similar to: ADPCM 4-bit, 6 kHz

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

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

I am projecting to migrate applications written with the Dialogic API to Asterisk. There are a few things that I do with Dialogic that I am still not sure are possible with Asterisk : - play a file from an offset (so a user can press a key to rewind 3 sec., pause, etc.) - dynamic volume control (the user can press a key to increase or decrease the volume) Thanks if anyone has a clue... Yves

When I try to call iaxtel it goes to codec ADPCM even though I have define in iax.conf gsm Call accepted by 69.73.19.178 (format ADPCM) -- Format for call is ADPCM My settings: [general] port=4569 register => xxxx:xxxx@iaxtel.com bandwidth=high jitterbuffer=no tos=lowdelay [voipjet] type=peer host= xxx.xxx.xxx.xx secret= xxx auth=md5 notransfer=yes context=incoming disallow=all ;

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 All; I need to buy one IAXy device, but I discovered that it supports only g711 and ADPCM codec, so I was wonder that it does not support g729 or GSM?! Anyway, what is that ADPCM and how much it consumes bandwitdh? Also, asterisk support such codec? What its name in the configuration? Any advise? Regards Bilal

Greetings: It would be nice if Icecast supported RTSP; however I would appreciate any suggestions for a small RTSP/RTP solution to encode 8kHz mono audio in GSM or ADPCM and service multiple unicast client connections. The ideal would be a black-box hardware solution with an audio input and ethernet interface similar to broadcast studio IP audio links or the network audio capabilities of certain

Hi everyone, I'm using the IAXy boxes and i'm having some trouble when I use it with the ADPCM codec. When I use the ADPCM codec only one person (out of the two of the conversation) is able to hear the other, but when I switch to ULAW codec everybody can hear the other. The ULAW codec is too heavy for my bandwidth (64Kbits/s) and its sounds choppy, the ADPCM codec sounds good but only

Thomas B. Ruecker wrote: > Michael Grigoni wrote: > >>Greetings: >> >>It would be nice if Icecast supported RTSP; > > It probably never will > >>however I would >>appreciate any suggestions for a small RTSP/RTP solution to >>encode 8kHz mono audio in GSM or ADPCM and service multiple >>unicast client connections. > > why not use

I'm copying the flac-dev list to see if anyone has any feedback also... --- Juhana Sadeharju <kouhia@nic.funet.fi> wrote: > Hello again. I had time to check the paper out. I have filled the > steps given in the paper with formulae, and then written a piece of > C code. It is not complete code, but could be a reasonable start. > Maybe there is one typo in the paper -- I have

Le mar 18/11/2003 à 17:39, Allen Drennan a écrit : > Speaking of video conferencing in conjunction with Speex, we are > currently beta testing a solution we developed that offers multi-point, > multi-party video and audio using the Speex engine for voice. > > http://www.wiredred.com/downloads/ecsetup.exe > > The fair and good audio settings are Speex narrowband, high quality

I found out the answer to why the bad password count was failing by following the stack trace If anyone has the same problem, this was caused by my access rights being too restrictive. source/passdb/Init_sam_from_ldap.c:1064 => There is a "goto fn_exit;" in case ldapsam_get_entry_timestamp() fails before the cache is read via login_cache_read. This function will fail if it cannot

Hello, What do you think of this project www.openpbx.org ? Something like ser and openser ! Kinds Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger T?l?chargez cette version sur http://fr.messenger.yahoo.com

I want to record audio in ADPCM format. According to the "show codecs" output of Asterisk, it looks like it supports adpcm. But I do not know what to tell the "RECORD FILE" directive in my AGI script. The RECORD FILE command usually has this form: RECORD FILE <filename> <format> <timeout> [BEEP] It records fine in WAV or GSM if I enter "wav" or

Hi everyone, I'm using the IAXy boxes and i'm having some trouble when I use it with the ADPCM codec. When I use the ADPCM codec only one person (out of the two of the conversation) is able to hear the other, but when I switch to ULAW codec everybody can hear the other. The ULAW codec is too heavy for my bandwidth (64Kbits/s) and its sounds choppy, the ADPCM codec sounds good but only

>From: Josh Coalson <j_coalson@yahoo.com> > >I'm copying the flac-dev list to see if anyone has any >feedback also... I'm supposed to be there myself since yesterday but have not got the first digest yet. >First, the results they show are for compression of data >that has already been lossily quantized to fewer bits per >sample, e.g. u-Law and A-Law are

Greetings, I'm trying to run Microsoft 3D Movie Maker. An older Win95 series game. I followed the error message to Microsoft itself at: http://support.microsoft.com/kb/149823 Need to know how to get this Codec Installed into Wine, so I can run this Old Game. Plus, I'm not a full expert at Linux, so if a solution is found, a bit of guidance of the installation process would be greatly

Hi, I presently have 6 PRIs of IVR traffic that I am planning to migrate from Dialogic on SCO-Unix to Digium-Asterisk on Debian. Here is the general description of the traffic in question : - IVR system, 138 PRI channels (6 PRIs, multiple D-channel) - Some traffic from TV ads, so all traffic typically arrives within a few seconds - Up to 12-15 simul. outbound (external) conferencing calls for

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