Voicemail call is cut by Originator due to lack of RTP/audio
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Revision as of 08:24, 31 January 2022 by Nerijuss (talk | contribs) (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...")
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 being recorded. It is done by uncommenting/adding an option
transmit_silence = yes
to
/etc/asterisk/asterisk.conf
and restarting Asterisk service (drops ongoing calls):
service asterisk restart
