similar to: First time user, problems on FreeBSD

Michael, I think you are right. However, I cannot successfully download and install icecast from icecast.org. When I do so, gzip complains: -- bash-2.05a$ tar fxz icecast-1.3.12.tar.gz gzip: stdin: not in gzip format tar: child returned status 1 -- Is the distro corrupt? or am I missing something? Is there a way to generate the crypted passwords? I could not get makepasswd.c to compile as

I have gotten icecast and live ice both up and running, but I cannot get them to play nicely together. I have the following in my configs: icecast: -- encoder_password test admin_password test oper_password test port 8000 port 8001 -- liveice: -- SERVER 192.168.1.2 PORT 8001 PASSWORD test #ICY_LOGIN X_AUDIOCAST_LOGIN -- but when I run both of them using: ./icecast ./liveice I get (from

@ben: try "tar xzf foo.tar.gz" daniel ditgens <p><p> Ben Timby <btimby_indy@ An: icecast@xiph.org

I am trying for the first time to get wine to build on my FreeBSD machine. I have XFree86 4 and FBSD 4.7, I tried building from source, using the sequence: ./configure make depend make install and also, using the ./tools/wineinstall script. Both of them produce the following error: -- cd `dirname avifil32/__install__` && make install ../../tools/mkinstalldirs /usr/local/lib/wine

My guess is that you have a space at the end of your password. 'Carl <p>-----Original Message----- From: Ben Timby [mailto:btimby_indy at yahoo.com] Sent: Wednesday, November 06, 2002 11:22 PM To: icecast at xiph.org Subject: [icecast] icecast + liveice won't play nicely <p>I have gotten icecast and live ice both up and running, but I cannot get them to play nicely together.

Hello List Asterisk 13.14.1 in use with pjsip stack. On the remote side is a SBC which performs some 'nat' detection. I suppose this means the SBC listens from where it is getting RTP data and then replies to that ip. As long as the asterisk is initiating the call this is fine, the asterisk start sending RTP to the media IP of the SBC and the SBC is sending media back. Now I want to do

Hi, I manage an SBC which stands between my company server farm and some SIP telco trunks. The system works fine, for inbound and outbound calls. Now I've configured the SBC to also act as a registration proxy, forwarding SIP registrations coming from the Internet to my asterisk servers. It all seems fine, but it doesn't work well, because by the time at least one phone registers through

*Hi All, I have starting to reading About SBC and found one artical reagding SBC and they gives a solutions like this. i want to know is this true in realtime sceanario while we think of an big implementation and is it possible with cloud computing. i have found from http://www.smartvox.co.uk/products_gateways_explained.htm Asterisk as a Session Border Controller* Equip the Asterisk server

Hi List I have been pondering over a problem to use an asterisk server behind an SBC unable to successfully handle registrations. Now I observed something strange which I suspect might be a bug on the asterisk side. The SBC originates is register from Port 6011 to Port 5060 on the Asterisk. The Contact Header of the REGISTER contains: Contact: user at SBC-IP:6011 The Asterisk is sending the

Hello, I would like to know the difference between encrypt the rtp and signaling between two asterisks, or putting an SBC in front of each Asterisk pbx. I'm trying to understand whether an SBC could fit an Asterisk deployment Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello List I work at an SIP Provider and we have added and SBC in front of our Voice Switch to protect it. This requires all our SIP Trunk customers to register via a 'proxy'. I struggle with Asterisk to work over a proxy. This is what I have done so far. register => username at sip.example.com:password at sbc.example.com This works fine, asterisk is sending registrations via the

Hi.. I am about to replace my aging Nortel Venture system with an Asterisk system and 6 Polycom IP 501 phones, and a couple sipura 841's for less used areas. We have 3 phone lines here. One is SBC, one Vonage, and one Voipjet... One hangup is that I can't figure out how to pass through a voicemail waiting indication from SBC. This is important because my wife and her family all

We're running Asterisk 1.6.1.17 for our campus voicemail server and Juniper M120s as our SBC. Unanswered calls, which arrive via the SBC, are diverted to voicemail using a 302 redirect when the called party doesn't answer. In this case the caller is able to hear the greetings and begin to leave a message only to have Asterisk terminate the call mid-recording. We're uncertain why

Why doesn't the CDR show outgoing numbers? I need a record of outbound digits dialed to reconcile my phone bills. __________________________________ Do you Yahoo!? SBC Yahoo! DSL - Now only $29.95 per month! http://sbc.yahoo.com

Ben, This is what I have in my liveice.cfg file. #SERVER xx.xx.xx.xx SERVER xxxxx.xxxx.xxx PORT 8000 PASSWORD xxxx USE_LAME3 /usr/local/bin/lame ENCODER_ARGS --lowpass 6 --silent --noshort -q2 SOUNDCARD HALF_DUPLEX SAMPLE_RATE 48000 BITRATE 64000 STEREO X_AUDIOCAST_LOGIN NAME KSL Radio 1160 GENRE KSL's Testing Internet Radio URL http://xxxxxx.xxx.xxxx/ DESCRIPTION Utah's Home for News,

Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress enabled in * we are having a few disconnects while calls are in session (about 2 reported in first 5 days of use). I have talked both to

Hi I'm trying to get the rebuilt parking functionality to work in Asterisk 12.0.0. In Asterisk 11.6.0 I managed to get a call to get parked by adding a dynamic feature in features.conf for the DMTF sequence *# which called a macro in extensions.conf, which then runned the ParkAndAnnounce application, and the call got parked. The syntax for ParkAndAnnounce I used was this (I don't

Hey List I sometimes use our asterisk server to do some debugging or other PBX and SBC. Now we have a case where a PBX is replying an incomming invite with 180 ringing immediately. It looks like the SBC does not accept this. According to my understanding of the RFC 3261 any provisional (aka 1XX) reply should be good enough to make the sender stop re-sending invites and accept this as a reply

Hi, As many of you may know, we are undertaking several tests in order to test the interoperability between several PBX IP from different vendors. Until now, we were trusting that the VoIP IP PBX were good enough to be interconnected directly, however, one of the vendors have presented the "SBC" concept. The "SBC" (Session Border Controller) is not a new concept since we

I have a pretty strange problem. I have about 100 DID's that come down a PRI from SBC in the United States. On Friday afternoon, one of my DID's flipped out. When you call it, the SBC operator comes on and says that the line has been disconnected. I contacted them and they ran test and they are telling me the problem has to be on my end. My problem is that the CLI never shows the