search for: codec's

...A2000. 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 16:06:59 NOTICE[10010661]: Unknown RTP codec 60 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 60 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 56 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 94 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 60 received Jan 21 16:06:59 NOTICE[10010661...

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 ----------...

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...

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...

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 sess...

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....

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 in...

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/pipe...

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...

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 f31f00...

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...

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 it will be more easy then to contribute for people who want and have time. For now, I focus on high compression settings, for me it's very complex, but currently h...

...c index ec84bc4c3a6e77..9ac9b58d7c8cdd 100644 --- a/sound/mips/hal2.c +++ b/sound/mips/hal2.c @@ -441,7 +441,8 @@ static inline void hal2_stop_adc(struct snd_hal2 *hal2) hal2->adc.pbus.pbus->pbdma_ctrl = HPC3_PDMACTRL_LD; } -static int hal2_alloc_dmabuf(struct snd_hal2 *hal2, struct hal2_codec *codec) +static int hal2_alloc_dmabuf(struct snd_hal2 *hal2, struct hal2_codec *codec, + enum dma_data_direction buffer_dir) { struct device *dev = hal2->card->dev; struct hal2_desc *desc; @@ -449,15 +450,15 @@ static int hal2_alloc_dmabuf(struct snd_hal2 *hal2, struct hal2_codec *codec...

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 oh...

...39;: Found == Parsing '/etc/asterisk/asterisk.conf': Found == Manager registered action DBGet == Manager registered action DBPut == Manager registered action DBDel == Manager registered action DBDelTree PBX UUID: 36c0be35-766e-46c4-be1f-96479afa1e2b == Registered 'audio' codec 'g723' at sample rate '8000' with id '1' == Created cached format with name 'g723' == Registered 'audio' codec 'ulaw' at sample rate '8000' with id '2' == Created cached format with name 'ulaw' == Registered 'audio&...