player (with the patch for playing ogg). You mean you're using it for ogg-streaming? If I remember correctly just a few mails ago somebody said it only plays static files. Which plugin (patch???) are you using? Regards, Stefan --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project

Hi All, Now I configured a linux box as a router. And I installed Asterisk on it. My problem is whenever the WAN is offline all the sip extensions will logon failed. My sip extensions are connected to Asterisk through LAN. Why the LAN side sip phones cannot logon when WAN is offline. Who knows the reason and solution? Thank you so much. I really appreciate. Be Well. lanfei

please help!!! I am dialing into our asterisk server(TDM400p) from the psnt. I hear our voicemail message and I press the extention 1000. The Polycom ip phone in the office rings. I pickup but neither side can hear one another. What have I done wrong? thanks sip.conf: [general] context=local-access ; Default context for incoming calls bindport=5060

Hi, Porblem 1 ok this time i have checked every thing still it is giving me this same error. [root@niit125 root]# kinit junaid@NIIT.EDU.PK Password for junaid@NIIT.EDU.PK: kinit(v5): Clock skew too great while getting initial credentials can any one tell me how to enable NTP on a windows domain controller cause when i run the command ntpdate [root@niit125 root]# ntpdate 10.10.11.1 20 May

It's 40 kbps/22 kHz stereo. I tried to convince them to raise the sampling rate to 44.1 kHz, but they're determined to keep it within reach of urban dialup modem users. This aught to increase awareness of Ogg among artists and record collectors in NYC and beyond who would not have otherwise heard of Ogg Vorbis. 'FMU has a huge presence on the web. WFMU is a non-commercial, freeform

Using SAmba 3.0.2 and Cups 1.1.17. How does one configure a W98 box to print to a samba shared printer? Per my previous postings I can't figure it out, and the methods I've used on older Sambas don't work. I've been thru the lengthy discussion in the samba with cups documentation, which tells how to do it for NT or later, but not 9x. I've posted the smb.conf before,

I would like to have asterisk pass along the caller ID phone number to a database server on a my local network (the same network that the * server resides on ) so that our customer service app. can pull up customer data automatially. Asterisk passes along caller ID to the phones fine, can someone tell me how to make it pass this info to my database server? Any suggestions would be greatly

We have replaced the simple contact management system in Asterisk@Home with SugarCRM a full CRM system. This might seem like over kill for a home PBX but Sugar has the best contact management we have seen. With click to dial functionality and the ability to import data from other contact managers it’s a great fit for Asterisk@Home. We have also added new version of the usual Asterisk software AMP

We had added a lot to this release to our one button install of Asterisk. Now you can have even more features automatically installed and configured. Asterisk 1.0.7 AMP 1-10-007 Flash Operator Panel 0.20 Redesigned WebMeetme weather agi scripts Midnight Commander We have added some of our most requested features. - Web Meetme is now installed by default and the meetme2

I recently upgraded my debian sid install to the latest version of wine- 20050310. Unfortunately it broke Office 97. When I try to select a file in the file open or save dialogs, there is a Paragraph character in the box already. This makes it impossible to open a new file. Suggestions? Thanks, Aaron __________________________________ Do you Yahoo!? Yahoo! Small Business - Try our new

Hi I have tried to setup freebsd (with samba3 ports), backend is openldap with smbldap-tools. After a windows client joined the domain, I tried to right click a folder to set Security/permission. The windows client has a popup windows which display: The program cannot open the required dialog box because it cannot determine whether the computer named "DOMAIN" is joined to a domain I