search for: codecing

Hello, I'm running the 1/14/04 CVS and am having problems with faxes. I've tried faxing through both a Cisco ATA186 and Sipura SPA2000. I have Asterisk, the ATA186 and Sipura all hard coded to G711ulaw. When the fax machine trains, I get a stream of errors in the log file. The log is attached below. If you have any suggestions they would be GREATLY appreciated. Thank you!!! Jan 21

Hi everyone, Here's a follow-up on the two BoFs we've had about doing royalty-free codecs at the IETF. Well, the good new is that the proposal is now in IETF last call until January 20th (see below). No final decision has been made, so it's important to get as much support as possible for the Working Group proposal. You can see the ongoing discussion on the mailing list archive:

Hi everyone, Here's a follow-up on the two BoFs we've had about doing royalty-free codecs at the IETF. Well, the good new is that the proposal is now in IETF last call until January 20th (see below). No final decision has been made, so it's important to get as much support as possible for the Working Group proposal. You can see the ongoing discussion on the mailing list archive:

Hi Everyone, 1- Are the Siren7 and Siren14 the G.722 HD voice codecs? 2- Are these codecs only for Polycom units or are they universal across all other SIP phones that advertise the HD voice codec like Aastra? 3- What is the main difference between the two and is it advisable to run these over the INTERnet (not INTRAnet)? Thanks -------------- next part -------------- An HTML attachment was

libswfdec/swfdec_codec_ffmpeg.c | 23 +++++++++++++++++++---- 1 files changed, 19 insertions(+), 4 deletions(-) New commits: diff-tree 27f2f8fe232d392ddf50e97116724f507cc05a43 (from ed66672b6bd13a613aaf875d2f898cc2f16a47b9) Author: Benjamin Otte <otte at gnome.org> Date: Tue Jun 26 12:32:15 2007 +0200 add support for VP6 to ffmpeg diff --git a/libswfdec/swfdec_codec_ffmpeg.c

Hi all, i have purchased the g.729 codec from digium. The registration was successful. (with the "old" binary) But there're a few questions: - should not the codec listed in the codec list when i enter "show codecs" ? - the codec is named with g729b but if i enter show codecs there is a codec g729a listed also the g729b is not installed. what is the difference between

I'm building asterisk 11 beta 2. I've been using silk a lot. I don't see silk listed in menuselect as a codec. But I also don't see an asterisk 11 silk codec on http://downloads.digium.com/pub/telephony/codec_silk. Do we use the asterisk 10 codec_silk.so ? sean

Hi, I'm having problems with an asterisk server that's not offering Codecs for ulaw and alaw as it should. I've three servers in total: a1, a2 and "b" A1 and A2 have pretty much the same config files, except IP address info changes Server B is configured to accept all inbound invites. Calls from A1 to B, all work fine, and in a sip debug session I can see

I'm having a small issue with the oh323 implementation when it comes to codec selection. Version info: CVS Head 6/30/2004 OH323 0.6.3 OpenPhone for windows version 1.8.1 Asterisk is configured as a h323 endpoint which either terminates to the PSTN locally through a PRI or terminates the h323 call to an IAX provider remotely. Asterisk also has G729 licences installed. in oh323.conf we

Hi all I have a problem using the Speex voice codecs when using GnomeMeeting on one side and NetMeeting on the other side. I use GnomeMeeting under Suse Linux 9.0 to communicate with a friend working under Windows XP and using NetMeeting 3.0. Under Windows XP / NetMeeting we have installed and registered the Speex voice codec. (You can find more information how we have registered the Speex codec

Are there any other codecs that can be used with the 7960 and the ATA-186? I have been using the default gsm codec and wanted to see if I could make use of something a little less bandwidth intensive. Kim Callis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030723/34e950e0/attachment.htm

I have seen the list of codecs for the ATA 186's but not sure if it was 100% or not. I want to know really is it possible to run GSM or ilbc on them or is a G729 lic the only way to get a low bandwidth codec? This is the list of codecs that I have seen. RxCodec and TxCodec?Configure the codec ID. * G.723.1?Codec ID 0 * G.711a?Codec ID 1 * G.711u?codec ID 2 * G.729a?codec

libswfdec/swfdec_audio_flv.c | 5 +- libswfdec/swfdec_audio_stream.c | 2 libswfdec/swfdec_codec.c | 27 +++++++----- libswfdec/swfdec_codec.h | 31 +++++++------- libswfdec/swfdec_codec_adpcm.c | 2 libswfdec/swfdec_codec_ffmpeg.c | 85 ++++++++++++++++------------------------ libswfdec/swfdec_codec_gst.c | 32 ++++++++++++--- libswfdec/swfdec_codec_mad.c

libswfdec/swfdec_codec_audio.c | 72 ++++++++++++++++++++++++++++++++--------- libswfdec/swfdec_codec_gst.c | 11 +++--- libswfdec/swfdec_codec_video.c | 64 ++++++++++++++++++++++++++++++------ libswfdec/swfdec_sound.c | 1 4 files changed, 117 insertions(+), 31 deletions(-) New commits: commit f31f005192c5b59e3fa793fef78cc8f06a30ba1b Author: Benjamin Otte <otte at

Does anyone know if it's possible to set the codecs for a number via an Asterisk command? Ie, yes you can set the codecs in sip.conf for a user, but I'd like to have a command that can set the same thing so that it can be done without having to change sip.conf. Essentially I want the user to be able to prefix a code to their dialled number to set their preferred codec for that call.

Hi! The encoding, decoding and recoding cost cpu time, that's sure. But does this time differs much depending on the used codec? Is - for example - a G729 faster than a GSM codec? Bye! Michael

Hi David! Many thanks for your very encouraging and kind feedback!!! (Hope you don't mind that I made your post public on the Theora channel). I try to be the more sincere as I can in my posts.Some people suggested me to make demo pages to explain how the codec works step by step, and I realize that it would be good, because certainly it will show that the NHW codec is not that complex and

Use the new non-coherent DMA API including proper ownership transfers. This also means we can allocate the buffer memory with the proper direction instead of bidirectional. Signed-off-by: Christoph Hellwig <hch at lst.de> --- sound/mips/hal2.c | 58 ++++++++++++++++++++++------------------------- 1 file changed, 27 insertions(+), 31 deletions(-) diff --git a/sound/mips/hal2.c

Hello, all How can I write sound file with external G723.1 codec ( actually I have CISCO that can make H323 call to Asterisk box with G723.1 or G729 codec ) I am trying to start Record application by specifying in extensions.conf [writesound] exten => s,1, Answer exten => s,2,Record(soundexample:g723sf) or ...... ( soundexample:g729) I'am using oh323 channel driver, in oh323.conf

ive installed asterisk but below is what am getting proces gets killed.please help [root at localhost sounds]# asterisk -vvvvc Asterisk 13.13.1, Copyright (C) 1999 - 2014, Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under