search for: sln16

It looks like there's a problem with the location or naming of the Extra SLN16 sounds: --14:11:43-- http://downloads.digium.com/pub/telephony/sounds/releases/asterisk-extra-sounds-fr-SLN16-1.4.9.tar.gz Resolving downloads.digium.com... 76.164.171.232 Connecting to downloads.digium.com|76.164.171.232|:80... connected. HTTP request sent, awaiting response....

...o an endpoint and to send over a remote file (HTTP). The issue is that I am experiencing an audio quality issue. I have tried encoding the file differently, but everytime Asterisk is cutting the audio frequencies above 4Khz. The call is established with G.722 and the audio file is mono 16Khz 16 bit sln16 extension. What can I do to improve the sound quality? Is there any way to not have asterisk cut the audio frequencies? best regards Tiago -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170512/1e6c7...

hi, which file extensios are supported in mixmonitor application? https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor can i record to Opus? -- --------------------------------------- Marek Cervenka =======================================

Hi all, If I use a SIP softphone and set to gsm codec clearly I get a 8K sample... if I change that to something like opus I get a much better sounding input... How do I get a "better" than 8K sampled input ? I "desire" to have that input be from a pipe. I have cd quality audio in this pipe - I would like to get that audio into asterisk to then send out to endpoints. How do I

Hello list, I have a file to be played in wav-format. I thought Asterisk would automatically take the wav-file and translate it to the codec used, but I see this : [Aug 28 11:16:29] WARNING[2705]: file.c:664 ast_openstream_full: File /var/lib/asterisk/sounds/vprompts/*zip-code.wav* does not exist in any format [Aug 28 11:16:29] WARNING[2705]: file.c:991 ast_streamfile: Unable to open

...talled in your operating system; e.g german= Europe/Berlin|'vm-received' a d b Y 'digits/at' kN So far. In asterisk.conf, the directory for sounds data points to /pool/asterisk/share/asterisk/sounds which is comprised of ./en/ ./de_DE/ de_DE contails all the .sln16 and .gsm files, owned by asterisk:asterisk, from the source above. Since the prompting of numbers and the date works well, but not the voicemail prompt, there is something fishy I can not fathom. For your help I'd like to thank in advance, Oliver -- O. Hartmann Ich widerspreche der Nutzung...