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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.
Recording Audio for Asterisk | Redial
http://www.itp-redial.com/class/weekly-notes/week2-notes/recording-audio-for-asterisk
Recording Audio for Asterisk. The audio standard for most Voice over IP is 8000Hz, 16Bit, Mono. Asterisk is going to expect that all audio conforms to this standard. There are a lot of codecs that are used to compress the audio, the most common being ULAW, GSM, and G729. The PSTN will only accept ULAW or G729.
No Sound on External SIP in Asterisk|FreePBX
https://www.sysfix.co.uk/Blog/No-sound-on-external-sip-asterisk.html
SIP One Way Audio Troubleshooting. Once these ports have been forwarded to the IP of your Asterisk server, give your router a reboot. Re-Login to your router just to reconfirm that the posts are in fact pointing to the correct IP of the Asterisk Server. The next step is to ensure that you configure your NAT settings on the Asterisk server correctly.
How To Convert Audio Files for Use With Asterisk
http://www.verterion.com/asterisk/convert-audio-files-use-asterisk/
Although Asterisk has many built-in audio files, most often you will want to have custom audio files for prompts for your auto attendants. You can either record them using the phone system itself (often times this can be somewhat confusing), or you can record them with conventional computer recording software and convert them into the proper format.
SIP with NAT or Firewalls - Asterisk Guru
http://asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html
Asterisk will send the audio to the port and ip where its receiving the audio from. Instead of relying on the addresses in the SIP and SDP messages. This will only work if the phone behind nat send and receive audio on the same port and if they send and receive the signaling on the same port. (The signaling port does not have to be the same as ...
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