similar to: Adpcm, 6KHz codec

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

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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 ;

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

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

Hi Steve and all, 1. Faxing from asterisk back to the same asterisk (from one Zap channel to another) doesn't work for us. Txfax called with the 'caller' parameter issues CED, while the receiving side needs CNG in order to switch to fax extension with rxfax. 2. This is probably the reason why J2 and our UC don't recognize incoming fax. Thank you. Alex Zarubin Webley Systems

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

Hello, CVS 07/31/03. Test with 130+ PSTN-to-SIP calls. Asterisk gets locked ... grep -e "Error" -e "eventually" p-console chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got

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

Hello, The v1-0_stable cvs release doesn't include the recent change ('poll' instead of 'select') in channel.c. Will it end up there any time soon, or we need to use cvs head to pick up this change? Thank you. Alex Zarubin Webley Systems -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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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

Hi We are having a significant (> 1 sec) delay in a multi-asterisk conference, with IAX2 legs connecting meetme on different boxes. All the other legs are PSTN (TE410P). The example configuration Slave box 1 meetme <--- IAX2 ---> Master box meetme <--- IAX2 ---> Slave box 2 meetme The delay is between Slave box 1 and Slave box 2 The primary suspect is our iax configuration

Hi all, After a long time having no time, I have finally done some fresh work on my software fax machine. I have replaced the original carrier tracking with something more robust. I have also added 4800, and 2400 bits per second modes, and cleaned up a few bugs in areas like superfine mode operation. I apologise for this update taking so long. At ftp://ftp.opencall.org/pub/spandsp you will

If there are multiple NICs in the box, how do we specify the local IP address to be used for RTP? Anything in rtp.conf ? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030609/d151f190/attachment.htm

Can T400P be configured for nfas (one d-channel providing signaling for more than one span)? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030522/f2b637a9/attachment.htm

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