Voicemail call is cut by Originator due to lack of RTP/audio

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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.

Changes from MOR

Transmit silence

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

RTP keepalive

Another option would be to enable RTP keepalive. To do this, uncomment ;rtpkeepalive=<secs> line in 

/etc/asterisk/sip.conf

and set it to some small value (less than the time after which the call is cut) 

rtpkeepalive=20

To apply changes, reload sip 

asterisk -rx 'sip reload keeprt'
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