Difference between revisions of "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...") |
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The best solution would be to inform the originator about that and ask to adjust the RTP timeout accordingly. | 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 | 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 | ||
| Line 14: | Line 15: | ||
service asterisk restart | 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' | |||
Latest revision as of 06:33, 26 September 2022
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'
