similar to: Virtual fax on the Asterisk box

I know speex is not supposed to do a great job compressing music, but I've noticed that the new VBR code chokes completely when you try to compress horns. I've placed a particularly offensive example up at http://www.utdallas.edu/~matthias/ . Take a look at a-16m*{ogg,spx}. a.ogg is the first minute of an ogg created from the source media (in 44khz stereo). The rest have been mixed down

Is there any way to access only the narrowband portion of a wideband stream? I'd like to be able to encode the audio only once, but allow members in a conference to have some rough selection of bandwidth, and allow them to move to a lower-bitrate stream if there is a need to do so. Thanks, Matthias -- Matthias Granberry matthias@utdallas.edu (469) 371-0596 --- >8 ---- List archives:

Hi, I try to use Speex codec into Win32 platforms. However, I find the CPU bandwidth usage is very heavy on a Pentium 3 machine. Compare to Microsoft's G723.1 codec, speex 8k is using more than 20% cpu bandwidth. Does anyone know what is the best version of Speex to "beat" the Microsoft's G723.1's on CPU bandwidth usage? Does Speex have MMX-enabled codes? thanks very

I am eager to begin using speex for downloadable audio on my church's website - but first a good win32 player is needed given that windows has a 94% domination... Are there any plans for an up-to-date speex player for windows? (that winamp plugin is hopelessly old) Building this into zinf (fmr. freeamp) might be an option, but their code is one big mess. --- >8 ---- List archives:

I'm going to use speex to compress sound that was record from a sound board. What container should I use so it will be able to play in some sound player available for windows, ogg or wave? If I choose to use wav container I simple have to put the speex stream into the data chunk of the wave file? I heard that there is a codec to put speex in a wav file so it can be played in windows media

Hi, Just to let you know that Speex 1.0 RC3 is out. It should be the last RC until 1.0 is released (tentative release date is March 16th). Please give RC3 a try and report bugs so they can be fixed before 1.0. As usual, download from http://www.speex.org/download.html Jean-Marc -- Jean-Marc Valin, M.Sc.A. LABORIUS (http://www.gel.usherb.ca/laborius) Université de Sherbrooke, Québec,

At 12:35 PM 10/6/2004, Matthias Granberry wrote: >There is some SSE assembly >language, but it's in GCC/AT&T syntax rather than the windows-standard >Intel syntax, so you might have to do some of your own translation to >something your compiler understands. We submitted an intel patch a ways back- maybe it is in the archives somewhere... Tom -- Tom Harper -

Hi all, Which is the practical (from your experience) limit of the number of X100P cards installed in a single Asterisk box? Asterisk can work reliable with 6 X100P cards in the same box? Anyone know when the 4 ports FXO Digium card will be available on the market? Many thanks, Dan P.S. Please do not aswer with RTFG ...tried before without success...:-))

Hi, Anyone knows a software fax device which can act as a SIP UA? I want to have a SIP based FAX machine (sofware) on a PC associated with an Asterisk extension. Thanks, Dan

Hi all, There is something special I must configure in order to get the voice mssage by mail? In voicemail.conf I have: serveremail=asterisk@mydomain.ro attach=yes [default] 301 => 6535,Home Mailbox,dtoma@fx.ro I have tried to let a message for 301, but this message is not forwarded by mail. I am missing something? Thanks, Dan

I've been given the task to hunt down a codec that out solar car team can use to transmit and receive data over a radio link between a chase car and the solar car its self. As of now I do not know the exact specifications of the internal hardware of the car, what I do know is they are looking at transmitting across a 2400baud connection, the unit in the car is homemade from our labs here.

Has anyone written a wrapper to use with .NET for the speex Win32 library? I had started to work on it but thought best to ask first. :) Thanks, Jessica <p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'speex-dev-request@xiph.org' containing only the word

On Wed, Mar 31, 2004 at 10:51:20AM +0300, George Bratis wrote: > I think that if you want to use a low cost dsp you must rewrite ti in > assebmly. > In our tests you will need 4-5 times more MIPS just compiling C code. This was also our experience porting the Tremor vorbis decoder to a TI dsp. The vendor toolchain is useless; someone really needs to pay cygnus to port gcc. -r ---

Hi, There is any possibility to dial a specific extension and then enter in your own mailbox (the one defined for that specific SIP phone) without asking for the exxtension number but only for the password? I want to be the same extension for all phones, not a specific one for each of them. It is possible to have a time stamp in the recorded message? I want to know when the message has been

Hi all, A new version (0.9.6) of DIAX is available for download at: http://www.laser.com/dante or http://www.geocities.com/tdanro There are no new functions, but some bugs fixed: What's new in version 0.9.6: - add Default_user locales as new language. The program language can be automatically selected based on default user locales on your system. You still can manually select the language,

Hi, It would be great if the IAX protocol will be able to tranfer fax data (even converted in another format) between Asterisk boxes, using low bandwidth codecs like GSM. I know that this is possible only with the G.711 now (passing faxes using the audio stream), but.... maybe in the future...some native support will permit this. Just a thought, Dan

Hi, I have a FAX machine connected to a TDM400 card FXS port. When I receive a fax call through X100 and transfer it to that extension, the FAX machine display REC, but nothing happen (no fax received). There is something special to be done for this configuration? Thanks, Dan

Hi to all, I've just received my Cisco 7905G ipphone. I want to connect it to asterisk server but it looks that it has been preloaded with sccp protocol, so I guess I need H.323 or SIP firmware image of some kind. I have a working tftp server on my asterisk box also....What do I need to do now to get things wokring? Thanx in advance, Victor...

Has anybody had experience with installing Supervoice (2.6) fax/voice software in wine? I've installed it apparently OK (although I do seem to remember getting some error on installation) - it seems to start to run OK, but with some glitches! Each time, it wants to re-detect and configure the modem - why?! Also, it shows me the messages (fax and voice) which were stored in the modem (USR 56K

I am having a problem with softphone registration, having read the list and watched it for a while for similar problems I just cant seem to figure out the problem. Using SJPHONE or DIAX , I can make outgoing calls but I can't get them to register with asterisk, I have other sip devices registering OK-7940's. From the list and the digium web site this seems to be a straight forward set up