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4) Subtract 1 from the keyframe, then repeat step 3). 5) Begin reading from the frame discovered in step 4. Drop any packets which are output on the first page. Count down until we reach the keyframe, dropping packets until then. 6) Continue counting down until we reach the target frame, we are now decoding each frame/packet. At the target frame produce the YUV output. Steps 4 and 5 are

One of the issues I've consistently run into with the cortado player app is seek behavior, so I was curious to see how cortado handles video encoded with the latest ffmpeg2theora (0.26) and the new -seek-index option, so I tried converting an h.264 video: ffmpeg2theora tronlegacy-tsr1_480p.mov --seek-index This output the following advisory messages: > Allocated 372 bytes for theora

Hi all, looking at the theora spec, it says that it is possible to encode a video with a frame offset for the actual image. How is it possible to do that ? I don't see any option for it in the encoder_example. Regards, Salsaman. http://lives.sourceforge.net https://www.ohloh.net/accounts/salsaman -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I just did a brief test with 3 different codec combinations: 1) ffmpeg sorenson/flv, mp3 audio in flv container 2) ffmpeg h264/vorbis audio/matroska 3) ogg/theora/vorbis using encoder_example Here are the results: 10721201 2010-10-16 14:46 origem1.flv 20731108 2010-09-13 23:04 origem1.mkv 33101703 2010-10-03 11:24 origem1.ogv as you can see, flv wins hands down, and it doesn't look

an external program, which at this stage, is not customizable ... I don't know if alternatives (LiMO, Android, ...) would be more open to this customization but for Symbian, not only Nokia SIP client wouldn't let you autoanswer to SIP calls, but any other SIP client complying to Symbian design wouldn't support autoanswer. PS: Please, note that I'm far from being an expert in GSM

such file or directory" on pure-IP platform in which I installed asterisk-libpri-dahdi trilogy. Maybe, it's me while following README instructions, maybe README instructions could be improved or maybe it's wrongly labeled messages ? That's why I told myself : I'm waiting too much from doc ? is a pure-IP platform too specific ? what is the official policy ? README starts with

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 The attached patch allows frames to be dropped in Cortado: Why: Currently, frames cannot truly be dropped in Cortado. If the decoded YUV arrives too late at the videosink, it will not be converted to RGB, but every frame must still be decoded. On slow systems, once the queue of undecoded Theora packets fills up, the video decoder will block the ogg

That's the main reason I opened this thread as it surprised me a bit ... > > > Any 2-wire analog leg will be a source of echo. Many, many, many calls > do not have a 2-wire leg. Even in handset audio circuit ? I was thinking that any handset is a potential echo source due to this audio circuit ... Do you agree ? > Think cell/mobile or endpoints with PRI or T-1. > >

... Thanks, anyway for telling as at least, it reflects your needs. > > > You want NT PtMP and i second that, > not being limited on the asterisk > side is a must in the > telephony ecosystem, since the legacy PABX aren't alwsys easy to > reconfigure. > > _______________________________________________ > -- Bandwidth and Colocation Provided by

Here's the problem: I've rewritten large parts of the output code of cortado 0.1.0 with the following aims: 1. output to something SWT-like instead of AWT. 2. Speed it up a bit. In the rewrites I haven't restricted myself to java 1.1; the fact that it uses SWT all by itself makes it useless for applets anyway. The speedup worked very nicely; it's almost twice as fast as

connecting legacy PBX to Asterisk (for the very same reason, those PBX use TE-PTMP). If others could join this thread and say if they agree or not with NT-PTMP being the 2nd most needed mode, would be great. Please, do not hesitate to comment. > > > Right now, I would not preclude the possibility that NT-PTMP support > might be added, but I could not give you a concrete time at which

supported, beside Idle, On call and Ringing ? Can we expect this list to match DEVICE_STATE's one (UNKNOWN | NOT_INUSE | INUSE | BUSY | INVALID | UNAVAILABLE | RINGING | RINGINUSE | ONHOLD) > Might be worth seeing if other phones do the same. > > S > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by

tnet.itand SIP register messages are not replied. I suggested to check if your Asterisk box is really sending SIP messages, you can use a net sniffer. Did you alerady used different sip client with the same sip account of your Asterisk box? Did you use zoiper from the same box? Marino p.s. Are you Italian? On Mon, Jul 14, 2008 at 5:27 PM, Giorgio Incantalupo < gincantalupo at

2010/2/17 Arnaud Quette > 2010/2/17 Arjen de Korte: > > Citeren Charles Lepple: > > > >> I wonder if there are any cross-compilation targets we could use to test > >> some of the word-size assumptions. Also, we could add in some static > >> analysis tools. > > FYI, I submitted NUT to the Coverity Scan program > (http://scan.coverity.com) last

could be "hot". Is there any chance this would cause the card to fail after a while? It appears this site just had 4 port Digium card fail today. > Also, I am trying to cross connect with another Asterisk system with > > the normal LBO setting (i.e. span=1,1,0,esf,b8zs) but as of yet the > > systems aren't seeing each other at all. Could the side with the high >

/var/lib/asterisk/sounds/soundfile.alaw /var/lib/asterisk/sounds/soundfile.wav to go from alaw to mp3, first convert to wav, then use lame <options> /var/lib/asterisk/sounds/soundfile.wav /var/lib/asterisk/sounds/soundfile.mp3 sox looks like it can ogg/vorbis, but mine doesn't list mp3. You might fetch the source for sox and see if it can do mp3; lame is probably just as easy to obtain

This mail announces the release of Flumotion 0.5.2 'Can Tomas'. Flumotion is a GPL streaming media server written in Python. It is distributed and component-based: every step in the streaming process (production, conversion, consumption) can be run inside a separate process on separate machines. Flumotion uses Twisted and GStreamer. Twisted enables the high-level functionality,

mailboxes). Are you certain that removing either 612 or 610 mailbox would keep Asterisk from complaining ? > > However, the MWI does not indicate voice mails for 610 and I keep seeing > this error message: > > ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox > 610 in context a10 > > However, mailbox 610 is clearly defined in voicemail.conf: >

... Activating "sip debug" shows the register packets but nothing in return. ... I think that this is a network related issue, but you have to solve it by using a Asterisk config file. Unfortunately I think that the faster way to solve your problem is trying to understand if sip messages are correctly sent to tnet. I strongly suggest to use http://www.wireshark.org/ previoulsly named

at least once a week I receive such an attack coming from a different ip. I will read the articles. Thanks again to everyone. Regards, Rodrigo Lang. 2010/6/29 Kenny Watson <kwatson at geniusgroupltd.com> > Hi, you can use fail2ban >