[Resolved] Freepbx/Asterisk 15 minute call drop

[Jos]

Going for test on your config
Thank you very much

[Jos]

The result is

[2020-03-31 09:43:06] NOTICE[10863]: chan_sip.c:29865 check_rtp_timeout: Disconnecting call 'SIP/FreePhoneLine-00001178' for lack of RTP activity in 31 seconds

noting in the log about losing the audio.

the audio is lost 30 secconds before I have the notice in log

[Jos]

@mkaye it works now to pass the 15 minutes, the error I made is when I used the copy and paste command I did not include the last command
canreinvite=no

Thank you very much.

[Liptonbrisk]

@mkaye it works now to pass the 15 minutes, the error I made is when I used the copy and paste command I did not include the last command
canreinvite=no

Oh, interesting. Thanks for posting. I think I have a better understanding of what's happening. With that setting set to yes (or enabled), there's a SIP re-invite that occurs in an attempt to make RTP media bypass Asterisk/Freepbx, so that the stream travels peer to peer. The re-negotiation fails, and the call drops.