similar to: Help me with not normal vox pls

I found some posts regarding this issue dating of September 2003, but no real answer. The ADPCM format supported by Asterisk (the .vox files) is 4-bit, 8 kHz. I need 4-bit, 6 kHz, which is also a widespread Dialogic format, to help migration. Is there an existing format/codec for this? If not, can I make myself a shared object in /usr/lib/asterisk/modules? Is this easy??? :-( Thanks, Yves

I have been looking at the medium-rate codecs in Asterisk - ADPCM and G.726. Both of these are adaptive PCM codecs - the G.726 one is a little more expensive in processing power, however both are 32k bit-rate. I am experiencing problems using G.726 where the audio level is high. It produces loud clicks as if clipping. For quiet audio however, it seems fine. ADPCM (Digilogic VOX?) seems to be

I''ve got something that I''m not sure is actually doable. I have a very complex model relationship with a lot of parent/child/grandchild/ greatgrandchild etc stuff going on. Is there a way to do a route like this? tld.com/ projects/:project_name/:sequence_acronym/:shot_number/:department/:eleme nt_name/:version/ where: project is the parent of sequence sequence is the

All, I am getting an undefined method exception while trying to use a helper method in a xerb file. I am running version 0.0.8 module Merb module FredHelper def blee( args ) .... end end end In the xerb template I have xml.bobo blee( args ) At runtime I am getting undefined method ''blee'' I tryied

Hi, I've already sent this to feedback@vorbis.com, but I got no response and this might be more correctly placed here anyway, so here is a revised vesrion. I have one thing to criticize, which is the file extension *.OGG. It's ambigous (the Netrek meaning) and using it for both video and audio seems confusing. Plus, there are a lot of OGG files floating around that are generated by

Hi, I've already sent this to feedback@vorbis.com, but I got no response and this might be more correctly placed here anyway, so here is a revised vesrion. I have one thing to criticize, which is the file extension *.OGG. It's ambigous (the Netrek meaning) and using it for both video and audio seems confusing. Plus, there are a lot of OGG files floating around that are generated by

Is there a way to play adpcm, 6KHz in asterisk? If yes, where can we get this codec? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030916/b8be2453/attachment.htm

Is there any particular protocol we should use to avoid stomping on each others toes when working on particular patches? It is quite easy to generate double nasty conflicts when working with patches of patches. I'm currently working through the patch set starting at the top (my patch tool was unhappy with 002-sync-bitops), working down my todo list. bobo@linux:~/paravirt> hg status M

Dear R-devel, My current research focuses upon sensitivity analyses which require [1] clusterings of patients in a baseline covariate X-space and [2] examining the distribution of within-cluster treatment differences in outcome. I have generated some primitive R code for this, but I really need to be able to interact with graphical displays of within-cluster information (local average

Hello, Is there a way to change the format from the default? 'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format ${MEETME_RECORDINGFORMAT}). Default filename is meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav. - requires chan_zap.so Many thanks ******************************************************************** This email and any attachments

Please, try exten => 99,1,Wait,1 exten => 99,2,Record,/tmp/pcmfile:pcm exten => 99,3,Wait,1 exten => 99,4,Playback,/tmp/pcmfile exten => 99,5,Wait,1 exten => 99,6,Record,/tmp/voxfile:vox exten => 99,7,Wait,1 exten => 99,8,Playback,/tmp/voxfile (put your own extension). Pcm recording is OK, playback is OK. Adpcm recording is noticeably worse. Adpcm playback is very

Hi, I have a peculiar problem with my Samba installation. I have a directory with lots of files that I want to make available via Samba. I can connect just fine, but I quickly noticed a lot of files seemed to be missing. After some testing I found out that the directory listing was simple cut off, right in the middle of a filename even. I created a test-directory with 1000 random files in

I was just wondering if someone has experiences to use Asterisk in an existing Ericsson MD110 environment. Particulary I'd like to know if it is possible to use the MD110's system phones directly connected to Asterisk. I'm not very familiar with it but would it be possible to use ADSI with these phones? Are they more like analog or more like digital phones or is the protocol even more

Hi, Ive been struggling with asterisk for a few days now. I understand pretty much how it works and how to tie things together (SIP -> SIP internally works fine etc), but just cannot get asterisk to work with an X100P clone (its a Ambient MD3200, if that means anything to you guys). I have tried (initially) asterisk 1.07 with zaptel 1.07, and now Asterisk CVS-HEAD with zaptel cvs. Both give

Hello Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0 and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd like to play PCM WAV files instead of eg. GSM. Per... www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk ... I recorded a sample of my voice using XP's Sound Recorder, then ran the following : sox test_wav.wav -r

I have a site that I don''t think "returns" a basic_auth request, but is able to use basic_auth. In the past on 0.6.8, I could use the following code: require ''rubygems'' # gem ''mechanize'', ''=0.6.8'' require ''mechanize'' agent = WWW::Mechanize.new agent.basic_auth("username", "password")

Could anyone please direct me on how to make a nicer table in R? THANKS FOR ALL THE HELP! I would like to make a table with the following in it: estimate, t value, significance, beta, standard errors, adjusted r squared, and residual standard error (3 decimal points if possible, but I can do it by hand). Also I'd love to include the source of my information on the bottom of the table if

Greetings! We have a somewhat noisy background in our call center, and I'd like to reduce this. Obviously, we could plaster the walls with sound absorbing material, but is there anything we can do in software either using any algorithms for our open source-based SIP library or inside Asterisk itself? Related to this, anyone have a good source for good panels? We are using

Hi Users, I was wondering if someone of you have the same thing on CentOS 64x? asterisk01*CLI> core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723

Hi everyone, We are using the IAXy boxes and Asterisk over the internet and I was wondering if Asterisk can do a codec translation during a call in order to lower the bandwidth that the comunications consumes? I mean, the IAXy boxes only support the ADPCM and uLAW codecs, but for a certain number of calls our bandwidth runs out, then I think if Asterisk can convert the signal that comes in ADPCM