similar to: Always Be Conferencing v16l "Looking Back to ASTERISK 13 Users Edition"

/**************************************************************************** * * * Always Be Conferencing (ABC) * * * * Creator: chris @ Penguin PBX Solutions * *

i am getting this error in ices: root@yurt:~/ices-0.2.3# ices -c /etc/icecast/ices.conf Cannot use config file (no XML support). Ices Exiting... o i tried to recompile it with XML support but... root@yurt:~/ices-0.2.3# make install Making install in src make[1]: Entering directory `/root/ices-0.2.3/src' Making install in playlist make[2]: Entering directory

my icecast install was working fine until today when it died inexplicably. now it says peter@yurt:~$ /etc/init.d/icecast-server start Starting /usr/sbin/icecast... start-stop-daemon: Unable to set gid to 102 what the HELL is gid thanking you in advance for the help, peter --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To

>From owner-icecast@xiph.org Sat Jan 19 07:40:36 2002 Return-Path: <owner-icecast@xiph.org> Delivered-To: icecast@xiph.org Received: from smtpout.mac.com (unknown [204.179.120.86]) by motherfish.xiph.org (Postfix) with ESMTP id 91B1918D08A for <icecast@xiph.org>; Sat, 19 Jan 2002 07:40:35 -0800 (PST) Received: from smtp-relay02.mac.com (server-source-si02

hi. is anyone out there familiar with running liveice with a usb audio input device? we want to do live encoding on machines other than big clunky towers with PCI slots, ie imacs, laptops, etc. there must be someone out there like us. please help!!!!! -yurt media center, amherst, ma <p>--- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage:

On 2020-01-15 11:24, Administrator wrote: 8<'s > One of the provider took a pcap and told us that expiration was set to 0 > that's why they don't accept the registration. We took a pcap on our > side when SIP packet goes out of our server and we see that the > expiration parameter is setted to 3600 ! Howdy, Maybe the clipping of your SIP packet is occurring on

On 2020-05-28 10:15, Doug Lytle wrote: > Everybody, > > I've had a request from my manager that I figure out how to get our Asterisk 13.x system using chan_sip to be able to display on the Polycom VVX series phone display (firmware 5.9.5), when an extension is called and the person on the other end is on the phone. > > He said, "Our old Analog phone system could do it,

On 2020-05-28 11:10, Doug Lytle wrote: >>>> But if you've already got the caller on the phone, then you might consider the CONNECTEDLINE function in Asterisk... > > And that we don't. > > It's the third party that would like the notification the the destination phone is currently busy with another call. CONNECTEDLINE only functions after a channel has been

On 2020-08-18 13:00, Carlos Chavez wrote: > users complain that confbridge calls end after about 30 minutes or so You might want to turn up SIP debug logging -- could be a re-INVITE is getting dropped, NAT pin-hole is closing, or some other network issue. -- 🤠 C. Maj, Technology Captain @ Penguin PBX Solutions 📞 USA Toll Free 1-833-PNGNPBX (1-833-764-6729) 🤙 International & SMS Texting

I had a similiar problem, but with calls dropping after 30 sec. It turned out that Android didn't support RP-CID (Reverse Party Caller ID) so when I sent the name of the callee to the caller (as some sort of "centralized phonebook function") it caused calls to be dropped as android refused to reply on the packets or sent rejections back. Check if you have some equipment on the line

My wife bought an inexpensive store brand web cam, so she can do video conferencing with people using Microsoft NetMeeting. Probably it is not supported in Linux and we will need to buy a supported web cam. I installed the Ekiga package. Is there something easier to configure and use or is Ekiga the best way to go? TIA! -- Lanny --------------------------------------------------------- Over 800

Could anyone recommend a package, preferably available via yum, to host web conferencing / collaboration? I am interested in being able to share an XP desktop with up to 5 other users. Any remote control capabilities would be a plus. Am currently running a Centos 4.4 server. Thanks, GB

Any one knows how to make Meet Me video conferencing room. Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily

UNICOM, Video Conferencing in Pakistan http://www.unicom.net.pk/ We at Unicom are pleased to inform you that we have expanded our network of video conferencing studios in all major cities of Pakistan including Karachi, Lahore, Peshawar, Islamabad, Rawalpindi, Quetta, Hayderabad, Nawabshah, Muzafarabad, Sialkot and now in Faisalabad and Gujrawala . All of our studio are equipped with professional

Hello , I am a newbie to * and have just been able to call a sip User Agent on a different machine thru *. I was trying to set up conferencing between 3 sip useragent on different macines at my worplace but was not able to figure out the procedure. I made the changes in meetme.conf and extension.conf as specified by someone in this mailing list, but * giving some error, " No ISA Tormeta

Hi this is just a heads up about an opportunity for commercial Asterisk experts. I don't know if this even possible but don't see why not and it is way beyond my capabilities so thought I would pass it out to the list. I've been looking into Microsoft Live Communications Server over the past few months for one of my clients, it's the same as ms messenger but for closed user

Ok, maybe a _wee_ bit esoteric, but... I've setup a developemnt system on a Sun Netra T1 running Aurora Linux (rh-7.3 for sparc64) -- just a few minor Makefile changes in codecs necessary. All in all, it's running nice. My only problem is lack of meetme or comprable features. I can't get zapdummy (these boxes use usb-ohci, not usb-uhci) to compile, and zaprtc sure won't work

http://www.voip-info.org/tiki-index.php?page=Asterisk%20bounty%20Meet%20 Me%20video%20conferencing I posted this bounty for $US2,000 some months ago. Basically I needed the ability for 4 or 5 of us to conference on a weekly basis which is why I was happy to offer this bounty, however I have only had 2 people make brief inquiries and no one has really offered any substantial indication they

I'm currently writing some code to support conferencing in Asterisk without using the Meetme application. The conference runs in its own thread and every new inbound or outbound channel that is created is passed to it. This thread runs the conference loop reading and writing frames to each channel. I'm writing this as if it were a bridge with more than two channels, and I'm not

Hi We are having a significant (> 1 sec) delay in a multi-asterisk conference, with IAX2 legs connecting meetme on different boxes. All the other legs are PSTN (TE410P). The example configuration Slave box 1 meetme <--- IAX2 ---> Master box meetme <--- IAX2 ---> Slave box 2 meetme The delay is between Slave box 1 and Slave box 2 The primary suspect is our iax configuration