All public logs

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Combined display of all available logs of Kolmisoft Wiki. You can narrow down the view by selecting a log type, the username (case-sensitive), or the affected page (also case-sensitive).

Logs
  • 08:24, 31 January 2022 Nerijuss talk contribs created page Voicemail call is cut by Originator due to lack of RTP/audio (Created page with "By default, Asterisk is not transmitting RTP packets while Voicemail message is being recorded. If the originator (DID provider in most cases) has an RTP timeout mechanism active, the originator might drop the call to Voicemail. The best solution would be to inform the originator about that and ask to adjust the RTP timeout accordingly. Alternatively, if the originator is not capable to resolve it, Asterisk can be enabled to send fake RTP while a Voicemail message is b...")