Going for test on your config
Thank you very much
[Resolved] Freepbx/Asterisk 15 minute call drop
Re: Calls are mute after 15 minutes
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
[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
Re: Calls are mute after 15 minutes
@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.
canreinvite=no
Thank you very much.
- Liptonbrisk
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Re: Calls are mute after 15 minutes
Jos wrote:@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.
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