similar to: [bug?] strange colors

Stock icecast, Sightly tweaked GStreamer (waiting for some bug fixes to get packaged). A script. An HDV cam. HTML webpages that remote attendees go to to click the video and watch. Done :-) Happy to send the script... Monty

Hi all, I was looking to build a SIP-to-PRI gateway using Asterisk (as in my other post), but there is also an alternative of using a Cisco router with something like an NM-HDV module with a T1 VIC module and DSP channel banks. The question is, would it be more reliable to offload all dahdi/zaptel/libpri type stuff to a dedicated gateway device (Asterisk or Cisco) and have the Asterisk PBX only

If you have a PRI terminated in a Cisco router talking SIP to * and would be willing to share your Cisco config, please respond. Also, I would be interested in knowing what version of IOS you are using. We are using an NM-HDV in a 3640. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

I planned to switch to linux ubuntu but I wanted to know some information on some windows apps I want to keep using. I looked through the app database and there were no results for the apps. The apps were: Wax 2.0, HDV split, and Sony vegas movie studio 7. Also, what are some linux alternatives to dvd architect?

Does anyone have a config they'd like to share w/ the above hardware doing termination for asterisk? I've got one coming in tomorrow along w/ some DSP's and would like to not have to create the config from scratch to start testing. W. Kevin Hunt

Hello, I have a linux box with the following S/W versions: CentOS release 5.2 (Final) Linux ns.hostname.ro 2.6.18-92.1.13.el5 #1 SMP Wed Sep 24 19:32:05 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux Version 3.0.28-1.el5_2.1 My server has 2 IP interfaces, eth0: with the public ip address and br0: for the private subnet. Br0 being a bridge ip interface between eth1 and tap0 ( for openvpn ). Now my

All, I know you can use, say, a 2620 w/2 port FXO card as a SIP gateway. Clearly you can use the 5300, 5800, and MGX8850 too. Does anyone know which cards, if any, exist for a 7206VXR to act in a similar capacity, either as a T1/PRI, DS3, or POTS FXO/FXS? What other Cisco routers can act as SIP gateways today? Thanks, Dave

So what are people using these days for SIP or IAX to PSTN gateways. 1. Do any of the standard companies (Packet8, Broadvox, Vonage, etc.) allow you to use your own SIP device (phone or something like *) instead of the interface hardware they usually provide? 2. What about latency and reliability? 3. Finally, do any of the providers deliver more than one call via SIP? In otherwords, if

Hi, We're currently editing a video advertizing Firefox and plan to release it exclusively encoded with Theora during a couple of days (before it hits YouTube, DailyMotion...). Editing is done with Final Cut Pro so sources are QT files. What is your experience of encoding QT to Theora? For the time being we are not down to testing xiphQT so we're looking for general advice. Thank you

On Mon, May 24, 2010 at 4:18 PM, Pablo Baena <pbaena at gmail.com> wrote: > html 5 doesn't support live video AFAIK. Interesting. I use it on a regular basis... it's how we're doing videoconferencing in RedHat right now. Is FF just going above and beyond? Monty

Helo, I have problem with Digium TE110P connected to CISCO 3640 (port on NM-HDV-2E1-60) wth PRI E1 link. I use CISCO now for testing but when I tried with real PBX problem was exactly same. I have this messages in Asterisk conole and log sometimes: NOTICE[1115] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Usually 2-5 such messages in series, can be repeated after 10

Hey all, I have an as5300 I use for dial in customers, we have 4 PRIs on it. We have a few free channels on it. I'm wondering if I setup SIP on the as5300 I can have asterisk use the free channels for dial out. I'd still have to use my TDM04B for incoming calls, but at least I can expand my outgoing. Anyone done anything like this before? I've never messed with VoIP on Cisco

Hello, I am using a Cisco-2,811 router with PRI as a gateway between Asterisk and Nortel TX-1. I had problems with name transfer and with the help of Cisco support I've fixed it. Enclosed here are the definitions needed for it. BTW, Cisco's CCM is using MGCP thus the Q.sig is handled by CCM. Here I am using SIP so the router must decode/encode the Q.sig. The Nortel should be defined

[moved to asterisk-users, as this is not a development question] At 1:40 PM -0400 on 4/24/04, Sudhir Kumar wrote: >I would like to hear from any of you who has done any kind of >benchmarking on a robust hardware that can handle large call volume, >preferably with G.729 codec involved. > >We are in the process of putting together a system that should have a >quad E1 card, G.729

2 Asterisk servers 1.2.12.1 Connected via IAX2, same switch, GigE, no packet loss, etc 1 with a Sangoma A101 for a PRI to the PSTN Ulaw QoS enabled NAT for the registered ATA boxes, no nat between the * servers Faxing inbound: Call from PRI hits the first Asterisk server Then talks to the 2nd via IAX2 NVFaxDetect receives the fax, converts to PDF and emails it out Works great!

hi i'm looking for information about the multichannel spec. of vorbis. 1/ the documentation specifies that 256 channels are available ( -c 256) ... it sounds very well but how may i specify the original files ??? (i haven't 256 physical entries on my soundcard ... arffff) so is there a way to declare something like oggenc file1.wav file2.wav file3.wav -c3 final.ogg ??? 2/ panoramics are