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Configuring Asterisk
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceConfig_id216341.html
The last two options, disallow and allow (sip.conf), are used to control which audio codecs are accepted from and offered to the telephone. By defining disallow=all first, we’re telling Asterisk to reset any previously defined codec definitions in the [general] section (or the internal defaults); then we explicitly declare which codecs we’ll accept (and the order we prefer).
Change Audio Codec - Asterisk Support - Asterisk Community
https://community.asterisk.org/t/change-audio-codec/83146
sampling rate is higher (44.1kHz) Why: If you want a high definition (HD) audio codec, you go for (for example) G.722, AMR-WB, or Opus. G.722 is build-in. Opus can be retrieved via the command ‘make menuselect’ or if you do not build Asterisk yourself, downloaded as binary file: Sangoma → Add-on Voice Codecs → Opus. Then, you add one of those audio codecs at …
Configuracion de Asterisk 1.6.X - Segunda Parte - sip.conf ...
https://www.voztovoice.org/?q=node/46
Asterisk tiene problemas para la negociacion audio/video (en el sentido que lo hace muy mal). En el sip.conf para cada extension definene un solo codec video (ej. H264) y hacen lo mismo en los softphone. Luego recargan la configuracion y intentan hacer una llamada video. Si no les funciona pueden poner (siempre en la configuracion de las ...
Asterisk Audio and Video Capabilities - Asterisk Project ...
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Audio+and+Video+Capabilities
Asterisk supports a variety of audio and video media. Asterisk provides CODEC modules to facilitate encoding and decoding of audio streams. Additionally file format modules are provided to handle writing to and reading from the file-system. The tables on this page describe what capabilities Asterisk supports and specific details for each format.
Asterisk 16 Configuration_res_pjsip - Asterisk Project ...
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Configuration_res_pjsip
Since this essentially replaces the underlying 'g726' codec with 'g726aal2' then 'g726aal2' needs to be specified in the endpoint's allowed codec list. inband_progress If set to yes , chan_pjsip will send a 183 Session Progress when told to indicate ringing and will immediately start sending ringing as audio.
Configuracion de Asterisk - quinta parte - iax.conf ...
https://www.voztovoice.org/?q=node/76
Actualizado 11 Septiembre 2009 Ya hemos visto como configurar extensiones SIP en nuestro Asterisk. Ahora vamos a ver como configurar más extensiones usando el protocolo IAX2. El archivo que tenemos que modificar el iax.conf y lo encontramos, como los demás, en la carpeta /etc/asterisk. Pues vamos a empezar. Indicaré solamente los parámetros más importantes de …
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