search for: asterisk_g729

According to digium's site, "Note: Please do not attempt to use the G.729 code in a SCSI-only system. We are currently working with VoiceAge to correct this issue." (found at http://www.digium.com/index.php?menu=asterisk_g729). Does anyone know what these issues are? Can anyone define SCSI-only system? I know this sounds kinda dumb, but I have a server with SCSI and IDE interfaces, but no IDE drives. Is that SCSI only? Thanks for your help, David Gomillion

.... A typical call consumes 64Kbps of voice bandwidth. G.729 reduces the call to 8Kbps (of course normal IP overhead adds to this number). Many people are using Asterisk with G.729 to replace expensive gateways. For more information, visit the following link: http://www.digium.com/index.php?menu=asterisk_g729 thevoice.digium.com ------------------- Digium is proud to announce our latest website, http://thevoice.digium.com/, for obtaining professional voice recordings for your Asterisk IVR prompts. In conjunction with Allison Smith we have created an easy and cost effective way to have customized recor...

I' got a problem, using asterisk-rc2 :IVR functions (Background...Playback...etc) doesn't works : Executing Background("OH323/RXXXXX", "vm-extension") in new stack channel.c:1650 ast_set_write_fornat: Unable to find path from GSM to G729A---Asterisk box supplied only with network adapter.---Asterisk box registered in Mera (soft-switch with H323 protocol) and doing

Hi, I tried to play the standard demo-echotest file !. It works when i use an ip-phone (like x-lite or kphone), but as far as i use an PSTN Gateway (from an VOIP Provider) to call my phone - i get the following error: Sep 8 14:58:33 NOTICE[-182461520]: channel.c:1691 ast_set_write_format: Unable to find a path from GSM to G729A Sep 8 14:58:33 WARNING[-182461520]: file.c:779 ast_streamfile:

Somebody must have! Please, send me a g723 and/or g729 (for Asterisk) to pisac@hotmail.com (antispam subject: codec) Thanks, thanks, thanks... :-)

Hello everyone can anyonone of you send me the file codec_g729.so this file has to be inserted in /urs/lib/asterisk/modules Thank You

Has anyone out there successfully set up their * box to terminate their VONAGE calls? I (and I am sure lots of others) would love to hear how you did it. I'd like to be able to get rid of the extra hardware I have hanging around here and use the ASTERISK machine to handle the SIP termination instead of needing to have a Linksys modem (w/phone) and an additional X100P card. Thanks.

Hi there, Here at my company we are willing to use the asterisk IVR system. The problem we are having rigth now is that all our GWs use G729. I've read that in order to asterisk be able to make transcoding from the GSM audio files to G.729, it is necesary to purchase a license from digium. Is this correct? I've seen that licenses are purchased on a per-channel basis. Could

I can't seem to get music on hold working, it tries to work, but I just hear strange noises on the extension.. Here is some debug info. Looks like mpg123 starts fine, but I hear nothing. I'm on todays CVS build. -- Executing Answer("SIP/2203-062c", "") in new stack -- Executing MusicOnHold("SIP/2203-062c", "default") in new stack --