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Good day all I want to know what the best codec is to use for asteris for VOIP We have two towns connected with a 64k line that's going to do VOIP with astersik.At the moment with the default installation the quality is bad and the bandwith is high. Is this even a codec problem Pleas help ALtus

...ahdi channel to another dahdi channel? 2) I'm getting the following error on the Asterisk CLI unless I disable the codec_dahdi module: ERROR[18854]: codec_dahdi.c:399 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory Is this because I do not have a hardware trancoding device? Can I safely ignore this error or is it a bug? Brandon. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090225/11fb3dfe/attachment.htm

...a good idea? On Fri, 04 Mar 2016 14:48:41 +0100, you wrote: >Great tool to do this: liquidsoap > >Keep in mind that transcoding degrades the quality of tour stream dramaticly. You can avoid this by feeding liquidsoap or stream transcoder with a highquality or even transparant stream and trancode this to the different streaming formats you like. I used to do this by feeding a flac stream to liquidsoap and transcode this to 5 different stream formats i needed. But you might be fine if you feed your transcoder with 320 kbps AAC and transcoder this to lower formats. Maybe experiment with a hi...

All the broadcasters on the server which I support deliver their content in MP3 format. Recently, there's been interest in supplying a second AAC stream at half the bandwidth but with the same audio quality (64kbps AAC versus 128kbps MP3) like TuneInRadio does for delivering their content regardless of the source. I've thought of using a third-party product called Stream Transcoder, but am

...hat's not worth it. > So why can't support for mp3 be added to ices2? From a technical > point, I mean? Somebody said "It's not and never will be". Any good All transcoding suffers from this. it doesn't matter if you do it before hand converting all your files or trancoding on-the-fly inside something like ices. Nothing is stopping it technically, but it's the non-technical points that are stopping this, there are patents on mp3 that people need to consider, ices2 is a xiph project so producing ogg streams comes before sorting out mp3 and of course some develo...

...> > So why can't support for mp3 be added to ices2? From a technical > > point, I mean? Somebody said "It's not and never will be". Any good > > All transcoding suffers from this. it doesn't matter if you do it > before hand converting all your files or trancoding on-the-fly inside > something like ices. > > Nothing is stopping it technically, but it's the non-technical points > that are stopping this, there are patents on mp3 that people need to > consider, ices2 is a xiph project so producing ogg streams comes > before sorting out...

...ages; or any other signaling gateway > e.g. socketio > 2. Capable to generate "answer-sdp" and return backing using same > signaling gateway. (make sure that ICE are gathered & included in the > anwer-sdp) > 3. For WebRTC, media server MUST support DTSL/SRTP to capture, trancode > and forward encrypted RTP/RTCP pakcets > > I don't know if icecast2 has such functionalities built-in. It currently > accepts PUT requests only on HTTP. > > ?You can install STUN server side by side with icecast2 on a separate > port.? > > ?I think a gateway (a mid...

Great tool to do this: liquidsoap Keep in mind that transcoding degrades the quality of tour stream dramaticly. You can avoid this by feeding liquidsoap or stream transcoder with a highquality or even transparant stream and trancode this to the different streaming formats you like. I used to do this by feeding a flac stream to liquidsoap and transcode this to 5 different stream formats i needed. But you might be fine if you feed your transcoder with 320 kbps AAC and transcoder this to lower formats. Maybe experiment with a hi...

...anding of T.38. 1. What T38FAX_VERSION_0 or T38FAX_VERSION_1 in chan_sip.c means ? voip-info.org implies one has to change values in chan_sip.c to make it work. Shall I set T38FAX_VERSION_1 or leave T38FAX_VERSION_0 in global_t38_capability ? Source code says "This is default: NO MMR and JBIG trancoding, NO fill bit removal, transferredTCF TCF, UDP FEC, Version 0 and 9600 max fax rate" What does "Version 0" relates to in this aboxe line ? MMR and JBIG transcoding ? 2. A softphone like Zoiper is able to receive voice or fax calls. When setting those values (see bellow) in sip.co...

...4 Initially I had no txgain=5.0, but users complains that they hear not as good as before asterisk was added into this scheme. We even tested and found that yes, asterisk decreases volume. Could you tell me why? As I understand this is digital connection and there is the same alaw codec, so no trancoding. Why can be sound volume changed during chan_dahdi native bridging? Thank you!

Need some help understanding codec preferences: I have 2 asterisk servers. Server 1 sends calls to the PSTN and has allow=g729 allow=gsm and allow=ulaw in iax.conf Server 2 receives calls and routes them to server 1. It has the same allow lines. We receive calls from a phone co and route them via server 2 to server 1. The calls originate in g729 and everything works fine. Now I want to take

Hi i'm trying to connect to the PSTN in the following way sip ATA -> * -> gnugk -> Cisco AS5300 -> PSTN I'm using asterisk CVS-HEAD-06/01/05-14:33:15 running over RH EL3 Asterisk-Oh323 0.7.2 pre1 Open H323 v1.13.5 pwlib v1.6.6 and I'm having a lot of trouble, gnugk and * both have public ips and are not behind any type of firewall, the sip ATA is behind a firewall and

> Best wishes > Stefan (on second thought ) :: my last post assumed that you had .OGG files to begin with .. depending on the size of your library, you may want to convert your mp3s to ogg . i ended up doing this , and it made things much simpler on the web end . also ; check out dir2ogg .. echo "dir2ogg written by Darren Kirby; linuxbot@shaw.ca; www.badcomputer.no-ip.com/"

Hi all - I am using asterisk 13.27.0 with Linphone. I turned off all codes on linphone except the one I want to try. For example: opus and speex (so only one enabled at a time). Then did this same on asterisk for the linphone extension. disallow=all allow=speex (for example). Then I place my call and the call fails. if I enable something like gsm, ulaw, alaw the call works fine. Why does the